
From mallman@icir.org  Wed Jul  1 06:41:56 2009
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To: "Eddy, Wesley M. (GRC-MS00)[Verizon]" <wesley.m.eddy@nasa.gov>
From: Mark Allman <mallman@icir.org>
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Subject: Re: [tcpm] WGLC on draft-ietf-tcpm-early-rexmt-01
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> (1) In the Introduction, paragraph 2, we have latex-style
>     double single-quotes instead of normal double quotes
> 
> (2) "When using small congestion windows"
>     should more correctly be
>     "When the congestion window is small".
> 
> (3) "Small windows can occur" ->
>     "Small congestion windows can occur"
> 
> (4) "Consider a congestion window (cwnd)"
>     The document only uses "cwnd" in one other place, in the
>     Related Work, so probably doesn't need to define an abbreviation
>     for congestion window here.
> 
> (5) Need to either define SMSS in this document or cite where we
>     pull the definition of the acronym from.  Section 2 possibly
>     should have a sentence that says we're going to reuse the
>     set of terminology from the 2581bis draft or something like
>     that so we don't have to worry about explaining all the state
>     variables here.

OK, I fixed all these things in my working version.

Also, I replaced all references to RFC2581 to the bis document since it
has been forwarded and seemingly would not block this document.  If
needed we can revert this change if ER somehow passes 2581bis in the
processing. 

> (6) "good to bad retransmits" ->
>     "needed to unneeded retransmissions" ???

I used the language from the cited paper ([Pax97]) and am inclined to
leave it because I think it is clear.  If you want to insist I'll change
it because it doesn't seem worth fighting over.

> (7) check IDNITS :)

Added intended status (experimental).

Added a couple missing refs.

A bunch of the refs flagged as having no reference do in fact have a
reference but the tool doesn't seem to grok [ref1,ref2,ref3] sorts of
constructs.

RFC3522 is not normative and so I moved it.

Updated the NewReno reference to RFC 3782.

There is new boilerplate since we submitted -01.  Big surprise.  This
will be fixed in -02, obviously.

Many thanks!

allman




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From wesley.m.eddy@nasa.gov  Wed Jul  1 07:03:11 2009
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From: "Eddy, Wesley M. (GRC-MS00)[Verizon]" <wesley.m.eddy@nasa.gov>
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>-----Original Message-----
>From: mallman@icir.org [mailto:mallman@icir.org]
>Sent: Wednesday, July 01, 2009 9:41 AM
>
>> (6) "good to bad retransmits" ->
>>     "needed to unneeded retransmissions" ???
>
>I used the language from the cited paper ([Pax97]) and am inclined to
>leave it because I think it is clear.  If you want to insist I'll change
>it because it doesn't seem worth fighting over.
>


This works for me; I didn't realize it had been quoted.


---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------

From fernando@gont.com.ar  Wed Jul  1 09:38:52 2009
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Subject: [tcpm] Handling of malformed options (was: Re: poll for adopting draft-gont-tcp-security)
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Hello, Joe,

>>>> This suggests that extending SACK while still using the same option type
>>>> is not a good idea, or at least not a backwards-compatible one.
>>> ...
>>>
>>> It may also suggest a bug.
>> Where's the bug? The spec says the length field is e.g. 2. If you
>> implement the standard, and receive an option with a length field that
>> is different than 2, how are you supposed to process the option? If you
>> don't know how to process it, it seems sensible to ignore the option.
> 
> OK, since this is supposed to be a security doc, can you explain why you
> take a segment with a mis-formed option and ignore it, even when the
> length - the field that tells you where the next option starts - is
> nonsensical?
> 
> Why don't you drop the segment or send a RST?

draft-gont-tcp-security recommends to silently drop the segment, *NOT*
to silently ignore the option.

>From Section 4.4.1 (page 43) of draft-gont-tcp-security-00:

"  The SACK-permitted option is composed by an option-kind octet (which
   must be 4), and an option-length octet which must be 2.  Therefore,
   the following check should be performed on the option:

   option-length == 2

   If the option does not pass this check, the TCP segment carrying the
   option should be silently dropped."


This should be a clear example that this document is not simply turning
"current implementations" into "advice". i.e., the advice provided
sometimes differs from what many implementations are doing (as in this
case).

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





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Subject: Re: [tcpm] Handling of malformed options
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Fernando Gont wrote:
> Hello, Joe,
> 
>>>>> This suggests that extending SACK while still using the same option type
>>>>> is not a good idea, or at least not a backwards-compatible one.
>>>> ...
>>>>
>>>> It may also suggest a bug.
>>> Where's the bug? The spec says the length field is e.g. 2. If you
>>> implement the standard, and receive an option with a length field that
>>> is different than 2, how are you supposed to process the option? If you
>>> don't know how to process it, it seems sensible to ignore the option.
>> OK, since this is supposed to be a security doc, can you explain why you
>> take a segment with a mis-formed option and ignore it, even when the
>> length - the field that tells you where the next option starts - is
>> nonsensical?
>>
>> Why don't you drop the segment or send a RST?
> 
> draft-gont-tcp-security recommends to silently drop the segment, *NOT*
> to silently ignore the option.

OK, but why not make a more general recommendation for all options in
this regard? And why does the doc not indicate that this does differ
from what is widely deployed, and discuss that issue?

Let's look at one other:

3.6.7 present Ramaiah's suggestions on RST handling, but lacks the
advice on which components were already standard (silently dropping RSTs
out of window) and the conditions under which Ramaiah's extensions are
recommended.

I continue to believe the doc needs a scrub on this point. It should be
much more clear about the difference between:

	- recommendations already required by standards

	- recommendations currently deployed but not standard,
	and the WG's position on each (recommended, recommended
	in some deployments, not recommended, or no position)

	- additional recommendations, and whether they are
	consistent with or diverge from standards, and
	whether they have been deployed or differ from
	deployments

If we are going to work with this doc as a starting point, having a
dozen people replicating that effort isn't useful.

Joe

Joe
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Cc: Matt Mathis <mathis@psc.edu>, "tcpm-chairs@tools.ietf.org" <tcpm-chairs@tools.ietf.org>, tcpm Extensions WG <tcpm@ietf.org>, Matt Mathis <matt.mathis@gmail.com>
Subject: Re: [tcpm] Handling of malformed options
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Hello, Joe,

>>> Why don't you drop the segment or send a RST?
>> draft-gont-tcp-security recommends to silently drop the segment, *NOT*
>> to silently ignore the option.
> 
> OK, but why not make a more general recommendation for all options in
> this regard? 

Yes, I guess that such a general recommendation would make sense. The
reason for which I repeated this recommendation separately for each
option is that I wanted the option-specific checks to be within their
respective sections.

Had I made a general recommendation, it would have been something like
"if the option-length doesn't match the expected option length, the
corresponding segment should be silently dropped". But then that would
require the implementer to go back to the spec to check which value
should be checked, etc. Also, for some options there's more than one
check that you should perform on the option length (e.g., see SACK).

That said, there *is* a Section (3.10) with general recommendations for
handling TCP options. So I guess that one could keep the option-specific
checks in each of the different sections, but also add a general
recommendation in Section 3.10.. I have no problem with that.



> And why does the doc not indicate that this does differ
> from what is widely deployed, and discuss that issue?

It's just missing. As I said, I'm not saying there is not room for
improvement. And I agree that this discussion that you are suggesting
should be incorporated in the document (i.e., will do).



> Let's look at one other:
> 
> 3.6.7 present Ramaiah's suggestions on RST handling, but lacks the
> advice on which components were already standard (silently dropping RSTs
> out of window) 

Well, isn't this the case for any TCP segment? In-window checks are a
required part of RFC 793 itself. That's why I didn't include it here
explicitly.



> and the conditions under which Ramaiah's extensions are
> recommended.

Well, the document is about improving TCP's resiliency/security.
Therefore, it is assumed that at least from that pov, you will implement
Ramaiah's extensions.

Are are you meaning that I should state that IETF-wise, Ramaiah's
extensions come with an applicability statement, etc.?



> I continue to believe the doc needs a scrub on this point.
[...]

This may be necessary. However, I think the wg is currently making a
more meta-level decision, i.e., whether to adopt draft-gont-tcp-security
as a wg item.

That said, I have no problem with working on this particular point
you've raised.

Thanks!

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





From kojo@cs.helsinki.fi  Wed Jul  1 14:23:41 2009
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From: Markku Kojo <kojo@cs.helsinki.fi>
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Hi,

TCPM still seems to be scheduled for Friday on the draft agenda from 
this (Wed) morning. Any feedback from the secreteriat yet? I strongly
support your suggestion to be moved to Monday, or any slot other than
Friday afternoon. 

/Markku

On Mon, 22 Jun 2009, Eddy, Wesley M. (GRC-MS00)[Verizon] wrote:

> FYI, a DRAFT agenda for IETF 75 is published:
> http://www.ietf.org/meetings/75/75-Draft-Agenda.pdf
> 
> Note that TCPM is currently put on the agenda for Friday
> afternoon, though this seems really undesirable, so I have
> a suggestion in that we be moved to the Monday 1520-1720
> slot where there are no other TSV meeting scheduled ... I
> think this would be a lot better given the typical skeleton
> crew left by Friday afternoons ...
> 
> ---------------------------
> Wes Eddy
> Network & Systems Architect
> Verizon FNS / NASA GRC
> Office: (216) 433-6682
> ---------------------------
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 

From wesley.m.eddy@nasa.gov  Wed Jul  1 15:51:57 2009
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Subject: [tcpm] moving TCP to Monday for IETF 75
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Hello Secretariat,

For IETF 75, the TCPM meeting is currently on the draft
agenda for Friday afternoon.  This doesn't work well for
several of the WG participants, but many have noted that
the Monday afternoon slot where there are no other TSV
meetings (1520-1720) would work well, if there's a room
available.  We asked once before, but is it possible to
change times?

---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------


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Subject: [tcpm] I-D Action:draft-ietf-tcpm-tcp-auth-opt-05.txt
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--NextPart

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


	Title           : The TCP Authentication Option
	Author(s)       : J. Touch, et al.
	Filename        : draft-ietf-tcpm-tcp-auth-opt-05.txt
	Pages           : 46
	Date            : 2009-07-03

This document specifies the TCP Authentication Option (TCP-AO), which
obsoletes the TCP MD5 Signature option of RFC-2385 (TCP MD5). TCP-AO
specifies the use of stronger Message Authentication Codes (MACs),
protects against replays even for long-lived TCP connections, and
provides more details on the association of security with TCP
connections than TCP MD5. TCP-AO is compatible with either static
master key tuple (MKT) configuration or an external, out-of-band MKT
management mechanism; in either case, TCP-AO also protects
connections when using the same MKT across repeated instances of a
connection, using traffic keys derived from the MKT, and coordinates
MKT changes between endpoints. The result is intended to support
current infrastructure uses of TCP MD5, such as to protect long-lived
connections (as used, e.g., in BGP and LDP), and to support a larger
set of MACs with minimal other system and operational changes. TCP-AO
uses its own option identifier, even though used mutually exclusive
of TCP MD5 on a given TCP connection. TCP-AO supports IPv6, and is
fully compatible with the requirements for the replacement of TCP
MD5.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-auth-opt-05.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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From L.Wood@surrey.ac.uk  Fri Jul  3 16:11:03 2009
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On 30 Jun 2009, at 19:32, Joe Touch wrote:
>
> I've repeatedly said why I don't want to proceed on this path. It puts
> the WG in the position of repeating ourselves on issues we've already
> decided. Specific example - ICMP in-window checking. We acknowledge
> systems do this,

where is it acknowledged?

thanks,

L.

> but do NOT recommend it as a security check. I've said
> why in at least 3 different IETF meetings, as have others. Yet this  
> doc
> recommends it. That exceeds advice of the WG.
>
> If you want this doc to be the basis of work in this WG, why don't
> **YOU** start by revising it to at least apply the existing WG  
> positions
> on previously discussed advice?


but Joe, you've obviously got more free time...

DTN work: http://info.ee.surrey.ac.uk/Personal/L.Wood/saratoga/

<http://info.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>






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Lloyd Wood wrote:
> 
> On 30 Jun 2009, at 19:32, Joe Touch wrote:
>>
>> I've repeatedly said why I don't want to proceed on this path. It puts
>> the WG in the position of repeating ourselves on issues we've already
>> decided. Specific example - ICMP in-window checking. We acknowledge
>> systems do this,
> 
> where is it acknowledged?

In Gont's ID.

Joe
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From L.Wood@surrey.ac.uk  Fri Jul  3 22:53:09 2009
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Cc: tcpm Extensions WG <tcpm@ietf.org>, Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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On 4 Jul 2009, at 01:51, Joe Touch wrote:
> Lloyd Wood wrote:
>>
>> On 30 Jun 2009, at 19:32, Joe Touch wrote:
>>>
>>> I've repeatedly said why I don't want to proceed on this path. It  
>>> puts
>>> the WG in the position of repeating ourselves on issues we've  
>>> already
>>> decided. Specific example - ICMP in-window checking. We acknowledge
>>> systems do this,
>>
>> where is it acknowledged?
>
> In Gont's ID.

What, in the same individual ID you're spending so much energy  
suggesting
shouldn't even exist, and which you're dismissing as implementation  
advice
outside of the scope of the IETF and have otherwise been trying to  
prevent
being adopted as a WG draft?

How can this be the WG repeating itself on a decided issue, if the WG  
has not
yet acknowledged this in a WG document in the first place?

Obviously, it would be entirely unreasonable for someone to argue  
against
this document's adoption as a WG item while simultaneously using this  
same
document as an example of acknowledgement of decided WG issues not  
written
anywhere else, for that would be... well, hypocrisy.

L.

"Publishing recommendations without validating them isn't wasting  
time."?
DTN work: http://info.ee.surrey.ac.uk/Personal/L.Wood/saratoga/

<http://info.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>






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Lloyd Wood wrote:
> On 4 Jul 2009, at 01:51, Joe Touch wrote:
>> Lloyd Wood wrote:
>>>
>>> On 30 Jun 2009, at 19:32, Joe Touch wrote:
>>>>
>>>> I've repeatedly said why I don't want to proceed on this path. It puts
>>>> the WG in the position of repeating ourselves on issues we've already
>>>> decided. Specific example - ICMP in-window checking. We acknowledge
>>>> systems do this,
>>>
>>> where is it acknowledged?
>>
>> In Gont's ID.
> 
> What, in the same individual ID you're spending so much energy suggesting
> shouldn't even exist, and which you're dismissing as implementation advice
> outside of the scope of the IETF and have otherwise been trying to prevent
> being adopted as a WG draft?

draft-ietf-tcpm-icmp-attacks, not this one.

> How can this be the WG repeating itself on a decided issue, if the WG
> has not
> yet acknowledged this in a WG document in the first place?

The WG acknowledged the icmp-attacks ID, not this one.

> Obviously, it would be entirely unreasonable for someone to argue against
> this document's adoption as a WG item while simultaneously using this same
> document as an example of acknowledgement of decided WG issues not written
> anywhere else, for that would be... well, hypocrisy.

If this document contained only information of decided WG issues,
including WG consensus on those issues, you would be correct. I have
pointed out several cases where that is not the case - and that has been
(and remains) my concern.

Joe
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From fernando@gont.com.ar  Fri Jul  3 23:54:42 2009
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Cc: tcpm Extensions WG <tcpm@ietf.org>, Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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Joe Touch wrote:

> If this document contained only information of decided WG issues, 
> including WG consensus on those issues, you would be correct.

A very large part of draft-gont-tcp-security is about stuff that has
never been discussed by this wg, but that you dismiss with arguments
such as "who TCP needs not be implemented in a secure way", etc.

Also, draft-gont-tcp-security is based on the idea of giving advice for
implementing TCP as good resiliency as possible. And one might argue
that this is a different context as that of draft-ietf-icmp-attacks,
which I guess that was meant for the general case ("general case"
probably being that case in which we are happy to be vulnerable to
obvious  and trivial attacks, I guess).


> I have pointed out several cases where that is not the case - and
> that has been (and remains) my concern.

The only case for which you raised this is for the ICMP attacks stuff. 
Will you support draft-gont-tcp-security if, say, I were to remove the 
entire ICMP section, or replace it with a summary of what's in
draft-ietf-icmp-attacks?


<sidetrack>
As for the ICMP stuff itself: there have been arguments against checking
  ICMPs to be "in window". However, I don't seem to recall people arguing
against the idea of "hard errors should be processes as soft errors when
received for connections in any of the synchronized states" (*this* is
what has been implemented in virtually all stacks for more than 20 years
as the main counter-measure for icmp-based reset attacks). Has anybody
argued against ignoring ICMP source quenches meant for TCP connections?
(this is the main countermeasure for source quench attacks, and what has
been implemented in most stacks). The "in-window" check is useful as a
mitigation technique for the PMTU counter-measure -- however, the one
described in the I-D is mainly based on the idea of "check for progress
on the connection", which I believe parallels the idea in PLPMTUD. --
and I don't recall people arguing against that idea, either.
Also, see the reviewers in the Acks of the I-D.. that's people from most
of the major TCP implementations. And the ideas in the I-D are the
running code we have and we use. However, we are so full of ourselves
that we tell implementers that the code they have implemented, which has
been running the Internet for the last 20 years or so, is "buggy",
backed on how we think the code should be (despite it doesn't run in
that way, and in some cases has not done it that way for ages (if it 
ever did)).

Want to now what the industry did in response? -- The have ignored our
consensus.
</sidetrack>

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





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Cc: tcpm Extensions WG <tcpm@ietf.org>, Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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Joe Touch wrote:

>> Obviously, it would be entirely unreasonable for someone to argue against
>> this document's adoption as a WG item while simultaneously using this same
>> document as an example of acknowledgement of decided WG issues not written
>> anywhere else, for that would be... well, hypocrisy.
> 
> If this document contained only information of decided WG issues,
> including WG consensus on those issues, you would be correct. I have
> pointed out several cases where that is not the case - and that has been
> (and remains) my concern.

On April 13 
(http://www.ietf.org/mail-archive/web/tcpm/current/msg04501.html) you 
said: "Implementation advice is outside the scope of the IETF. It's not 
even operational, IMO.". i.e., you argued that we should *dismiss* this 
I-D (draft-gont-tcp-security).

So...if your opinion has now changed, and the only problem you have with 
this I-D is the section on "ICMP processing", would you support 
draft-gont-tcp-security (or at least not object its adoption as a wg 
item) were I to, e.g., remove the section on ICMP processing, replace it 
with a summary of what's in draft-ietf-icmp-attacks, or e.g., note that 
the provided advice is only for TCP implementations that desire good 
resiliency?

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





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Cc: tcpm Extensions WG <tcpm@ietf.org>, Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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Fernando Gont wrote:
...
> <sidetrack>
...
> Has anybody
> argued against ignoring ICMP source quenches meant for TCP connections?

Source quenches were supposed to be deprecated a while back, but never
made it into a doc.

> The "in-window" check is useful as a
> mitigation technique for the PMTU counter-measure -- however, the one
> described in the I-D is mainly based on the idea of "check for progress
> on the connection", which I believe parallels the idea in PLPMTUD. --
> and I don't recall people arguing against that idea, either.

Ignoring certain ICMPs when the connection is continuing to make
progress might be reasonable.

However, this has *nothing* to do with checking the in-window of the
ICMP payload, which is inappropriate. As I and others have pointed out
many times, routers are not required to send ICMPs in a timely basis, so
there's no reason that an out-of-window ICMP should be treated any
differently from an in-window one.

> Also, see the reviewers in the Acks of the I-D.. that's people from most
> of the major TCP implementations. And the ideas in the I-D are the
> running code we have and we use. However, we are so full of ourselves
> that we tell implementers that the code they have implemented, which has
> been running the Internet for the last 20 years or so, is "buggy",

OK, this is the sort of argument that holds no water. Convince us that
something is useful and correct. Pleaase cease claiming that
implementations define standards.

...
> Want to now what the industry did in response? -- The have ignored our
> consensus.

We need to continue to pull in the direction of standards, regardless of
what implementers do. If you care that much about the implementations,
then change them. It'd be more productive than simply documenting what
has been implemented instead.

Joe
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Fernando Gont wrote:
> Joe Touch wrote:
> 
>>> Obviously, it would be entirely unreasonable for someone to argue
>>> against
>>> this document's adoption as a WG item while simultaneously using this
>>> same
>>> document as an example of acknowledgement of decided WG issues not
>>> written
>>> anywhere else, for that would be... well, hypocrisy.
>>
>> If this document contained only information of decided WG issues,
>> including WG consensus on those issues, you would be correct. I have
>> pointed out several cases where that is not the case - and that has been
>> (and remains) my concern.
> 
> On April 13
> (http://www.ietf.org/mail-archive/web/tcpm/current/msg04501.html) you
> said: "Implementation advice is outside the scope of the IETF. It's not
> even operational, IMO.". i.e., you argued that we should *dismiss* this
> I-D (draft-gont-tcp-security).

That's correct, but please re-read what I wrote above. The doc does not
contain only info the WG has consensus on.

> So...if your opinion has now changed, and the only problem you have with
> this I-D is the section on "ICMP processing", would you support
> draft-gont-tcp-security (or at least not object its adoption as a wg
> item) were I to, e.g., remove the section on ICMP processing, replace it
> with a summary of what's in draft-ietf-icmp-attacks, or e.g., note that
> the provided advice is only for TCP implementations that desire good
> resiliency?

It may be useful to explain how to efficiently implement reordering and
reassembly, but not in an RFC.

Joe
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Cc: tcpm Extensions WG <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>, Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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>> OK, this is the sort of argument that holds no water. Convince us  
>> that
> something is useful and correct. Pleaase cease claiming that
> implementations define standards.

It's rough consensus and running code, Joe.

(Who died and made you king?)

>> Want to now what the industry did in response? -- The have ignored  
>> our
>> consensus.
>
> We need to continue to pull in the direction of standards,  
> regardless of
> what implementers do.

Well then, head on back to the happy perfect world of OSI, which  
implementers ignored.


> If you care that much about the implementations,
> then change them. It'd be more productive than simply documenting what
> has been implemented instead.

Implementation experience is an important input to developing and  
refining an IETF standard.

The IETF standard can't be defined wholly on paper theoretically de  
jure, or wholly
in implementations de facto. There's a meeting in the middle - hence  
consensus and code.

I know the M in TCP stands for minor, but really - why are we even  
bringing up standards arguments, when an informational doc would  
suffice?

L.

DTN work: http://info.ee.surrey.ac.uk/Personal/L.Wood/saratoga/

<http://info.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>

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Lloyd Wood wrote:
> 
>>> OK, this is the sort of argument that holds no water. Convince us that
>> something is useful and correct. Pleaase cease claiming that
>> implementations define standards.
> 
> It's rough consensus and running code, Joe.

It's rough consensus AND running code, Lloyd.

Running code isn't enough. Lots of running code doesn't define consensus.

>> If you care that much about the implementations,
>> then change them. It'd be more productive than simply documenting what
>> has been implemented instead.
> 
> Implementation experience is an important input to developing and
> refining an IETF standard.
> 
> The IETF standard can't be defined wholly on paper theoretically de
> jure, or wholly
> in implementations de facto. There's a meeting in the middle - hence
> consensus and code.

Please review sec 9.1 of the TAO of the IETF. The running code is to
prove that the standards are viable. It's the two together that are
meaningful. I.e., standards AND running code, not running code THEN
standards based on them.

> I know the M in TCP stands for minor, but really - why are we even
> bringing up standards arguments, when an informational doc would suffice?

draft-gont-tcp-security reads like BCP, and recommends changes that
would require it to be standards-track.

Is informational the intended direction? Or BCP? Or changes to standards?

Joe
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Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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On 4 Jul 2009, at 20:27, Joe Touch wrote:
>
>>> If you care that much about the implementations,
>>> then change them. It'd be more productive than simply documenting  
>>> what
>>> has been implemented instead.
>>
>> Implementation experience is an important input to developing and
>> refining an IETF standard.
>>
>> The IETF standard can't be defined wholly on paper theoretically de
>> jure, or wholly in implementations de facto. There's a meeting in  
>> the middle - hence
>> consensus and code.
>
> Please review sec 9.1 of the TAO of the IETF.

You might want to reread that. From section 9.1 of the Tao of the IETF:

'One of the oft-quoted tenets of the IETF is "running code wins"'

> The running code is to
> prove that the standards are viable.

No. 'One of the oft-quoted tenets of the IETF is "running code wins"'

The code is rather the _whole point_ of the exercise. No  
implementations,
no point.


> It's the two together that are
> meaningful. I.e., standards AND running code, not running code THEN
> standards based on them.

and not, as your 'change the implementations' line at top suggests,
standards THEN running code. It's a collaborative feedback loop,
not a one way process from document to code carried down the mountain
to the poor saps at the bottom.

Documenting what is implemented and used widely matters more than
standards that aren't implemented.

Running code wins.

>> I know the M in TCP stands for minor, but really - why are we even
>> bringing up standards arguments, when an informational doc would  
>> suffice?
>
> draft-gont-tcp-security reads like BCP, and recommends changes that
> would require it to be standards-track.

It recommends changes only to what was previously documented,
which has long been superseded by implementations. Not to reality.
(I suggest informational as a workaround, so that the standards
wonks can continue to believe all is right in their world.)

Running code wins.

(If TCPM doesn't take on this work, then TCPM is irrelevant, and the
IETF likely abdicates any authority it had on TCP. Still, there's
always adding new stuff to SCTP, eh?)

L.

DTN work: http://info.ee.surrey.ac.uk/Personal/L.Wood/saratoga/

<http://info.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>

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Lloyd Wood wrote:
> On 4 Jul 2009, at 20:27, Joe Touch wrote:
>>
>>>> If you care that much about the implementations,
>>>> then change them. It'd be more productive than simply documenting what
>>>> has been implemented instead.
>>>
>>> Implementation experience is an important input to developing and
>>> refining an IETF standard.
>>>
>>> The IETF standard can't be defined wholly on paper theoretically de
>>> jure, or wholly in implementations de facto. There's a meeting in the
>>> middle - hence
>>> consensus and code.
>>
>> Please review sec 9.1 of the TAO of the IETF.
> 
> You might want to reread that. From section 9.1 of the Tao of the IETF:
> 
> 'One of the oft-quoted tenets of the IETF is "running code wins"'

You need to quote the entire passage:

Implement -- Write programs that use the current Internet standards. The
standards aren't worth much unless they are available to Internet users.
Implement even the "minor" standards, since they will become less minor
if they appear in more software. Report any problems you find with the
standards to the appropriate Working Group so that the standard can be
clarified in later revisions. One of the oft-quoted tenets of the IETF
is "running code wins", so you can help support the standards you want
to become more widespread by creating more running code.

I.e., to support the standards, make running code. Notice it doesn't say
doing things the other way around.

...
> (If TCPM doesn't take on this work, then TCPM is irrelevant, and the
> IETF likely abdicates any authority it had on TCP. Still, there's
> always adding new stuff to SCTP, eh?)

You're basically claiming that RFC2525 was a waste of time. I disagree.

The WG needs to decide what we want (consensus - which means my voice
counts as much as yours or Fernando's), and decide what position we
should take. No, I don't think TCPM's charter is to run around trying to
standardize or, worse, document without taking a stand, every place
where implementation differs from standard.

In 1999 we had a backbone and decided between deployey* bugs (per 2525)
and things we wanted to change that weren't deployed (e.g., the TOS bit,
which was changed in RFC2873). All I'm asking is that we do the same
now, not just claim that deployed code is the basis of an appropriate
position.

Joe
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On 4 Jul 2009, at 22:01, Joe Touch wrote:
> Lloyd Wood wrote:
>> On 4 Jul 2009, at 20:27, Joe Touch wrote:
>>>
>>>>> If you care that much about the implementations,
>>>>> then change them. It'd be more productive than simply  
>>>>> documenting what
>>>>> has been implemented instead.
>>>>
>>>> Implementation experience is an important input to developing and
>>>> refining an IETF standard.
>>>>
>>>> The IETF standard can't be defined wholly on paper theoretically de
>>>> jure, or wholly in implementations de facto. There's a meeting in  
>>>> the
>>>> middle - hence
>>>> consensus and code.
>>>
>>> Please review sec 9.1 of the TAO of the IETF.
>>
>> You might want to reread that. From section 9.1 of the Tao of the  
>> IETF:
>>
>> 'One of the oft-quoted tenets of the IETF is "running code wins"'
>
> You need to quote the entire passage:

Quoting it at all rather misses the philosophical point that Tao can  
never
be adequately expressed in words, what? [section A.1 barely touches
on this paradox.]


> Implement -- Write programs that use the current Internet standards.  
> The
> standards aren't worth much unless they are available to Internet  
> users.
> Implement even the "minor" standards, since they will become less  
> minor
> if they appear in more software. Report any problems you find with the
> standards to the appropriate Working Group so that the standard can be
> clarified in later revisions. One of the oft-quoted tenets of the IETF
> is "running code wins", so you can help support the standards you want
> to become more widespread by creating more running code.
>
> I.e., to support the standards, make running code. Notice it doesn't  
> say
> doing things the other way around.

It does: "Report any problems you find with the standards to the  
appropriate
Working Group so that the standard can be clarified in later revisions."

And those problems are found with the implementations.
The standard is not immutable. The standard is not set in stone.
The standard can be revised. (Much as the Tao of the IETF is
revised.) There's a feedback loop. And, in that loop,
running code wins.

