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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Nandita Dukkipati" <nanditad@google.com>, "Matthew Mathis" <mattmathis@google.com>
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Cc: Arvind Jain <arvind@google.com>, tcpm@ietf.org, David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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Hi Nandita, Matt,
=20
As mentioned in the session of ICCRG WG (and to not loose this item from
discussion here at TCPM):


I think it would be good, to break out the results you have gathered so
far (flow completion time in high percentiles, across multiple regions)
also into SACK vs non-SACK flows.

Furthermore, as you explained, IW10 (IW16) has only been verified in
your DC with a SACK-enabled sender, which follows Matt's original
intention of using SACK information to recover from lost
retransmissions. No other stack (that I'm familiar with; opensolaris,
{free,open,net}bsd, windows, aix, hpux, is currently doing that, afaik).


My concern here is, that an umbrella recommendation without any strong
remarks about that fact, will led to implementatiors of other stacks to
increase the IW accordingly.

However, the ensuing burst loss will probably cancel out any gains due
to the possibility of fast recovery. (This is why I would like to see
the flow-competion time for non-SACK flows broken out from your
numbers).


Don't understand me wrong here - I feel absolutely positive that SACK
enabled stacks should be encouraged to make use of the information
contained within SACK, to recover fast from lost retransmissions, and
that most stacks should nowadays always use SACK.

However, the Linux Stack is the only one which is currently doing this
(even though there are simple means by which any SACK stack could be
enabled to gain much of the functionality without the fancy scoreboard
implementation of Linux; I think, even the lwIP SACK patch could be made
to do that under a majority of cases).

Perhaps you should write a seperate draft on that aspect, to extend
RFC2018...


Richard Scheffenegger=20


From lastewart@swin.edu.au  Tue Aug  3 01:19:28 2010
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Date: Tue, 03 Aug 2010 18:19:49 +1000
From: Lawrence Stewart <lastewart@swin.edu.au>
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Cc: David Hayes <dahayes@swin.edu.au>, grenville armitage <garmitage@swin.edu.au>
Subject: [tcpm] Software for FreeBSD TCP R&D
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Hi all,

(apologies for any duplicates you receive and please trim the CC list
appropriately if you reply to this email)

We're pleased to announce the release of a substantial set of new
and updated BSD licenced software for TCP experimentation and research
using FreeBSD:

 - "CAIA-Hamilton Delay" Congestion Control Algorithm v0.1 (new)
 - "Hamilton Delay" Congestion Control Algorithm v0.2 (update)
 - Vegas Congestion Control Algorithm v0.2 (update)
 - H-TCP Congestion Control Algorithm v0.12 (update)
 - CUBIC Congestion Control Algorithm v0.10 (update)
 - Statistical Information For TCP Research (SIFTR) v1.2.3 (update)
 - Enhanced Round Trip Time (ERTT) Khelp Module v0.2 (update)
 - Khelp Framework for FreeBSD v0.1.1 (update)
 - Modular TCP Congestion Control for FreeBSD v0.10.0 (update)

A more detailed summary of the various tools/patches is included at the
end of this email.

Everything mentioned above along with other papers, patches and software
relevant/useful for protocol analysis, debugging and experimental
research is available from:

http://caia.swin.edu.au/urp/newtcp/tools.html

The software was developed as part of the NewTCP research
project at Swinburne University's Centre for Advanced Internet
Architectures, made possible in part by a grant from the Cisco
University Research Program Fund at Community Foundation Silicon
Valley. Testing and development was further assisted by a grant
from the FreeBSD Foundation.

We welcome any feedback and hope you enjoy playing with the code
and tools.

Cheers,
Lawrence Stewart, David Hayes and Grenville Armitage

http://caia.swin.edu.au





NB: "CAIA-Hamilton Delay" and "Hamilton Delay" are our working names for
algorithms which do not have established names in the literature.


"CAIA-Hamilton Delay" Congestion Control Algorithm v0.1 [1,2]
----------------------------------------------------------------------
A FreeBSD loadable kernel module that implements an experimental delay
based congestion control algorithm proposed by the Hamilton Institute
[3] but with modifications made by CAIA [4]. It builds on Hamilton's
initial proposal by adding tolerance to non-congestion related losses
and still aims to allow delay- and loss-based algorithms
to fairly coexist.


"Hamilton Delay" Congestion Control Algorithm v0.2 [5,6]
----------------------------------------------------------------------
A FreeBSD loadable kernel module that implements an experimental delay
based congestion control algorithm proposed by the Hamilton Institute
[3]. It provides a first step toward the ability of delay based
algorithms to fairly coexist with loss based algorithms.


Vegas Congestion Control Algorithm v0.2 [7,8]
----------------------------------------------------------------------
A FreeBSD loadable kernel module that implements the Vegas delay-based
congestion control algorithm [9].


H-TCP Congestion Control Algorithm v0.12 [10,11]
----------------------------------------------------------------------
A FreeBSD loadable kernel module that implements the H-TCP loss based
TCP congestion control algorithm [12].


CUBIC Congestion Control Algorithm v0.10 [13,14]
----------------------------------------------------------------------
A FreeBSD loadable kernel module that implements the CUBIC loss based
TCP congestion control algorithm [15]. CUBIC is the current default
algorithm used by Linux.


Statistical Information For TCP Research (SIFTR) v1.2.3 [16,17]
----------------------------------------------------------------------
A FreeBSD 6/7/8/9 kernel module that logs a range of statistics on
active TCP connections to a log file. It provides the ability to make
highly granular measurements of TCP connection state, aimed at system
administrators, developers and researchers. NB: SIFTR has been imported
into FreeBSD's "head" svn branch (a.k.a. 9-CURRENT) as revision 209662
and will be backported to 8-STABLE in time for 8.2-RELEASE.


Enhanced Round Trip Time (ERTT) Khelp Module v0.2 [18,19]
----------------------------------------------------------------------
A FreeBSD loadable kernel module that provides enhanced RTT
measurements for use by delay and rate based TCP congestion control
algorithms. Robust estimates of RTT are provided even when the receiver
uses delayed acknowledgements, TCP segmentation offload (TSO) or
Selective Acknowledgements (SACK). This module is required by the delay
based congestion control algorithms.


Khelp Framework for FreeBSD v0.1.1 [20,21]
----------------------------------------------------------------------
A FreeBSD kernel patch that provides support for generic kernel modules
known as "helpers" to hook into arbitrary points within the kernel and
provide service(s) to the running system. This forms the foundation for
the ERTT Khelp module.


Modular TCP Congestion Control for FreeBSD v0.10.0 [22,23]
-------------------------------------------------
This revision of the modular TCP congestion control framework
patch provides a current snapshot of the project's development
branch. Some fairly major changes and refactoring have gone into this
release, along with numerous tweaks, bug fixes and enhancements to
support the new delay based congestion control algorithms we have
been working on. In theory, the framework now also supports sharing of
congestion control algorithms between congestion aware transport
protocols like TCP and SCTP, though this integration work is yet to be done.