(There's another argument on how the Tao is written for the  
inexperienced
in the IETF, and how citing it in the first place either indicates
inexperience, or that far too much faith is placed in the written word,
which is _so_ not Taoist, as noted above.

And, of course, the Tao is informational, and thus has no bearing on  
standards-
track docs, so a self-respecting standards wonk wouldn't use it in the  
first
place. Much less quote it. Quoting something that recognises that it
can't be encapsulated in words is really missing the point.

And, reflecting on the Tao, one can observe that the above paragraph  
sounds
awfully like the evangelism of Stallman's manifestos. Lots of do-this- 
for-us
imperatives. It's laying out the appropriate puritan work ethic to
be followed, which is certainly not Taoist. The document's a sham!)


>> (If TCPM doesn't take on this work, then TCPM is irrelevant, and the
>> IETF likely abdicates any authority it had on TCP. Still, there's
>> always adding new stuff to SCTP, eh?)
>
> You're basically claiming that RFC2525 was a waste of time.

I claimed no such thing. (And in 1999, when RFC2525 was published, the  
IETF
was reaching its peak meeting attendance, indicating that it was more
relevant as an organisation And TCPM didn't yet exist.)

> I disagree.

You're disagreeing with a strawman position that you invented for me.

RFC2525 is informational, which is an approach that draft-gont could  
take.
The difference here is that we're documenting problems with the written
documentation, not with the implementations - i.e. the inverse of 2525.
The feedback loop also goes the other way. The aim is to keep documents
and code close together. Either can be changed. In this case, changing
the documentation to match widespread practice in a mature
protocol makes a lot of sense.


> The WG needs to decide what we want (consensus - which means my voice
> counts as much as yours or Fernando's),

rough consensus, not vote counting. I think Fernando's voice, and his
experience in working with others to put together this and other  
documents
in the face of consistent opposition from you, speaks far louder than  
either
of our voices. He's far more credible than we are.


> and decide what position we
> should take. No, I don't think TCPM's charter is to run around  
> trying to
> standardize or, worse, document without taking a stand, every place
> where implementation differs from standard.

Surprised you didn't quote the charter here.


> In 1999 we had a backbone and decided between deployey* bugs (per  
> 2525)
> and things we wanted to change that weren't deployed (e.g., the TOS  
> bit,
> which was changed in RFC2873). All I'm asking

Constantly using the imperative tense is not asking.


> is that we do the same
> now, not just claim that deployed code is the basis of an appropriate
> position.

Deployed code is, at least, testable science.
A piece of a standard without code implementing that piece as  
specified is
merely religious dogma.
And yet you prefer the latter position?

(RFC2525 didn't touch on the TOS bits. False argument there.)

L.

DTN work: http://info.ee.surrey.ac.uk/Personal/L.Wood/saratoga/

<http://info.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>

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Joe Touch wrote:

>> So...if your opinion has now changed, and the only problem you have
>> with this I-D is the section on "ICMP processing", would you
>> support draft-gont-tcp-security (or at least not object its
>> adoption as a wg item) were I to, e.g., remove the section on ICMP
>> processing, replace it with a summary of what's in
>> draft-ietf-icmp-attacks, or e.g., note that the provided advice is
>> only for TCP implementations that desire good resiliency?
> 
> It may be useful to explain how to efficiently implement reordering
> and reassembly, but not in an RFC.

Is this a joke?

RFC 816: "MODULARITY AND EFFICIENCY IN PROTOCOL IMPLEMENTATION"
RFC 815: "IP DATAGRAM REASSEMBLY ALGORITHMS" (see Section 4)

and,

RFC 1936: "Implementing the Internet Checksum in Hardware" (of which you
are on of the co-authors)

You argue against an I-D that gives implementation advice, yet you have
co-authored such an I-D? Should we take your opinion seriously?

Do you do this intentionally so that we waste cycles in these pointless
discussions instead of getting work done?

Sorry, but I will not play your game anymore.

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





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Fernando Gont wrote:
> Joe Touch wrote:
> 
>>> So...if your opinion has now changed, and the only problem you have
>>> with this I-D is the section on "ICMP processing", would you
>>> support draft-gont-tcp-security (or at least not object its
>>> adoption as a wg item) were I to, e.g., remove the section on ICMP
>>> processing, replace it with a summary of what's in
>>> draft-ietf-icmp-attacks, or e.g., note that the provided advice is
>>> only for TCP implementations that desire good resiliency?
>> It may be useful to explain how to efficiently implement reordering
>> and reassembly, but not in an RFC.
> 
> Is this a joke?
> 
> RFC 816: "MODULARITY AND EFFICIENCY IN PROTOCOL IMPLEMENTATION"

That's 817, FWIW.

> RFC 815: "IP DATAGRAM REASSEMBLY ALGORITHMS" (see Section 4)
> 
> and,
> 
> RFC 1936: "Implementing the Internet Checksum in Hardware" (of which you
> are on of the co-authors)
> 
> You argue against an I-D that gives implementation advice, yet you have
> co-authored such an I-D? Should we take your opinion seriously?

I didn't give advice, I gave an example, as do many RFCs. I don't agree
that efficiency is an appropriate focus of an RFC, and no, that doesn't
stop them from being written by others. There is a difference which is
relevant to some (granted, not all) recommendations in this document.

It is important to deciding whether you're describing (informationally)
or recommending (BCP-ish).

Joe
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Lloyd Wood wrote:
> On 4 Jul 2009, at 22:01, Joe Touch wrote:
>> Lloyd Wood wrote:
>>> On 4 Jul 2009, at 20:27, Joe Touch wrote:
>>>>
>>>>>> If you care that much about the implementations,
>>>>>> then change them. It'd be more productive than simply documenting
>>>>>> what
>>>>>> has been implemented instead.
>>>>>
>>>>> Implementation experience is an important input to developing and
>>>>> refining an IETF standard.
>>>>>
>>>>> The IETF standard can't be defined wholly on paper theoretically de
>>>>> jure, or wholly in implementations de facto. There's a meeting in the
>>>>> middle - hence
>>>>> consensus and code.
>>>>
>>>> Please review sec 9.1 of the TAO of the IETF.
>>>
>>> You might want to reread that. From section 9.1 of the Tao of the IETF:
>>>
>>> 'One of the oft-quoted tenets of the IETF is "running code wins"'
>>
>> You need to quote the entire passage:
> 
> Quoting it at all rather misses the philosophical point that Tao can never
> be adequately expressed in words, what? [section A.1 barely touches
> on this paradox.]

So you complain when I quote it in its entirety, but respond by further
quoting it out of context?

>> Implement -- Write programs that use the current Internet standards. The
>> standards aren't worth much unless they are available to Internet users.
>> Implement even the "minor" standards, since they will become less minor
>> if they appear in more software. Report any problems you find with the
>> standards to the appropriate Working Group so that the standard can be
>> clarified in later revisions. One of the oft-quoted tenets of the IETF
>> is "running code wins", so you can help support the standards you want
>> to become more widespread by creating more running code.
>>
>> I.e., to support the standards, make running code. Notice it doesn't say
>> doing things the other way around.
> 
> It does: "Report any problems you find with the standards to the
> appropriate
> Working Group so that the standard can be clarified in later revisions."

Clarifying a standard is what happens when the standard has an
ambiguity. Problems you find can be found in many ways.

None of that says "change the standards to match implementations". None
of that says that "code wins over standards".

> And those problems are found with the implementations.

That's one of many ways problems have been found.

> The standard is not immutable. The standard is not set in stone.
> The standard can be revised. (Much as the Tao of the IETF is
> revised.) There's a feedback loop. And, in that loop,
> running code wins.

Please re-read the paragraph above. It says to write code to support the
standards you want to become more widespread, not to write code to
support changes to the standard that you want to them justify as
evidence that the standard should be changed.

...
>>> (If TCPM doesn't take on this work, then TCPM is irrelevant, and the
>>> IETF likely abdicates any authority it had on TCP. Still, there's
>>> always adding new stuff to SCTP, eh?)
>>
>> You're basically claiming that RFC2525 was a waste of time.
> 
> I claimed no such thing. (And in 1999, when RFC2525 was published, the IETF
> was reaching its peak meeting attendance, indicating that it was more
> relevant as an organisation And TCPM didn't yet exist.)
> 
>> I disagree.
> 
> You're disagreeing with a strawman position that you invented for me.
> 
> RFC2525 is informational, which is an approach that draft-gont could take.

2525 doesn't need to be more than informational; it didn't change the
standards (it reiterated them, essentially). Straying from the standard
means either standards track (to change the standard) or BCP (to explain
ways to apply SHOULDs/alternates in the standard to an environment). And
2525 talks about how implementations vary from the standard, not
implementation issues that a standard never addresses (nor should it).

> The difference here is that we're documenting problems with the written
> documentation, not with the implementations - i.e. the inverse of 2525.
> The feedback loop also goes the other way. The aim is to keep documents
> and code close together. Either can be changed. In this case, changing
> the documentation to match widespread practice in a mature
> protocol makes a lot of sense.

...
>> and decide what position we
>> should take. No, I don't think TCPM's charter is to run around trying to
>> standardize or, worse, document without taking a stand, every place
>> where implementation differs from standard.
> 
> Surprised you didn't quote the charter here.

I'll wait for you to show me where in the charter it explains that we're
here to document and standardize implementations, rather than to decide
what's better for the future of TCP.

Joe
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Cc: tcpm Extensions WG <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>, Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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> I'll wait for you to show me where in the charter it explains that  
> we're
> here to document and standardize implementations, rather than to  
> decide
> what's better for the future of TCP.

The future of TCP only becomes real when implemented.

If you ignore implementations, you have no business "deciding the  
future of TCP".

Implementers detect pomposity, and route around it.

L.

DTN work: http://info.ee.surrey.ac.uk/Personal/L.Wood/saratoga/

<http://info.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>






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Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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On Wed, 24 Jun 2009, Eddy, Wesley M. (GRC-MS00)[Verizon] wrote:

> TCPMers, there was a thread a while ago about working on
> draft-gont-tcp-security in this working group that didn't
> conclusively give us a feeling one way or other:
> http://www.ietf.org/mail-archive/web/tcpm/current/msg04489.html
>
> Basically, my understanding is that there are at least a
> handful of people in the WG that think it should be done
> here as a WG item (more likely for Informational rather
> than BCP), and there are also some expressed opinions on
> why it shouldn't.
>
> Given the raw size of the document, if the WG intends to
> take this document on, then we need some people to clearly
> commit to putting cycles into review and contributions to
> the document.  Since it is quite large, and to my knowledge,
> there hasn't been a specific technical review of the content
> on this list, but just discussions about if the idea in
> general is a good or bad thing, we still need to know if
> people are willing to invest their time and energy in this.

I think the further development of this document with the balanced 
approach (cf: 'the requirements' mention and evaluating the mitigations 
with respect to them) is a useful activity, and as such, would like to 
commit cycles to it if it gets adopted.

thanks,
andrew

>
> Please let us know if there is traction for this in the
> near term, and/or we can also discuss it in Stockholm.
>
> ---------------------------
> Wes Eddy
> Network & Systems Architect
> Verizon FNS / NASA GRC
> Office: (216) 433-6682
> ---------------------------
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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Here's a draft agenda for the TCPM meeting in Stockholm; please let David a=
nd I know if we missed anyone that asked for a slot, or if there are other =
people that want slots ... there should be a little bit of room left given =
our 2 hour timeslot that was allocated.



TCP Maintenance & Minor Extensions Agenda
           IETF 75 - Stockholm, Sweden
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D

Note Well, Agenda Bashing
WG Status
10 minutes

Current WG Items
----------------
    TCP Authentication Option Changes & Status
    Joe Touch
    20 minutes

Other Items
-----------
    draft-gont-tcp-security
    WG Discussion
    15 minutes

    Make TCP More Robust to Long Connectivity Disruptions
    Alexander Zimmerman
    15 minutes

    Lowering the Initial RTO
    Jerry Chu
    15 minutes



---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------=

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Apologies for lack of subject line ...

________________________________________
From: Eddy, Wesley M. (GRC-MS00)[Verizon]
Sent: Tuesday, July 07, 2009 8:45 PM
To: tcpm@ietf.org
Subject:

Here's a draft agenda for the TCPM meeting in Stockholm; please let David a=
nd I know if we missed anyone that asked for a slot, or if there are other =
people that want slots ... there should be a little bit of room left given =
our 2 hour timeslot that was allocated.



TCP Maintenance & Minor Extensions Agenda
           IETF 75 - Stockholm, Sweden
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D

Note Well, Agenda Bashing
WG Status
10 minutes

Current WG Items
----------------
    TCP Authentication Option Changes & Status
    Joe Touch
    20 minutes

Other Items
-----------
    draft-gont-tcp-security
    WG Discussion
    15 minutes

    Make TCP More Robust to Long Connectivity Disruptions
    Alexander Zimmerman
    15 minutes

    Lowering the Initial RTO
    Jerry Chu
    15 minutes



---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------=

From gregory.ietf@gmail.com  Sat Jul 11 16:45:55 2009
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Gents,
We probably need 5-10 min in Stockholm to review the updated (will 
come out in next 48 hrs) AO-crypto-helper draft.

Suggested text for agenda:

    * Gregory Lebovitz                                       (10 minutes)
              "Cryptographic Algorithms, Use, & Implementation
              Requirments for TCP Authentication Option"
                 <http://tools.ietf.org/html?draft=draft-lebovitz-ietf-tcpm-tcp-ao-crypto-00>draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01

Thoughts?
Gregory.


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--=====================_6631703==.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<body>
Gents,<br>
We probably need 5-10 min in Stockholm to review the updated (will come
out in next 48 hrs) AO-crypto-helper draft.<br><br>
Suggested text for agenda:<br><br>
<pre>&nbsp;&nbsp; * Gregory
Lebovitz&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(10 minutes)
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&quot;Cryptographic Algorithms, Use, &amp; Implementation
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Requirments for TCP Authentication Option&quot;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<a href="http://tools.ietf.org/html?draft=draft-lebovitz-ietf-tcpm-tcp-ao-crypto-00">
draft-lebovitz-ietf-tcpm-tcp-ao-crypto-0</a>1

</pre><font face="Courier New, Courier">Thoughts?<br>
Gregory.<br><br>
</font><x-sigsep><p></x-sigsep>
+++++++++++++++++++++++<br>
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--=====================_6631703==.ALT--


From gregory.ietf@gmail.com  Sat Jul 11 23:56:11 2009
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To: Eric Rescorla <ekr@rtfm.com>,Joe Touch <touch@ISI.EDU>, Tim Polk <tim.polk@nist.gov>,Brian Weis <bew@cisco.com>, Bill Atwood <bill@cse.concordia.ca>,David McGrew <mcgrew@cisco.com>
From: "Gregory M. Lebovitz" <gregory.ietf@gmail.com>
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Cc: David Borman <david.borman@windriver.com>, tcpm@ietf.org, Wesley Eddy <weddy@grc.nasa.gov>
Subject: [tcpm] Fwd: New Version Notification for draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01
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Folks,
I just submitted the update for this document. Please have a quick 
review and provide feedback to me, cc'ing the tcpm@ietf.org list.

Wes and David B - the draft is posted, so you can link it in the WG agenda now.

Hope it helps,
Gregory.

>Delivered-To: gregory.ietf@gmail.com
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>From: IETF I-D Submission Tool <idsubmission@ietf.org>
>To: gregory.ietf@gmail.com
>Subject: New Version Notification for
>          draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01
>Date: Sat, 11 Jul 2009 23:48:20 -0700 (PDT)
>
>
>A new version of I-D, draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01.txt 
>has been successfuly submitted by Gregory Lebovitz and posted to the 
>IETF repository.
>
>Filename:       draft-lebovitz-ietf-tcpm-tcp-ao-crypto
>Revision:       01
>Title:          Cryptographic Algorithms, Use, & Implementation 
>Requirments for TCP Authentication Option
>Creation_date:  2009-07-11
>WG ID:          Independent Submission
>Number_of_pages: 16
>
>Abstract:
>The TCP Authentication Option, TCP-AO, relies on security algorithms
>to provide authentication between two end-points.  There are many
>such algorithms available, and two TCP-AO systems cannot interoperate
>unless they are using the same algorithm(s).  This document specifies
>the algorithms and attributes that can be used in TCP-AO's current
>manual keying mechanism.
> 
>
>
>
>The IETF Secretariat.

+++++++++++++++++++++++
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Juniper Networks
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From touch@ISI.EDU  Sun Jul 12 08:36:07 2009
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Gregory,

The document needs to be updated to align with AO-05 as follows:

	- remove the reference to the TAPD; use Master Key Tuple (MKT)
	instead

	- check spelling (TPC vs TCP)

	- AO uses the term data_block, not conn_block, for KDF input
	(conn_block could refer to the TCB from RFC 793,
	so I avoided that term)

	- the output of the PRF should be called the traffic key,
	not the connection key; similarly for the input to the MAC alg
		if a shorter term is needed in this doc, it
		needs to be defined in this doc (e.g., traff_key),
		because TCP-AO-05 doesn't define any notation
		other than "traffic key"

	- check formatting, esp. the line starting "where FR is always"

	- the tips for UI are useful, but 'tips' aren't a specification,
	so SHOULDs need to be removed

Joe


Gregory M. Lebovitz wrote:
> Folks,
> I just submitted the update for this document. Please have a quick
> review and provide feedback to me, cc'ing the tcpm@ietf.org list.
> 
> Wes and David B - the draft is posted, so you can link it in the WG
> agenda now.
> 
> Hope it helps,
> Gregory.
> 
>> Delivered-To: gregory.ietf@gmail.com
>> Authentication-Results: mx.google.com; spf=neutral (google.com:
>> 64.170.98.32 is neither permitted nor denied by best guess record for
>> domain of wwwrun@core3.amsl.com) smtp.mail=wwwrun@core3.amsl.com
>> From: IETF I-D Submission Tool <idsubmission@ietf.org>
>> To: gregory.ietf@gmail.com
>> Subject: New Version Notification for
>>          draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01
>> Date: Sat, 11 Jul 2009 23:48:20 -0700 (PDT)
>>
>>
>> A new version of I-D, draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01.txt
>> has been successfuly submitted by Gregory Lebovitz and posted to the
>> IETF repository.
>>
>> Filename:       draft-lebovitz-ietf-tcpm-tcp-ao-crypto
>> Revision:       01
>> Title:          Cryptographic Algorithms, Use, & Implementation
>> Requirments for TCP Authentication Option
>> Creation_date:  2009-07-11
>> WG ID:          Independent Submission
>> Number_of_pages: 16
>>
>> Abstract:
>> The TCP Authentication Option, TCP-AO, relies on security algorithms
>> to provide authentication between two end-points.  There are many
>> such algorithms available, and two TCP-AO systems cannot interoperate
>> unless they are using the same algorithm(s).  This document specifies
>> the algorithms and attributes that can be used in TCP-AO's current
>> manual keying mechanism.
>>
>>
>>
>>
>> The IETF Secretariat.
> 
> +++++++++++++++++++++++
> IETF-related email from
> Gregory M. Lebovitz
> Juniper Networks
> g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m
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From wesley.m.eddy@nasa.gov  Mon Jul 13 06:53:38 2009
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I've uploaded a draft TCPM agenda for IETF 75 at:
http://www.ietf.org/proceedings/09jul/agenda/tcpm.txt

I don't recall seeing an I-D or mail list thread for
the topic Jerry Chu wants to discuss (lowering the
initial RTO), so if he can give the mail list a bit
of a description of his proposal that would be
helpful.

---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------


From alexander.zimmermann@nets.rwth-aachen.de  Mon Jul 13 13:43:14 2009
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Hi folks,

I upload a new version of our draft to the IETF repository:
http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-01.txt

Comments are more than welcome.
I'm looking forward to seeing you in Stockholm.

Alex


Filename:	 draft-zimmermann-tcp-lcd
Revision:	 01
Title:		 Make TCP more Robust to Long Connectivity Disruptions
Creation_date:	 2009-07-13
WG ID:		 Independent Submission
Number_of_pages: 16

Abstract:
Disruptions in end-to-end path connectivity which last longer than
one retransmission timeout cause suboptimal TCP performance.  The
reason for the performance degradation is that TCP interprets segment
loss induced by connectivity disruptions as a sign of congestion,
resulting in repeated backoffs of the retransmission timer.  This
leads in turn to a deferred detection of the re-establishment of the
connection since TCP waits until the next retransmission timeout
occurs before attempting the retransmission.

This document describes how standard ICMP messages can be exploited
to disambiguate true congestion loss from non-congestion loss caused
by long connectivity disruptions.  Moreover, a revert strategy of the
retransmission timer is specified that enables a more prompt
detection of whether the connectivity to a previously disconnected
peer node has been restored or not.  The specified algorithm is a TCP
sender-only modification that effectively improves TCP performance in
presence of connectivity disruptions.

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From touch@ISI.EDU  Mon Jul 13 13:58:05 2009
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Hi, all,

Based on a discussion on a different list with Dan Wing, I'd like to
revisit thinking about NATs in the context of the current AO, which uses
traffic keys derived from the socket pair and ISN pair. We might be able
to now support NATs as follows, e.g.:

	add the following flags to the MKT:

		localNAT flag - indicates whether the local IP/port are
			zeroed before MAC calculation

		remoteNAT flag - indicates whether the remote IP/port
			are zeroed before MAC calculation

	add steps to the incoming/outgoing processing:
		zero the corresponding IP/port when the flag indicates,
		both on outgoing and incoming MAC calculation

That's basically it. A client behind a NAT would have a MKT with
localNAT true, and a server for that client would need to have a MKT
with remoteNAT true. This does require careful MKT configuration.
Although I wouldn't expect both localNAT and remoteNAT to be true, there
isn't a particular reason it needs to be prohibited.

Here's the security impact:

	- no impact to other connections (AFAICT)

	- SYN/SYN-ACKs limited only as much as the MKT TCP connection
	ID is, i.e., as a small range or single value rather than
	wildcard for either address or port

	- connections are reasonably well-protected once established,
	much like BTNS (due to use of both ISNs in the traffic keys)

	- reduced entropy for the traffic keys from a given MKT,
	since their input could be limited to the ISN pair in the
	worst case

I did NOT include this in TCP-AO-05, but it's simple enough to add if
useful. I'd like to ask that we think about this before Stockholm, and
discuss it on the list if possible in advance.

Joe
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	Title           : TCP Options and MSS
	Author(s)       : D. Borman
	Filename        : draft-ietf-tcpm-tcpmss-01.txt
	Pages           : 3
	Date            : 2009-07-13

This memo discusses what value to use with the TCP MSS option.

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From david.borman@windriver.com  Mon Jul 13 15:55:45 2009
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Hi,

I've just posted an update to the TCP MSS draft.  I made a mistake and  
posted a -01 version that didn't have the change, and so I had to  
follow it with a -02 version that has the actual change.

The change was to add a reference to the original e-mail on the TCP  
Large Windows mailing list where this information was first posted.  I  
believe that was the only change request that came out of the last  
IETF meeting.

It's a pretty simple document and I think it is ready for last call,  
so please speak up if you disagree.
	
			-David

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title           : TCP Options and MSS
	Author(s)       : D. Borman
	Filename        : draft-ietf-tcpm-tcpmss-02.txt
	Pages           : 3
	Date            : 2009-07-13

This memo discusses what value to use with the TCP MSS option.

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Joe,

If TCP-AO were for the exclusive use of BGP, I wouldn't worry so much
about NAT. Nobody (to the best of my knowlege) runs BGP across a NAT
boundary.

If TCP-AO is being designed for non-BGP applications also, the tweaks
that Joe describes make perfect sense. They appear to be simple enough
and shouldn't take too long to document.

                                   Ron

Joe Touch wrote:
> Hi, all,
> 
> Based on a discussion on a different list with Dan Wing, I'd like to
> revisit thinking about NATs in the context of the current AO, which uses
> traffic keys derived from the socket pair and ISN pair. We might be able
> to now support NATs as follows, e.g.:
> 
> 	add the following flags to the MKT:
> 
> 		localNAT flag - indicates whether the local IP/port are
> 			zeroed before MAC calculation
> 
> 		remoteNAT flag - indicates whether the remote IP/port
> 			are zeroed before MAC calculation
> 
> 	add steps to the incoming/outgoing processing:
> 		zero the corresponding IP/port when the flag indicates,
> 		both on outgoing and incoming MAC calculation
> 
> That's basically it. A client behind a NAT would have a MKT with
> localNAT true, and a server for that client would need to have a MKT
> with remoteNAT true. This does require careful MKT configuration.
> Although I wouldn't expect both localNAT and remoteNAT to be true, there
> isn't a particular reason it needs to be prohibited.
> 
> Here's the security impact:
> 
> 	- no impact to other connections (AFAICT)
> 
> 	- SYN/SYN-ACKs limited only as much as the MKT TCP connection
> 	ID is, i.e., as a small range or single value rather than
> 	wildcard for either address or port
> 
> 	- connections are reasonably well-protected once established,
> 	much like BTNS (due to use of both ISNs in the traffic keys)
> 
> 	- reduced entropy for the traffic keys from a given MKT,
> 	since their input could be limited to the ISN pair in the
> 	worst case
> 
> I did NOT include this in TCP-AO-05, but it's simple enough to add if
> useful. I'd like to ask that we think about this before Stockholm, and
> discuss it on the list if possible in advance.
> 
> Joe
_______________________________________________
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.


From david.borman@windriver.com  Mon Jul 13 16:12:02 2009
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Subject: Re: [tcpm] 1323bis - PAWS check's problem with reodering on the reverse path
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I've been thinking about this recently, and I think that while it may  
be a real issue, it's not one that we need to worry about.  I believe  
that the problem that the change fixes is a more likely scenario, so  
if you have to choose between the two, that's the one to pick.

Also, for this issue that Alexander brings up, the timestamp has to be  
changing between ACKs.  If the timestamp hasn't changed between the re- 
ordered ACKs, then the "SEG.TSval >= TS.recent" check will succeed.   
Generally, the TS clock shouldn't be ticking that often, if it is  
changing with every ACK then it might be going faster than it needs to.

So, this issue only occurs when ACKs are re-ordered, *and* the  
Timestamp clock has ticked between the reordered ACKs.  I'm thinking  
that 1323bis should just note that this situation does exist.

What does everyone else think?  Shall we leave the text as is and just  
document this edge case, or do people disagree with me and feel that  
this is a real problem that needs to be addressed, and is more  
important than the problems that the original change fixes?

			-David

On Mar 26, 2009, at 6:10 PM, Alexander Zimmermann wrote:

> Hi David, hi TCPM Folks,
>
> in the 1323bis draft the PAWS check has a problem (discarding ACKs)  
> if reordering is present on the
> reverse path and no data is piggybacked. Actually, it's not the PAWS  
> check itself, it's the modification
> of section 3.4: "Which Timestamp to Echo".
>
> The important parts of the RFC1323 and the draft (in this order) are
>
> ---
> (2) If Last.ACK.sent falls within the range of sequence numbers
>      of an incoming segment:
>
>         SEG.SEQ <= Last.ACK.sent < SEG.SEQ + SEG.LEN
>
>      then the TSval from the segment is copied to TS.Recent;
>      otherwise, the TSval is ignored.
>
> ---
>
> (2) If:
>
>       SEG.TSval >= TS.recent and SEG.SEQ <= Last.ACK.sent
>
>       then SEG.TSval is copied to TS.Recent; otherwise, it is
>       ignored.
> ---
>
> The addition of "SEG.TSval >= TS.recent" is not important for us at  
> moment. As the appendix said,
> this check is included for consistency with the PAWS test only.
>
> The relevant paragraph is the change from "SEG.SEQ <= Last.ACK.sent  
> < SEG.SEQ + SEG.LEN" to the
> reduced form "SEG.SEQ <= Last.ACK.sent". Again, the appendix said  
> that this change was made
> since the original algorithm to control which timestamp is echoed  
> was incorrect in two regards:
>
> (1) It failed to update TSrecent for a retransmitted segment that  
> resulted from a lost ACK.
>
> (2) It failed if SEG.LEN = 0.
>
> It's right that the 1323bis algo fix these two problems, however,  
> it's introduced the new problem mentioned
> already above.
>
> Following situation:
> - One-way flow from TCP A to TCP B
> - Reordering on the reverse path
>
> Consequence
> - TCP A will discard all ACKs with were delayed
> - Harmful in the case ABC (RFC 3465) is off, since we count ACKs for  
> slow-start and congestion avoidance
>
> Examples are attached.
>
> Alex
>
> <PAWS 1323.txt><PAWS 1323bis.txt>
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22220
> // email: zimmermann@cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>


From hkchu@google.com  Mon Jul 13 16:18:46 2009
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Subject: [tcpm] Tuning TCP parameters for the 21st century
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Sorry I don't have the I-D ready for submission yet (and the deadline
has passed anyway) so I'll try to describe what we are up to below.
I'll have more data to present in Stockholm.

At Google we have embarked on a mission to make the web faster, not
just for services provided by Google, but for every web user. We started
with TCP because TCP performance has always been a key component of
the end2end web performance. There are a number of default TCP parameter
settings, that, although conservative, have served us well over the
years. We believe the time has come to tune some of the parameters to
get more speed out of a much faster Internet than 10-20 years ago. The
current list includes:

Lowering initRTO
Increasing initcwnd
Lowering minRTO
Lowering delayed ack timer value

I'll start with Lowering initRTO.

RFC1122 contains the following paragraph:

The following values SHOULD be used to initialize the
estimation parameters for a new connection:

(a)  RTT = 0 seconds.

(b)  RTO = 3 seconds.  (The smoothed variance is to be
initialized to the value that will result in this RTO).

The "3secs SHOULD" is reaffirmed in RFC2988.

>From our own measurement of world wide RTT distribution to Google servers
we believe 3secs is too conservative, and like to propose it to be reduced
to 1sec.

Why does it matter?

We have seen SYN-ACK retransmission rates upto a few percentage points to
some of our servers. We also have indirect data showing the SYN (client
side) retransmission to be non-negligible (~1.42% worldwide). At a rate >
1% a large RTO value can have a significant negative impact on the average
end2end latency, hence the user experience. This is especially true for
short connections, including much of the web traffic.

What's the downside?

For those users behind a slow (e.g., dialup, wireless) link, the RTT may
still go up to > 1 sec. We believe a small amount of supriously
retransmitted SYN/SYN-ACK packets should not be a cause for concern (e.g.,
inducing more congestion,...) In some rare case the TCP performance may be
negatively affected by false congestion backoff, resulted from dupacks
caused by multiple spuriously retransmitted SYN/SYN-ACK packets. We believe
there are techniques to detect and mitigate these cases.

I'd like to hear feedback from this list on the feasibility of reducing
initRTO. For some other items (e.g., increasing IW/RW...) unfortunately I
don't have data ready to share yet but if you have an opinion on them I'd
love to hear it too.

Thanks,

Jerry

On Mon, Jul 13, 2009 at 6:54 AM, Eddy, Wesley M. (GRC-MS00)[Verizon]
<wesley.m.eddy@nasa.gov> wrote:
>
> I've uploaded a draft TCPM agenda for IETF 75 at:
> http://www.ietf.org/proceedings/09jul/agenda/tcpm.txt
>
> I don't recall seeing an I-D or mail list thread for
> the topic Jerry Chu wants to discuss (lowering the
> initial RTO), so if he can give the mail list a bit
> of a description of his proposal that would be
> helpful.
>
> ---------------------------
> Wes Eddy
> Network & Systems Architect
> Verizon FNS / NASA GRC
> Office: (216) 433-6682
> ---------------------------
>

From mahesh@cisco.com  Mon Jul 13 22:38:22 2009
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Subject: [tcpm] New I-D posted: draft-ananth-tcpm-persist
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A revised version of the draft has been posted and can be viewed here. 

http://www.ietf.org/internet-drafts/draft-ananth-tcpm-persist-01.txt

Filename:	 draft-ananth-tcpm-persist
Revision:	 01
Title:		 Clarification of sender behaviour in persist condition.
Creation_date:	 2009-07-10
WG ID:		 Independent Submission
Number_of_pages: 11

Abstract:
This document attempts to clarify the notion of the Zero Window
Probes (ZWP) described in RFC 1122 [RFC1122].  In particular, it
clarifies the actions that can be taken on connections which are
experiencing the ZWP condition.  The motivation for this document
stems from the belief that TCP implementations strictly adhering to
the current RFC language have the potential to become vulnerable to
Denial of Service (DoS) scenarios.

-- mj

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Jerry Chu wrote:

> From our own measurement of world wide RTT distribution to Google servers
> we believe 3secs is too conservative, and like to propose it to be reduced
> to 1sec.
 >
> Why does it matter?
> 
> We have seen SYN-ACK retransmission rates upto a few percentage points to
> some of our servers. We also have indirect data showing the SYN (client
> side) retransmission to be non-negligible (~1.42% worldwide). At a rate >
> 1% a large RTO value can have a significant negative impact on the average
> end2end latency, hence the user experience. This is especially true for
> short connections, including much of the web traffic.

For short web traffic I'd assume many of the connections go to a server 
with which the client has recently had a tcp connection.

What would be the pros and cons of using the cached rtt measurements 
from previous connections for the SYN?

    Erik

From touch@ISI.EDU  Tue Jul 14 07:16:27 2009
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-----BEGIN PGP SIGNED MESSAGE-----
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Erik Nordmark wrote:
> Jerry Chu wrote:
> 
>> From our own measurement of world wide RTT distribution to Google servers
>> we believe 3secs is too conservative, and like to propose it to be
>> reduced
>> to 1sec.
>>
>> Why does it matter?
>>
>> We have seen SYN-ACK retransmission rates upto a few percentage points to
>> some of our servers. We also have indirect data showing the SYN (client
>> side) retransmission to be non-negligible (~1.42% worldwide). At a rate >
>> 1% a large RTO value can have a significant negative impact on the
>> average
>> end2end latency, hence the user experience. This is especially true for
>> short connections, including much of the web traffic.
> 
> For short web traffic I'd assume many of the connections go to a server
> with which the client has recently had a tcp connection.
> 
> What would be the pros and cons of using the cached rtt measurements
> from previous connections for the SYN

You can use more than cached RTT; the technique and some other cacheable
items are described in RFC2140.

The pro is more rapid convergence to an accurate RTT; the con is that
you're using a potentially invalid RTT, but then that's what you do when
you start without knowing the RTT at all anyway.

It has also been implemented in Linux; see RFC4614.

Joe
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From hkchu@google.com  Tue Jul 14 12:28:32 2009
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Date: Tue, 14 Jul 2009 11:55:57 -0700
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Hi Erik,

It's good to hear from you!

On Tue, Jul 14, 2009 at 2:46 AM, Erik Nordmark<erik.nordmark@sun.com> wrote=
:
> Jerry Chu wrote:
>
>> From our own measurement of world wide RTT distribution to Google server=
s
>> we believe 3secs is too conservative, and like to propose it to be reduc=
ed
>> to 1sec.
>
>>
>>
>> Why does it matter?
>>
>> We have seen SYN-ACK retransmission rates upto a few percentage points t=
o
>> some of our servers. We also have indirect data showing the SYN (client
>> side) retransmission to be non-negligible (~1.42% worldwide). At a rate =
>
>> 1% a large RTO value can have a significant negative impact on the avera=
ge
>> end2end latency, hence the user experience. This is especially true for
>> short connections, including much of the web traffic.
>
> For short web traffic I'd assume many of the connections go to a server w=
ith
> which the client has recently had a tcp connection.
>
> What would be the pros and cons of using the cached rtt measurements from
> previous connections for the SYN?

Yes this is one of the things we've experiemented with. How effective the c=
ache
will depend on how much cache hit and the numbers for us were not very good=
.
Also Linux dst cache has some implementation/performance issues.

In order to increase the effectiveness of the cache, we're experiementing w=
ith a
cache based on /24 (or some other prefix length) rather than
individual dest IP addr.

Note that most likely whatever scheme that may work for the server
side won't work
for the client side but we'd like to see the client side (SYN) to
speed up as well,
hence the proposal to reduce initRTO.

Hope to see you at Stockholm!

Jerry

>
> =A0 Erik
>

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...
>>> >> What would be the pros and cons of using the cached rtt measurements
>>> >> from previous connections for the SYN
>> >
>> > You can use more than cached RTT; the technique and some other cacheable
>> > items are described in RFC2140.
> 
> Yes we are very aware of your proposal. I think the most effective one to speed
> up short connections is the snd_cwnd reuse, but unfortunately it's also the most
> dicey due to cwnd's highly temporal nature. Is this why snd_cwnd reuse was not
> implemented per rfc2140? Any insight and suggestion will be much appreciated.

We didn't distribute our implementations; the Linux code was developed
separately. We did do some tests using predictive SND_CWND learning.
There are two parts; one is "how fast would a connection speed up if
guessing were perfect", and the other is "how close to that perfect goal
can be achieved using basic prediction".

Granted, our web server didn't have your traffic, but we found that the
impact of SND_CWND was small, and limited to only a small subset of
connections. Connections too short don't benefit from a large window.
Connections too long have little benefit from increasing startup behavior.

(see the tech report at http://www.isi.edu/aln)

>> > The pro is more rapid convergence to an accurate RTT; the con is that
>> > you're using a potentially invalid RTT, but then that's what you do when
>> > you start without knowing the RTT at all anyway.
>> >
>> > It has also been implemented in Linux; see RFC4614.
> 
> Yes Linux stack maintains a "destination cache" of ssthresh and RTT on
> a per dest IP address basis. It doesn't currently use snd_cwnd, probably
> out of the same concerned mentioned above?

Lack of benefit is also an issue.

> Also it is very conservative
> hence won't use the cached RTT value unless the timestamp option is
> on.
> 
> Also the dst cache is only consulted after 3WHS. So for SYN/SYN-ACK
> the initRTO is still set to 3secs (don't know why).

If you spin the initRTO down too far, you end up resending the SYN
needlessly, no?

Joe

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On Tue, Jul 14, 2009 at 7:15 AM, Joe Touch<touch@isi.edu> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
>
>
> Erik Nordmark wrote:
>> Jerry Chu wrote:
>>
>>> From our own measurement of world wide RTT distribution to Google servers
>>> we believe 3secs is too conservative, and like to propose it to be
>>> reduced
>>> to 1sec.
>>>
>>> Why does it matter?
>>>
>>> We have seen SYN-ACK retransmission rates upto a few percentage points to
>>> some of our servers. We also have indirect data showing the SYN (client
>>> side) retransmission to be non-negligible (~1.42% worldwide). At a rate >
>>> 1% a large RTO value can have a significant negative impact on the
>>> average
>>> end2end latency, hence the user experience. This is especially true for
>>> short connections, including much of the web traffic.
>>
>> For short web traffic I'd assume many of the connections go to a server
>> with which the client has recently had a tcp connection.
>>
>> What would be the pros and cons of using the cached rtt measurements
>> from previous connections for the SYN
>
> You can use more than cached RTT; the technique and some other cacheable
> items are described in RFC2140.

Yes we are very aware of your proposal. I think the most effective one to speed
up short connections is the snd_cwnd reuse, but unfortunately it's also the most
dicey due to cwnd's highly temporal nature. Is this why snd_cwnd reuse was not
implemented per rfc2140? Any insight and suggestion will be much appreciated.

>
> The pro is more rapid convergence to an accurate RTT; the con is that
> you're using a potentially invalid RTT, but then that's what you do when
> you start without knowing the RTT at all anyway.
>
> It has also been implemented in Linux; see RFC4614.

Yes Linux stack maintains a "destination cache" of ssthresh and RTT on
a per dest IP address basis. It doesn't currently use snd_cwnd, probably
out of the same concerned mentioned above? Also it is very conservative
hence won't use the cached RTT value unless the timestamp option is
on.

Also the dst cache is only consulted after 3WHS. So for SYN/SYN-ACK
the initRTO is still set to 3secs (don't know why).

Jerry

>
> Joe
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On Tue, Jul 14, 2009 at 1:00 PM, Joe Touch<touch@isi.edu> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> ...
>>>> >> What would be the pros and cons of using the cached rtt measurements
>>>> >> from previous connections for the SYN
>>> >
>>> > You can use more than cached RTT; the technique and some other cacheable
>>> > items are described in RFC2140.
>>
>> Yes we are very aware of your proposal. I think the most effective one to speed
>> up short connections is the snd_cwnd reuse, but unfortunately it's also the most
>> dicey due to cwnd's highly temporal nature. Is this why snd_cwnd reuse was not
>> implemented per rfc2140? Any insight and suggestion will be much appreciated.
>
> We didn't distribute our implementations; the Linux code was developed
> separately. We did do some tests using predictive SND_CWND learning.
> There are two parts; one is "how fast would a connection speed up if
> guessing were perfect", and the other is "how close to that perfect goal
> can be achieved using basic prediction".
>
> Granted, our web server didn't have your traffic, but we found that the
> impact of SND_CWND was small, and limited to only a small subset of
> connections. Connections too short don't benefit from a large window.
> Connections too long have little benefit from increasing startup behavior.

Hmm, your conclusion seems a bit counter-intuitive - for a typical web object
of 5-8 pkts in size, starting with an initcwnd of 6, e.g., from the cache will
require only half of the round trips to complete the http transaction compared
to initcwnd = 3.

>
> (see the tech report at http://www.isi.edu/aln)
>
>>> > The pro is more rapid convergence to an accurate RTT; the con is that
>>> > you're using a potentially invalid RTT, but then that's what you do when
>>> > you start without knowing the RTT at all anyway.
>>> >
>>> > It has also been implemented in Linux; see RFC4614.
>>
>> Yes Linux stack maintains a "destination cache" of ssthresh and RTT on
>> a per dest IP address basis. It doesn't currently use snd_cwnd, probably
>> out of the same concerned mentioned above?
>
> Lack of benefit is also an issue.

See above. I thought this should be the one that will give the largest beneift
(in terms of latency reduction) for the web traffic, if done (i.e.,
predicted) correctly.

>
>> Also it is very conservative
>> hence won't use the cached RTT value unless the timestamp option is
>> on.
>>
>> Also the dst cache is only consulted after 3WHS. So for SYN/SYN-ACK
>> the initRTO is still set to 3secs (don't know why).
>
> If you spin the initRTO down too far, you end up resending the SYN
> needlessly, no?

Correct so there is a fine line to walk. But if > 98% of all TCP connections
experience RTT << 1 sec, it just seems too conservative to have a global
initRTO == 3secs just to avoid spurious retransmission in the < 2% category.

Jerry

>
> Joe
>
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> On Tue, Jul 14, 2009 at 1:00 PM, Joe Touch<touch@isi.edu> wrote:
...
>> > Granted, our web server didn't have your traffic, but we found that the
>> > impact of SND_CWND was small, and limited to only a small subset of
>> > connections. Connections too short don't benefit from a large window.
>> > Connections too long have little benefit from increasing startup behavior.
> 
> Hmm, your conclusion seems a bit counter-intuitive - for a typical web object
> of 5-8 pkts in size, starting with an initcwnd of 6, e.g., from the cache will
> require only half of the round trips to complete the http transaction compared
> to initcwnd = 3.

First, you get to send 4K or 4 packets (whichever is smaller) at the
start of a connection anyway. Second, RTT is only a fraction of the
overall delays experienced by a message. Finally, connections already
eat 2 RTTs before you do anything to get a response, due to the SYN
handshake followed by the request-response of HTTP. You might be eating
a single RTT off of a 3-4 RTT exchange. Your RTTs may go down by 30-50%,
but since they're not 100% of your delay, you end up benefitting only
around 20% or so.

Larger messages benefit as you open the window to burst the whole thing
out. Here's a back of the envelope:

# packets	was RTTs:	is RTTs:	benefit
	4	2		2		0%
 	10 	3		2		33%
	19	4		2		50%
	32	5		2		60%
	52	6		2		67%
	82	7		2		71%

Note though that at some point your SND_CWND will be opened only as far
as the path allows, at which point sending more packets just takes more
RTTs, so you start to lose. 90% of our max CWNDs were below 9K, which
means we only saw benefits in the 20-30% range (i.e., the first two rows
above). Sure, if you see CWNDs much larger, you might benefit from
reusing old values...
	
>> > (see the tech report at http://www.isi.edu/aln)
>> >
>>>>> >>> > The pro is more rapid convergence to an accurate RTT; the con is that
>>>>> >>> > you're using a potentially invalid RTT, but then that's what you do when
>>>>> >>> > you start without knowing the RTT at all anyway.
>>>>> >>> >
>>>>> >>> > It has also been implemented in Linux; see RFC4614.
>>> >>
>>> >> Yes Linux stack maintains a "destination cache" of ssthresh and RTT on
>>> >> a per dest IP address basis. It doesn't currently use snd_cwnd, probably
>>> >> out of the same concerned mentioned above?
>> >
>> > Lack of benefit is also an issue.
> 
> See above. I thought this should be the one that will give the largest beneift
> (in terms of latency reduction) for the web traffic, if done (i.e.,
> predicted) correctly.

It would if most connections's max CWND was larger than 10, and if most
connections sent more than 20 packets, but neither was the case for us.

>>> >> Also it is very conservative
>>> >> hence won't use the cached RTT value unless the timestamp option is
>>> >> on.
>>> >>
>>> >> Also the dst cache is only consulted after 3WHS. So for SYN/SYN-ACK
>>> >> the initRTO is still set to 3secs (don't know why).
>> >
>> > If you spin the initRTO down too far, you end up resending the SYN
>> > needlessly, no?
> 
> Correct so there is a fine line to walk. But if > 98% of all TCP connections
> experience RTT << 1 sec, it just seems too conservative to have a global
> initRTO == 3secs just to avoid spurious retransmission in the < 2% category.

It doesn't matter much if 99.99% of connections wouldn't benefit from
having the initRTO go off earlier anyway. That's the tradeoff. Don't
know if it helps your case, though.

Joe

>> > Joe
>> >
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From michael.scharf@ikr.uni-stuttgart.de  Wed Jul 15 02:35:00 2009
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On Tue, 14 Jul 2009 at 16:02:39, Joe Touch wrote:
> Larger messages benefit as you open the window to burst the whole thing
> out. Here's a back of the envelope:
> 
> # packets	was RTTs:	is RTTs:	benefit
> 	4	2		2		0%
>  	10 	3		2		33%
> 	19	4		2		50%
> 	32	5		2		60%
> 	52	6		2		67%
> 	82	7		2		71%
> 
> Note though that at some point your SND_CWND will be opened only as far
> as the path allows, at which point sending more packets just takes more
> RTTs, so you start to lose. 90% of our max CWNDs were below 9K, which
> means we only saw benefits in the 20-30% range (i.e., the first two rows
> above). Sure, if you see CWNDs much larger, you might benefit from
> reusing old values...

A large initial SND_CWND might not result in such a large benefit if
the receiver still announces a small receive window (e. g., the Linux
stack initially announces a receive window of about 6K only). This
issue is documented in an (expired) ID:

  http://tools.ietf.org/html/draft-scharf-tcpm-flow-control-quick-start-00

An example for a limitation by the receive window can also be seen on
slide 19 of

  http://www.ietf.org/proceedings/08nov/slides/iccrg-2.pdf

Note that, at least in case of Linux, it is possible to mitigate this
limitation by an ugly hack: The sender could ignore the announced
receive window during the first RTT of the Slow-Start and
optimistically send more data. As the receiver exponentially opens the
receive window, the received packets are always within the current
receive window - if the receiver does not run out of memory. But of
course this hack violates RFC 793.

BTW: My experiments with different fast startup schemes (with and
without rate pacing) also show that the absolute benefit of a faster
startup is small for many typical Internet workloads. Yet,
applications that frequently transfer mid-sized amounts of data can
indeed benefit. And since only few applications actually use a large
initial SND_CWND, the risk of a moderate increase of the initial
window appears to be rather small. As also mentioned in my ICCRG
presentation, the "Jump-Start" idea of M. Allman et al. seems to have
a quite reasonable performance in many scenarios.

Unfortunately, I cannot back these statements by large-scale
experimental data so far. But if someone wanted to perform some
experiments, e. g., with the "Jump-Start" scheme, I could provide
corresponding Linux kernel patches.

Michael

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Jerry Chu wrote:
...
>> We're taking one more look at T/TCP as well (probably need to start a different
>> mail thread on T/TCP too).

T/TCP is deprecated due to a fairly large security issue that has not
been solved, AFAIK. It'd be useful to have a solution to that issue
before proceeding with it as an approach.

...
>> The SYN/SYN-ACK retransmission rate we measured turned out to be >> 1%
>> in many cases (a bit surprising to us) hence the benefit.

That points to some other problem going on. As a packet loss rate,
that's quite high and causes problems elsewhere in TCP. It would be
useful to understand where the drops are and why. Retransmitting SYNs to
a busy endpoint whose queue is overflowing doesn't help things, e.g.

Joe
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Subject: Re: [tcpm] Tuning TCP parameters for the 21st century
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>>> The SYN/SYN-ACK retransmission rate we measured turned out to be >> 1%
>>> in many cases (a bit surprising to us) hence the benefit.
>
> That points to some other problem going on. As a packet loss rate,
> that's quite high and causes problems elsewhere in TCP. It would be

It doesn't have to be a packet loss rate, maybe it was a
per-connection measurement. We also found this to occur
for > 0.5% of connections in our own measurement study:

Dragana Damjanovic, Philipp Gschwandtner, Michael Welzl: "Why is this Web
Page coming up so slow? Investigating the Loss of SYN Packets", IFIP/TC6
NETWORKING 2009, 11-15 May 2009, Aachen, Germany.
http://www.welzl.at/research/publications/networking2009.pdf


> useful to understand where the drops are and why. Retransmitting SYNs to
> a busy endpoint whose queue is overflowing doesn't help things, e.g.

I understand that now, too (thanks to Lachlan who
explained it to me), and hence apologize for having
been ignorant to this rather obvious issue in the
above paper.

- but what about retransmitting SYN/ACKs faster?
Since the other side already sent a SYN, it probably
isn't overloaded.

Cheers,
Michael



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michawe@ifi.uio.no wrote:
...
>> useful to understand where the drops are and why. Retransmitting SYNs to
>> a busy endpoint whose queue is overflowing doesn't help things, e.g.
> 
> I understand that now, too (thanks to Lachlan who
> explained it to me), and hence apologize for having
> been ignorant to this rather obvious issue in the
> above paper.
> 
> - but what about retransmitting SYN/ACKs faster?
> Since the other side already sent a SYN, it probably
> isn't overloaded

That presumes the other side already sent a SYN and it's been received,
and that the SYN/ACK is what's getting lost/delayed/retransmitted.  If
not, that won't help.

First find where things are going wrong empirically ;-)

Joe
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On Tue, Jul 14, 2009 at 4:02 PM, Joe Touch<touch@isi.edu> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
>> On Tue, Jul 14, 2009 at 1:00 PM, Joe Touch<touch@isi.edu> wrote:
> ...
>>> > Granted, our web server didn't have your traffic, but we found that t=
he
>>> > impact of SND_CWND was small, and limited to only a small subset of
>>> > connections. Connections too short don't benefit from a large window.
>>> > Connections too long have little benefit from increasing startup beha=
vior.
>>
>> Hmm, your conclusion seems a bit counter-intuitive - for a typical web o=
bject
>> of 5-8 pkts in size, starting with an initcwnd of 6, e.g., from the cach=
e will
>> require only half of the round trips to complete the http transaction co=
mpared
>> to initcwnd =3D 3.
>
> First, you get to send 4K or 4 packets (whichever is smaller) at the
> start of a connection anyway. Second, RTT is only a fraction of the
> overall delays experienced by a message.

Sure and we are looking at other layers (especially HTTP) for performance
optimization as well.

>Finally, connections already
> eat 2 RTTs before you do anything to get a response, due to the SYN

We're taking one more look at T/TCP as well (probably need to start a diffe=
rent
mail thread on T/TCP too).

> handshake followed by the request-response of HTTP. You might be eating
> a single RTT off of a 3-4 RTT exchange. Your RTTs may go down by 30-50%,
> but since they're not 100% of your delay, you end up benefitting only
> around 20% or so.
>
> Larger messages benefit as you open the window to burst the whole thing
> out. Here's a back of the envelope:
>
> # packets =A0 =A0 =A0 was RTTs: =A0 =A0 =A0 is RTTs: =A0 =A0 =A0 =A0benef=
it
> =A0 =A0 =A0 =A04 =A0 =A0 =A0 2 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 0%
> =A0 =A0 =A0 =A010 =A0 =A0 =A03 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 33%
> =A0 =A0 =A0 =A019 =A0 =A0 =A04 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 50%
> =A0 =A0 =A0 =A032 =A0 =A0 =A05 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 60%
> =A0 =A0 =A0 =A052 =A0 =A0 =A06 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 67%
> =A0 =A0 =A0 =A082 =A0 =A0 =A07 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 71%
>
> Note though that at some point your SND_CWND will be opened only as far
> as the path allows, at which point sending more packets just takes more
> RTTs, so you start to lose. 90% of our max CWNDs were below 9K, which
> means we only saw benefits in the 20-30% range (i.e., the first two rows
> above). Sure, if you see CWNDs much larger, you might benefit from
> reusing old values...

Correct. We are trying to pilot our cache code and will find out our snd_cw=
nd
distribution, and how much it helps (or hurts if it triggers more pkt drops=
).

>
>>> > (see the tech report at http://www.isi.edu/aln)
>>> >
>>>>>> >>> > The pro is more rapid convergence to an accurate RTT; the con =
is that
>>>>>> >>> > you're using a potentially invalid RTT, but then that's what y=
ou do when
>>>>>> >>> > you start without knowing the RTT at all anyway.
>>>>>> >>> >
>>>>>> >>> > It has also been implemented in Linux; see RFC4614.
>>>> >>
>>>> >> Yes Linux stack maintains a "destination cache" of ssthresh and RTT=
 on
>>>> >> a per dest IP address basis. It doesn't currently use snd_cwnd, pro=
bably
>>>> >> out of the same concerned mentioned above?
>>> >
>>> > Lack of benefit is also an issue.
>>
>> See above. I thought this should be the one that will give the largest b=
eneift
>> (in terms of latency reduction) for the web traffic, if done (i.e.,
>> predicted) correctly.
>
> It would if most connections's max CWND was larger than 10, and if most
> connections sent more than 20 packets, but neither was the case for us.
>
>>>> >> Also it is very conservative
>>>> >> hence won't use the cached RTT value unless the timestamp option is
>>>> >> on.
>>>> >>
>>>> >> Also the dst cache is only consulted after 3WHS. So for SYN/SYN-ACK
>>>> >> the initRTO is still set to 3secs (don't know why).
>>> >
>>> > If you spin the initRTO down too far, you end up resending the SYN
>>> > needlessly, no?
>>
>> Correct so there is a fine line to walk. But if > 98% of all TCP connect=
ions
>> experience RTT << 1 sec, it just seems too conservative to have a global
>> initRTO =3D=3D 3secs just to avoid spurious retransmission in the < 2% c=
ategory.
>
> It doesn't matter much if 99.99% of connections wouldn't benefit from
> having the initRTO go off earlier anyway. That's the tradeoff. Don't
> know if it helps your case, though.

The SYN/SYN-ACK retransmission rate we measured turned out to be >> 1%
in many cases (a bit surprising to us) hence the benefit.

Jerry

>
> Joe
>
>>> > Joe
>>> >
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On 14 Jul 2009, at 15:15, Joe Touch wrote:

> It has also been implemented in Linux; see RFC4614.

And FreeBSD, BTW.

--
Rui Paulo


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Cc: 'tcpm Extensions WG' <tcpm@ietf.org>
Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On 
> Behalf Of Andrew Yourtchenko
> Sent: Monday, July 06, 2009 9:31 AM
> To: Eddy, Wesley M. (GRC-MS00)[Verizon]
> Cc: tcpm Extensions WG
> Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
> 
> 
> 
> On Wed, 24 Jun 2009, Eddy, Wesley M. (GRC-MS00)[Verizon] wrote:
> 
> > TCPMers, there was a thread a while ago about working on
> > draft-gont-tcp-security in this working group that didn't
> > conclusively give us a feeling one way or other:
> > http://www.ietf.org/mail-archive/web/tcpm/current/msg04489.html
> >
> > Basically, my understanding is that there are at least a
> > handful of people in the WG that think it should be done
> > here as a WG item (more likely for Informational rather
> > than BCP), and there are also some expressed opinions on
> > why it shouldn't.
> >
> > Given the raw size of the document, if the WG intends to
> > take this document on, then we need some people to clearly
> > commit to putting cycles into review and contributions to
> > the document.  Since it is quite large, and to my knowledge,
> > there hasn't been a specific technical review of the content
> > on this list, but just discussions about if the idea in
> > general is a good or bad thing, we still need to know if
> > people are willing to invest their time and energy in this.
> 
> I think the further development of this document with the balanced 
> approach (cf: 'the requirements' mention and evaluating the 
> mitigations 
> with respect to them) is a useful activity, and as such, 
> would like to 
> commit cycles to it if it gets adopted.

Agreed.  And I will contribute cycles as well.

-d


> thanks,
> andrew
> 
> >
> > Please let us know if there is traction for this in the
> > near term, and/or we can also discuss it in Stockholm.
> >
> > ---------------------------
> > Wes Eddy
> > Network & Systems Architect
> > Verizon FNS / NASA GRC
> > Office: (216) 433-6682
> > ---------------------------
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 


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Date: Wed, 15 Jul 2009 14:20:15 -0700
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On Wed, Jul 15, 2009 at 7:25 AM, Joe Touch<touch@isi.edu> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
>
>
> Jerry Chu wrote:
> ...
>>> We're taking one more look at T/TCP as well (probably need to start a different
>>> mail thread on T/TCP too).
>
> T/TCP is deprecated due to a fairly large security issue that has not
> been solved, AFAIK. It'd be useful to have a solution to that issue
> before proceeding with it as an approach.

Understood. On the high level it seems to me a nonce/cookie exchanged
over one reqular TCP connection (w/ 3WHS) upfront will always be required
but should be sufficient to cover later T/TCP sessions and address the security
concern? (Oh perhaps the cookie can be passed around by attackers to mount
a DDOS attack?)

> ...
>>> The SYN/SYN-ACK retransmission rate we measured turned out to be >> 1%
>>> in many cases (a bit surprising to us) hence the benefit.
>
> That points to some other problem going on. As a packet loss rate,
> that's quite high and causes problems elsewhere in TCP. It would be
> useful to understand where the drops are and why. Retransmitting SYNs to
> a busy endpoint whose queue is overflowing doesn't help things, e.g.

One set of our data show our TCP pkt rexmit rate ranges from 0.8% - 2.4%
globally. Even at 1% drop rate 3secs is a noticeable bump for the average
latency.