[1] http://caia.swin.edu.au/urp/newtcp/tools/cc_chd-0.1.tar.gz

[2] http://caia.swin.edu.au/urp/newtcp/tools/cc_chd-readme-0.1.txt

[3] L. Budzisz, R. Stanojevic, R. Shorten, and F. Baker, "A strategy for
fair coexistence of loss and delay-based congestion control algorithms",
IEEE Commun. Lett., vol. 13, no. 7, pp. 555-557, Jul. 2009.

[4] D. A. Hayes and G. Armitage, "Improved coexistence and loss
tolerance for delay based TCP congestion control", in Proceedings of
35th Annual IEEE Conference on Local Computer Networks (LCN 2010),
Denver, Colorado, USA, 11-14 October 2010.

[5] http://caia.swin.edu.au/urp/newtcp/tools/cc_hd-0.2.tar.gz

[6] http://caia.swin.edu.au/urp/newtcp/tools/cc_hd-readme-0.2.txt

[7] http://caia.swin.edu.au/urp/newtcp/tools/cc_vegas-0.2.tar.gz

[8] http://caia.swin.edu.au/urp/newtcp/tools/cc_vegas-readme-0.2.txt

[9] L. S. Brakmo and L. L. Peterson, "TCP Vegas: end to end
congestion avoidance on a global internet", IEEE J. Sel. Areas
Commun., vol. 13, no. 8, pp. 1465-1480, Oct. 1995.

[10] http://caia.swin.edu.au/urp/newtcp/tools/cc_htcp-0.12.tar.gz

[11] http://caia.swin.edu.au/urp/newtcp/tools/cc_htcp-readme-0.12.txt

[12] D. Leith, R. Shorten, "H-TCP: TCP for high-speed and long-distance
networks", in Second International Workshop on Protocols for Fast
Long-Distance Networks, Argonne, Illinois USA, February 2004.

[13] http://caia.swin.edu.au/urp/newtcp/tools/cc_cubic-0.10.tar.gz

[14] http://caia.swin.edu.au/urp/newtcp/tools/cc_cubic-readme-0.10.txt

[15] I. Rhee, L. Xu, S. Ha, "CUBIC for Fast Long-Distance Networks",
Internet Draft, draft-rhee-tcpm-cubic-02.txt, August 2008.

[16] http://caia.swin.edu.au/urp/newtcp/tools/siftr-1.2.3.tar.gz

[17] http://caia.swin.edu.au/urp/newtcp/tools/siftr-readme-1.2.3.txt

[18] http://caia.swin.edu.au/urp/newtcp/tools/h_ertt-0.2.tar.gz

[19] http://caia.swin.edu.au/urp/newtcp/tools/h_ertt-readme-0.2.txt

[20]
http://caia.swin.edu.au/urp/newtcp/tools/caia_khelp_framework_v0.1.1_9.x.r209905.patch

[21] http://caia.swin.edu.au/urp/newtcp/tools/khelp-readme-0.1.1.txt

[22]
http://caia.swin.edu.au/urp/newtcp/tools/caia_modularcc_v0.10.0_9.x.r209905.patch

[23] http://caia.swin.edu.au/urp/newtcp/tools/modularcc-readme-0.10.0.txt

From mattmathis@google.com  Wed Aug  4 13:39:27 2010
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From: Matt Mathis <mattmathis@google.com>
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Cc: Arvind Jain <arvind@google.com>, tcpm@ietf.org, David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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Although I think this is an important point and does need to be
followed up, it is actually inverted relative to the question at hand.

Content providers who are using non-Linux might find some conditions
under which raising IW reduces their performance.  If so they have
grounds to be more cautious about raising IW. In either case, SACK is
(nearly always) work conserving, so the inability to retransmit lost
retransmissions is extremely unlikely to cause additional pain for IW3
users (the fairness question).

If we had done the experimentation with non-Linux, then the
possibility exists that the inability to recover from lost
retransmissions would be protecting IW3 users from our traffic, and
the results couldn't be trusted to apply to Linux.

(I guess this argument does still apply to any other bugs in the Linux
TCP recovery code).

Thanks,
--MM--

On Sun, Aug 1, 2010 at 8:14 AM, Scheffenegger, Richard <rs@netapp.com> wrot=
e:
> Hi Nandita, Matt,
>
> As mentioned in the session of ICCRG WG (and to not loose this item from
> discussion here at TCPM):
>
>
> I think it would be good, to break out the results you have gathered so
> far (flow completion time in high percentiles, across multiple regions)
> also into SACK vs non-SACK flows.
>
> Furthermore, as you explained, IW10 (IW16) has only been verified in
> your DC with a SACK-enabled sender, which follows Matt's original
> intention of using SACK information to recover from lost
> retransmissions. No other stack (that I'm familiar with; opensolaris,
> {free,open,net}bsd, windows, aix, hpux, is currently doing that, afaik).
>
>
> My concern here is, that an umbrella recommendation without any strong
> remarks about that fact, will led to implementatiors of other stacks to
> increase the IW accordingly.
>
> However, the ensuing burst loss will probably cancel out any gains due
> to the possibility of fast recovery. (This is why I would like to see
> the flow-competion time for non-SACK flows broken out from your
> numbers).
>
>
> Don't understand me wrong here - I feel absolutely positive that SACK
> enabled stacks should be encouraged to make use of the information
> contained within SACK, to recover fast from lost retransmissions, and
> that most stacks should nowadays always use SACK.
>
> However, the Linux Stack is the only one which is currently doing this
> (even though there are simple means by which any SACK stack could be
> enabled to gain much of the functionality without the fancy scoreboard
> implementation of Linux; I think, even the lwIP SACK patch could be made
> to do that under a majority of cases).
>
> Perhaps you should write a seperate draft on that aspect, to extend
> RFC2018...
>
>
> Richard Scheffenegger
>
>



--=20
--MM--
The best way to predict the future is to create it. =A0- Peter Drucker

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Thread-Topic: [tcpm] Increasing TCP's Initial Congestion Window
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References: <AANLkTi=+7r2tqHap5pSPEwUPHOWDb+0VwX1yfA6gkwxc@mail.gmail.com><C304DB494AC0C04C87C6A6E2FF5603DB48131ED3A5@NDJSSCC01.ndc.nasa.gov><AANLkTinXx=m+DO8gwqmXH+cC1kMi-T5uVxet_y0Kh7nL@mail.gmail.com><5FDC413D5FA246468C200652D63E627A09C48990@LDCMVEXC1-PRD.hq.netapp.com> <AANLkTinSUSJCOecf0D89dHuXAbkRtPWeeFr+bedUSD5i@mail.gmail.com>
From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Matt Mathis" <mattmathis@google.com>
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Cc: Arvind Jain <arvind@google.com>, tcpm@ietf.org, David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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Hi Matt,


I believe the "inability to retransmit lost retransmissions", which is =
the current default behavior (except for Linux), DOES permanently cause =
every receiver pain in the form of lengthy RTO recoveries and long flow =
completion times. And not necessarily only during slow start and the =
initial few segments of the flow, but also during steady state, when =
some other flow's burst causes this effect for some victim flow.