Jerry

>
> Joe
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Date: Wed, 15 Jul 2009 15:34:19 -0700
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Subject: Re: [tcpm] Tuning TCP parameters for the 21st century
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On Wed, Jul 15, 2009 at 2:15 AM, Michael
Scharf<michael.scharf@ikr.uni-stuttgart.de> wrote:
> On Tue, 14 Jul 2009 at 16:02:39, Joe Touch wrote:
>> Larger messages benefit as you open the window to burst the whole thing
>> out. Here's a back of the envelope:
>>
>> # packets =A0 =A0 was RTTs: =A0 =A0 =A0 is RTTs: =A0 =A0 =A0 =A0benefit
>> =A0 =A0 =A0 4 =A0 =A0 =A0 2 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 0%
>> =A0 =A0 =A0 10 =A0 =A0 =A03 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 33%
>> =A0 =A0 =A0 19 =A0 =A0 =A04 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 50%
>> =A0 =A0 =A0 32 =A0 =A0 =A05 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 60%
>> =A0 =A0 =A0 52 =A0 =A0 =A06 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 67%
>> =A0 =A0 =A0 82 =A0 =A0 =A07 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 71%
>>
>> Note though that at some point your SND_CWND will be opened only as far
>> as the path allows, at which point sending more packets just takes more
>> RTTs, so you start to lose. 90% of our max CWNDs were below 9K, which
>> means we only saw benefits in the 20-30% range (i.e., the first two rows
>> above). Sure, if you see CWNDs much larger, you might benefit from
>> reusing old values...
>
> A large initial SND_CWND might not result in such a large benefit if
> the receiver still announces a small receive window (e. g., the Linux
> stack initially announces a receive window of about 6K only). This
> issue is documented in an (expired) ID:
>
> =A0http://tools.ietf.org/html/draft-scharf-tcpm-flow-control-quick-start-=
00

Yes I remember Linux stack also "slow starts" the receive window.
The good news is that for us the receiver is on the client side dominated b=
y
the Windows stack, which is probably not sophisticated enough to do this.
(Need double check.)

>
> An example for a limitation by the receive window can also be seen on
> slide 19 of
>
> =A0http://www.ietf.org/proceedings/08nov/slides/iccrg-2.pdf
>
> Note that, at least in case of Linux, it is possible to mitigate this
> limitation by an ugly hack: The sender could ignore the announced
> receive window during the first RTT of the Slow-Start and
> optimistically send more data. As the receiver exponentially opens the
> receive window, the received packets are always within the current
> receive window - if the receiver does not run out of memory. But of
> course this hack violates RFC 793.
>
> BTW: My experiments with different fast startup schemes (with and
> without rate pacing) also show that the absolute benefit of a faster
> startup is small for many typical Internet workloads. Yet,
> applications that frequently transfer mid-sized amounts of data can
> indeed benefit. And since only few applications actually use a large
> initial SND_CWND, the risk of a moderate increase of the initial
> window appears to be rather small. As also mentioned in my ICCRG
> presentation, the "Jump-Start" idea of M. Allman et al. seems to have
> a quite reasonable performance in many scenarios.

Thanks for the pointers! This seems very relevant to our on-going struggle
with TCP slow start :-).

>
> Unfortunately, I cannot back these statements by large-scale
> experimental data so far. But if someone wanted to perform some
> experiments, e. g., with the "Jump-Start" scheme, I could provide
> corresponding Linux kernel patches.

We're planning for some experiement in this area too. Will let you know
how it goes.

Jerry

>
> Michael
>

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Jerry Chu wrote:
> On Wed, Jul 15, 2009 at 7:25 AM, Joe Touch<touch@isi.edu> wrote:
> 
> 
> Jerry Chu wrote:
> ...
>>>>> We're taking one more look at T/TCP as well (probably need to start a different
>>>>> mail thread on T/TCP too).
> T/TCP is deprecated due to a fairly large security issue that has not
> been solved, AFAIK. It'd be useful to have a solution to that issue
> before proceeding with it as an approach.
> 
>> Understood. On the high level it seems to me a nonce/cookie exchanged
>> over one reqular TCP connection (w/ 3WHS) upfront will always be required
>> but should be sufficient to cover later T/TCP sessions and address the security
>> concern? (Oh perhaps the cookie can be passed around by attackers to mount
>> a DDOS attack?)

Yeah - that's the problem. See http://www.mid-way.org/doc/ttcp-sec.txt

Nonces are addressed in Sec 5, and there doesn't appear to be a solution.

Joe
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From michawe@ifi.uio.no  Thu Jul 16 01:39:25 2009
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> > - but what about retransmitting SYN/ACKs faster?
> > Since the other side already sent a SYN, it probably
> > isn't overloaded
> 
> That presumes the other side already sent a SYN and it's been received,
> and that the SYN/ACK is what's getting lost/delayed/retransmitted.  If
> not, that won't help.

Right -


> First find where things are going wrong empirically ;-)

We've done that. See, for example, fig. 2 (a) of
http://www.welzl.at/research/publications/networking2009.pdf
(page 7 of the pdf)

Since our goal in this paper was to document general failures
of connection setup, we did not strictly split our results
into the case of lost SYNs or lost SYN/ACKs in it.
Section 3 (end of page 3 / beginning of page 4) explains
how lost SYN/ACKs were incorporated in our measurements.


This paper is based on data produced by a student. The
data are more thoroughly documented in his (alas, German)
bachelor thesis. This document is here:
http://www.welzl.at/research/tools/syn-retransmit/benjamin_kaser_bak1.pdf
and the most relevant information is table 2 on page 47.
(Table 1 is similar, just for SYNs, not SYN/ACKs).

If you look at it, with or without German knowledge
you will be able to see the connection to table 1 in
the paper, where we exchanged columns with lines and
omitted some details - apparently just the crucial
ones  :-(    Anyway, the listed measurements are just
the same, and hence you can read the details in the
paper if you're interested in how we obtained our
data.

Table 2 on page 47 of the bachelor thesis has several
details about duplicate SYN/ACKs that we've seen,
e.g. classified by eventually successful and unsuccessful
connections. As a quick summary, in 3432744 connection
attempts, we have seen 56586 duplicate SYN/ACKs in total
(this is a cumulative number, i.e. 4 SYN/ACKs are counted
as 3). Counting only the occurrence of duplicate SYN/ACKs,
we saw them for 0,55% of the 3432744 connection attempts.
In general, the difference between duplicate SYN/ACK
occurrences and duplicate SYN occurrences is negligible
in our measurements.

As I just looked at the data again, I noticed a mistake:
in the paper, we wrote that the problem only happens for
around 0.5% of the connections, but this is really only
from the SYN table, not the SYN/ACK table in the bachelor
thesis. The SYN/ACK table covers the same measurement sets.
Hence, in total, it seems that we also saw > 1% lost SYNs
*or* SYN/ACKs, which exactly matches what Jerry Chu said.
 
BTW I maintain a small website about the whole thing:
http://www.welzl.at/research/tools/syn-retransmit/index.html

Cheers,
Michael



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On Thu, Jul 16, 2009 at 2:09 AM, David Malone<David.Malone@nuim.ie> wrote:
>> We have seen SYN-ACK retransmission rates upto a few percentage points t=
o
>> some of our servers.
>
> Note, that I've seen Google reported to have unusually SYN-ACK
> retransmission rates in this paper:
>
> =A0 =A0 =A0 =A0http://www.springerlink.com/content/t033171209815745/
>
> Are Google already doing something unusual with their SYN-ACK
> retransmissions?

I can't access the above link so don't know what the unusual SYN-ACK
retransmission rate is. We are experiementing with using RTT history
on a handful of servers as mentioned before. Otherwise our SYN-ACK
retransmit rate seems pretty much on par with the general pkt retransmit
rate, and can go up to a few percentage in regions like India, China...

Jerry

>
> =A0 =A0 =A0 =A0David.
>

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Subject: Re: [tcpm] Tuning TCP parameters for the 21st century
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On Thu, Jul 16, 2009 at 1:39 AM, Michael Welzl<michawe@ifi.uio.no> wrote:
>> > - but what about retransmitting SYN/ACKs faster?
>> > Since the other side already sent a SYN, it probably
>> > isn't overloaded
>>
>> That presumes the other side already sent a SYN and it's been received,
>> and that the SYN/ACK is what's getting lost/delayed/retransmitted. =A0If
>> not, that won't help.
>
> Right -
>
>
>> First find where things are going wrong empirically ;-)
>
> We've done that. See, for example, fig. 2 (a) of
> http://www.welzl.at/research/publications/networking2009.pdf
> (page 7 of the pdf)
>
> Since our goal in this paper was to document general failures
> of connection setup, we did not strictly split our results
> into the case of lost SYNs or lost SYN/ACKs in it.
> Section 3 (end of page 3 / beginning of page 4) explains
> how lost SYN/ACKs were incorporated in our measurements.
>
>
> This paper is based on data produced by a student. The
> data are more thoroughly documented in his (alas, German)
> bachelor thesis. This document is here:
> http://www.welzl.at/research/tools/syn-retransmit/benjamin_kaser_bak1.pdf
> and the most relevant information is table 2 on page 47.
> (Table 1 is similar, just for SYNs, not SYN/ACKs).
>
> If you look at it, with or without German knowledge
> you will be able to see the connection to table 1 in
> the paper, where we exchanged columns with lines and
> omitted some details - apparently just the crucial
> ones =A0:-( =A0 =A0Anyway, the listed measurements are just
> the same, and hence you can read the details in the
> paper if you're interested in how we obtained our
> data.
>
> Table 2 on page 47 of the bachelor thesis has several
> details about duplicate SYN/ACKs that we've seen,
> e.g. classified by eventually successful and unsuccessful
> connections. As a quick summary, in 3432744 connection
> attempts, we have seen 56586 duplicate SYN/ACKs in total
> (this is a cumulative number, i.e. 4 SYN/ACKs are counted
> as 3). Counting only the occurrence of duplicate SYN/ACKs,
> we saw them for 0,55% of the 3432744 connection attempts.
> In general, the difference between duplicate SYN/ACK
> occurrences and duplicate SYN occurrences is negligible
> in our measurements.
>
> As I just looked at the data again, I noticed a mistake:
> in the paper, we wrote that the problem only happens for
> around 0.5% of the connections, but this is really only
> from the SYN table, not the SYN/ACK table in the bachelor
> thesis. The SYN/ACK table covers the same measurement sets.
> Hence, in total, it seems that we also saw > 1% lost SYNs
> *or* SYN/ACKs, which exactly matches what Jerry Chu said.
>
> BTW I maintain a small website about the whole thing:
> http://www.welzl.at/research/tools/syn-retransmit/index.html

Our overall pkt retransmission rate often goes over 1%. I was
wondering if SYN/SYN-ACK pkts are less likely to be dropped
by some routers due to their smaller size so we collected traces
and computed SYN-ACK retransmissions rate on some servers.
We confirmed it to be consistent with the overall pkt drop rate,
i.e., > 1% often.

FYI,

Jerry

>
> Cheers,
> Michael
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From michawe@ifi.uio.no  Thu Jul 16 23:30:44 2009
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From: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [tcpm] Tuning TCP parameters for the 21st century
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So, after some data have been mentioned, can we
get back to this question:

On Thu, 2009-07-16 at 19:36 -0700, Jerry Chu wrote:
> On Thu, Jul 16, 2009 at 1:39 AM, Michael Welzl<michawe@ifi.uio.no> wrote:
> >> > - but what about retransmitting SYN/ACKs faster?
> >> > Since the other side already sent a SYN, it probably
> >> > isn't overloaded

?

Cheers,
Michael



From gregory.ietf@gmail.com  Wed Jul 22 18:59:31 2009
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Date: Wed, 22 Jul 2009 18:59:18 -0700
To: Eric Rescorla <ekr@rtfm.com>,David McGrew <mcgrew@cisco.com>, Brian Weis <bew@cisco.com>,Tim Polk <tim.polk@nist.gov>
From: "Gregory M. Lebovitz" <gregory.ietf@gmail.com>
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Cc: tcpm@ietf.org
Subject: [tcpm] handling variable length keys with AES-128-CMAC in TCP-AO-crypto draft
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Team,
So, Dave McGrew doesn't like the idea of using the variable length 
master key in the input portion of AES-128-CMAC, and using some fixed 
128 bit blob in the key portion in a key-sizing pre-run, in order to 
get a 128 bit key from a variable length key in order to feed that 
back into the AES-128-CMAC a second time to get the traffic key. 
(i.e. the method currently documented in 
draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01, and used in rfc 4615)

We are trying to handle the case where MK is the *variable length* 
master key, and sometimes MK < 128 bits, and sometimes MK > 128 bits, 
and sometimes VK = 128 bits.

If we constrain the problem to:
  - we need to show people what to do if they get variable length keys
  - we don't want to use any other PRF in the AES-128-CMAC case (i.e. 
mandating the use of yet another algo like SHA256 increases the 
burden on implementations, and, for agility reasons we don't want the 
AES KDF to be built atop SHA1),
  - the same variable length master key entered on two different 
implementations MUST produce the same traffic key for any given connection,

then, could we solve it like this:
  - Normative text:  AES-128-CMAC needs a 128-bit key. Therefore 
administrators SHOULD use a suitable key of exactly 128 bits, as 
represented by a 16 character ASCII string. Implementation UI's 
SHOULD force the administrators to enter a 128-bit key when selecting 
AES-128-CMAC as the KDF. UI's SHOULD reject with an error any key not 
equal in length to 128 bits. If, for whatever reason this is not 
acceptable for certain environments, implementations MAY consider 
accepting keys from the UI with lengths not equal to 128bits and 
using AES-128-CMAC as described in appendix A to create a 128-bit key 
from a non-128-bit string.

- Appendix A (non-Normative):
   - if the UI will allow strings of length not equal to 128 bits, and
   - if you get a string > 128-bit or <128-bit, then
      - if > 128-bit, truncate the string to 128-bits, and use them as the key
      - if < 128-bit, pad the end of the string with zeros until the 
string length equals 128-bit, and use that as the key string.

I realize this isn't great, but I'm not sure how else to solve the 
problem that hasn't already been shot down. If neither the method in 
draft-01 nor the above work for you, then PLEASE send text for 
something better. Remember, we need to get this done asap, and I'd 
like to have a scrubbed proposal for the TCPM WG meeting in Stockholm.

Thanks,

Gregory.

+++++++++++++++++++++++
IETF-related email from
Gregory M. Lebovitz
Juniper Networks
g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m 


From ekr@networkresonance.com  Wed Jul 22 23:32:35 2009
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From: Eric Rescorla <ekr@networkresonance.com>
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Subject: Re: [tcpm] handling variable length keys with AES-128-CMAC in TCP-AO-crypto draft
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At Wed, 22 Jul 2009 18:59:18 -0700,
Gregory M. Lebovitz wrote:
> 
> Team,
> So, Dave McGrew doesn't like the idea of using the variable length 
> master key in the input portion of AES-128-CMAC, and using some fixed 
> 128 bit blob in the key portion in a key-sizing pre-run, in order to 
> get a 128 bit key from a variable length key in order to feed that 
> back into the AES-128-CMAC a second time to get the traffic key. 
> (i.e. the method currently documented in 
> draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01, and used in rfc 4615)
> 
> We are trying to handle the case where MK is the *variable length* 
> master key, and sometimes MK < 128 bits, and sometimes MK > 128 bits, 
> and sometimes VK = 128 bits.
> 
> If we constrain the problem to:
>   - we need to show people what to do if they get variable length keys
>   - we don't want to use any other PRF in the AES-128-CMAC case (i.e. 
> mandating the use of yet another algo like SHA256 increases the 
> burden on implementations, and, for agility reasons we don't want the 
> AES KDF to be built atop SHA1),
>   - the same variable length master key entered on two different 
> implementations MUST produce the same traffic key for any given connection,
> 
> then, could we solve it like this:
>   - Normative text:  AES-128-CMAC needs a 128-bit key. Therefore 
> administrators SHOULD use a suitable key of exactly 128 bits, as 
> represented by a 16 character ASCII string. Implementation UI's 
> SHOULD force the administrators to enter a 128-bit key when selecting 
> AES-128-CMAC as the KDF. UI's SHOULD reject with an error any key not 
> equal in length to 128 bits. If, for whatever reason this is not 
> acceptable for certain environments, implementations MAY consider 
> accepting keys from the UI with lengths not equal to 128bits and 
> using AES-128-CMAC as described in appendix A to create a 128-bit key 
> from a non-128-bit string.
> 
> - Appendix A (non-Normative):
>    - if the UI will allow strings of length not equal to 128 bits, and
>    - if you get a string > 128-bit or <128-bit, then
>       - if > 128-bit, truncate the string to 128-bits, and use them as the key
>       - if < 128-bit, pad the end of the string with zeros until the 
> string length equals 128-bit, and use that as the key string.
> 
> I realize this isn't great, but I'm not sure how else to solve the 
> problem that hasn't already been shot down. If neither the method in 
> draft-01 nor the above work for you, then PLEASE send text for 
> something better. Remember, we need to get this done asap, and I'd 
> like to have a scrubbed proposal for the TCPM WG meeting in Stockholm.

This seems to me to be a bad approach; it creates a different interface
for CMAC and HMAC. I would prefer that we abandon the requirement to use
only AES and use an HMAC-based PRF as a whitening function. I don't
think it matters whether it's SHA-1 or SHA-256--SHA-1 is plenty strong
for this, but SHA-256 seems more popular at this point.

-Ekr


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Subject: [tcpm] SACK-based recovery entry I-D updated (not in std repo yet)
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Hello all,

We've updated draft-jarvinen-tcpm-sack-recovery-entry but missed the 
deadline so the updated version is not available through standard 
repository just yet. Therefore I've put the updated version available at:

http://www.cs.helsinki.fi/group/wiseciti/I-Ds/draft-jarvinen-tcpm-sack-recovery-entry-01a.txt

As promised, we've now included the paraller dupACK based fallback that 
solves small segment sender problem into the algorithm itself and 
doing corresponding removal of its description from the textual side
(the biggest changes from -00 are due to that). Besides that, a 
placeholder for SACK splitting got added. The rest of the changes
are quite minor. Any feedback is appreciated.

-- 
  i.

From ilpo.jarvinen@helsinki.fi  Thu Jul 23 01:06:42 2009
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On Tue, 7 Jul 2009, Eddy, Wesley M. (GRC-MS00)[Verizon] wrote:

> Here's a draft agenda for the TCPM meeting in Stockholm; please let 
> David and I know if we missed anyone that asked for a slot, or if there 
> are other people that want slots ... there should be a little bit of 
> room left given our 2 hour timeslot that was allocated.

If there's still room on the agenda, could you please allocate a slot for
draft-jarvinen-tcpm-sack-recovery-entry (10mins is fine). Thanks.


-- 
  i.

From wesley.m.eddy@nasa.gov  Thu Jul 23 07:54:26 2009
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>-----Original Message-----
>From: Ilpo J=E4rvinen [mailto:ilpo.jarvinen@helsinki.fi]
>Sent: Thursday, July 23, 2009 4:06 AM
>To: Eddy, Wesley M. (GRC-MS00)[Verizon].; David Borman
>Cc: tcpm@ietf.org
>Subject: Re: [tcpm] Draft Agenda for IET5 75 TCPM Meeting
>
>If there's still room on the agenda, could you please allocate a slot
>for
>draft-jarvinen-tcpm-sack-recovery-entry (10mins is fine). Thanks.
>

Ok; I've updated the schedule and placed this at the end.

---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------

From narten@us.ibm.com  Fri Jul 24 12:53:59 2009
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Date: Fri, 24 Jul 2009 15:52:22 -0400
From: Thomas Narten <narten@us.ibm.com>
Subject: [tcpm] Nagle Algorithm and Delayed ACKs
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It has been long known that the Nagle Algorithm and Delayed ACKs
combine in ways that can be problematic. Googling around, I found
Stewart Chesire's writeup
(http://www.stuartcheshire.org/papers/NagleDelayedAck/) as well as an
ancient (!) ID from Greg Minshall
(http://lists.w3.org/Archives/Public/ietf-discuss/1998Dec/0025.html)

The TCP_NODELAY socket option can be used to turn off the Nagle
Algorithm, but applications have to know to enable it -- which they
often don't know, until after they've run into problems and traced
them down to the specific problem.  This can be a long and expensive
process (e.g., involve help desk calls).

What is the current thinking on this problem? Am I right in assuming
that:

 - the Nagle Algorithm is still a SHOULD to implement
 - Delayed ACKs are also a SHOULD to implement?
 - Most everyone provides a TCP_NODELAY type socket option?

Is that all there is to it? Is this considered good enough?

Have there not been in any attempts over the last ten years to update
the above recommendations to make them work together better without an
application needing to set the TCP_NODELAY option, in those cases
where it matters?

Thomas

From Michael.Tuexen@lurchi.franken.de  Fri Jul 24 13:47:39 2009
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From: =?ISO-8859-1?Q?Michael_T=FCxen?= <Michael.Tuexen@lurchi.franken.de>
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Hi Thomas,

interesting you bring this up...

For SCTP we have the same problem and we therefore submitted
draft-tuexen-tsvwg-sctp-sack-immediately-02.txt
which was presented at the last TSVWG session.

However, the result of the discussion was that the issue
is minor and there was not much interest in the WG to
adopt the ID.

Best regards
Michael

On Jul 24, 2009, at 9:52 PM, Thomas Narten wrote:

> It has been long known that the Nagle Algorithm and Delayed ACKs
> combine in ways that can be problematic. Googling around, I found
> Stewart Chesire's writeup
> (http://www.stuartcheshire.org/papers/NagleDelayedAck/) as well as an
> ancient (!) ID from Greg Minshall
> (http://lists.w3.org/Archives/Public/ietf-discuss/1998Dec/0025.html)
>
> The TCP_NODELAY socket option can be used to turn off the Nagle
> Algorithm, but applications have to know to enable it -- which they
> often don't know, until after they've run into problems and traced
> them down to the specific problem.  This can be a long and expensive
> process (e.g., involve help desk calls).
>
> What is the current thinking on this problem? Am I right in assuming
> that:
>
> - the Nagle Algorithm is still a SHOULD to implement
> - Delayed ACKs are also a SHOULD to implement?
> - Most everyone provides a TCP_NODELAY type socket option?
>
> Is that all there is to it? Is this considered good enough?
>
> Have there not been in any attempts over the last ten years to update
> the above recommendations to make them work together better without an
> application needing to set the TCP_NODELAY option, in those cases
> where it matters?
>
> Thomas
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From cabo@tzi.org  Fri Jul 24 22:17:55 2009
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On Jul 24, 2009, at 21:52, Thomas Narten wrote:

> TCP_NODELAY

My view would be that this is a well-documented option in the socket  
interface.
Even though the interaction with delayed ACKs is rarely explained  
correctly, there are lots of "Socket FAQ" and "Tuning Guide" documents  
that explain "Nagling" (ouch) and the TCP_NODELAY socket option (I get  
117000 google hits...).
TCP_NODELAY certainly is standard fare in any reasonable socket API  
course.
People already need to think about the number of write/send operations  
they perform for one message.
(Note that Linux also has TCP_CORK.)

In essence, the kernel needs to know whether the programmer was not  
considering the issue (making it likely there will be lots of silly  
small writes) or whether the issue was considered and the program is  
optimized with respect to the writes.
A socket option defaulting to the first case sounds like the right way  
to communicate that information, which is hard to infer.

What would you want to improve?

Gruesse, Carsten


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Subject: [tcpm] TCP-AO - Working session to close open issues?
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Hey guys,
In the spirit of trying to get to working group last call asap, what 
do you think about having a working session on Sunday evening or 
monday to try to close out some of the open items on TCP-AO BEFORE 
our monday 17:40 TCPM session?

Do any of the following work?
  - Sunday 21:00 - 23:00
  - Monday 15:10 - 17:30

If we can't get together before the TCPM mtg on Monday, how about we 
get a quick bite together on Monday after the meeting, and then spend 
a bit of time together on the document's open issues?

(note: EKR's not here this week.)

Gregory.


Gregory.

+++++++++++++++++++++++
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Juniper Networks
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Subject: Re: [tcpm] possible NAT support for TCP-AO
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inline...

At 01:58 PM 7/13/2009, Joe Touch wrote:
>-----BEGIN PGP SIGNED MESSAGE-----
>Hash: SHA1
>
>Hi, all,
>
>Based on a discussion on a different list with Dan Wing, I'd like to
>revisit thinking about NATs in the context of the current AO,

I'm not against the idea. I see some issues (see more below). But I 
think it could work. My one big concern is that the BGP community has 
been waiting a VERY long time for this update, and I'd hate to drag 
out a last call, or RFC, for this issue which they don't care much 
about. But I think solving it in the long run is important. As a 
compromise between these two competing needs, might we consider this:
  - proceed to WG LC & RFC without the NAT solution.
  - in parallel, work on text for the NAT solution
  - do a rather quick rev of the first RFC to include the NAT 
solution (and the few bugs I'm sure we'll find as we move to implementations)
??

>which uses
>traffic keys derived from the socket pair and ISN pair. We might be able
>to now support NATs as follows, e.g.:
>
>         add the following flags to the MKT:
>
>                 localNAT flag - indicates whether the local IP/port are
>                         zeroed before MAC calculation
>
>                 remoteNAT flag - indicates whether the remote IP/port
>                         are zeroed before MAC calculation

This will be tough, as you won't always know. We could solve this 
later in a KMP system that provides NAT detection, and hands the 
answer down to the TCP stack config (I'm thinking of the way that 
NAT-T for IKE works wrt IPsec/ESP). But it could work.


>         add steps to the incoming/outgoing processing:
>                 zero the corresponding IP/port when the flag indicates,
>                 both on outgoing and incoming MAC calculation

If both directions are behind NATs (a very probably case for 
non-routing protocol usage), then the uniqueness data defining the 
connection is reduced greatly, i.e. only the ISNs remain as unique. 
This affects the entropy of the KDF, where the connection data is 
used as the input to the function, in order to derive the traffic 
keys. If I'm not mistaken, it means that, in the unlikely event that 
the same ISN's are chosen for a different connection 5-tuple, then a 
replay attack could be accomplished.

I can live with that risk trade-off, personally. But I'd want to make 
sure we call it out explicitly the NAT-T section where the mechanism 
is described, and then go into it more completely in the security 
considerations. section.


>That's basically it. A client behind a NAT would have a MKT with
>localNAT true, and a server for that client would need to have a MKT
>with remoteNAT true. This does require careful MKT configuration.
>Although I wouldn't expect both localNAT and remoteNAT to be true, there
>isn't a particular reason it needs to be prohibited.
>
>Here's the security impact:
>
>         - no impact to other connections (AFAICT)
>
>         - SYN/SYN-ACKs limited only as much as the MKT TCP connection
>         ID is, i.e., as a small range or single value rather than
>         wildcard for either address or port
>
>         - connections are reasonably well-protected once established,
>         much like BTNS (due to use of both ISNs in the traffic keys)
>
>         - reduced entropy for the traffic keys from a given MKT,
>         since their input could be limited to the ISN pair in the
>         worst case

I think what I described above fits under this point, if I understood 
Joe's point correctly.

That's my .02, after a very quick read and response,
Gregory.


>I did NOT include this in TCP-AO-05, but it's simple enough to add if
>useful. I'd like to ask that we think about this before Stockholm, and
>discuss it on the list if possible in advance.
>
>Joe
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From touch@ISI.EDU  Sat Jul 25 22:32:18 2009
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Subject: Re: [tcpm] TCP-AO - Working session to close open issues?
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Please see the current doc - there are no open issues I am aware of  
except NAT support, which can be closed during the WG mtg. I can meet  
after the WG only, fwiw.

Joe

Sent from my iPhone

On Jul 26, 2009, at 12:26 AM, "Gregory M. Lebovitz" <gregory.ietf@gmail.com 
 > wrote:

> Hey guys,
> In the spirit of trying to get to working group last call asap, what  
> do you think about having a working session on Sunday evening or  
> monday to try to close out some of the open items on TCP-AO BEFORE  
> our monday 17:40 TCPM session?
>
> Do any of the following work?
> - Sunday 21:00 - 23:00
> - Monday 15:10 - 17:30
>
> If we can't get together before the TCPM mtg on Monday, how about we  
> get a quick bite together on Monday after the meeting, and then  
> spend a bit of time together on the document's open issues?
>
> (note: EKR's not here this week.)
>
> Gregory.
>
>
> Gregory.
>
> +++++++++++++++++++++++
> IETF-related email from
> Gregory M. Lebovitz
> Juniper Networks
> g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m

From touch@ISI.EDU  Sun Jul 26 03:51:42 2009
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Subject: Re: [tcpm] TCP-AO - Working session to close open issues?
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Sent from my iPhone

On Jul 26, 2009, at 7:30 AM, Joe Touch <touch@ISI.EDU> wrote:

> Please see the current doc - there are no open issues I am aware of  
> except NAT support, which can be closed during the WG mtg. I can  
> meet after the WG only, fwiw.