Or, put differently, IW10 may increase that problem (due to a higher =
probability of lost retransmissions) over IW3 - and you'd be actually =
penalizing IW10 flows (running into RTO) and the this proposal may see =
the same fate as ECN deployment.

(Due to some broken algorithms/implementations somewhere else, a very =
good idea would be blamed).

Therefore I would think, had you done your expirimentation with a =
"standard" SACK stack, you would have never bothered writing an I-D, as =
you wouldn't have seen any merit. At least that is what I would have =
expected as the outcome...


Thus I would like to see some strong recommendations, or an add-on RFC =
to 2018 /5681, to explicitly recommend lost retransmission detected =
*before* IW10 is implemented in a stack...

(Perhaps you should have been more insistent back then on how to deal =
with the wealth of info given by SACK to the sender ;) ).


Richard Scheffenegger


=20

> -----Original Message-----
> From: Matt Mathis [mailto:mattmathis@google.com]=20
> Sent: Mittwoch, 4. August 2010 22:40
> To: Scheffenegger, Richard
> Cc: Nandita Dukkipati; tcpm@ietf.org; David Borman; Arvind=20
> Jain; Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
> Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
>=20
> Although I think this is an important point and does need to=20
> be followed up, it is actually inverted relative to the=20
> question at hand.
>=20
> Content providers who are using non-Linux might find some=20
> conditions under which raising IW reduces their performance. =20
> If so they have grounds to be more cautious about raising IW.=20
> In either case, SACK is (nearly always) work conserving, so=20
> the inability to retransmit lost retransmissions is extremely=20
> unlikely to cause additional pain for IW3 users (the fairness=20
> question).
>=20
> If we had done the experimentation with non-Linux, then the=20
> possibility exists that the inability to recover from lost=20
> retransmissions would be protecting IW3 users from our=20
> traffic, and the results couldn't be trusted to apply to Linux.
>=20
> (I guess this argument does still apply to any other bugs in=20
> the Linux TCP recovery code).
>=20
> Thanks,
> --MM--
>=20
> On Sun, Aug 1, 2010 at 8:14 AM, Scheffenegger, Richard=20
> <rs@netapp.com> wrote:
> > Hi Nandita, Matt,
> >
> > As mentioned in the session of ICCRG WG (and to not loose this item=20
> > from discussion here at TCPM):
> >
> >
> > I think it would be good, to break out the results you have=20
> gathered=20
> > so far (flow completion time in high percentiles, across multiple=20
> > regions) also into SACK vs non-SACK flows.
> >
> > Furthermore, as you explained, IW10 (IW16) has only been=20
> verified in=20
> > your DC with a SACK-enabled sender, which follows Matt's original=20
> > intention of using SACK information to recover from lost=20
> > retransmissions. No other stack (that I'm familiar with;=20
> opensolaris,=20
> > {free,open,net}bsd, windows, aix, hpux, is currently doing=20
> that, afaik).
> >
> >
> > My concern here is, that an umbrella recommendation without=20
> any strong=20
> > remarks about that fact, will led to implementatiors of=20
> other stacks=20
> > to increase the IW accordingly.
> >
> > However, the ensuing burst loss will probably cancel out=20
> any gains due=20
> > to the possibility of fast recovery. (This is why I would=20
> like to see=20
> > the flow-competion time for non-SACK flows broken out from your=20
> > numbers).
> >
> >
> > Don't understand me wrong here - I feel absolutely positive=20
> that SACK=20
> > enabled stacks should be encouraged to make use of the information=20
> > contained within SACK, to recover fast from lost=20
> retransmissions, and=20
> > that most stacks should nowadays always use SACK.
> >
> > However, the Linux Stack is the only one which is currently=20
> doing this=20
> > (even though there are simple means by which any SACK stack=20
> could be=20
> > enabled to gain much of the functionality without the fancy=20
> scoreboard=20
> > implementation of Linux; I think, even the lwIP SACK patch could be=20
> > made to do that under a majority of cases).
> >
> > Perhaps you should write a seperate draft on that aspect, to extend=20
> > RFC2018...
> >
> >
> > Richard Scheffenegger
> >
> >
>=20
>=20
>=20
> --
> --MM--
> The best way to predict the future is to create it. =A0- Peter Drucker
>=20

From mattmathis@google.com  Wed Aug  4 16:59:03 2010
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From: Matt Mathis <mattmathis@google.com>
To: "Scheffenegger, Richard" <rs@netapp.com>
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Cc: Arvind Jain <arvind@google.com>, tcpm@ietf.org, David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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On Wed, Aug 4, 2010 at 7:17 PM, Scheffenegger, Richard <rs@netapp.com> wrot=
e:
> I believe the "inability to retransmit lost retransmissions", which is th=
e current default behavior (except for Linux), DOES permanently cause every=
 receiver pain in the form of lengthy RTO recoveries and long flow completi=
on times. And not necessarily only during slow start and the initial few se=
gments of the flow, but also during steady state, when some other flow's bu=
rst causes this effect for some victim flow.
>
Point taken, although the WG concern is mostly about what happens to
somebody else's flows.   Content providers should always regression
test new OS rollouts, and if by raising IW they hurt themselves (say
because their web page opens 83 concurrent connections), they should
fix something.   Yes, it is possible with a different stack we would
have chosen a different number, and we don't know if the details of
the SACK implementation affect our results much.

> Or, put differently, IW10 may increase that problem (due to a higher prob=
ability of lost retransmissions) over IW3 - and you'd be actually penalizin=
g IW10 flows (running into RTO) and the this proposal may see the same fate=
 as ECN deployment.
>

It is not the same as ECN at all, because it permits up to IW10 -
users (content providers, OS vendors) can choose their own values.
There should be a sweet spot at 5 or 6 as well (because Fast
Retransmit starts working), which is actually a logical intermediate
step for a large content provider or OS vendor who wants to do
something, but is extra cautious or otherwise not totally confident of
the measurement studies.

> (Due to some broken algorithms/implementations somewhere else, a very goo=
d idea would be blamed).
>
> Therefore I would think, had you done your expirimentation with a "standa=
rd" SACK stack, you would have never bothered writing an I-D, as you wouldn=
't have seen any merit. At least that is what I would have expected as the =
outcome...
>
>
> Thus I would like to see some strong recommendations, or an add-on RFC to=
 2018 /5681, to explicitly recommend lost retransmission detected *before* =
IW10 is implemented in a stack...
>
> (Perhaps you should have been more insistent back then on how to deal wit=
h the wealth of info given by SACK to the sender ;) ).
>
I actually abandoned the ID for a reason.  Here at Google, I am
considering re-opening it.   2581/5681 actually forbid the correct
answer, which I could not argue when we dropped the FACK ID.   Now I
can make a better argument....  It would get ugly because it conflicts
with many existing standards.   And Fred would not be very happy at
all.

(Search for mathis congestion control easter egg )

Thanks,
--MM--

From mallman@icir.org  Thu Aug 12 06:16:48 2010
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Folks-

I have just read through the thread on seeding the RTO during the 3WHS
when using larger initial windows.  The notes I sent a few weeks back
still apply in my mind.  But, I think there is a big picture issue we
need to puzzle through before we consider when to seed the RTO ...

Whether we seed in the 3WHS or the first flight of data seems immaterial
if we reduce the default initial RTO from 3sec to 1sec (a proposal that
is out there in draft-paxson-tcpm-rfc2988bis-00.txt).  I.e., regardless
of whether you seed in the 3WHS or not the RTO is going to end up in the
same place when transmitting the first data segment.  

So, it seems to me that before we try to puzzle through this seeding
business we should figure out if we just want to make the change to the
default value.  If we do then lets forget this seeding business.  If we
don't then we can consider whether this seeding in the 3WHS is a good
idea or not.

My two bits ...

allman


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




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From: Jerry Chu <hkchu@google.com>
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Subject: Re: [tcpm] reset: seeding RTO in 3WHS
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Mark,

On Thu, Aug 12, 2010 at 6:17 AM, Mark Allman <mallman@icir.org> wrote:
>
> Folks-
>
> I have just read through the thread on seeding the RTO during the 3WHS
> when using larger initial windows. =A0The notes I sent a few weeks back
> still apply in my mind. =A0But, I think there is a big picture issue we
> need to puzzle through before we consider when to seed the RTO ...
>
> Whether we seed in the 3WHS or the first flight of data seems immaterial
> if we reduce the default initial RTO from 3sec to 1sec (a proposal that
> is out there in draft-paxson-tcpm-rfc2988bis-00.txt). =A0I.e., regardless
> of whether you seed in the 3WHS or not the RTO is going to end up in the
> same place when transmitting the first data segment.
>
> So, it seems to me that before we try to puzzle through this seeding
> business we should figure out if we just want to make the change to the
> default value. =A0If we do then lets forget this seeding business. =A0If =
we
> don't then we can consider whether this seeding in the 3WHS is a good
> idea or not.

Agreed 100%. So the question really is if the WG is ready to adopt
draft-paxson-tcpm-rfc2988bis-00.txt. I think the only outstanding question =
on
the initRTO=3D1sec proposal is why not try to reduce minRTO as well and I
was asking why the two things need to go together (see my 6/28/10 email).

Since you have done some study on the overall impact of minRTO perhaps
you can share your thought on this?

Thanks,

Jerry

>
> My two bits ...
>
> allman
>
>
> --
> Mark Allman -- ICSI -- http://www.icir.org/mallman/
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>

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To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Thu, 12 Aug 2010 11:48:36 -0500
Thread-Topic: TCP persist timer min and max values (specified in RFC?)
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Subject: [tcpm] FW: TCP persist timer min and max values (specified in RFC?)
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Asked on the IETF main list; kevin is at kfall@cs.berkeley.edu for
replies:

>-----Original Message-----
>From: ietf-bounces@ietf.org [mailto:ietf-bounces@ietf.org] On Behalf Of
>Kevin Fall
>Sent: Wednesday, August 11, 2010 10:31 PM
>To: ietf@ietf.org
>Subject: TCP persist timer min and max values (specified in RFC?)
>
>
>Quick and simple question---
>
>I'm trying to determine whether the range for the TCP persist timer
>([5,60] seconds commonly) is specified by RFC, or instead more of a
>historical artifact based on implementation.  If it is specified by RFC,
>could somebody please point me at it?
>
>thanks,
>- K
>_______________________________________________
>Ietf mailing list
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From mallman@icir.org  Thu Aug 12 13:15:18 2010
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Jerry-

> I think the only outstanding question on the initRTO=1sec proposal is
> why not try to reduce minRTO as well and I was asking why the two
> things need to go together (see my 6/28/10 email).
> 
> Since you have done some study on the overall impact of minRTO perhaps
> you can share your thought on this?

I have done so previously (Feb/2010 and at other times) and just went
and grabbed the text.  Basically, I think changing the initial RTO (as
in our draft) is fine, but the min seems like a much murkier situation
to me.  So, I'd rather not block on changing the min to see the initial
RTO changed.  

The old text:

  One problem with the minimum is we don't know what to set it at [*].
  The finding in [AP99] is that as you reduce the min you take more
  spurious timeouts.  The tradeoff seemed fundamental to us.  But, given
  our methodology (trace driven simulation) we could not really tell the
  impact of the spurious timeouts.  So, we just sought to minimize them
  in the spec.  I know some TCP implementations are being pretty
  aggressive with the min these days.  But, it is not at all clear to me
  whether this is a net win or net lose.  I.e., is the win in terms of
  peppiness to needed retransmits still a win when it is played against
  the consequences of a spurious transmit (i.e., setting cwnd = 1)?
  Perhaps there is a sweet spot that is < 1sec whereby this tradeoff
  better balances and we should be using that.  But, it'd take a big
  experiment to show that, I think.

  [*] Maybe there is work that I just don't know about and if so I'd
      like to be pointed at it.

Also in that Feb email was a wondering I have off-and-on had ...

  Another thing I have wondered about from time to time .... Much like
  Early Retransmit could we have some sort of early RTO when the cwnd
  size is small?  When the window size is small it seems that the
  downside of making a bad rexmt decision is not as bad as when the cwnd
  is large so it might pay in terms of responsiveness to RTO sooner.
  Also, as the cwnd gets larger we have better loss recovery techniques
  that should come into play that can repair lots of loss without
  involving the RTO and so perhaps by being overly aggressive in the RTO
  we just hurt ourselves.  Finally, as the cwnd gets bigger that means
  we are transferring more data.  And, the longer a connection goes the
  less important the fine-grained responsiveness matters.  Just a random
  thought ...

(Note, I would not advocate that for inclusion in an 2988bis.  If there
was interest in working this out then that'd seem to me to be something
experimental at first.)

allman




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Hi Jerry,

I had the same discussion with Mark a while ago, whether or not ever
we should include the reducing minRTO stuff into the 2988bis doc or
not. Mark persuaded my that it is better to do not do so in order to
not block the doc.

However, since feb, nothing is happened... So I think we should decide
NOW if the WG would like to adopt the doc as it is (without minRTO
adjustment) or not.

Alex

Am 12.08.2010 um 22:15 schrieb Mark Allman:

> 
> Jerry-
> 
>> I think the only outstanding question on the initRTO=1sec proposal is
>> why not try to reduce minRTO as well and I was asking why the two
>> things need to go together (see my 6/28/10 email).
>> 
>> Since you have done some study on the overall impact of minRTO perhaps
>> you can share your thought on this?
> 
> 
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] TCP persist timer min and max values (specified in RFC?)
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Kevin Fall wrote:

> I'm trying to determine whether the range for the TCP persist timer
> ([5,60] seconds commonly) is specified by RFC, or instead more of a
> historical artifact based on implementation.  If it is specified by
> RFC, could somebody please point me at it?