Ps - only until the homegate bar bof. (or whatever the tsv barbof of  
the night is)

>
> Joe
>
> Sent from my iPhone
>
> On Jul 26, 2009, at 12:26 AM, "Gregory M. Lebovitz" <gregory.ietf@gmail.com 
> > wrote:
>
>> Hey guys,
>> In the spirit of trying to get to working group last call asap,  
>> what do you think about having a working session on Sunday evening  
>> or monday to try to close out some of the open items on TCP-AO  
>> BEFORE our monday 17:40 TCPM session?
>>
>> Do any of the following work?
>> - Sunday 21:00 - 23:00
>> - Monday 15:10 - 17:30
>>
>> If we can't get together before the TCPM mtg on Monday, how about  
>> we get a quick bite together on Monday after the meeting, and then  
>> spend a bit of time together on the document's open issues?
>>
>> (note: EKR's not here this week.)
>>
>> Gregory.
>>
>>
>> Gregory.
>>
>> +++++++++++++++++++++++
>> IETF-related email from
>> Gregory M. Lebovitz
>> Juniper Networks
>> g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m

From gregory.ietf@gmail.com  Sun Jul 26 05:07:51 2009
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Subject: Re: [tcpm] TCP-AO - Working session to close open issues?
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At 10:30 PM 7/25/2009, Joe Touch wrote:
>Please see the current doc - there are no open issues I am aware of
>except NAT support, which can be closed during the WG mtg.

Hey Joe.
It was while reading draft-05 that I wrote my email. Here are the 
things I'd like to work through, just to be sure we have closure:
  - step by step through rolling from one key to a new key
  - step by step through the removal of a key
  - step by step through the matching overlap problem
  - clarity on KDF / HMAC language
  - clarify the current crypto issue with variable length MasterKeys 
and AES-128-CMAC

If tonight won't work, when, even if it's after the meeting another day?
Gregory

>I can meet
>after the WG only, fwiw.
>
>Joe
>
>Sent from my iPhone
>
>On Jul 26, 2009, at 12:26 AM, "Gregory M. Lebovitz" 
><gregory.ietf@gmail.com > wrote:
>
>>Hey guys,
>>In the spirit of trying to get to working group last call asap, what
>>do you think about having a working session on Sunday evening or
>>monday to try to close out some of the open items on TCP-AO BEFORE
>>our monday 17:40 TCPM session?
>>
>>Do any of the following work?
>>- Sunday 21:00 - 23:00
>>- Monday 15:10 - 17:30
>>
>>If we can't get together before the TCPM mtg on Monday, how about we
>>get a quick bite together on Monday after the meeting, and then
>>spend a bit of time together on the document's open issues?
>>
>>(note: EKR's not here this week.)
>>
>>Gregory.
>>
>>
>>Gregory.
>>
>>+++++++++++++++++++++++
>>IETF-related email from
>>Gregory M. Lebovitz
>>Juniper Networks
>>g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m


From gregory.ietf@gmail.com  Sun Jul 26 12:43:50 2009
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To: "Polk, William T." <william.polk@nist.gov>, "mcgrew@cisco.com" <mcgrew@cisco.com>
From: "Gregory M. Lebovitz" <gregory.ietf@gmail.com>
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Subject: Re: [tcpm] handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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Tim,
This is helpful, thanks. We will proceed toward WG LC with the 
mechanism as represented in draft v-01.

If later, after further discussions within the cryptographic 
community, it is decided that the randomness extraction mechanism to 
go from a variable-length user-given string to a 128-bit key would be 
better accomplished some other way, then we can always do a quick 
-bis on the RFC and update the mechanism. That's easy enough for a 
small RFC like this.

Gregory.

At 08:40 AM 7/26/2009, Polk, William T. wrote:
>Gregory,
>
>As requested, Pasi and I took a look at how 
>draft-lebovitz-ietf-tcpm-tcp-ao-crypto
>handles the AES-CMAC case when the master key is not 128 bits.
>
>The method currently used (using AES-CMAC with a known fixed key as a
>"randomness extractor") is basically the same as used in IKEv2 (RFC
>4615 and 4434), except IKEv2 uses all-zeroes key (but the specific
>value of the known fixed key shouldn't matter much -- just pick one).
>
>Using the same method in TCP-AO would be acceptable us, and seems
>strongly preferable to requiring network admins to configure exactly
>128 bit keys.
>
>Other methods of getting from "what the admin configures" to 128 bits
>could work, too. Since the output of this step is *only* used as the
>key for further PRF calculations (and for *nothing* else), the
>requirements for it are very limited. If all keys were <= 128 bits,
>even something trivial such as padding with zeroes would work -- so
>this step doesn't even have to be one-way.  Since we need to handle
>also keys >128 bits, simple padding isn't enough, of course. If the WG
>wants AES-only method, the one that's in the draft now seems OK to us.
>
>I hope this was helpful.
>
>Tim Polk
>Pasi Eronen


From ekr@networkresonance.com  Sun Jul 26 16:22:50 2009
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Date: Sun, 26 Jul 2009 16:26:24 -0700
From: Eric Rescorla <ekr@networkresonance.com>
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Subject: Re: [tcpm] handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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At Sun, 26 Jul 2009 12:43:37 -0700,
Gregory M. Lebovitz wrote:
> 
> Tim,
> This is helpful, thanks. We will proceed toward WG LC with the 
> mechanism as represented in draft v-01.

I'll be sending my comments on v-01 shortly, but I object to
the magic number in the current mechanism. If we're going to
use a magic key, it should be zero or some other anchor point.
The use of a random number pulled out of the air is highly
inappropriate in cryptographic systems [I refer you to the
contrroversy over the DES S-boxes].

-Ekr


 
> If later, after further discussions within the cryptographic 
> community, it is decided that the randomness extraction mechanism to 
> go from a variable-length user-given string to a 128-bit key would be 
> better accomplished some other way, then we can always do a quick 
> -bis on the RFC and update the mechanism. That's easy enough for a 
> small RFC like this.


> Gregory.
> 
> At 08:40 AM 7/26/2009, Polk, William T. wrote:
> >Gregory,
> >
> >As requested, Pasi and I took a look at how 
> >draft-lebovitz-ietf-tcpm-tcp-ao-crypto
> >handles the AES-CMAC case when the master key is not 128 bits.
> >
> >The method currently used (using AES-CMAC with a known fixed key as a
> >"randomness extractor") is basically the same as used in IKEv2 (RFC
> >4615 and 4434), except IKEv2 uses all-zeroes key (but the specific
> >value of the known fixed key shouldn't matter much -- just pick one).
> >
> >Using the same method in TCP-AO would be acceptable us, and seems
> >strongly preferable to requiring network admins to configure exactly
> >128 bit keys.
> >
> >Other methods of getting from "what the admin configures" to 128 bits
> >could work, too. Since the output of this step is *only* used as the
> >key for further PRF calculations (and for *nothing* else), the
> >requirements for it are very limited. If all keys were <= 128 bits,
> >even something trivial such as padding with zeroes would work -- so
> >this step doesn't even have to be one-way.  Since we need to handle
> >also keys >128 bits, simple padding isn't enough, of course. If the WG
> >wants AES-only method, the one that's in the draft now seems OK to us.
> >
> >I hope this was helpful.
> >
> >Tim Polk
> >Pasi Eronen
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From ekr@networkresonance.com  Sun Jul 26 16:23:38 2009
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Subject: [tcpm] Review of draft-ietf-tcpm-tcp-auth-opt-05
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$Id: draft-ietf-tcpm-tcp-auth-opt-05-rev-real.txt,v 1.1 2009/07/26 22:18:40 ekr Exp $

This draft seems to resolve all my major technical issues.
The comments below are primarily editorial.


TECHNICAL
   The relationship between MKTs and traffic keys is shown in Figure
   Figure 3. Traffic keys are indicated with a "*". Note that every MKT
   derives all four traffic keys, regardless of whether all four are
   needed for a given connection. Section 7.2 provides further details
   on how traffic keys are derived.

This seems like an unwarranted requirement. The keys are generated
independently, so why make implementations gnerate keys they don't
need?



S 8.2.
   When a segment is received, the following algorithm (written in C)
   computes the SNE used in the MAC; an equivalent algorithm can be
   applied to the "SND" side:

I know it seems like I'm nitpicking about this, but this algorithm is 
*not* valid C: # is not a comment character and C identifiers may not
contain .s in the name. I suppose for the latter you could be arguing that these
are structs, but then where's the defn of those? This seems like
perfectly legitimate pseudocode, but if you're going to claim it's C, 
it really needs to be valid C. I would remove the claim.

S 9.5.
As I read this description, if an implementation receives a TCP segment
with TCP-AO but it doesn't match any existing MKT, it should silently
accept it. I don't agree with this. It should discard it. It's clearly
not valid.

           i. If lengths differ, silently discard the segment. Log
               and/or signal the event as indicated in Section 9.3.

I observe that this logging suggestion has the possibility to be
a DoS in and of itself. If we're going to recommend logging,
we need to recommend log throttling.








EDITORIAL
Abstract
   protects against replays even for long-lived TCP connections, and

s/protects against replays/detects replayed segments/

   set of MACs with minimal other system and operational changes. TCP-AO
   uses its own option identifier, even though used mutually exclusive
   of TCP MD5 on a given TCP connection. TCP-AO supports IPv6, and is

I would just remove "even though..." This wording is clumsy and
it probably isn't necessary at the level of the abstract


S 2.
   Section 9.6). This document obsoletes the TCP MD5 option with a more
   general TCP Authentication Option (TCP-AO), to support the use of

s/obsoletes/replaces/


   This document is not intended to replace the use of the IPsec suite
   (IPsec and IKE) to protect TCP connections [RFC4301][RFC4306]. In
   fact, we recommend the use of IPsec and IKE, especially where IKE's
   level of existing support for parameter negotiation, session key
   negotiation, or rekeying are desired. TCP-AO is intended for use only
   where the IPsec suite would not be feasible, e.g., as has been
   suggested is the case to support some routing protocols, or in cases
   where keys need to be tightly coordinated with individual transport
   sessions [Be07].

I would remove this paragraph entirely, but if you with to retain
it, I don't think it does that great a job of explaining the relationship
here. Fundamentally people have complained that IPSec is too 
heavyweight. I would rewrite this as:

   This document is not intended to replace the use of the IPsec suite
   (IPsec and IKE) to protect TCP connections [RFC4301][RFC4306]. Rather,
   it is intended for the limited goal of providing traffic integrity
   and denial of service (DoS) protection for TCP connections in
   settings where a simpler solution than IPsec/IKE is desired.


S 2.1
   o  TCP-AO ensures per-connection traffic keys as unique as the TCP
      connection itself, using TCP's initial sequence numbers (ISNs) for

s/ensures/provides/
s/as unique/which are as unique/


   o  TCP-AO forces a change of computed MACs when a connection
      restarts, even when reusing a TCP socket pair (IP addresses and
      port numbers) [Be07].

I think I know what this is supposed to mean, but I think it's
getting off into the weeds. However, if you want to say it,
try:

   TCP-AO does not allow packets from one TCP connection to
   be replayed on another TCP connection, even if the
   connection parameters (IP addess and port numbers)
   are the same.


S 4.2.
   The new TCP-AO option provides a superset of the capabilities of TCP
   MD5, and is minimal in the spirit of SP4 [SDNS88]. TCP-AO uses a new
   Kind field, and similar Length field to TCP MD5, a KeyID field, and a
   NextKeyID field as shown in Figure 2.

Isn't a similar length field a requirement of the TCP options layout.
I would also observe that SP4 is now over 20 years old and was never
widely deployed even then, so references to the spirit of SP4 feel
anachronistic.

      Values of 4 and other small values larger than 4 (e.g., indicating
      MACs of very short length) are of dubious utility but are not
      specifically prohibited.

It's probably worth stating explicitly here that the MAC length is
dictated by the MKT, not the option.



S 5.

I would rewrite this first paragraph as follows:

TCP-AO makes use of two sets of keys:

- Master Key Tuples (MKTs) are managed either manually or by a
  (future) key management protocols. They comprise a master key, the
  set of cryptographic algorithms to be used with that key, and the
  TCP connection identifiers to which the MKT applies. MKTs are never
  used directly to protect data but rather are used to derive traffic
  keys.  Multiple MKTs may apply to a given connection, and are
  distinguished by a key identifier (key-id).

- Traffic keys are used to compute the MAC for each TCP segment.  Each
  TCP connection has a fresh set(s) of traffic keys which are
  automatically derived from the appropriate MKT(s) for that
  connection. The traffic key derivation function includes the TCP
  Initial Sequence numbers, resulting in a high probability that
  traffic keys will be unique, even if the IP address and TCP port are
  reused.


S 5.1.
      >> MKT IDs MUST support any value, 0-255 inclusive. There are no
      reserved ID values.

MKTs don't support anything. TCP-AO implementations do. This construct
appears in a number of places in the document and should be replaced
with "Implementations MUST suupport"


      Implementations are advised to keep master key values in a
      private, protected area of memory or other storage.

I'm curious if anyone has a concrete notion of what this means. As I
understand the situation, Cisco routers (for example) are a single
monolothic process. How would you do this for a Cisco?


   >> The IDs of MKTs MUST NOT overlap where their TCP connection
   identifiers overlap.

IDs aren't ranges. I would rewrite as:

Any two MKTs with overlapping TCP connection identifiers [could we
use "specifications" here more generally] MUST have distinct
MKT IDs.


S 5.2.
MKTs don't derive (see above).

I'm not sure this taxonomy and the use of send/receive is that
useful here, since each side has its own send and receive.
It gets doubly confusing when you say

   Note that the keys are directional. A given connection typically uses
   only three of these keys, because only one of the SYN keys is
   typically used. All four are used only when a connection goes through
   'simultaneous open' [RFC793].

But as a practical matter, if Alice does an active open (connect) to Bob,
she will generate:

   o  Send_SYN_traffic_key

   o  Send_other_traffic_key

   o  Receive_other_traffic_key


And Bob will generate:

   o  Receive_SYN_traffic_key

   o  Send_other_traffic_key

   o  Receive_other_traffic_key

But Bob's Receive keys are Alice's send keys and vice versa.

So, I think for clarity, this should be rewritten not to talk
about the connection but about the implementation's view.
E.g.,

   Each side of the connection needs up to four traffic keys:

   o  Send_SYN_traffic_key

   o  Receive_SYN_traffic_key

   o  Send_other_traffic_key

   o  Receive_other_traffic_key
   
   The only time when the *_SYN keys are used is when sending
   a SYN before having seen any packets from the other side.
   
   A given side typically uses only three of these keys, because only
   one of the SYN keys is typically used: either that node will be
   doing an active open and will need the Send_SYN_traffic key or will
   be doing a passive open and will need the Send_SYN_traffic key. All
   four are used only when a connection goes through 'simultaneous
   open' [RFC793].



S 5.3.
This whole discussion of how many MKTs can match seems confusing
and self-contradictory.

My understanding of the rules is as follows:

- Any outgoing packet, which has not yet been processed by TCP-AO,
  may match any number of MKTs, but each MKT must have a distinct
  key-id. Once that packet has been processed by AO, it must
  match only one MKT, determined by key-id.

- Any incoming packet must match either one or zero MKTs.

Is there any disagreement that this is correct? If so, I recommend
rewriting the text to contain more or less just this.



S 6.
Current_Key and Next_Key seem confusing here because we're punning
on the use of key.

- Conenctions must store the current traffic keys (or enough information
  to regenerate them). They must also store the key-id needed to indicate
  those traffic keys.

- The SNEs.

There is no need to *store* Next_Key. It can be regenerated every time.
What's required is simply that it appear on the wire. I don't see any
need to store the MKTs with the connection either. They're "associated"
in that they match, I suppose...


S 7.1.

   o  MAC_truncation - the number of bytes to truncate the output of the 
      MAC to. This is indicated by the MAC algorithm, as specified in 
      [ao-crypto]. 

I don't think it makes sense to talk about truncation here.
The MACs produce a value that is crammed into the packet.
If there's a separate spec that describes the MACs, it's not
AO's business whether those MACs were produced by a truncation process.



S 7.2.
   o  Output_length - The desired output length of the KDF, i.e., the
      length to which the KDF's output will be truncated. In TCP-AO, the
      output_length is the PRF_truncation value of the MKT. This is
      specified by the KDF as described in [ao-crypto].

Everything from i.e., seems (1) superfluous and (2) isnufficiently
general. Who says the PRF is truncated?


   The data used as input to the KDF combines TCP socket pair with the
   endpoint initial sequence numbers (ISNs) of a connection. This

s/TCP socket/the TCP socket/


   Traffic keys are directional, so "source" and "destaination" are

s/destaination/destination/


S 7.3.
   of the TCP-AO option), or for coordinating rekeying during a
   connection. We assume out-of-band mechanisms for MKT establishment,

It does now...


   We encourage users of TCP-AO to apply known techniques for generating
   appropriate MKTs, including the use of reasonable master key lengths,
   limited traffic key sharing, and limiting the duration of MKT use
   [RFC3562]. This also includes the use of per-connection nonces, as
   suggested in Section 7.2.

This seems like old text. We don't encoruage people to use per-connection
nonces, it's part of the protocol. I don't agree that a recommendation
to limit lifetime is really that sensible. I would simply remove this entirely.


S 9.5.
                     a. Optionally, set a timer on the MKT indicated by
                       the current_key and segment socket pair to
                       ensure that the MKT cannot be deleted for 2*MSL.
                       If a timer has already been set, reset the
                       timer.

                       This timer is advisory only. Removing MKTs with
                       unexpired timers can result in a user-level
                       warning, but is not prohibited. Implementation
                       of timers is not required.

                     b. Set Current_key to the NextKeyID MKT.

I would reverse the order of these, since (a) is required and (b) is
optional.


S 10.
Was TCP-MD5 Mandatory to Implement? If not, this shouldn't be either.
Even if so, this seems like kind of an unwarranted requirement.

-Ekr

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$Id: draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01-rev.txt,v 1.1 2009/07/26 22:40:47 ekr Exp $

TECHNICAL
S 3.1.
This mixing of the KDF interface and the NIST 800-108 interface
is extraordinarily confusing.


To recap:
The input to the KDF requires:

- Master Key
- Context (i.e., the conn-data)
- Output length

Now, as it happens, you're constructing your KDFs a la SP800-108,
so you then use the relevant PRF with an input block constructed
as:

           ( i || Label || Context || Output_Length)

But that is not a meaningful part of the interface to the KDF
as far as AO is concerned. If you were (for instance) to define
a new TLS PRF based KDF, you would not use i internally because
the TLS PRF generates an arbitrary length string. Again, it's
important to distinguish between the TCP-AO requirements and
what this document happens to do. 


S 3.1.3.
Wait, I don't remember agreeing to this. I'm happy to use these
as labels in soem IANA registry, byt people's UI is not our
business.


EDITORIAL
S 2.2.

   The security issues driving the migration from SHA-1 to SHA-256 for
   digital signatures [HMAC-ATTACK] [HMAC-ATTACK]do not immediately
   render SHA-1 weak for this application of SHA-1 in HMAC mode.  The
   security strength of SHA-1 HMACs should be sufficient for the
   foreseeable future, especially given that the tags are truncated to
   96 bits.  However, while it's clear that the attacks aren't practical
   on SHA-1, these types of analysis are mounting and could potentially
   pose a concern for HMAC forgery if they were significantly improved,
   over time.  In anticipation of SHA-1's growing less dependable over
   time, but given its wide deployment and current strength, it is a
   "MUST" for TCP-AO today.  AES-128 CMAC is considered to be far
   stronger algorithm, but may not yet have very wide implementation.
   It is also a "MUST" to implement, in order to drive vendors toward
   its use.

This is simply not true. It's not at all clear that AES-128-CMAC is
stronger than HMAC-SHA1, and I don't believe you could get anything
like a consensus of cryptographers to say that it was. The
MUST-MUST was taken on the basis of a WG hum. I'm fine with that,
but this justification needs to be removed.




S 2.3.
       (1)  Key Derivation Functions (KDFs) which name a pseudorandom
            function (PRF) and use a Master_Key and some connection-
            specific Input with that PRF to produce Conn_Keys, the keys
            suitable for authenticating and integrity checking
            individual TCP segments.

This is confusing. KDFs may or may not be constructed with PRFs.
However, THESE KDFs are so constructed. If you're giong to have
a document which applies to future KDFs, don't say PRF here.



S 3.1.
                      An octet of all zeros.  An optional data field
                      used to indicate a separation of different
                      variable length data fields.

This is just dangling.


   When invoked, a KDF runs a certain PRF, using the Master_Key as the
   seed, and Input as the message input and produces a result of
   Output_Length bits.  This result may then be used as a cryptographic
   key for any algorithm which takes an Output_Length length key as its
   seed.  A KDF MAY specify a maximum Output_Length parameter.

Again, misplaced concreteness.


Woah, I think my numbered list cut and paste here seems needs some
major re-editing. It was intended for the mailing list, not for
direct import into the document. 


S 3.1.1.

It's a mistake to talk about "Output Length" as a property of the KDF.
In principle, this KDF could generate any output length of your
choice by using i=0, i=1, etc.. Why are you nailing it down?


S 3.1.2.
I strongly object to the use of this magic key string here. 



S 3.2.
This whole Conn_Key(KDF) thing is confusing. What's it doing?




From ekr@networkresonance.com  Sun Jul 26 17:58:30 2009
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Greg and I spent a while discussing the construction of the
KDF in draft-lebovitz-ietf-tcpm-ao-crypto-01, and I think 
there's some confusion about what the right interfaces are.

This document does two things:

- Defines the interface and requirements for all KDFs
- Provides two specific KDFs.


INTERFACE REQUIREMENTS
As far as the first job goes, the basic interface you want for a KDF is:

     KDF(Master_Key, Context, Output_Length)


Where Output_Length describes the size of the key you're deriving with
the KDF. A KDF of this sort can be used in a mix-and-match fashion
with any target cryptographic algorithm (i.e., can generate keys for
any algorithm) because it can generate an arbitrary length key.  For
instance, you could use an HMAC-SHA based KDF to generate keys for AES
or RC4. In fact, this is precisely what both TLS and IPsec do because
they need to generate keys for both a MAC algorithm and an encryption
algorithm. 


So, KDFs for TCP-AO need to meet the above interface, with:

- Master_Key      == the key from the MKT
- Context         == the data block
- Output_length   == the right size key for the target integrity
  		     algorithm.


Aside from the security requirements for a KDF, this is all you need
to say in terms of requirements on future KDFs.



SPECIFIC KDFS
With that in mind, we now need to define concrete KDFs that meet the
above interface spec. For concreteness, let's focus on the
HMAC-SHA-1-based KDF.

This KDF needs to match the above interface but we're otherwise
unconstrained. However, there seems to be a desire to have a
KDF that matches the (IMHO over-engineered) SP800-108 definitions, and this
is quite possible within this framework. We simply need to arrange for
the correct input values for the PRF that is the core of the KDF.

The pseudocode for this looks like:

    hmac_sha1_kdf(MK, context, output_length) {
      needed = output_length
      i = 0
      result = ""
    
      while(needed) {
        blk = i + "TCP-AO" + context + output_length
     
        result += HMAC-SHA1(MK, blk)
        needed -= 160
      } 
      
      return result
    }

Note that if you generate a 160-bit key, this is computationally
compatible with what's defined in 3.1.1. We just hide the Input
construction under the covers of the KDF. This means we could do
a non SP800-108 KDF, such as the TLS PRF, simply by saying

TLS_PRF_KDF(MK, context, output_length) = TLS_PRF(MK, context)[0..output_length-1]


Is this clear to people? Are there objections to this overall strategy?
If not, I can provide text that implements it.

-Ekr



































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inline...

At 04:46 PM 7/26/2009, Eric Rescorla wrote:
>$Id: draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01-rev.txt,v 1.1 
>2009/07/26 22:40:47 ekr Exp $
>
>TECHNICAL
>S 3.1.
>This mixing of the KDF interface and the NIST 800-108 interface
>is extraordinarily confusing.

I'm not sure it's "mixing". Isn't one is a subset of the other? The 
KDF interface is an interface. The NIST thing is the actual 
concatenated construct which fulfills the Context.



>To recap:
>The input to the KDF requires:
>
>- Master Key
>- Context (i.e., the conn-data)

it's not just the Data_Block (updated lingo, per auth-opt-05). In 
order to follow SP800-180, the context is a few things concatenated, 
and Data_Block is one of the things.

>- Output length
>
>Now, as it happens, you're constructing your KDFs a la SP800-108,
>so you then use the relevant PRF with an input block constructed
>as:
>
>            ( i || Label || Context || Output_Length)
>
>But that is not a meaningful part of the interface to the KDF
>as far as AO is concerned. If you were (for instance) to define
>a new TLS PRF based KDF, you would not use i internally because
>the TLS PRF generates an arbitrary length string. Again, it's
>important to distinguish between the TCP-AO requirements and
>what this document happens to do.

I think this document is NOT just the generic interface - that's 
covered in auth-opt-05. This document reminds us of the generic 
interface, and then specifically defines how to do the SHA1 and 
AES-128 KDFs themselves. That's why we made it specific. Maybe I'm 
not understanding you?



>S 3.1.3.
>Wait, I don't remember agreeing to this. I'm happy to use these
>as labels in soem IANA registry, byt people's UI is not our
>business.

This is the years of IKE interop work which motivates this. 
Clarifying the UI representations to make it very easy on operators 
is a good thing. Others opinions on this UI guidance?



>EDITORIAL
>S 2.2.
>
>    The security issues driving the migration from SHA-1 to SHA-256 for
>    digital signatures [HMAC-ATTACK] [HMAC-ATTACK]do not immediately
>    render SHA-1 weak for this application of SHA-1 in HMAC mode.  The
>    security strength of SHA-1 HMACs should be sufficient for the
>    foreseeable future, especially given that the tags are truncated to
>    96 bits.  However, while it's clear that the attacks aren't practical
>    on SHA-1, these types of analysis are mounting and could potentially
>    pose a concern for HMAC forgery if they were significantly improved,
>    over time.  In anticipation of SHA-1's growing less dependable over
>    time, but given its wide deployment and current strength, it is a
>    "MUST" for TCP-AO today.  AES-128 CMAC is considered to be far
>    stronger algorithm, but may not yet have very wide implementation.
>    It is also a "MUST" to implement, in order to drive vendors toward
>    its use.
>
>This is simply not true. It's not at all clear that AES-128-CMAC is
>stronger than HMAC-SHA1, and I don't believe you could get anything
>like a consensus of cryptographers to say that it was. The
>MUST-MUST was taken on the basis of a WG hum. I'm fine with that,
>but this justification needs to be removed.

Since this is others' text, I'll let those others respond.





>S 2.3.
>        (1)  Key Derivation Functions (KDFs) which name a pseudorandom
>             function (PRF) and use a Master_Key and some connection-
>             specific Input with that PRF to produce Conn_Keys, the keys
>             suitable for authenticating and integrity checking
>             individual TCP segments.
>
>This is confusing. KDFs may or may not be constructed with PRFs.
>However, THESE KDFs are so constructed. If you're giong to have
>a document which applies to future KDFs, don't say PRF here.

it does say "which name", not "MUST name". Doesn't this cover your point?




>S 3.1.
>                       An octet of all zeros.  An optional data field
>                       used to indicate a separation of different
>                       variable length data fields.
>
>This is just dangling.

Done.



>    When invoked, a KDF runs a certain PRF, using the Master_Key as the
>    seed, and Input as the message input and produces a result of
>    Output_Length bits.  This result may then be used as a cryptographic
>    key for any algorithm which takes an Output_Length length key as its
>    seed.  A KDF MAY specify a maximum Output_Length parameter.
>
>Again, misplaced concreteness.
>
>
>Woah, I think my numbered list cut and paste here seems needs some
>major re-editing. It was intended for the mailing list, not for
>direct import into the document.

send text.



>S 3.1.1.
>
>It's a mistake to talk about "Output Length" as a property of the KDF.
>In principle, this KDF could generate any output length of your
>choice by using i=0, i=1, etc.. Why are you nailing it down?
>
>
>S 3.1.2.
>I strongly object to the use of this magic key string here.

discussed with Tim Polk. done. We're back to 0^128 as the key in -02.




>S 3.2.
>This whole Conn_Key(KDF) thing is confusing. What's it doing?

Honestly, I thought that was the notation you requested upon review 
of -00... I'll ping you offline.

I'm posting -02 now, because there was naming I needed to update to 
stay in sync with auth-opt-05 anyway.

Thx for the review,
Gregory.




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


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Subject: Re: [tcpm] handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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ack. change made in crypto-02.
(I'm feeling like a ping-pong ball here -- take it out... no, put it 
in... no, out... no, in... -- but such is life for an I-D editor ;-)
Gregory.