>From Section 4.2.2.17 of RFC 1122:

>             The transmitting host SHOULD send the first zero-window
>             probe when a zero window has existed for the retransmission
>             timeout period (see Section 4.2.2.15), and SHOULD increase
>             exponentially the interval between successive probes.
> 
>             DISCUSSION:
>                  This procedure minimizes delay if the zero-window
>                  condition is due to a lost ACK segment containing a
>                  window-opening update.  Exponential backoff is
>                  recommended, possibly with some maximum interval not
>                  specified here.  This procedure is similar to that of
>                  the retransmission algorithm, and it may be possible to
>                  combine the two procedures in the implementation.

Then from Section 4.2.3.1 of the same RFC:

>             The recommended upper and lower bounds on the RTO are known
>             to be inadequate on large internets.  The lower bound SHOULD
>             be measured in fractions of a second (to accommodate high
>             speed LANs) and the upper bound should be 2*MSL, i.e., 240
>             seconds.

Thus that the range *is* specified by the RFC (not hardcoded, but
derived from the RTO).

Regarding the upper limit of the RTO (and hence the persist timer), in
theory it is derived from the MSL, which in turn is derived from the
recommended TTL (60). However, most implementations use different values
for the TTL (either 255 or a power of of 2, such as 64 or 128). This,
together with the fact that the TTL is assumed to be a hop limit (rather
than a time count) might leave some room for using a different upper
limit for the RTO (and hence the persist timer).

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





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Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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Hi Richard,

We monitor the "lost retransmissions" reported by Linux
(LINUX_MIB_TCPLOSTRETRANSMIT). They consist of less than <0.5% of all
retransmissions in both IW3 and IW10 flows in our experiments.
Therefore I don't think this feature is critical.

Yuchung

On Wed, Aug 4, 2010 at 4:17 PM, Scheffenegger, Richard <rs@netapp.com> wrot=
e:
>
> Hi Matt,
>
>
> I believe the "inability to retransmit lost retransmissions", which is th=
e current default behavior (except for Linux), DOES permanently cause every=
 receiver pain in the form of lengthy RTO recoveries and long flow completi=
on times. And not necessarily only during slow start and the initial few se=
gments of the flow, but also during steady state, when some other flow's bu=
rst causes this effect for some victim flow.
>
> Or, put differently, IW10 may increase that problem (due to a higher prob=
ability of lost retransmissions) over IW3 - and you'd be actually penalizin=
g IW10 flows (running into RTO) and the this proposal may see the same fate=
 as ECN deployment.
>
> (Due to some broken algorithms/implementations somewhere else, a very goo=
d idea would be blamed).
>
> Therefore I would think, had you done your expirimentation with a "standa=
rd" SACK stack, you would have never bothered writing an I-D, as you wouldn=
't have seen any merit. At least that is what I would have expected as the =
outcome...
>
>
> Thus I would like to see some strong recommendations, or an add-on RFC to=
 2018 /5681, to explicitly recommend lost retransmission detected *before* =
IW10 is implemented in a stack...
>
> (Perhaps you should have been more insistent back then on how to deal wit=
h the wealth of info given by SACK to the sender ;) ).
>
>
> Richard Scheffenegger
>
>
>
>
>> -----Original Message-----
>> From: Matt Mathis [mailto:mattmathis@google.com]
>> Sent: Mittwoch, 4. August 2010 22:40
>> To: Scheffenegger, Richard
>> Cc: Nandita Dukkipati; tcpm@ietf.org; David Borman; Arvind
>> Jain; Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
>> Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
>>
>> Although I think this is an important point and does need to
>> be followed up, it is actually inverted relative to the
>> question at hand.
>>
>> Content providers who are using non-Linux might find some
>> conditions under which raising IW reduces their performance.
>> If so they have grounds to be more cautious about raising IW.
>> In either case, SACK is (nearly always) work conserving, so
>> the inability to retransmit lost retransmissions is extremely
>> unlikely to cause additional pain for IW3 users (the fairness
>> question).
>>
>> If we had done the experimentation with non-Linux, then the
>> possibility exists that the inability to recover from lost
>> retransmissions would be protecting IW3 users from our
>> traffic, and the results couldn't be trusted to apply to Linux.
>>
>> (I guess this argument does still apply to any other bugs in
>> the Linux TCP recovery code).
>>
>> Thanks,
>> --MM--
>>
>> On Sun, Aug 1, 2010 at 8:14 AM, Scheffenegger, Richard
>> <rs@netapp.com> wrote:
>> > Hi Nandita, Matt,
>> >
>> > As mentioned in the session of ICCRG WG (and to not loose this item
>> > from discussion here at TCPM):
>> >
>> >
>> > I think it would be good, to break out the results you have
>> gathered
>> > so far (flow completion time in high percentiles, across multiple
>> > regions) also into SACK vs non-SACK flows.
>> >
>> > Furthermore, as you explained, IW10 (IW16) has only been
>> verified in
>> > your DC with a SACK-enabled sender, which follows Matt's original
>> > intention of using SACK information to recover from lost
>> > retransmissions. No other stack (that I'm familiar with;
>> opensolaris,
>> > {free,open,net}bsd, windows, aix, hpux, is currently doing
>> that, afaik).
>> >
>> >
>> > My concern here is, that an umbrella recommendation without
>> any strong
>> > remarks about that fact, will led to implementatiors of
>> other stacks
>> > to increase the IW accordingly.
>> >
>> > However, the ensuing burst loss will probably cancel out
>> any gains due
>> > to the possibility of fast recovery. (This is why I would
>> like to see
>> > the flow-competion time for non-SACK flows broken out from your
>> > numbers).
>> >
>> >
>> > Don't understand me wrong here - I feel absolutely positive
>> that SACK
>> > enabled stacks should be encouraged to make use of the information
>> > contained within SACK, to recover fast from lost
>> retransmissions, and
>> > that most stacks should nowadays always use SACK.
>> >
>> > However, the Linux Stack is the only one which is currently
>> doing this
>> > (even though there are simple means by which any SACK stack
>> could be
>> > enabled to gain much of the functionality without the fancy
>> scoreboard
>> > implementation of Linux; I think, even the lwIP SACK patch could be
>> > made to do that under a majority of cases).
>> >
>> > Perhaps you should write a seperate draft on that aspect, to extend
>> > RFC2018...
>> >
>> >
>> > Richard Scheffenegger
>> >
>> >
>>
>>
>>
>> --
>> --MM--
>> The best way to predict the future is to create it. =A0- Peter Drucker
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From wesley.m.eddy@nasa.gov  Tue Aug 17 08:10:22 2010
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From: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
To: Joe Touch <touch@isi.edu>, "tcpm@ietf.org" <tcpm@ietf.org>
Date: Tue, 17 Aug 2010 10:10:48 -0500
Thread-Topic: [tcpm] Increasing TCP's Initial Congestion Window
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Cc: Arvind Jain <arvind@google.com>, Mathis <mattmathis@google.com>
Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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>-----Original Message-----
>From: Joe Touch [mailto:touch@isi.edu]
>Sent: Thursday, July 29, 2010 5:42 AM
>To: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
>Cc: Nandita Dukkipati; tcpm@ietf.org; David Borman; Arvind Jain; Mathis;
>Matthew@core3.amsl.com
>Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
>
>
>
>On 7/23/2010 12:37 PM, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
>wrote:
>...
>> David and I just talked about this and propose to add a milestone to
>> the charter for late 2011, if the working group agrees, to publish
>> this document on increasing TCP's initial window, based on the large
>> amount of discussion both in the meetings and on the list.  If anyone
>> would object to this, please let us know in the next 2 weeks.
>>
>> One thing I still don't understand, is how we know that the intended
>> status for standards track is appropriate at this point ... from
>> talking to David it sounds like the working group can determine that
>> after deciding to adopt it, and we don't have to have it decided ahead
>> of time.
>
>I don't think that's appropriate process. We need to decide when we
>adopt an item what its intended status will be, IMO.
>
>Otherwise, it's too early to consider adopting it as a work item IMO.
>(that's not to say we can't continue to discuss it, though).