At 11:40 PM 7/26/2009, Polk, William T. wrote:
>Ekr,
>
>Your objections on the magic number are well taken.  Since zeroes would
>be a holdover from 4615 and 4434, that value won't introduce any new
>issues.  I think that would be the best course of action.
>
>Tim
>
>________________________________________
>From: Eric Rescorla [ekr@networkresonance.com]
>Sent: Sunday, July 26, 2009 7:26 PM
>To: Gregory M. Lebovitz
>Cc: Polk, William T.; mcgrew@cisco.com; housley@vigilsec.com; 
>tcpm@ietf.org; Pasi.Eronen@nokia.com
>Subject: Re: [tcpm] handling variable length keys in 
>draft-lebovitz-ietf-tcpm-tcp-ao-crypto
>
>At Sun, 26 Jul 2009 12:43:37 -0700,
>Gregory M. Lebovitz wrote:
> >
> > Tim,
> > This is helpful, thanks. We will proceed toward WG LC with the
> > mechanism as represented in draft v-01.
>
>I'll be sending my comments on v-01 shortly, but I object to
>the magic number in the current mechanism. If we're going to
>use a magic key, it should be zero or some other anchor point.
>The use of a random number pulled out of the air is highly
>inappropriate in cryptographic systems [I refer you to the
>contrroversy over the DES S-boxes].
>
>-Ekr
>
>
>
> > If later, after further discussions within the cryptographic
> > community, it is decided that the randomness extraction mechanism to
> > go from a variable-length user-given string to a 128-bit key would be
> > better accomplished some other way, then we can always do a quick
> > -bis on the RFC and update the mechanism. That's easy enough for a
> > small RFC like this.
>
>
> > Gregory.
> >
> > At 08:40 AM 7/26/2009, Polk, William T. wrote:
> > >Gregory,
> > >
> > >As requested, Pasi and I took a look at how
> > >draft-lebovitz-ietf-tcpm-tcp-ao-crypto
> > >handles the AES-CMAC case when the master key is not 128 bits.
> > >
> > >The method currently used (using AES-CMAC with a known fixed key as a
> > >"randomness extractor") is basically the same as used in IKEv2 (RFC
> > >4615 and 4434), except IKEv2 uses all-zeroes key (but the specific
> > >value of the known fixed key shouldn't matter much -- just pick one).
> > >
> > >Using the same method in TCP-AO would be acceptable us, and seems
> > >strongly preferable to requiring network admins to configure exactly
> > >128 bit keys.
> > >
> > >Other methods of getting from "what the admin configures" to 128 bits
> > >could work, too. Since the output of this step is *only* used as the
> > >key for further PRF calculations (and for *nothing* else), the
> > >requirements for it are very limited. If all keys were <= 128 bits,
> > >even something trivial such as padding with zeroes would work -- so
> > >this step doesn't even have to be one-way.  Since we need to handle
> > >also keys >128 bits, simple padding isn't enough, of course. If the WG
> > >wants AES-only method, the one that's in the draft now seems OK to us.
> > >
> > >I hope this was helpful.
> > >
> > >Tim Polk
> > >Pasi Eronen
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm


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TCPM,
I did a quick turn last night to clean up the ao-crypto draft before 
the WG today. Here is the changes (pasted from Sect 4, Change 
History, of the I-D:
  lebovitz...-02 - 3rd submission

    o  cleaned up explanation in 3.1.2
    o  in 3.1.2, changed the key extractor mechanism back, from using an
       alphanumeric string for the fixed key C to use 0^128 as the key
       (as it was in -00) (Polk & Ekr)
    o  deleted cut-and-paste error text from sect 3.1 between "label" and
       "context"
    o  changed "conn_key" to "traffic_key" throughout, to follow auth-
       opt-05
    o  changed "tsad" to "mkt" throughout, to follow auth-opt-05
    o  changed "conn_block" to "data_block" throughout, to follow auth-
       opt-05

See you later today,
Gregory.



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>From: IETF I-D Submission Tool <idsubmission@ietf.org>
>To: gregory.ietf@gmail.com
>Subject: New Version Notification for
>          draft-lebovitz-ietf-tcpm-tcp-ao-crypto-02
>Date: Mon, 27 Jul 2009 01:44:11 -0700 (PDT)
>
>
>A new version of I-D, draft-lebovitz-ietf-tcpm-tcp-ao-crypto-02.txt 
>has been successfuly submitted by Gregory Lebovitz and posted to the 
>IETF repository.
>
>Filename:       draft-lebovitz-ietf-tcpm-tcp-ao-crypto
>Revision:       02
>Title:          Cryptographic Algorithms, Use, & Implementation 
>Requirments for TCP Authentication Option
>Creation_date:  2009-07-27
>WG ID:          Independent Submission
>Number_of_pages: 16
>
>Abstract:
>The TCP Authentication Option, TCP-AO, relies on security algorithms
>to provide authentication between two end-points.  There are many
>such algorithms available, and two TCP-AO systems cannot interoperate
>unless they are using the same algorithm(s).  This document specifies
>the algorithms and attributes that can be used in TCP-AO's current
>manual keying mechanism.
> 
>
>
>
>The IETF Secretariat.

+++++++++++++++++++++++
IETF-related email from
Gregory M. Lebovitz
Juniper Networks
g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m 


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

Hi, Eric,

Agreed on the editorial feedback, and thanks. Notes on the technical
issues below - I think we agree on most.

Joe

Eric Rescorla wrote:
> $Id: draft-ietf-tcpm-tcp-auth-opt-05-rev-real.txt,v 1.1 2009/07/26 22:18:40 ekr Exp $
> 
> This draft seems to resolve all my major technical issues.
> The comments below are primarily editorial.
> 
> 
> TECHNICAL
>    The relationship between MKTs and traffic keys is shown in Figure
>    Figure 3. Traffic keys are indicated with a "*". Note that every MKT
>    derives all four traffic keys, regardless of whether all four are
>    needed for a given connection. Section 7.2 provides further details
>    on how traffic keys are derived.
> 
> This seems like an unwarranted requirement. The keys are generated
> independently, so why make implementations gnerate keys they don't
> need?

Agreed - I was trying to say that every MKT *can* derive any of the four
traffic keys, not that all four *are* always derived.

...
> S 9.5.
> As I read this description, if an implementation receives a TCP segment
> with TCP-AO but it doesn't match any existing MKT, it should silently
> accept it. I don't agree with this. It should discard it. It's clearly
> not valid.

Agreed. The original intent was to allow communication with an endpoint
that didn't have an MKT, but that clearly won't work because MKTs cover
both directions, and thus the ACKs won't be received anyway.

>            i. If lengths differ, silently discard the segment. Log
>                and/or signal the event as indicated in Section 9.3.
> 
> I observe that this logging suggestion has the possibility to be
> a DoS in and of itself. If we're going to recommend logging,
> we need to recommend log throttling.

I thought we had; I'll check and add or add a reminder.

> EDITORIAL
...
> S 2.
>    Section 9.6). This document obsoletes the TCP MD5 option with a more
>    general TCP Authentication Option (TCP-AO), to support the use of
> 
> s/obsoletes/replaces/

I thought we needed to use the term "obsoletes" somewhere. Maybe this is
the right sentence to do so, maybe not.

...
>       Values of 4 and other small values larger than 4 (e.g., indicating
>       MACs of very short length) are of dubious utility but are not
>       specifically prohibited.
> 
> It's probably worth stating explicitly here that the MAC length is
> dictated by the MKT, not the option.

The length of the MAC field is dictated by the option. If the option
says the MAC is 70 bytes, and the MKT says it should be 72, then the
packet would fail authentication.

I'll correct to call this "indicating MAC fields of very short length"

...
>       Implementations are advised to keep master key values in a
>       private, protected area of memory or other storage.
> 
> I'm curious if anyone has a concrete notion of what this means. As I
> understand the situation, Cisco routers (for example) are a single
> monolothic process. How would you do this for a Cisco?

Keep the keys in an area that non-operator processes can't get to
without explicit permission or override.

...
...
> S 5.3.
> This whole discussion of how many MKTs can match seems confusing
> and self-contradictory.
> 
> My understanding of the rules is as follows:
> 
> - Any outgoing packet, which has not yet been processed by TCP-AO,
>   may match any number of MKTs, but each MKT must have a distinct
>   key-id. Once that packet has been processed by AO, it must
>   match only one MKT, determined by key-id.

If it matches multiple MKTs, the question is whether there is a
preferred one or not. I had thought there was, but if not then this text
can be used.

If it matches no MKTs, TCP-AO is not used.

> - Any incoming packet must match either one or zero MKTs.

If it matches zero MKTs and has TCP-AO option, then it must be dropped.

If it matches one or more MKTs and has no TCP-AO option, then it must be
dropped.

> Is there any disagreement that this is correct? If so, I recommend
> rewriting the text to contain more or less just this.

As per above, the full set of cases needs to be discussed here. Granted
the text can be revised to avoid "preferred" keys as per above.

...
> S 6.
> Current_Key and Next_Key seem confusing here because we're punning
> on the use of key.

Agreed - should change that to "current_MKT" and "next_MKT".

> - Conenctions must store the current traffic keys (or enough information
>   to regenerate them). They must also store the key-id needed to indicate
>   those traffic keys.
> 
> - The SNEs.
> 
> There is no need to *store* Next_Key. It can be regenerated every time.
> What's required is simply that it appear on the wire. I don't see any
> need to store the MKTs with the connection either. They're "associated"
> in that they match, I suppose...

Next_key is a value indicated by the user. It needs to be stored somewhere.

Connections need to either store the traffic keys or the info to
generate them. The latter would include the ISNs. I'll list the info
needed to generate them instead of storing the keys.

> S 7.1.
> 
>    o  MAC_truncation - the number of bytes to truncate the output of the 
>       MAC to. This is indicated by the MAC algorithm, as specified in 
>       [ao-crypto]. 
> 
> I don't think it makes sense to talk about truncation here.
> The MACs produce a value that is crammed into the packet.
> If there's a separate spec that describes the MACs, it's not
> AO's business whether those MACs were produced by a truncation process.

We need to describe the function as a call; the details are provided in
Gregory's doc. Since truncation is an argument to the call, I need to
indicate it.

Otherwise, we need two versions:
	
	1) the API between TCP-AO and Gregory's doc

	2) the API between Gregory's doc and the MAC alg.

Gregory's doc focuses on parameters; TCP-AO describes the algs and
interfaces, so I think it's important to include this here - even though
it's an opaque 'pass-through' value.

> S 7.2.
>    o  Output_length - The desired output length of the KDF, i.e., the
>       length to which the KDF's output will be truncated. In TCP-AO, the
>       output_length is the PRF_truncation value of the MKT. This is
>       specified by the KDF as described in [ao-crypto].
> 
> Everything from i.e., seems (1) superfluous and (2) isnufficiently
> general. Who says the PRF is truncated?

Again, I was describing a general alg from Gregory's doc. If that's
never an arg to a PRF, then we can omit it in TCP-AO.

...
>    We encourage users of TCP-AO to apply known techniques for generating
>    appropriate MKTs, including the use of reasonable master key lengths,
>    limited traffic key sharing, and limiting the duration of MKT use
>    [RFC3562]. This also includes the use of per-connection nonces, as
>    suggested in Section 7.2.
> 
> This seems like old text. We don't encoruage people to use per-connection
> nonces, it's part of the protocol.
...

"It" here is TCP-AO, not the user. That should be revised to be more clear.

> S 9.5.
>                      a. Optionally, set a timer on the MKT indicated by
>                        the current_key and segment socket pair to
>                        ensure that the MKT cannot be deleted for 2*MSL.
>                        If a timer has already been set, reset the
>                        timer.
> 
>                        This timer is advisory only. Removing MKTs with
>                        unexpired timers can result in a user-level
>                        warning, but is not prohibited. Implementation
>                        of timers is not required.
> 
>                      b. Set Current_key to the NextKeyID MKT.
> 
> I would reverse the order of these, since (a) is required and (b) is
> optional.

The timer is optional, and shifting keyIDs is required.

If we shift the order, we would need an intermediate value for
current_key, since we operate on current_key in (a). The order described
avoids that. I don't see why we'd reverse the order of these as a result.

> S 10.
> Was TCP-MD5 Mandatory to Implement? If not, this shouldn't be either.
> Even if so, this seems like kind of an unwarranted requirement.

Hard to say. RFC793 says all options must be implemented, and separates
that claim from the list of "currently specified" options. Whatever the
WG feels is appropriate here is fine.

IMO, it can't just be optional; it should be manditory at least in some
environments.

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


	Title           : TCP Congestion Control
	Author(s)       : M. Allman, et al.
	Filename        : draft-ietf-tcpm-rfc2581bis-06.txt
	Pages           : 17
	Date            : 2009-07-27

This document defines TCP's four intertwined congestion control
 algorithms: slow start, congestion avoidance, fast retransmit, and
 fast recovery.  In addition, the document specifies how TCP should
 begin transmission after a relatively long idle period, as well as
 discussing various acknowledgment generation methods.  This document
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From touch@ISI.EDU  Mon Jul 27 05:56:22 2009
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Hi, all,

Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.
> 
> 
> 	Title           : TCP Congestion Control
> 	Author(s)       : M. Allman, et al.
> 	Filename        : draft-ietf-tcpm-rfc2581bis-06.txt
> 	Pages           : 17
> 	Date            : 2009-07-27
> 
> This document defines TCP's four intertwined congestion control
>  algorithms: slow start, congestion avoidance, fast retransmit, and
>  fast recovery.  In addition, the document specifies how TCP should
>  begin transmission after a relatively long idle period, as well as
>  discussing various acknowledgment generation methods.  This document
>  obsoletes RFC 2581.

FWIW, regarding the discussion of the long idle period:

The algorithm used is based on a footnote in the appendix of an
unpublished extension of Van's 1988 Sigcomm paper. Amy Hughes, John
Heidemann, and I noted this a few years ago on this list in the
following I-D:
http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt

Here is a link to the 1988 Sigcomm paper:
http://www.acm.org/pubs/articles/proceedings/comm/52324/p314-jacobson/p314-jacobson.pdf

The Sigcomm version was only 16 pages long. The word "idle" does not
appear anywhere in it.

FWIW, see also our I-D for a few ways to deal with the issue. The
problem is really that the sender is accumulating permission to burst,
which just using a timer cannot address.

Joe



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At Mon, 27 Jul 2009 04:34:23 -0700,
> > This seems like an unwarranted requirement. The keys are generated
> > independently, so why make implementations gnerate keys they don't
> > need?
> 
> Agreed - I was trying to say that every MKT *can* derive any of the four
> traffic keys, not that all four *are* always derived.

OK. So this may be editorial only.

> > EDITORIAL
> ...
> > S 2.
> >    Section 9.6). This document obsoletes the TCP MD5 option with a more
> >    general TCP Authentication Option (TCP-AO), to support the use of
> > 
> > s/obsoletes/replaces/
> 
> I thought we needed to use the term "obsoletes" somewhere. Maybe this is
> the right sentence to do so, maybe not.

Well, I think we could rewrite this as.

"This document obsoletes the TCP MD5 option. It replaces it with..."


> ...
> >       Values of 4 and other small values larger than 4 (e.g., indicating
> >       MACs of very short length) are of dubious utility but are not
> >       specifically prohibited.
> > 
> > It's probably worth stating explicitly here that the MAC length is
> > dictated by the MKT, not the option.
> 
> The length of the MAC field is dictated by the option. If the option
> says the MAC is 70 bytes, and the MKT says it should be 72, then the
> packet would fail authentication.
> 
> I'll correct to call this "indicating MAC fields of very short length"

AGreed. I'm just mentioning this so people don't think that you
can specify a shorter MAC in the option and have it validate.


> ...
> >       Implementations are advised to keep master key values in a
> >       private, protected area of memory or other storage.
> > 
> > I'm curious if anyone has a concrete notion of what this means. As I
> > understand the situation, Cisco routers (for example) are a single
> > monolothic process. How would you do this for a Cisco?
> 
> Keep the keys in an area that non-operator processes can't get to
> without explicit permission or override.

Sure. I'm just observing that this isn't true on some systems
which only have one level of securuty.



> > S 5.3.
> > This whole discussion of how many MKTs can match seems confusing
> > and self-contradictory.
> > 
> > My understanding of the rules is as follows:
> > 
> > - Any outgoing packet, which has not yet been processed by TCP-AO,
> >   may match any number of MKTs, but each MKT must have a distinct
> >   key-id. Once that packet has been processed by AO, it must
> >   match only one MKT, determined by key-id.
> 
> If it matches multiple MKTs, the question is whether there is a
> preferred one or not. I had thought there was, but if not then this text
> can be used.

I had sort of assumed that this was an invariant--that the system
had to pick a preference if there were any.


> > There is no need to *store* Next_Key. It can be regenerated every time.
> > What's required is simply that it appear on the wire. I don't see any
> > need to store the MKTs with the connection either. They're "associated"
> > in that they match, I suppose...
> 
> Next_key is a value indicated by the user. It needs to be stored somewhere.

Sure, but it could be a global setting.


> > S 7.1.
> > 
> >    o  MAC_truncation - the number of bytes to truncate the output of the 
> >       MAC to. This is indicated by the MAC algorithm, as specified in 
> >       [ao-crypto]. 
> > 
> > I don't think it makes sense to talk about truncation here.
> > The MACs produce a value that is crammed into the packet.
> > If there's a separate spec that describes the MACs, it's not
> > AO's business whether those MACs were produced by a truncation process.
> 
> We need to describe the function as a call; the details are provided in
> Gregory's doc. Since truncation is an argument to the call, I need to
> indicate it.

Right, I don't think it should be an argument to the call--so it's
Greg's fault :)



> > S 9.5.
> >                      a. Optionally, set a timer on the MKT indicated by
> >                        the current_key and segment socket pair to
> >                        ensure that the MKT cannot be deleted for 2*MSL.
> >                        If a timer has already been set, reset the
> >                        timer.
> > 
> >                        This timer is advisory only. Removing MKTs with
> >                        unexpired timers can result in a user-level
> >                        warning, but is not prohibited. Implementation
> >                        of timers is not required.
> > 
> >                      b. Set Current_key to the NextKeyID MKT.
> > 
> > I would reverse the order of these, since (a) is required and (b) is
> > optional.
> 
> The timer is optional, and shifting keyIDs is required.
> 
> If we shift the order, we would need an intermediate value for
> current_key, since we operate on current_key in (a). The order described
> avoids that. I don't see why we'd reverse the order of these as a result.

OK.

I just think mndatory before optional.


> > S 10.
> > Was TCP-MD5 Mandatory to Implement? If not, this shouldn't be either.
> > Even if so, this seems like kind of an unwarranted requirement.
> 
> Hard to say. RFC793 says all options must be implemented, and separates
> that claim from the list of "currently specified" options. Whatever the
> WG feels is appropriate here is fine.
> 
> IMO, it can't just be optional; it should be manditory at least in some
> environments.

I'm not going to go to the wall on this one. What do others
think?

-Ekr

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At Mon, 27 Jul 2009 01:24:21 -0700,
Gregory M. Lebovitz wrote:
> 
> inline...
> 
> At 04:46 PM 7/26/2009, Eric Rescorla wrote:
> >$Id: draft-lebovitz-ietf-tcpm-tcp-ao-crypto-01-rev.txt,v 1.1 
> >2009/07/26 22:40:47 ekr Exp $
> >
> >TECHNICAL
> >S 3.1.
> >This mixing of the KDF interface and the NIST 800-108 interface
> >is extraordinarily confusing.
> 
> I'm not sure it's "mixing". Isn't one is a subset of the other? The 
> KDF interface is an interface. The NIST thing is the actual 
> concatenated construct which fulfills the Context.

Well, as usual, the interface is abstract and the NIST thing is
a concrete implementation.


> >To recap:
> >The input to the KDF requires:
> >
> >- Master Key
> >- Context (i.e., the conn-data)
> 
> it's not just the Data_Block (updated lingo, per auth-opt-05). In 
> order to follow SP800-180, the context is a few things concatenated, 
> and Data_Block is one of the things.

Yes, that's what it is now and my point is that that's wrong.




> >- Output length
> >
> >Now, as it happens, you're constructing your KDFs a la SP800-108,
> >so you then use the relevant PRF with an input block constructed
> >as:
> >
> >            ( i || Label || Context || Output_Length)
> >
> >But that is not a meaningful part of the interface to the KDF
> >as far as AO is concerned. If you were (for instance) to define
> >a new TLS PRF based KDF, you would not use i internally because
> >the TLS PRF generates an arbitrary length string. Again, it's
> >important to distinguish between the TCP-AO requirements and
> >what this document happens to do.
> 
> I think this document is NOT just the generic interface - that's 
> covered in auth-opt-05. This document reminds us of the generic 
> interface, and then specifically defines how to do the SHA1 and 
> AES-128 KDFs themselves. That's why we made it specific. Maybe I'm 
> not understanding you?

Apparently not. My point is that you shouldn't be exporting
the SP800-108 stuff into the TCP-AO document, and the way you
have defined the generic interface does that.


> >S 3.1.3.
> >Wait, I don't remember agreeing to this. I'm happy to use these
> >as labels in soem IANA registry, byt people's UI is not our
> >business.
> 
> This is the years of IKE interop work which motivates this. 

Well, years of TLS interop suggest the exact contrary.
We're not in the business of mandating UI.


> Clarifying the UI representations to make it very easy on operators 
> is a good thing. Others opinions on this UI guidance?



> >S 2.3.
> >        (1)  Key Derivation Functions (KDFs) which name a pseudorandom
> >             function (PRF) and use a Master_Key and some connection-
> >             specific Input with that PRF to produce Conn_Keys, the keys
> >             suitable for authenticating and integrity checking
> >             individual TCP segments.
> >
> >This is confusing. KDFs may or may not be constructed with PRFs.
> >However, THESE KDFs are so constructed. If you're giong to have
> >a document which applies to future KDFs, don't say PRF here.
> 
> it does say "which name", not "MUST name". Doesn't this cover your point?

Not really, no. I think you need to separate the abstract requirements
from the concrete instantiations.


-Ekr

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Subject: Re: [tcpm] 1323bis - PAWS check's problem with reodering on the reverse path
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Hi David,

comments are inline.


Am 14.07.2009 um 01:12 schrieb David Borman:

> I've been thinking about this recently, and I think that while it may
> be a real issue, it's not one that we need to worry about.  I believe
> that the problem that the change fixes is a more likely scenario, so
> if you have to choose between the two, that's the one to pick.

Sure. As you mentioned offline last time after the meeting the new  
version
of the alg. is version that most OS implement. I fully agree with you  
that we
should adopt the updated version.

>
> Also, for this issue that Alexander brings up, the timestamp has to be
> changing between ACKs.  If the timestamp hasn't changed between the  
> re-
> ordered ACKs, then the "SEG.TSval >= TS.recent" check will succeed.
> Generally, the TS clock shouldn't be ticking that often, if it is
> changing with every ACK then it might be going faster than it needs  
> to.

However, *if* reordering is present in the network the case that I  
mentioned
is not so unlikely. We saw the phenomenon during some measurements/ 
dumps...

>
> So, this issue only occurs when ACKs are re-ordered, *and* the
> Timestamp clock has ticked between the reordered ACKs.  I'm thinking
> that 1323bis should just note that this situation does exist.

Definitely.
I know that for example Linux implement also an heuristic to detect  
reordered ACK.
Maybe we can put a point to that.

>
> What does everyone else think?  Shall we leave the text as is and just
> document this edge case, or do people disagree with me and feel that
> this is a real problem that needs to be addressed, and is more
> important than the problems that the original change fixes?
>
> 			-David

Alex

>
> On Mar 26, 2009, at 6:10 PM, Alexander Zimmermann wrote:
>
>> Hi David, hi TCPM Folks,
>>
>> in the 1323bis draft the PAWS check has a problem (discarding ACKs)
>> if reordering is present on the
>> reverse path and no data is piggybacked. Actually, it's not the PAWS
>> check itself, it's the modification
>> of section 3.4: "Which Timestamp to Echo".
>>
>> The important parts of the RFC1323 and the draft (in this order) are
>>
>> ---
>> (2) If Last.ACK.sent falls within the range of sequence numbers
>>     of an incoming segment:
>>
>>        SEG.SEQ <= Last.ACK.sent < SEG.SEQ + SEG.LEN
>>
>>     then the TSval from the segment is copied to TS.Recent;
>>     otherwise, the TSval is ignored.
>>
>> ---
>>
>> (2) If:
>>
>>      SEG.TSval >= TS.recent and SEG.SEQ <= Last.ACK.sent
>>
>>      then SEG.TSval is copied to TS.Recent; otherwise, it is
>>      ignored.
>> ---
>>
>> The addition of "SEG.TSval >= TS.recent" is not important for us at
>> moment. As the appendix said,
>> this check is included for consistency with the PAWS test only.
>>
>> The relevant paragraph is the change from "SEG.SEQ <= Last.ACK.sent
>> < SEG.SEQ + SEG.LEN" to the
>> reduced form "SEG.SEQ <= Last.ACK.sent". Again, the appendix said
>> that this change was made
>> since the original algorithm to control which timestamp is echoed
>> was incorrect in two regards:
>>
>> (1) It failed to update TSrecent for a retransmitted segment that
>> resulted from a lost ACK.
>>
>> (2) It failed if SEG.LEN = 0.
>>
>> It's right that the 1323bis algo fix these two problems, however,
>> it's introduced the new problem mentioned
>> already above.
>>
>> Following situation:
>> - One-way flow from TCP A to TCP B
>> - Reordering on the reverse path
>>
>> Consequence
>> - TCP A will discard all ACKs with were delayed
>> - Harmful in the case ABC (RFC 3465) is off, since we count ACKs for
>> slow-start and congestion avoidance
>>
>> Examples are attached.
>>
>> Alex
>>
>> <PAWS 1323.txt><PAWS 1323bis.txt>
>> //
>> // Dipl.-Inform. Alexander Zimmermann
>> // Department of Computer Science, Informatik 4
>> // RWTH Aachen University
>> // Ahornstr. 55, 52056 Aachen, Germany
>> // phone: (49-241) 80-21422, fax: (49-241) 80-22220
>> // email: zimmermann@cs.rwth-aachen.de
>> // web: http://www.umic-mesh.net
>> //
>>
>




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From william.polk@nist.gov  Sun Jul 26 08:44:04 2009
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From: "Polk, William T." <william.polk@nist.gov>
To: "gregory.ietf@gmail.com" <gregory.ietf@gmail.com>, "mcgrew@cisco.com" <mcgrew@cisco.com>
Date: Sun, 26 Jul 2009 11:40:50 -0400
Thread-Topic: handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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Subject: [tcpm] handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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Gregory,

As requested, Pasi and I took a look at how draft-lebovitz-ietf-tcpm-tcp-ao=
-crypto
handles the AES-CMAC case when the master key is not 128 bits.

The method currently used (using AES-CMAC with a known fixed key as a
"randomness extractor") is basically the same as used in IKEv2 (RFC
4615 and 4434), except IKEv2 uses all-zeroes key (but the specific
value of the known fixed key shouldn't matter much -- just pick one).

Using the same method in TCP-AO would be acceptable us, and seems
strongly preferable to requiring network admins to configure exactly
128 bit keys.

Other methods of getting from "what the admin configures" to 128 bits
could work, too. Since the output of this step is *only* used as the
key for further PRF calculations (and for *nothing* else), the
requirements for it are very limited. If all keys were <=3D 128 bits,
even something trivial such as padding with zeroes would work -- so
this step doesn't even have to be one-way.  Since we need to handle
also keys >128 bits, simple padding isn't enough, of course. If the WG
wants AES-only method, the one that's in the draft now seems OK to us.

I hope this was helpful.

Tim Polk
Pasi Eronen=

From william.polk@nist.gov  Sun Jul 26 23:45:25 2009
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From: "Polk, William T." <william.polk@nist.gov>
To: Eric Rescorla <ekr@networkresonance.com>, "Gregory M. Lebovitz" <gregory.ietf@gmail.com>
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Subject: Re: [tcpm] handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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Ekr,

Your objections on the magic number are well taken.  Since zeroes would
be a holdover from 4615 and 4434, that value won't introduce any new
issues.  I think that would be the best course of action.

Tim

________________________________________
From: Eric Rescorla [ekr@networkresonance.com]
Sent: Sunday, July 26, 2009 7:26 PM
To: Gregory M. Lebovitz
Cc: Polk, William T.; mcgrew@cisco.com; housley@vigilsec.com; tcpm@ietf.org=
; Pasi.Eronen@nokia.com
Subject: Re: [tcpm] handling variable length keys in draft-lebovitz-ietf-tc=
pm-tcp-ao-crypto

At Sun, 26 Jul 2009 12:43:37 -0700,
Gregory M. Lebovitz wrote:
>
> Tim,
> This is helpful, thanks. We will proceed toward WG LC with the
> mechanism as represented in draft v-01.

I'll be sending my comments on v-01 shortly, but I object to
the magic number in the current mechanism. If we're going to
use a magic key, it should be zero or some other anchor point.
The use of a random number pulled out of the air is highly
inappropriate in cryptographic systems [I refer you to the
contrroversy over the DES S-boxes].

-Ekr



> If later, after further discussions within the cryptographic
> community, it is decided that the randomness extraction mechanism to
> go from a variable-length user-given string to a 128-bit key would be
> better accomplished some other way, then we can always do a quick
> -bis on the RFC and update the mechanism. That's easy enough for a
> small RFC like this.


> Gregory.
>
> At 08:40 AM 7/26/2009, Polk, William T. wrote:
> >Gregory,
> >
> >As requested, Pasi and I took a look at how
> >draft-lebovitz-ietf-tcpm-tcp-ao-crypto
> >handles the AES-CMAC case when the master key is not 128 bits.
> >
> >The method currently used (using AES-CMAC with a known fixed key as a
> >"randomness extractor") is basically the same as used in IKEv2 (RFC
> >4615 and 4434), except IKEv2 uses all-zeroes key (but the specific
> >value of the known fixed key shouldn't matter much -- just pick one).
> >
> >Using the same method in TCP-AO would be acceptable us, and seems
> >strongly preferable to requiring network admins to configure exactly
> >128 bit keys.
> >
> >Other methods of getting from "what the admin configures" to 128 bits
> >could work, too. Since the output of this step is *only* used as the
> >key for further PRF calculations (and for *nothing* else), the
> >requirements for it are very limited. If all keys were <=3D 128 bits,
> >even something trivial such as padding with zeroes would work -- so
> >this step doesn't even have to be one-way.  Since we need to handle
> >also keys >128 bits, simple padding isn't enough, of course. If the WG
> >wants AES-only method, the one that's in the draft now seems OK to us.
> >
> >I hope this was helpful.
> >
> >Tim Polk
> >Pasi Eronen
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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From: Eric Rescorla <ekr@networkresonance.com>
To: "Polk, William T." <william.polk@nist.gov>
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Subject: Re: [tcpm] handling variable length keys in draft-lebovitz-ietf-tcpm-tcp-ao-crypto
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At Sun, 26 Jul 2009 11:40:50 -0400,
Polk, William T. wrote:
> 
> Gregory,
> 
> As requested, Pasi and I took a look at how draft-lebovitz-ietf-tcpm-tcp-ao-crypto
> handles the AES-CMAC case when the master key is not 128 bits.
> 
> The method currently used (using AES-CMAC with a known fixed key as a
> "randomness extractor") is basically the same as used in IKEv2 (RFC
> 4615 and 4434), except IKEv2 uses all-zeroes key (but the specific
> value of the known fixed key shouldn't matter much -- just pick one).
> 
> Using the same method in TCP-AO would be acceptable us, and seems
> strongly preferable to requiring network admins to configure exactly
> 128 bit keys.
> 
> Other methods of getting from "what the admin configures" to 128 bits
> could work, too. Since the output of this step is *only* used as the
> key for further PRF calculations (and for *nothing* else), the
> requirements for it are very limited. If all keys were <= 128 bits,
> even something trivial such as padding with zeroes would work -- so
> this step doesn't even have to be one-way.