Hi Joe, I checked with Lars on the process question of whether we
can adopt documents with undetermined target status.  He indicated
that it would be reasonable to add a milestone to determine what
the appropriate level to publish the potential document at is, followed
by a milestone to actually publish the document a short time later.

I propose that the first milestone to determine status between
Experimental and Proposed Standard could have a cutoff-date at which
time it would default to the lower option (Experimental) if there
isn't clear consensus and comfort with Proposed Standard at that time.
This assumes there were no problems discovered with deployment at that
level in the meantime.  If so, those problems would probably get
documented as Informational.  Experimental is an appropriate default
since it has nearly been deployed at that level already.

The main question I have, and would like feedback from the authors and
the rest of the working group on, is how far ahead to set that first
milestone to determine status?  Do we need to think about this for a
year (i.e. 3 more meetings)?  Shorter?  Longer?  I would guess after
determining the publication track, we'd only spend a month touching up
the document and last calling it for the 2nd milestone to publish the
document.

Any feedback on this would be appreciated.

--
Wes Eddy
MTI Systems


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Subject: Re: [tcpm] reset: seeding RTO in 3WHS
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Wes & David,

It looks like 2988bis is ready for adoption as a WG item. Please let us know
what to do next.

Notat that besides reducing minRTO, a couple of other issues have been
brought up and discussed, one from Gorry Fairhurst on the implication of
SYN RTO expiry, the other from Markku on slow links with a larger IW.
We (Mark/Vern/I) will revise the draft to capture both discussions in the near
future.

Thanks,

Jerry

On Fri, Aug 13, 2010 at 1:42 AM, Alexander Zimmermann
<alexander.zimmermann@comsys.rwth-aachen.de> wrote:
>
> Hi Jerry,
>
> I had the same discussion with Mark a while ago, whether or not ever
> we should include the reducing minRTO stuff into the 2988bis doc or
> not. Mark persuaded my that it is better to do not do so in order to
> not block the doc.
>
> However, since feb, nothing is happened... So I think we should decide
> NOW if the WG would like to adopt the doc as it is (without minRTO
> adjustment) or not.
>
> Alex
>
> Am 12.08.2010 um 22:15 schrieb Mark Allman:
>
> >
> > Jerry-
> >
> >> I think the only outstanding question on the initRTO=1sec proposal is
> >> why not try to reduce minRTO as well and I was asking why the two
> >> things need to go together (see my 6/28/10 email).
> >>
> >> Since you have done some study on the overall impact of minRTO perhaps
> >> you can share your thought on this?
> >
> >
> >
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>

From hkchu@google.com  Tue Aug 17 23:11:25 2010
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Cc: Arvind Jain <arvind@google.com>, "tcpm@ietf.org" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>, Mathis <mattmathis@google.com>
Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
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Hi Wes,

Thanks for trying to move this forward quickly. The two-milestone approach
sounds very reasonable to us. Regarding what the appropriate time line
might be for the first milestone, I think this mostly depends on how many
concerns we have successfully addressed vs how many still remaining.
It also depends on how much help folks on this list are willing to provide
to cover the remaining concerns.

To answer the question it may help to recount what we've gotten so far:

1. Results from a number of large scale experiments at Google's data
centers presented at IETF/Anaheim.

http://www.ietf.org/proceedings/77/slides/tcpm-4.pdf

2. Details from the above published in A SIGCOMM CCR paper in July.

http://ccr.sigcomm.org/online/?q=3Dnode/621

3. "A Simulation Study on Increasing TCP's IW" presented at
ICCRG/Maastricht by Ilpo Jarvinen on the effect of IW10 over wireless links=
.

4. Some fairness study by Michael Scharf as part of his PhD thesis that was
sent to the ICCRG list on 7/25/10.

5. Results from another round of experiments plus a testbed study by us in
an attempt to study and address concerns raised at Anaheim.

http://www.ietf.org/proceedings/78/slides/iccrg-3.pdf

Since our presentation was the last one at the ICCRG meeting and the
meeting adjourned shortly after, plus it was the last day of the IETF week,
we did not receive much feedback on how our data address, or even just
partially alleviate the concern on how IW10 flows affect other flows with a
smaller IW, for example. (There was some suggestion related to our testbed.
But unfortunately we have lost our intern back to school so not much has
been done since.)

Thus it will be helpful if people can throw feedbacks/comments/suggestions
at us now on our results so far so we can better sense where we are at
right now and how much further we need to go to get to Proposed Standard.