It seems like you're putting a lot of weight on AES to assume that
a key that is 24 random bits and 104 bits of zeros at the end
really has strength 2^24

-Ekr

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--NextPart

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


	Title           : TCP Congestion Control
	Author(s)       : M. Allman, et al.
	Filename        : draft-ietf-tcpm-rfc2581bis-07.txt
	Pages           : 17
	Date            : 2009-07-27

This document defines TCP's four intertwined congestion control
 algorithms: slow start, congestion avoidance, fast retransmit, and
 fast recovery.  In addition, the document specifies how TCP should
 begin transmission after a relatively long idle period, as well as
 discussing various acknowledgment generation methods.  This document
 obsoletes RFC 2581.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-rfc2581bis-07.txt

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Date: Mon, 27 Jul 2009 08:55:57 -0700
From: Eric Rescorla <ekr@networkresonance.com>
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Subject: Re: [tcpm] possible NAT support for TCP-AO
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At Mon, 13 Jul 2009 13:58:19 -0700,
Joe Touch wrote:
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi, all,
> 
> Based on a discussion on a different list with Dan Wing, I'd like to
> revisit thinking about NATs in the context of the current AO, which uses
> traffic keys derived from the socket pair and ISN pair. We might be able
> to now support NATs as follows, e.g.:
> 
> 	add the following flags to the MKT:
> 
> 		localNAT flag - indicates whether the local IP/port are
> 			zeroed before MAC calculation
> 
> 		remoteNAT flag - indicates whether the remote IP/port
> 			are zeroed before MAC calculation
> 
> 	add steps to the incoming/outgoing processing:
> 		zero the corresponding IP/port when the flag indicates,
> 		both on outgoing and incoming MAC calculation
> 
> That's basically it. A client behind a NAT would have a MKT with
> localNAT true, and a server for that client would need to have a MKT
> with remoteNAT true. This does require careful MKT configuration.
> Although I wouldn't expect both localNAT and remoteNAT to be true, there
> isn't a particular reason it needs to be prohibited.

So I'm going to jump in right here: it's not in general practical
to know whether you are behind a NAT. That's why the BEHAVE 
NAT discovery document is going to Experimental.

-Ekr

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Eric Rescorla wrote:
...
> So I'm going to jump in right here: it's not in general practical
> to know whether you are behind a NAT. That's why the BEHAVE 
> NAT discovery document is going to Experimental.

You're talking about NAT discovery; this mod doesn't talk about how you
figure that out, just how to support it if you do know it.

You can trivially know it in a key distro protocol if you send your
adr/port in the payload, so that the endpoint can see if it matches what
is on the outside. Granted that won't work in a backward-compatible way
(e.g., for BEHAVE), but key distro protocols don't have that limitation.

Joe
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From mallman@icir.org  Mon Jul 27 12:28:13 2009
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Subject: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21st century)
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Jerry-

A few random thoughts here ...

> I'll start with Lowering initRTO.
> 
> RFC1122 contains the following paragraph:
> 
> The following values SHOULD be used to initialize the
> estimation parameters for a new connection:
> 
> (a)  RTT = 0 seconds.
> 
> (b)  RTO = 3 seconds.  (The smoothed variance is to be
> initialized to the value that will result in this RTO).
> 
> The "3secs SHOULD" is reaffirmed in RFC2988.
> 
> From our own measurement of world wide RTT distribution to Google
> servers we believe 3secs is too conservative, and like to propose it
> to be reduced to 1sect.

I am not at all sure this is a good idea.  I have an easier time
believing the others on your list are perhaps reasonable.  But, this one
seems somewhat dubious to me.  A few things ...

  - The fundamental problem here is that you have *no* information.
    That is, we don't know how long the path is before we have done an
    exchange.  When you start from scratch you have nothing to go on
    except defaults.  So, it seems to me on those grounds alone
    conservativeness is fine.  Because,

  - If it was just an extra small packet or two that got sent out that
    doesn't seem like a Big Deal.  But, once you retransmit the SYN you
    no longer can take an RTT sample from the 3WHS per Karn's algorithm.
    So, if in fact the initial RTO is too short then it isn't just going
    to strobe out an extra packet, but what it means is that it's pretty
    likely that the packets in your initial window---after clumsily
    finishing the 3WHS---will likewise be retransmitted because the RTO
    estimate is low and we did not get the opportunity in the 3WHS to
    take an actual RTT sample to better seed the estimator.  This is RTO
    Hell.

  - At first blush timestamps might help here because if used then we
    don't have to use Karn's algorithm.  But, again, since we are just
    initiating a connection how do we know if the peer is going to use
    timestamps?  If the initiator sends a timestamp option then there is
    a chance that timestamps will be in use and therefore there is a
    chance you'll avoid RTO Hell.  But, there is also a chance you
    won't.  The 3WHS responder (sender of the SYN+ACK) will know if
    timestamps will be in use and therefore could perhaps lower the
    initial RTO (basically, this is the ECNSYN trick).  That doesn't
    seem all that unreasonable to me.

  - Now, if you track information across connections as others have
    noted then, sure.  It seems perfectly acceptable to take the view
    that with high confidence you understand that 1sec (or whatever)
    will be fine for an initial RTO over some path that you have
    transmitted traffic across in the recent past and so then you can
    use that.  In this case, you are picking an initial RTO for a
    connection but not flying completely in the dark.

  - It seems that (per the discussion in today's meeting) a naive
    lowering to 1sec is going to be problematic because we have
    bandwidth-on-demand networks, deep queues in access devices are not
    rare, etc.

In a subsequent email you note:

> Correct so there is a fine line to walk. But if > 98% of all TCP
> connections experience RTT << 1 sec, it just seems too conservative to
> have a global initRTO == 3secs just to avoid spurious retransmission
> in the < 2% category.

I agree that it is a fine line.  But, I think your 98%-vs-2% is far too
glib.  That is, we have to look at how bad we're making it for those 2%.
If we degraded each of those 2% by "a smidge" then who cares.  But, if
we really hose those connections (see second bullet above about RTO
Hell) it doesn't seem like a good tradeoff.  It's useful to remember
that TCP was designed to be general and not optimal.  Certainly we don't
want to unduly penalize most of the traffic (/users) to dogmatically
accommodate every last esoteric situation that might happen to crop up
on the third Tuesday of the 6th month following the most recent blue
moon.  But, we also can err on the other side, too.  I think simple
percentages as you have given are pretty superficial and we'd need to go
beyond that to really decide what line we wanted to walk.

Just my two bits ...

allman




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Date: Tue, 28 Jul 2009 08:32:02 -0700
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From: Jerry Chu <hkchu@google.com>
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Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21st century)
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Mark,

Thanks for your thoughtful comments. The good news is, I'm pretty much on
the same page as you. The problem has to be real and the solution must be
reasonable with acceptable downside if the latter is unavoidable. From our
applications folks the problem seems very real and the performance
difference is not negligible. So I'll focus more on how to rein in the down=
side.

On Mon, Jul 27, 2009 at 12:28 PM, Mark Allman <mallman@icir.org> wrote:
>
> Jerry-
>
> A few random thoughts here ...
>
> > I'll start with Lowering initRTO.
> >
> > RFC1122 contains the following paragraph:
> >
> > The following values SHOULD be used to initialize the
> > estimation parameters for a new connection:
> >
> > (a) =A0RTT =3D 0 seconds.
> >
> > (b) =A0RTO =3D 3 seconds. =A0(The smoothed variance is to be
> > initialized to the value that will result in this RTO).
> >
> > The "3secs SHOULD" is reaffirmed in RFC2988.
> >
> > From our own measurement of world wide RTT distribution to Google
> > servers we believe 3secs is too conservative, and like to propose it
> > to be reduced to 1sect.
>
> I am not at all sure this is a good idea. =A0I have an easier time
> believing the others on your list are perhaps reasonable. =A0But, this on=
e
> seems somewhat dubious to me. =A0A few things ...
>
> =A0- The fundamental problem here is that you have *no* information.
> =A0 =A0That is, we don't know how long the path is before we have done an
> =A0 =A0exchange. =A0When you start from scratch you have nothing to go on
> =A0 =A0except defaults. =A0So, it seems to me on those grounds alone
> =A0 =A0conservativeness is fine. =A0Because,

Agreed. The question is, how conservative one needs to be? Can we do
better for the majority and still not sacrificing the ones left behind? If
some cost is unavoidable, will the tradeoff be acceptable?

>
> =A0- If it was just an extra small packet or two that got sent out that
> =A0 =A0doesn't seem like a Big Deal. =A0But, once you retransmit the SYN =
you
> =A0 =A0no longer can take an RTT sample from the 3WHS per Karn's algorith=
m.
> =A0 =A0So, if in fact the initial RTO is too short then it isn't just goi=
ng
> =A0 =A0to strobe out an extra packet, but what it means is that it's pret=
ty
> =A0 =A0likely that the packets in your initial window---after clumsily
> =A0 =A0finishing the 3WHS---will likewise be retransmitted because the RT=
O
> =A0 =A0estimate is low and we did not get the opportunity in the 3WHS to
> =A0 =A0take an actual RTT sample to better seed the estimator. =A0This is=
 RTO
> =A0 =A0Hell.

Understood. I've also had this as one of the bullet points on the cost side
in my slides. But my gut feeling was this problem could be mitigated someho=
w.
(Guess now I'll need to demonstrate how :-)

>
> =A0- At first blush timestamps might help here because if used then we
> =A0 =A0don't have to use Karn's algorithm. =A0But, again, since we are ju=
st
> =A0 =A0initiating a connection how do we know if the peer is going to use
> =A0 =A0timestamps? =A0If the initiator sends a timestamp option then ther=
e is
> =A0 =A0a chance that timestamps will be in use and therefore there is a
> =A0 =A0chance you'll avoid RTO Hell. =A0But, there is also a chance you
> =A0 =A0won't.

So the active open side has retransmitted SYN, assuming initRTO of
1sec, and eventually received a SYN-ACK but the TS option was
denied so no good RTT sample can be taken. It will continue to send
the init data with the initRTO.

If the SYN retransmission has been unnecessary because the initRTO
has been too aggressive, more SYN-ACK will be triggered. Assuming
these dup SYN-ACKs are not lost, could they be used as a hint to the
active open side that the initRTO has been too short, so that the RTO
timer can be reverted back to 3secs immediately, in time to avoid further
RTO hell?

One price to pay in the above scenario compared to the default case
of 3secs is that one could have gotten a good RTT sample of say,
2secs but due to the choice of a more aggressive initRTO of 1sec,
one ends up with a fall-back 3secs again for the data phase so any
loss from there will incur more time (3 vs 2secs) to recover. But the
price seems acceptable, and is incurred only when there is immediate
further loss. At least the dreadful endless spurious retransmission,
which would've been unacceptable, is stopped.

>The 3WHS responder (sender of the SYN+ACK) will know if
> =A0 =A0timestamps will be in use and therefore could perhaps lower the
> =A0 =A0initial RTO (basically, this is the ECNSYN trick). =A0That doesn't
> =A0 =A0seem all that unreasonable to me.

Correct, but I'd really like to find a solution for the active open side
because there are other techniques the server side can employ (e.g.,
initialize the initRTO from the RTT history...) as a backup if this one
doesn't fly, but those techniques don't work well on the client side.

>
> =A0- Now, if you track information across connections as others have
> =A0 =A0noted then, sure. =A0It seems perfectly acceptable to take the vie=
w
> =A0 =A0that with high confidence you understand that 1sec (or whatever)
> =A0 =A0will be fine for an initial RTO over some path that you have
> =A0 =A0transmitted traffic across in the recent past and so then you can
> =A0 =A0use that. =A0In this case, you are picking an initial RTO for a
> =A0 =A0connection but not flying completely in the dark.

Correct. There is a certain amount of added complexity associated
with the cached RTT idea. As such, it is still nice if this reducing initRT=
O
proposal can fly, not just to cover the client side but also as a simpler
solution for those servers that can't afford to use the more complex
solution.

>
> =A0- It seems that (per the discussion in today's meeting) a naive
> =A0 =A0lowering to 1sec is going to be problematic because we have
> =A0 =A0bandwidth-on-demand networks, deep queues in access devices are no=
t
> =A0 =A0rare, etc.

Yes I heard it, loud and clear :-).

>
> In a subsequent email you note:
>
> > Correct so there is a fine line to walk. But if > 98% of all TCP
> > connections experience RTT << 1 sec, it just seems too conservative to
> > have a global initRTO =3D=3D 3secs just to avoid spurious retransmissio=
n
> > in the < 2% category.
>
> I agree that it is a fine line. =A0But, I think your 98%-vs-2% is far too
> glib. =A0That is, we have to look at how bad we're making it for those 2%=
.
> If we degraded each of those 2% by "a smidge" then who cares. =A0But, if
> we really hose those connections (see second bullet above about RTO
> Hell) it doesn't seem like a good tradeoff. =A0It's useful to remember
> that TCP was designed to be general and not optimal. =A0Certainly we don'=
t
> want to unduly penalize most of the traffic (/users) to dogmatically
> accommodate every last esoteric situation that might happen to crop up
> on the third Tuesday of the 6th month following the most recent blue
> moon. =A0But, we also can err on the other side, too. =A0I think simple
> percentages as you have given are pretty superficial and we'd need to go
> beyond that to really decide what line we wanted to walk.

Understood. See above.

Jerry

>
> Just my two bits ...
>
> allman
>
>
>

From mallman@icir.org  Tue Jul 28 11:36:05 2009
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Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21st century)
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Jerry-

> So the active open side has retransmitted SYN, assuming initRTO of
> 1sec, and eventually received a SYN-ACK but the TS option was
> denied so no good RTT sample can be taken. It will continue to send
> the init data with the initRTO.
> 
> If the SYN retransmission has been unnecessary because the initRTO has
> been too aggressive, more SYN-ACK will be triggered. Assuming these
> dup SYN-ACKs are not lost, could they be used as a hint to the active
> open side that the initRTO has been too short, so that the RTO timer
> can be reverted back to 3secs immediately, in time to avoid further
> RTO hell?

So, you're saying that if you send N SYNs (N>1) and you get N SYN+ACKs
then you take that to mean the initRTO was too short.

Yeah, perhaps ... 

Are you suggesting something like this ...

  (1) Use an initRTO of 1sec.

  (2) If the RTO fires and you retransmit the SYN back the RTO to 3sec.
      (And, if necessary you exponential backoff even more.)

  (3) After the 3WHS you will either (a) have a good RTT sample from the
      timestamp option and can therefore set the RTO reasonably or (b)
      you retain the 3sec RTO until you get a good RTT sample.

??

That means that if 1sec is not enough in the 3WHS you have to use 3sec
until you have some idea about the network.  However, you also get a
shot [*one shot*] to retransmit sooner based on the notion that most
RTTs are less than 1sec.  

I think I buy that notion ... that both allows for more aggressiveness
for the common case while also protecting against RTO Hell for the long
path cases.  Is that the right walking of the line?

allman




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From wwwrun@core3.amsl.com  Wed Jul 29 01:36:01 2009
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Subject: [tcpm] Protocol Action: 'TCP Congestion Control' to Draft Standard
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The IESG has approved the following document:

- 'TCP Congestion Control '
   <draft-ietf-tcpm-rfc2581bis-07.txt> as a Draft Standard


This document is the product of the TCP Maintenance and Minor Extensions Working Group. 

The IESG contact persons are Lars Eggert and Magnus Westerlund.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-rfc2581bis-07.txt

Technical Summary

  The congestion control algorithms described in RFC 2581 are widely
  implemented and used.  This document clarifies some specific points
  that have created questions for implementers in the past.

Working Group Summary

  The working group saw the need for this document based on a small
  number of commonly recurring questions about what is meant by certain
  wordings in RFC2581.  The working group was easily able to agree on
  the clarifications that this document provides.

Document Quality

  The document was reviewed for quality by a large number of TCPM
  WG members.

Personnel

  Wes Eddy (wesley.m.eddy@nasa.gov) is the document shepherd.
  Lars Eggert (lars.eggert@nokia.com) reviewed the document for the IESG.


From lars.eggert@nokia.com  Wed Jul 29 03:48:57 2009
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Subject: [tcpm] status of TCP-MD5 after TCP-AO publication
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Hi,

at the meeting, the question came up which status TCP-MD5 should have  
after TCP-AO is published. Specifically, whether it should be  
obsoleted by TCP-AO and/or if it should be reclassified as Historic. A  
related issue I came across while trying to form an opinion of the  
former issue is if the publication of TCP-AO means that we can lift  
the Standards Variance for TCP-MD5 introduced by RFC4728. (I'm CC'ing  
the IESG because of this latter point, because that Standards Variance  
came from the SEC and RTG areas.)

I looked at some other cases related to MD5 to see how we handled  
obsoleting or making things Historic in the past, with regards to TCP- 
MD5.

Example 1 is RFC2537 "RSA/MD5 KEYs and SIGs in the Domain Name System  
(DNS)", which is obsoleted by RFC3110 "RSA/SHA-1 SIGs and RSA KEYs in  
the Domain Name System (DNS)", but *not* moved to Historic. (Note that  
this change replaces MD5 by SHA1, i.e., isn't wire-compatible, similar  
to how moving from TCP-MD5 to TCP-AO isn't
wire-compatible, so that is apparently OK.)

Example 2 is RFC2082, which is also obsoleted (but not made Historic)  
by RFC4822 and also replaces MD5 with another algorithm that affects  
wire-compatibility.

So there is precedent for TCP-AO to obsolete TCP-MD5, which is also  
what the -AO draft currently contains.

Now, on the topic of moving TCP-MD5 to Historic, RFC2026 says that  
even when one spec obsoletes another, there may be cases where the  
obsoleted spec is kept at its original level (i.e., not made Historic)  
to honor the requirements of an installed base, if there is one. (See  
PS below for the actual text.) I believe that TCP-MD5 will still  
remain widely deployed initially even after TCP-AO is published, and  
so moving it to Historic is not something we should aim for at this  
time. Agreed?

Finally, on whether the publication of TCP-AO means that we can lift  
the Standards Variance for TCP-MD5 introduced by RFC4728, I'd be  
really interested in hearing your feedback.

Thanks,
Lars

PS: To save you all some scrolling, RFC2026 says this about Historic  
RFCs:

    4.2.4  Historic

    A specification that has been superseded by a more recent
    specification or is for any other reason considered to be obsolete  
is
    assigned to the "Historic" level.  (Purists have suggested that the
    word should be "Historical"; however, at this point the use of
    "Historic" is historical.)

and later

    6.3  Revising a Standard

    A new version of an established Internet Standard must progress
    through the full Internet standardization process as if it were a
    completely new specification.  Once the new version has reached the
    Standard level, it will usually replace the previous version, which
    will be moved to Historic status.  However, in some cases both
    versions may remain as Internet Standards to honor the requirements
    of an installed base.  In this situation, the relationship between
    the previous and the new versions must be explicitly stated in the
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one
    or more existing standards track specifications for the same  
function
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    with the same Last-Call and notification procedures used for any
    other standards action.  A request to retire an existing standard  
can
    originate from a Working Group, an Area Director or some other
    interested party.



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From iljitsch@muada.com  Wed Jul 29 04:02:07 2009
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Subject: Re: [tcpm] status of TCP-MD5 after TCP-AO publication
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On 29 jul 2009, at 12:48, Lars Eggert wrote:

> at the meeting, the question came up which status TCP-MD5 should  
> have after TCP-AO is published. Specifically, whether it should be  
> obsoleted by TCP-AO and/or if it should be reclassified as Historic.

First of all, I'd like to see some operational experience with TCP-AO.  
Don't throw away your old shoes until you know whether the new ones fit.

Second, it's not like TCP-MD5 isn't being used. As such, "historic"  
wouldn't apply. "Deprecated", maybe.

Third, why is it exactly that we can't simply move from MD5 to IPsec  
to protect BGP sessions??

From lars.eggert@nokia.com  Wed Jul 29 04:14:11 2009
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Hi,

On 2009-7-29, at 13:02, Iljitsch van Beijnum wrote:
>> at the meeting, the question came up which status TCP-MD5 should
>> have after TCP-AO is published. Specifically, whether it should be
>> obsoleted by TCP-AO and/or if it should be reclassified as Historic.
>
> First of all, I'd like to see some operational experience with TCP-AO.
> Don't throw away your old shoes until you know whether the new ones  
> fit.

so are you saying we should *neither* obsolete TCP-MD5 *nor* move it  
to Historic?

> Second, it's not like TCP-MD5 isn't being used. As such, "historic"
> wouldn't apply. "Deprecated", maybe.

There is no way to express this for an RFC. There are two options: "A  
obsoletes B", which is indicated in the document header, and "A is  
reclassified as Historic" in the RFC Editor database.

> Third, why is it exactly that we can't simply move from MD5 to IPsec
> to protect BGP sessions??

I refer you to the extensive discussion that happened when draft- 
bonica was first presented to the WG, which I have no interest in  
rehashing.

Lars
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Subject: Re: [tcpm] status of TCP-MD5 after TCP-AO publication
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On Jul 29, 2009, at 12:48 PM, Lars Eggert wrote:

> Hi,
>
> at the meeting, the question came up which status TCP-MD5 should  
> have after TCP-AO is published. Specifically, whether it should be  
> obsoleted by TCP-AO and/or if it should be reclassified as Historic.

1) TCP-AO should obsolete TCP-MD5.  That was the whole point of this  
exercise of creating TCP-AO.

2) Whether or not TCP-MD5 is moved to historic is a separate issue.   
If people are concerned about marking it historic at this point in  
time, then don't.  We can always come back in a year or two after TCP- 
AO has been deployed, and then mark TCP-MD5 historic.

As long as it is clear that TCP-AO obsoletes TCP-MD5, then I don't  
think it matters that much if/when TCP-MD5 is moved to historic.

			-David Borman


From touch@ISI.EDU  Wed Jul 29 04:45:11 2009
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Subject: Re: [tcpm] status of TCP-MD5 after TCP-AO publication
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Iljitsch van Beijnum wrote:
> On 29 jul 2009, at 12:48, Lars Eggert wrote:
> 
>> at the meeting, the question came up which status TCP-MD5 should have
>> after TCP-AO is published. Specifically, whether it should be
>> obsoleted by TCP-AO and/or if it should be reclassified as Historic.
> 
> First of all, I'd like to see some operational experience with TCP-AO.
> Don't throw away your old shoes until you know whether the new ones fit.
> 
> Second, it's not like TCP-MD5 isn't being used. As such, "historic"
> wouldn't apply. "Deprecated", maybe.
> 
> Third, why is it exactly that we can't simply move from MD5 to IPsec to
> protect BGP sessions??

This is addressed in the TCP-AO document:

- ---

   This document is not intended to replace the use of the IPsec suite
   (IPsec and IKE) to protect TCP connections [RFC4301][RFC4306]. In
   fact, we recommend the use of IPsec and IKE, especially where IKE's
   level of existing support for parameter negotiation, session key
   negotiation, or rekeying are desired. TCP-AO is intended for use only
   where the IPsec suite would not be feasible, e.g., as has been
   suggested is the case to support some routing protocols, or in cases
   where keys need to be tightly coordinated with individual transport
   sessions [Be07].

- ---

   This document differs from an IPsec/IKE solution in that TCP-AO as
   follows [RFC4301][RFC4306]:

   o  TCP-AO does not support dynamic parameter negotiation.

   o  TCP-AO uses TCP's socket pair (source address, destination
      address, source port, destination port) as a security parameter
      index, rather than using a separate field as an index (IPsec's
      SPI).

   o  TCP-AO forces a change of computed MACs when a connection
      restarts, even when reusing a TCP socket pair (IP addresses and
      port numbers) [Be07].

   o  TCP-AO does not support encryption.

   o  TCP-AO does not authenticate ICMP messages (some ICMP messages may
      be authenticated when using IPsec, depending on the
      configuration).

- ---

See also the Security Considerations section.
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Date: Wed, 29 Jul 2009 07:56:01 -0700
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Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21st century)
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Mark,

On Tue, Jul 28, 2009 at 11:35 AM, Mark Allman <mallman@icir.org> wrote:

>
> Jerry-
>
> > So the active open side has retransmitted SYN, assuming initRTO of
> > 1sec, and eventually received a SYN-ACK but the TS option was
> > denied so no good RTT sample can be taken. It will continue to send
> > the init data with the initRTO.
> >
> > If the SYN retransmission has been unnecessary because the initRTO has
> > been too aggressive, more SYN-ACK will be triggered. Assuming these
> > dup SYN-ACKs are not lost, could they be used as a hint to the active
> > open side that the initRTO has been too short, so that the RTO timer
> > can be reverted back to 3secs immediately, in time to avoid further
> > RTO hell?
>
> So, you're saying that if you send N SYNs (N>1) and you get N SYN+ACKs
> then you take that to mean the initRTO was too short.
>
> Yeah, perhaps ...
>
> Are you suggesting something like this ...
>
>  (1) Use an initRTO of 1sec.
>
>  (2) If the RTO fires and you retransmit the SYN back the RTO to 3sec.
>      (And, if necessary you exponential backoff even more.)
>
>  (3) After the 3WHS you will either (a) have a good RTT sample from the
>      timestamp option and can therefore set the RTO reasonably or (b)
>      you retain the 3sec RTO until you get a good RTT sample.
>
> ??
>
> That means that if 1sec is not enough in the 3WHS you have to use 3sec
> until you have some idea about the network.  However, you also get a
> shot [*one shot*] to retransmit sooner based on the notion that most
> RTTs are less than 1sec.
>
> I think I buy that notion ... that both allows for more aggressiveness
> for the common case while also protecting against RTO Hell for the long
> path cases.  Is that the right walking of the line?


I can think of a number of variations. The one-shot 1-sec-initRTO idea you
described above also came through my mind but the drawback is you only
get one-shot even though we know statistically > 98% of connections
have RTT < 1 sec so most likely the continuous use of 1-sec-initRTO will
turn
out to be better. (A counter argument might be one-shot is "good enough",
benefitting > 90% of the cases statistically...) The advantage of it is its
simplicity,
restricting the max # of spurious retransmissions caused by the reduced
initRTO
to 1, and obviously avoiding the RTO hell problem.

A different variation that is slightly more complex but allowing the reduced
initRTO to be used more than one-time (hence potentially providing more
benefit
to those > 98% connections) is to only terminate its use upon the detection
of spurious retransmission. Specifically -

1. initRTO = 1sec
2. if a valid pkt with the SYN bit on is received (so it could be either SYN
or
SYN-ACK) and the following two conditions are met:
  2a. the connection state is neither SYN-SENT nor SYN-RCVD
  2b. RTO == 1sec (or no good RTT sample has been taken)
Simply reset the RTO to 3secs.

On a second look it doesn't seem much more complex, although it may be
slightly more work to analyze its thoroughness (e.g., what if all the dup
SYN/SYN-ACK were lost, or all come back late...) Also it does incur
more spurious pkts for those connections with RTT > 2sec.

What do you think?