Thanks,

Jerry

On Tue, Aug 17, 2010 at 8:10 AM, Eddy, Wesley M. (GRC-MS00)[ASRC
AEROSPACE CORP] <wesley.m.eddy@nasa.gov> wrote:
>>-----Original Message-----
>>From: Joe Touch [mailto:touch@isi.edu]
>>Sent: Thursday, July 29, 2010 5:42 AM
>>To: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
>>Cc: Nandita Dukkipati; tcpm@ietf.org; David Borman; Arvind Jain; Mathis;
>>Matthew@core3.amsl.com
>>Subject: Re: [tcpm] Increasing TCP's Initial Congestion Window
>>
>>
>>
>>On 7/23/2010 12:37 PM, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
>>wrote:
>>...
>>> David and I just talked about this and propose to add a milestone to
>>> the charter for late 2011, if the working group agrees, to publish
>>> this document on increasing TCP's initial window, based on the large
>>> amount of discussion both in the meetings and on the list. =A0If anyone
>>> would object to this, please let us know in the next 2 weeks.
>>>
>>> One thing I still don't understand, is how we know that the intended
>>> status for standards track is appropriate at this point ... from
>>> talking to David it sounds like the working group can determine that
>>> after deciding to adopt it, and we don't have to have it decided ahead
>>> of time.
>>
>>I don't think that's appropriate process. We need to decide when we
>>adopt an item what its intended status will be, IMO.
>>
>>Otherwise, it's too early to consider adopting it as a work item IMO.
>>(that's not to say we can't continue to discuss it, though).
>
>
> Hi Joe, I checked with Lars on the process question of whether we
> can adopt documents with undetermined target status. =A0He indicated
> that it would be reasonable to add a milestone to determine what
> the appropriate level to publish the potential document at is, followed
> by a milestone to actually publish the document a short time later.
>
> I propose that the first milestone to determine status between
> Experimental and Proposed Standard could have a cutoff-date at which
> time it would default to the lower option (Experimental) if there
> isn't clear consensus and comfort with Proposed Standard at that time.
> This assumes there were no problems discovered with deployment at that
> level in the meantime. =A0If so, those problems would probably get
> documented as Informational. =A0Experimental is an appropriate default
> since it has nearly been deployed at that level already.
>
> The main question I have, and would like feedback from the authors and
> the rest of the working group on, is how far ahead to set that first
> milestone to determine status? =A0Do we need to think about this for a
> year (i.e. 3 more meetings)? =A0Shorter? =A0Longer? =A0I would guess afte=
r
> determining the publication track, we'd only spend a month touching up
> the document and last calling it for the 2nd milestone to publish the
> document.
>
> Any feedback on this would be appreciated.
>
> --
> Wes Eddy
> MTI Systems
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From wesley.m.eddy@nasa.gov  Thu Aug 19 18:40:58 2010
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To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Thu, 19 Aug 2010 20:41:31 -0500
Thread-Topic: initiating WGLC for draft-eggert-tcpm-historicize
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Subject: [tcpm] initiating WGLC for draft-eggert-tcpm-historicize
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Following up on the earlier thread to adopt draft-eggert-tcpm-historicize a=
s a TCPM working group item, there seems to be positive support and no obje=
ction to the document itself, though there was a lot of discussion on the a=
lternate checksum option going forward.

Based on the nature of this document (not a technical spec, just cleaning u=
p the status of other documents), the authors and chairs think we can just =
do a one-month working group last call in order to complete this activity.

The document can be found at:
http://tools.ietf.org/html/draft-eggert-tcpm-historicize-00

Please review it, and send your comments, concerns, and critiques to the TC=
PM list by September 17.  If this is too soon, and you think more time is w=
arranted, this would be a fair comment to raise as well.

Thanks in advance.=

From pekkas@netcore.fi  Thu Aug 19 22:50:49 2010
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On Thu, 19 Aug 2010, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP] wrote:
> Please review it, and send your comments, concerns, and critiques to 
> the TCPM list by September 17.  If this is too soon, and you think 
> more time is warranted, this would be a fair comment to raise as 
> well.

Support.

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

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--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           : On the implementation of the TCP urgent mechanism
	Author(s)       : F. Gont, A. Yourtchenko
	Filename        : draft-ietf-tcpm-urgent-data-06.txt
	Pages           : 13
	Date            : 2010-08-23

This document analyzes how current TCP implementations process TCP
urgent indications, and how the behavior of some widely-deployed
middle-boxes affect how urgent indications are processed by end
systems.  This document updates the relevant specifications such that
they accommodate current practice in processing TCP urgent
indications, provides advice to applications that make use of the
urgent mechanism, and raises awareness about the reliability of TCP
urgent indications in the current Internet.

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

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Subject: [tcpm] TCP window max size?
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--000e0cd2281efff3d3048e929bd3
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Hi Fellows.

(Apologies if this is not the correct forum to this e-mail, but the tcplw WG
and list are already concluded.)

The RFC-1323 specifies the TCP Window Scale Option. It states that the
maximum window size is 2^30 (1Gbyte), please see the last paragraph in the
page 10. It limits the shift.count <= 14, this way the requirement above is
guaranteed. Note that the maximum window size is one-quarter of the TCP
sequence number space (2^32 = 4Gbyte).

However the references below indicate that a Sliding Window protocol works
properly if the maximum window size is at most half of the sequence number
space available. The key is insure that once the receiver advances its
window, the new range of valid sequence numbers does not overlap with the
old one.

In this case the maximum window size of TCP could be 2Gbyte instead of
1Gbyte. Even with a window length of 2Gbyte it does not seem possible to
duplicate sequence numbers in the same window by overlap or segment/ack
losses. Duplicates are still possible due earlier packet incarnations,
however they must be handled by the Maximum Segment Lifetime or PAWS
extension.

I would be grateful it someone could explain me or recommend a reference to
understand the reason why the TCP window max size is limited to 1Gbyte
(2^30) instead 2Gb(2^31).

Thank you for any help you can provide.

           Vinicius M. Romao


References:
1 - Tanenbaum, Andrew. Computer Networks. 4Ed. Prentice Hall Pub, 2002.
Section: 3.4.3 A Protocol Using Selective Repeat.

2 - Peterson, Larry, and Davie, Bruce. Computer networks: A Systems
Approach. 3Ed. Boston: Morgan Kaufmann Pub, 2003. 104-105. Section: 2.5.2
Sliding Window.

--000e0cd2281efff3d3048e929bd3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Fellows.<br><br>(Apologies if this is not the correct forum to this e-ma=
il, but the tcplw WG and list are already concluded.)<br><br>The RFC-1323 s=
pecifies the TCP Window Scale Option. It states that the maximum window siz=
e is 2^30 (1Gbyte), please see the last paragraph in the page 10. It limits=
 the shift.count &lt;=3D 14, this way the requirement above is guaranteed. =
Note that the maximum window size is one-quarter of the TCP sequence number=
 space (2^32 =3D 4Gbyte).<br>

<br>However the references below indicate that a Sliding Window protocol wo=
rks properly if the maximum window size is at most half of the sequence num=
ber space available. The key is insure that once the receiver advances its =
window, the new range of valid sequence numbers does not overlap with the o=
ld one.<br>

<br>In this case the maximum window size of TCP could be 2Gbyte instead of =
1Gbyte. Even with a window length of 2Gbyte it does not seem possible to du=
plicate sequence numbers in the same window by overlap or segment/ack losse=
s. Duplicates are still possible due earlier packet incarnations, however t=
hey must be handled by the Maximum Segment Lifetime or PAWS extension.<br>

<br>I would be grateful it someone could explain me or recommend a referenc=
e to understand the reason why the TCP window max size is limited to 1Gbyte=
 (2^30) instead 2Gb(2^31).<br><br>Thank you for any help you can provide.<b=
r>
<br>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Vinicius M. Romao<br>
<br><br>References:<br>1 - Tanenbaum, Andrew. Computer Networks. 4Ed. Prent=
ice Hall Pub, 2002. Section: 3.4.3 A Protocol Using Selective Repeat.<br><b=
r>2 - Peterson, Larry, and Davie, Bruce. Computer networks: A Systems Appro=
ach. 3Ed. Boston: Morgan Kaufmann Pub, 2003. 104-105. Section: 2.5.2 Slidin=
g Window.<br>

<br>

--000e0cd2281efff3d3048e929bd3--

From wesley.m.eddy@nasa.gov  Tue Aug 24 13:39:02 2010
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From: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Tue, 24 Aug 2010 15:37:31 -0500
Thread-Topic: revising and advancing NewReno
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Subject: [tcpm] revising and advancing NewReno
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Based on Yoshifumi's work on NewReno in:
http://tools.ietf.org/html/draft-nishida-newreno-modification-02
combined with the existing Errata to RFC 3782, the possibility
of doing a revision to RFC 3782 has been proposed with the editors
from that document.  None were opposed, and the prospect of also
advancing NewReno from Proposed Standard to Draft Standard as part
of the revision was raised.