Jerry


>
> allman
>
>
>
>

--0016e64135dcbb6531046fd9623c
Content-Type: text/html; charset=ISO-8859-1
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Mark,<br><br><div class=3D"gmail_quote">On Tue, Jul 28, 2009 at 11:35 AM, M=
ark Allman <span dir=3D"ltr">&lt;<a href=3D"mailto:mallman@icir.org">mallma=
n@icir.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; p=
adding-left: 1ex;">
<br>
Jerry-<br>
<div class=3D"im"><br>
&gt; So the active open side has retransmitted SYN, assuming initRTO of<br>
&gt; 1sec, and eventually received a SYN-ACK but the TS option was<br>
&gt; denied so no good RTT sample can be taken. It will continue to send<br=
>
&gt; the init data with the initRTO.<br>
&gt;<br>
&gt; If the SYN retransmission has been unnecessary because the initRTO has=
<br>
&gt; been too aggressive, more SYN-ACK will be triggered. Assuming these<br=
>
&gt; dup SYN-ACKs are not lost, could they be used as a hint to the active<=
br>
&gt; open side that the initRTO has been too short, so that the RTO timer<b=
r>
&gt; can be reverted back to 3secs immediately, in time to avoid further<br=
>
&gt; RTO hell?<br>
<br>
</div>So, you&#39;re saying that if you send N SYNs (N&gt;1) and you get N =
SYN+ACKs<br>
then you take that to mean the initRTO was too short.<br>
<br>
Yeah, perhaps ...<br>
<br>
Are you suggesting something like this ...<br>
<br>
 =A0(1) Use an initRTO of 1sec.<br>
<br>
 =A0(2) If the RTO fires and you retransmit the SYN back the RTO to 3sec.<b=
r>
 =A0 =A0 =A0(And, if necessary you exponential backoff even more.)<br>
<br>
 =A0(3) After the 3WHS you will either (a) have a good RTT sample from the<=
br>
 =A0 =A0 =A0timestamp option and can therefore set the RTO reasonably or (b=
)<br>
 =A0 =A0 =A0you retain the 3sec RTO until you get a good RTT sample.<br>
<br>
??<br>
<br>
That means that if 1sec is not enough in the 3WHS you have to use 3sec<br>
until you have some idea about the network. =A0However, you also get a<br>
shot [*one shot*] to retransmit sooner based on the notion that most<br>
RTTs are less than 1sec.<br>
<br>
I think I buy that notion ... that both allows for more aggressiveness<br>
for the common case while also protecting against RTO Hell for the long<br>
path cases. =A0Is that the right walking of the line?</blockquote><div><br>=
I can think of a number of variations. The one-shot 1-sec-initRTO idea you<=
br>described above also came through my mind but the drawback is you only<b=
r>
get one-shot even though we know statistically &gt; 98% of connections<br>h=
ave RTT &lt; 1 sec so most likely the continuous use of 1-sec-initRTO will =
turn<br>out to be better. (A counter argument might be one-shot is &quot;go=
od enough&quot;,<br>
benefitting &gt; 90% of the cases statistically...) The advantage of it is =
its simplicity,<br>restricting the max # of spurious retransmissions caused=
 by the reduced initRTO<br>to 1, and obviously avoiding the RTO hell proble=
m.<br>
<br>A different variation that is slightly more complex but allowing the re=
duced<br>initRTO to be used more than one-time (hence potentially providing=
 more benefit<br>to those &gt; 98% connections) is to only terminate its us=
e upon the detection<br>
of spurious retransmission. Specifically -<br><br>1. initRTO =3D 1sec<br>2.=
 if a valid pkt with the SYN bit on is received (so it could be either SYN =
or<br>SYN-ACK) and the following two conditions are met:<br>=A0 2a. the con=
nection state is neither SYN-SENT nor SYN-RCVD<br>
=A0 2b. RTO =3D=3D 1sec (or no good RTT sample has been taken)<br>Simply re=
set the RTO to 3secs.<br><br>On a second look it doesn&#39;t seem much more=
 complex, although it may be<br>slightly more work to analyze its thoroughn=
ess (e.g., what if all the dup<br>
SYN/SYN-ACK were lost, or all come back late...) Also it does incur<br>more=
 spurious pkts for those connections with RTT &gt; 2sec.<br><br>What do you=
 think?<br><br>Jerry<br><br></div><blockquote class=3D"gmail_quote" style=
=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; p=
adding-left: 1ex;">
<br>
<font color=3D"#888888"><br>
allman<br>
<br>
<br>
<br>
</font></blockquote></div><br>

--0016e64135dcbb6531046fd9623c--

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> I can think of a number of variations. The one-shot 1-sec-initRTO idea
> you described above also came through my mind but the drawback is you
> only get one-shot even though we know statistically > 98% of
> connections have RTT < 1 sec so most likely the continuous use of
> 1-sec-initRTO will turn out to be better. (A counter argument might be
> one-shot is "good enough", benefitting > 90% of the cases
> statistically...) The advantage of it is its simplicity, restricting
> the max # of spurious retransmissions caused by the reduced initRTO to
> 1, and obviously avoiding the RTO hell problem.

Two responses to "one shot":

(1) Yes, one-shot ought to be enough.  Considering losing the SYN,
    retransmitting it using an initRTO of 1sec and reseting the initRTO
    to 3sec.  Now, if there is actually loss in the first RTT of data
    transmission talking about fine-grained performance (i.e., that we
    can get from using 1sec again instead of 3sec) doesn't make a lot of
    sense because 1sec vs. 3sec doesn't matter because performance is
    going to suck no matter what.  So, why bother with anything terribly
    "smart" here?

(2) Using the numbers on your slides it seems to me that the fraction of
    hosts with an RTT of > 1sec is roughly the same as the SYN
    retransmit rate (at an RTT of 3sec, I assume).  To me that says that
    if you use an initRTO of 1sec and then retransmit then the reason
    for that retransmit is just as likely to be loss as it is to be a
    long path.  So, your approach of preferring more than one-shot
    assumes loss.  But, I don't see the measurements you gave as
    suggesting that is the right approach.  The notion of going back to
    3sec just sort of punts.  I.e., the notion is that we have hit a
    situation whereby we don't know what is going on and so let's not
    dogmatically try to push forward, but let's throw up our hands and
    try to do things that ultimately will figure out what is going on.
    And, further, one mistake does not propagate.

So, for me one-shot is just about the right balance here.  Any more than
that we're getting into the corner of a corner case and further in that
corner the empirical evidence is not suggestive of a clear path.  So,
let's just do something that will allow the protocol to get a handle on
things as they are in the specific situation and not try to make guesses
that propagate further.

allman




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From L.Wood@surrey.ac.uk  Thu Jul 30 01:27:59 2009
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Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21st century)
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It's necessary to disambiguate which ACK is for which SYN -=20
i.e. if you have a _very_ long-delay path, you might send out
three SYNs before an ACK to the first SYN comes in. But it appears
at the time to be an ACK to the third SYN, with other
responses lost in transit. So SYN timestamps really are
necessary to disambiguate in this situation. I can see
a cascading sequence where the first ACK comes in just after say
the fifth SYN goes out, and it's immediately necessary to
bump the RTO up to tens of seconds.

related discussion of this is in our TCP Protocol Radius paper:
http://personal.ee.surrey.ac.uk/Personal/L.Wood/publications/index.html#p=
rotocol-radius
esp. Fig 2.

L.

<http://www.ee.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>



-----Original Message-----
From: tcpm-bounces@ietf.org on behalf of Mark Allman
Sent: Tue 2009-07-28 19:35
To: Jerry Chu
Cc: tcpm@ietf.org
Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the =
21st century)
=20

Jerry-

> So the active open side has retransmitted SYN, assuming initRTO of
> 1sec, and eventually received a SYN-ACK but the TS option was
> denied so no good RTT sample can be taken. It will continue to send
> the init data with the initRTO.
>=20
> If the SYN retransmission has been unnecessary because the initRTO has
> been too aggressive, more SYN-ACK will be triggered. Assuming these
> dup SYN-ACKs are not lost, could they be used as a hint to the active
> open side that the initRTO has been too short, so that the RTO timer
> can be reverted back to 3secs immediately, in time to avoid further
> RTO hell?

So, you're saying that if you send N SYNs (N>1) and you get N SYN+ACKs
then you take that to mean the initRTO was too short.

Yeah, perhaps ...=20

Are you suggesting something like this ...

  (1) Use an initRTO of 1sec.

  (2) If the RTO fires and you retransmit the SYN back the RTO to 3sec.
      (And, if necessary you exponential backoff even more.)

  (3) After the 3WHS you will either (a) have a good RTT sample from the
      timestamp option and can therefore set the RTO reasonably or (b)
      you retain the 3sec RTO until you get a good RTT sample.

??

That means that if 1sec is not enough in the 3WHS you have to use 3sec
until you have some idea about the network.  However, you also get a
shot [*one shot*] to retransmit sooner based on the notion that most
RTTs are less than 1sec. =20

I think I buy that notion ... that both allows for more aggressiveness
for the common case while also protecting against RTO Hell for the long
path cases.  Is that the right walking of the line?

allman






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<TITLE>RE: [tcpm] initial RTO (was Re: Tuning TCP parameters for the =
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<P><FONT SIZE=3D2>It's necessary to disambiguate which ACK is for which =
SYN -<BR>
i.e. if you have a _very_ long-delay path, you might send out<BR>
three SYNs before an ACK to the first SYN comes in. But it appears<BR>
at the time to be an ACK to the third SYN, with other<BR>
responses lost in transit. So SYN timestamps really are<BR>
necessary to disambiguate in this situation. I can see<BR>
a cascading sequence where the first ACK comes in just after say<BR>
the fifth SYN goes out, and it's immediately necessary to<BR>
bump the RTO up to tens of seconds.<BR>
<BR>
related discussion of this is in our TCP Protocol Radius paper:<BR>
<A =
HREF=3D"http://personal.ee.surrey.ac.uk/Personal/L.Wood/publications/inde=
x.html#protocol-radius">http://personal.ee.surrey.ac.uk/Personal/L.Wood/p=
ublications/index.html#protocol-radius</A><BR>
esp. Fig 2.<BR>
<BR>
L.<BR>
<BR>
&lt;<A =
HREF=3D"http://www.ee.surrey.ac.uk/Personal/L.Wood/">http://www.ee.surrey=
.ac.uk/Personal/L.Wood/</A>&gt;&lt;L.Wood@surrey.ac.uk&gt;<BR>
<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: tcpm-bounces@ietf.org on behalf of Mark Allman<BR>
Sent: Tue 2009-07-28 19:35<BR>
To: Jerry Chu<BR>
Cc: tcpm@ietf.org<BR>
Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the =
21st century)<BR>
<BR>
<BR>
Jerry-<BR>
<BR>
&gt; So the active open side has retransmitted SYN, assuming initRTO =
of<BR>
&gt; 1sec, and eventually received a SYN-ACK but the TS option was<BR>
&gt; denied so no good RTT sample can be taken. It will continue to =
send<BR>
&gt; the init data with the initRTO.<BR>
&gt;<BR>
&gt; If the SYN retransmission has been unnecessary because the initRTO =
has<BR>
&gt; been too aggressive, more SYN-ACK will be triggered. Assuming =
these<BR>
&gt; dup SYN-ACKs are not lost, could they be used as a hint to the =
active<BR>
&gt; open side that the initRTO has been too short, so that the RTO =
timer<BR>
&gt; can be reverted back to 3secs immediately, in time to avoid =
further<BR>
&gt; RTO hell?<BR>
<BR>
So, you're saying that if you send N SYNs (N&gt;1) and you get N =
SYN+ACKs<BR>
then you take that to mean the initRTO was too short.<BR>
<BR>
Yeah, perhaps ...<BR>
<BR>
Are you suggesting something like this ...<BR>
<BR>
&nbsp; (1) Use an initRTO of 1sec.<BR>
<BR>
&nbsp; (2) If the RTO fires and you retransmit the SYN back the RTO to =
3sec.<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (And, if necessary you exponential =
backoff even more.)<BR>
<BR>
&nbsp; (3) After the 3WHS you will either (a) have a good RTT sample =
from the<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; timestamp option and can therefore set =
the RTO reasonably or (b)<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; you retain the 3sec RTO until you get a =
good RTT sample.<BR>
<BR>
??<BR>
<BR>
That means that if 1sec is not enough in the 3WHS you have to use =
3sec<BR>
until you have some idea about the network.&nbsp; However, you also get =
a<BR>
shot [*one shot*] to retransmit sooner based on the notion that most<BR>
RTTs are less than 1sec.&nbsp;<BR>
<BR>
I think I buy that notion ... that both allows for more =
aggressiveness<BR>
for the common case while also protecting against RTO Hell for the =
long<BR>
path cases.&nbsp; Is that the right walking of the line?<BR>
<BR>
allman<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
</FONT>
</P>

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> It's necessary to disambiguate which ACK is for which SYN - i.e. if
> you have a _very_ long-delay path, you might send out three SYNs
> before an ACK to the first SYN comes in. But it appears at the time to
> be an ACK to the third SYN, with other responses lost in transit. So
> SYN timestamps really are necessary to disambiguate in this
> situation. I can see a cascading sequence where the first ACK comes in
> just after say the fifth SYN goes out, and it's immediately necessary
> to bump the RTO up to tens of seconds.

Sure.

I guess Jerry and I are talking about the case where with exceedingly
high probability the RTT is covered by the current 3sec RTO and the vast
majority of the connections have an RTT of less than 1sec.  (So, say,
99.99999% < 3sec and 98% < 1sec.)  So, how then do we use 1sec for the
common case, but still reasonably accommodate the fraction that have
RTTs > 1sec but < 3sec?

Would it be nice to keep pushing and accommodate connections with even
larger RTTs?  Ideally, yeah.  But, I am not sure imposing timestamps or
other mechanisms on everyone to deal with that case is useful.  At some
point you have to engineer your edge case for what it is.

allman




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SYN timestamps benefit _all_ cases...

<http://www.ee.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>



-----Original Message-----
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Sent: Thu 2009-07-30 12:41
To: Wood L Dr (Electronic Eng)
Cc: hkchu@google.com; tcpm@ietf.org
Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the =
21st century)=20
=20

> It's necessary to disambiguate which ACK is for which SYN - i.e. if
> you have a _very_ long-delay path, you might send out three SYNs
> before an ACK to the first SYN comes in. But it appears at the time to
> be an ACK to the third SYN, with other responses lost in transit. So
> SYN timestamps really are necessary to disambiguate in this
> situation. I can see a cascading sequence where the first ACK comes in
> just after say the fifth SYN goes out, and it's immediately necessary
> to bump the RTO up to tens of seconds.

Sure.

I guess Jerry and I are talking about the case where with exceedingly
high probability the RTT is covered by the current 3sec RTO and the vast
majority of the connections have an RTT of less than 1sec.  (So, say,
99.99999% < 3sec and 98% < 1sec.)  So, how then do we use 1sec for the
common case, but still reasonably accommodate the fraction that have
RTTs > 1sec but < 3sec?

Would it be nice to keep pushing and accommodate connections with even
larger RTTs?  Ideally, yeah.  But, I am not sure imposing timestamps or
other mechanisms on everyone to deal with that case is useful.  At some
point you have to engineer your edge case for what it is.

allman





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<TITLE>RE: [tcpm] initial RTO (was Re: Tuning TCP parameters for the =
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<P><FONT SIZE=3D2>SYN timestamps benefit _all_ cases...<BR>
<BR>
&lt;<A =
HREF=3D"http://www.ee.surrey.ac.uk/Personal/L.Wood/">http://www.ee.surrey=
.ac.uk/Personal/L.Wood/</A>&gt;&lt;L.Wood@surrey.ac.uk&gt;<BR>
<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Mark Allman [<A =
HREF=3D"mailto:mallman@icir.org">mailto:mallman@icir.org</A>]<BR>
Sent: Thu 2009-07-30 12:41<BR>
To: Wood L Dr (Electronic Eng)<BR>
Cc: hkchu@google.com; tcpm@ietf.org<BR>
Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the =
21st century)<BR>
<BR>
<BR>
&gt; It's necessary to disambiguate which ACK is for which SYN - i.e. =
if<BR>
&gt; you have a _very_ long-delay path, you might send out three =
SYNs<BR>
&gt; before an ACK to the first SYN comes in. But it appears at the time =
to<BR>
&gt; be an ACK to the third SYN, with other responses lost in transit. =
So<BR>
&gt; SYN timestamps really are necessary to disambiguate in this<BR>
&gt; situation. I can see a cascading sequence where the first ACK comes =
in<BR>
&gt; just after say the fifth SYN goes out, and it's immediately =
necessary<BR>
&gt; to bump the RTO up to tens of seconds.<BR>
<BR>
Sure.<BR>
<BR>
I guess Jerry and I are talking about the case where with =
exceedingly<BR>
high probability the RTT is covered by the current 3sec RTO and the =
vast<BR>
majority of the connections have an RTT of less than 1sec.&nbsp; (So, =
say,<BR>
99.99999% &lt; 3sec and 98% &lt; 1sec.)&nbsp; So, how then do we use =
1sec for the<BR>
common case, but still reasonably accommodate the fraction that have<BR>
RTTs &gt; 1sec but &lt; 3sec?<BR>
<BR>
Would it be nice to keep pushing and accommodate connections with =
even<BR>
larger RTTs?&nbsp; Ideally, yeah.&nbsp; But, I am not sure imposing =
timestamps or<BR>
other mechanisms on everyone to deal with that case is useful.&nbsp; At =
some<BR>
point you have to engineer your edge case for what it is.<BR>
<BR>
allman<BR>
<BR>
<BR>
<BR>
<BR>
</FONT>
</P>

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From dab@weston.borman.com  Thu Jul 30 05:46:35 2009
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Subject: Re: [tcpm] poll for adopting draft-gont-tcp-security
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Based on the response to this poll, and the subsequent discussion at  
the IETF meeting this week, there is clear consensus to take this on  
as a WG item.  We will be kicking off various discussions in separate  
messages to the TCPM list.

	-David Borman & Wes Eddy, TCPM co-chairs


On Jun 24, 2009, at 9:25 PM, Eddy, Wesley M. (GRC-MS00)[Verizon] wrote:

> TCPMers, there was a thread a while ago about working on
> draft-gont-tcp-security in this working group that didn't
> conclusively give us a feeling one way or other:
> http://www.ietf.org/mail-archive/web/tcpm/current/msg04489.html
>
> Basically, my understanding is that there are at least a
> handful of people in the WG that think it should be done
> here as a WG item (more likely for Informational rather
> than BCP), and there are also some expressed opinions on
> why it shouldn't.
>
> Given the raw size of the document, if the WG intends to
> take this document on, then we need some people to clearly
> commit to putting cycles into review and contributions to
> the document.  Since it is quite large, and to my knowledge,
> there hasn't been a specific technical review of the content
> on this list, but just discussions about if the idea in
> general is a good or bad thing, we still need to know if
> people are willing to invest their time and energy in this.
>
> Please let us know if there is traction for this in the
> near term, and/or we can also discuss it in Stockholm.
>
> ---------------------------
> Wes Eddy
> Network & Systems Architect
> Verizon FNS / NASA GRC
> Office: (216) 433-6682
> ---------------------------
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From hkchu@google.com  Thu Jul 30 09:38:08 2009
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On Wed, Jul 29, 2009 at 9:05 AM, Mark Allman<mallman@icir.org> wrote:
>
>> I can think of a number of variations. The one-shot 1-sec-initRTO idea
>> you described above also came through my mind but the drawback is you
>> only get one-shot even though we know statistically > 98% of
>> connections have RTT < 1 sec so most likely the continuous use of
>> 1-sec-initRTO will turn out to be better. (A counter argument might be
>> one-shot is "good enough", benefitting > 90% of the cases
>> statistically...) The advantage of it is its simplicity, restricting
>> the max # of spurious retransmissions caused by the reduced initRTO to
>> 1, and obviously avoiding the RTO hell problem.
>
> Two responses to "one shot":
>
> (1) Yes, one-shot ought to be enough. =A0Considering losing the SYN,
> =A0 =A0retransmitting it using an initRTO of 1sec and reseting the initRT=
O
> =A0 =A0to 3sec. =A0Now, if there is actually loss in the first RTT of dat=
a
> =A0 =A0transmission talking about fine-grained performance (i.e., that we
> =A0 =A0can get from using 1sec again instead of 3sec) doesn't make a lot =
of
> =A0 =A0sense because 1sec vs. 3sec doesn't matter because performance is
> =A0 =A0going to suck no matter what. =A0So, why bother with anything terr=
ibly
> =A0 =A0"smart" here?

Guess we have some disagreement here and it's not surprise - although
we seem to agree upon the general principles - keep it simple, focus on
the main benefit not the corner cases, but both principles involve some
degree of subjective calls. E.g., the exit condition to initRTO of 1sec
suggested in my previous email doesn't seem all that complex to me.
It only involves a simple check of dupack (or dup SYN/SYN-ACK).

Also I'm not sure why performance has to suck just because a connection
experience early (e.g., SYN/SYN-ACK) loss? I don't remember if TCP
spec requires a connection to go into congestion avoidance mode after
SYN/SYN-ACK loss? Even if so, for short lived connections with small
amount of data (e.g., HTTP/TCP), a speedy recovery from a more optimal
RTO value does still seem matter. Also the reason for a possible speedy
recovery for a loss event AFTER 3WHS seems equally valid as the one
for 3WHS, assuming these loss events are stochastically independent
of each other..,

>
> (2) Using the numbers on your slides it seems to me that the fraction of
> =A0 =A0hosts with an RTT of > 1sec is roughly the same as the SYN
> =A0 =A0retransmit rate (at an RTT of 3sec, I assume). =A0To me that says =
that
> =A0 =A0if you use an initRTO of 1sec and then retransmit then the reason
> =A0 =A0for that retransmit is just as likely to be loss as it is to be a
> =A0 =A0long path. =A0So, your approach of preferring more than one-shot
> =A0 =A0assumes loss. =A0But, I don't see the measurements you gave as
> =A0 =A0suggesting that is the right approach. =A0The notion of going back=
 to
> =A0 =A03sec just sort of punts. =A0I.e., the notion is that we have hit a
> =A0 =A0situation whereby we don't know what is going on and so let's not
> =A0 =A0dogmatically try to push forward, but let's throw up our hands and
> =A0 =A0try to do things that ultimately will figure out what is going on.
> =A0 =A0And, further, one mistake does not propagate.

Again I agree with your statements in general but I just feel one-shot
is a little weak, shy of being "good enough". How about allowing N
shots where N is some TBD (> 1) number? I suspect N=3D2 may satisfy
the "good enough" criterion, taking cue from MacOS (see my slides on
MacOS's repeated 1sec RTO 5 times...) There may be a couple of
different flavors of allowing N shots, either consecutive-only or not but
that's details.

>
> So, for me one-shot is just about the right balance here. =A0Any more tha=
n
> that we're getting into the corner of a corner case and further in that
> corner the empirical evidence is not suggestive of a clear path. =A0So,
> let's just do something that will allow the protocol to get a handle on
> things as they are in the specific situation and not try to make guesses
> that propagate further.

The N-shot solution doesn't add any complexity (just log(N) more bits to
the TCP structure) and seems a strong enough dose for those HTTP
connections suffering the dreadful 3sec pause.

Jerry

>
> allman
>
>
>
>

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Date: Thu, 30 Jul 2009 09:45:58 -0700
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Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21st century)
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On Thu, Jul 30, 2009 at 1:27 AM, <L.Wood@surrey.ac.uk> wrote:
> It's necessary to disambiguate which ACK is for which SYN -
> i.e. if you have a _very_ long-delay path, you might send out
> three SYNs before an ACK to the first SYN comes in. But it appears
> at the time to be an ACK to the third SYN, with other
> responses lost in transit. So SYN timestamps really are
> necessary to disambiguate in this situation. I can see
> a cascading sequence where the first ACK comes in just after say
> the fifth SYN goes out, and it's immediately necessary to
> bump the RTO up to tens of seconds.

One solution I proposed only requires the detection of  sprurious
retransmission so it doesn't need to match ACKs with their SYNs,
as long as any dupack is received...

Jerry

>
> related discussion of this is in our TCP Protocol Radius paper:
> http://personal.ee.surrey.ac.uk/Personal/L.Wood/publications/index.html#p=
rotocol-radius
> esp. Fig 2.
>
> L.
>
> <http://www.ee.surrey.ac.uk/Personal/L.Wood/><L.Wood@surrey.ac.uk>
>
>
>
> -----Original Message-----
> From: tcpm-bounces@ietf.org on behalf of Mark Allman
> Sent: Tue 2009-07-28 19:35
> To: Jerry Chu
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] initial RTO (was Re: Tuning TCP parameters for the 21=
st
> century)
>
>
> Jerry-
>
>> So the active open side has retransmitted SYN, assuming initRTO of
>> 1sec, and eventually received a SYN-ACK but the TS option was
>> denied so no good RTT sample can be taken. It will continue to send
>> the init data with the initRTO.
>>
>> If the SYN retransmission has been unnecessary because the initRTO has
>> been too aggressive, more SYN-ACK will be triggered. Assuming these
>> dup SYN-ACKs are not lost, could they be used as a hint to the active
>> open side that the initRTO has been too short, so that the RTO timer
>> can be reverted back to 3secs immediately, in time to avoid further
>> RTO hell?
>
> So, you're saying that if you send N SYNs (N>1) and you get N SYN+ACKs
> then you take that to mean the initRTO was too short.
>
> Yeah, perhaps ...
>
> Are you suggesting something like this ...
>
> =A0 (1) Use an initRTO of 1sec.
>
> =A0 (2) If the RTO fires and you retransmit the SYN back the RTO to 3sec.
> =A0=A0=A0=A0=A0 (And, if necessary you exponential backoff even more.)
>
> =A0 (3) After the 3WHS you will either (a) have a good RTT sample from th=
e
> =A0=A0=A0=A0=A0 timestamp option and can therefore set the RTO reasonably=
 or (b)
> =A0=A0=A0=A0=A0 you retain the 3sec RTO until you get a good RTT sample.
>
> ??
>
> That means that if 1sec is not enough in the 3WHS you have to use 3sec
> until you have some idea about the network.=A0 However, you also get a
> shot [*one shot*] to retransmit sooner based on the notion that most
> RTTs are less than 1sec.
>
> I think I buy that notion ... that both allows for more aggressiveness
> for the common case while also protecting against RTO Hell for the long
> path cases.=A0 Is that the right walking of the line?
>
> allman
>
>
>
>
>
>

From mallman@icir.org  Thu Jul 30 12:22:36 2009
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Jerry-

Let's put some (rough; based on your slides) numbers to things here ... 

  + With an initRTO of 3sec your data suggests that 98% of the
    connections complete the 3WHS without retransmitting.  So, in 2% of
    the connections we in fact lose a SYN.

  + Also, you note that something like 2% of the connections have an RTT
    longer than 1sec.  (And, I am making the assumption that is really >
    1sec and < 3sec.)

  + So, with an initRTO of 1sec we'd expect to see 2% of the connections
    experience loss, 2% of the connections have a long RTT and
    spuriously retransmit which leaves 96% of the connections Just
    Working.  (All in rough terms.)

  + Forget the 96%... they are good to go.  They got an RTT sample in
    the 3WHS and so presumably are working fine and no longer have to
    worry about the initRTO.

  + The 2% of the connections that experienced loss will have each saved
    2sec in the 3WHS by using an initRTO of 1sec vs. 3sec.  So, if we
    care about X connections that's an aggregate savings of X*0.02*2sec
    when using an initRTO of 1sec versus using an initRTO of 3sec (which
    yields 0sec of savings).

  + The connections that experienced loss will send data in the first
    RTT (say) and experience another 2% loss rate.  If we have a try-2
    approach and again use an initRTO of 1sec then this would save each
    of these connections 2sec over my notion of reverting the initRTO to
    3sec.  In the aggregate the savings here is X*0.0*0.02*2sec.

  + So, now we have saved X*0.02*2sec + X*0.02*0.02*2sec with a try-2
    approach vs. X*0.02*2sec with a try-1 approach.  For X=10K
    connections that is a difference of 8sec in the aggregate (400sec
    with try-1 vs. 408sec with try-2)---or, less than 1msec per
    connection on average if you'd like to do it that way.

  + Then there are the spurious RTOs caused by lowering the initRTO to
    1sec.  We'll have 2% of those in the 3WHS.  The problem is that
    keeping the initRTO at 1sec **ensures** a spurious retransmit in the
    first RTT of data transfer, too.  So, the cwnd will be reduced to an
    MSS, no RTT sample will be taken again, linear increase will be
    forced upon the connection, etc.

  + (Note, I am ignoring connections that use timestamps.  Connections
    that successfully use timestamps will have an RTT sample from the
    3WHS and therefore we don't have to worry about the initRTO
    further.) 

To me the tradeoff is clearly in favor of try-1.  For the advantage of a
*tiny* time savings to the 0.02*0.02 of connections that experience loss
in both the 3WHS and the initial window of data (i.e., what try-2 would
help) you pay by dooming 0.02 of the connections (that now work fine,
BTW) to no exponential ramp up.  That might be a tradeoff you are
personally willing to make---i.e., to sacrifice one type of connection
in favor of another.  But, I don't see that as a good tradeoff for the
standards to make.

Also, note, your scheme of counting SYNs is not overly complicated and
does not have overly onerous state requirements.  I didn't mean to
indicate either of those.  However, it isn't terribly robust either and
I am not ultimately sure how it'd play out.  So, say a connection has a
2sec RTT (works with others, too):
  
  0.0 xmit SYN
  1.0 RTO (==1sec), rexmit SYN
  2.0 rec SYN+ACK (from original transmit) / send ACK / send DATA
  3.0 resend DATA
  3.0 rec SYN+ACK (from retransmit)

Those last two events represent a race condition.  I.e., in this case,
we hope we get the SYN+ACK before we resend the data because then we can
use your scheme to revert to an initRTO of 3sec.  But, we might get it
in the order given above.  And, we might not get that packet at all.
So, it might work and it might not work.  But, the cost of not using it
(possibly saving X*0.02*0.02*2sec) is so small that it seems like
needless complication to me.

Now, there is something you can do here... If you wanted to take the
reception of the SYN+ACK and compare that to the *earliest* SYN
transmission and use that as an RTT sample and then use that to seed the
RTO estimator then fine.  I.e., in this case that'd (correctly) see an
RTT of 2sec.  And, if the original SYN was lost then the returning
SYN+ACK would yield an RTT sample of 3sec.  I.e., using this scheme
might overestimate the RTT, but you won't underestimate it.  If that is
less than 3sec you'd be better off for the first window of data and
you'd be protected against spurious retransmits (to the best of the
standard RTO estimator's abilities) by using a conservative RTT sample.

allman




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