Two questions to poll the working group on are:

1 - Would anyone in the working group be opposed to TCPM taking on
    a revision to RFC 3782 as a working group item?

2 - Does advancing NewReno to Draft Standard seem agreeable?

--
Wes Eddy
MTI Systems



From david.borman@windriver.com  Tue Aug 24 14:21:56 2010
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Cc: tcpm@ietf.org
Subject: Re: [tcpm] TCP window max size?
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This reasoning is explained in RFC 1323, section 2.3, at the bottom of
page 10 and the top of page 11:

     TCP determines if a data segment is "old" or "new" by testing
     whether its sequence number is within 2**31 bytes of the left edge
     of the window, and if it is not, discarding the data as "old".  To
     insure that new data is never mistakenly considered old and vice-
     versa, the left edge of the sender's window has to be at most
     2**31 away from the right edge of the receiver's window.
     Similarly with the sender's right edge and receiver's left edge.
     Since the right and left edges of either the sender's or
     receiver's window differ by the window size, and since the sender
     and receiver windows can be out of phase by at most the window
     size, the above constraints imply that 2 * the max window size
     must be less than 2**31, or

          max window < 2**30

     Since the max window is 2**S (where S is the scaling shift count)
     times at most 2**16 - 1 (the maximum unscaled window), the maximum
     window is guaranteed to be < 2*30 if S <=3D 14.  Thus, the shift
     count must be limited to 14 (which allows windows of 2**30 =3D 1
     Gbyte)...

			-David Borman

On Aug 24, 2010, at 9:30 AM, Vinicius Rom=E3o wrote:

> Hi Fellows.
>=20
> (Apologies if this is not the correct forum to this e-mail, but the =
tcplw WG
> and list are already concluded.)
>=20
> The RFC-1323 specifies the TCP Window Scale Option. It states that the
> maximum window size is 2^30 (1Gbyte), please see the last paragraph in =
the
> page 10. It limits the shift.count <=3D 14, this way the requirement =
above is
> guaranteed. Note that the maximum window size is one-quarter of the =
TCP
> sequence number space (2^32 =3D 4Gbyte).
>=20
> However the references below indicate that a Sliding Window protocol =
works
> properly if the maximum window size is at most half of the sequence =
number
> space available. The key is insure that once the receiver advances its
> window, the new range of valid sequence numbers does not overlap with =
the
> old one.
>=20
> In this case the maximum window size of TCP could be 2Gbyte instead of
> 1Gbyte. Even with a window length of 2Gbyte it does not seem possible =
to
> duplicate sequence numbers in the same window by overlap or =
segment/ack
> losses. Duplicates are still possible due earlier packet incarnations,
> however they must be handled by the Maximum Segment Lifetime or PAWS
> extension.
>=20
> I would be grateful it someone could explain me or recommend a =
reference to
> understand the reason why the TCP window max size is limited to 1Gbyte
> (2^30) instead 2Gb(2^31).
>=20
> Thank you for any help you can provide.
>=20
>          Vinicius M. Romao
>=20
>=20
> References:
> 1 - Tanenbaum, Andrew. Computer Networks. 4Ed. Prentice Hall Pub, =
2002.
> Section: 3.4.3 A Protocol Using Selective Repeat.
>=20
> 2 - Peterson, Larry, and Davie, Bruce. Computer networks: A Systems
> Approach. 3Ed. Boston: Morgan Kaufmann Pub, 2003. 104-105. Section: =
2.5.2
> Sliding Window.
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From david.borman@windriver.com  Tue Aug 24 15:07:20 2010
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Cc: "tcpm@ietf.org WG" <tcpm@ietf.org>
Subject: Re: [tcpm] TCP window max size?
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On Aug 24, 2010, at 4:22 PM, David Borman wrote:

> This reasoning is explained in RFC 1323, section 2.3, at the bottom of
> page 10 and the top of page 11:
>=20
>    TCP determines if a data segment is "old" or "new" by testing
>    whether its sequence number is within 2**31 bytes of the left edge
>    of the window, and if it is not, discarding the data as "old".  To
>    insure that new data is never mistakenly considered old and vice-
>    versa, the left edge of the sender's window has to be at most
>    2**31 away from the right edge of the receiver's window.
>    Similarly with the sender's right edge and receiver's left edge.
>    Since the right and left edges of either the sender's or
>    receiver's window differ by the window size, and since the sender
>    and receiver windows can be out of phase by at most the window
>    size, the above constraints imply that 2 * the max window size
>    must be less than 2**31, or
>=20
>         max window < 2**30
>=20
>    Since the max window is 2**S (where S is the scaling shift count)
>    times at most 2**16 - 1 (the maximum unscaled window), the maximum
>    window is guaranteed to be < 2*30 if S <=3D 14.  Thus, the shift
>    count must be limited to 14 (which allows windows of 2**30 =3D 1
>    Gbyte)...

And what I forgot to ask was whether you or someone else disagreed with
this logic, but this is why it is 14.  However, even if the logic is
faulty, the max value will remain at 14, the only way to change it would
be via a new TCP option number.

But mostly the place for trouble is at the boundary conditions, and if
you had a 2^31 window and the sender/receiver windows were fully skewed,
any outstanding packet from before left edge of the senders window could
show up at the receiver and be viewed as new data at the right edge of
the receivers window, which would be a bad thing.

1) Sender has a full 2^31 window of unacknowledged data,
   receiver is a full 2^31 ahead of the sender.
2) Sender retransmits a packet at left edge of it's window.
3) In-transit Ack from Receiver arrives, acks all data, and
   advances window by 2^31.
4) Sender sends a new packet.
5) New packet gets reordered in transit with the retransmitted packet.
6) New packet arrives, and receiver advances window.
7) Retransmitted packet arrives, and it now appears to be at the
   right edge of the receivers window as new data.

A 2^14 max window keeps things well away from those boundary conditions.


>=20
> 			-David Borman
>=20
> On Aug 24, 2010, at 9:30 AM, Vinicius Rom=E3o wrote:
>=20
>> Hi Fellows.
>>=20
>> (Apologies if this is not the correct forum to this e-mail, but the =
tcplw WG
>> and list are already concluded.)
>>=20
>> The RFC-1323 specifies the TCP Window Scale Option. It states that =
the
>> maximum window size is 2^30 (1Gbyte), please see the last paragraph =
in the
>> page 10. It limits the shift.count <=3D 14, this way the requirement =
above is
>> guaranteed. Note that the maximum window size is one-quarter of the =
TCP
>> sequence number space (2^32 =3D 4Gbyte).
>>=20
>> However the references below indicate that a Sliding Window protocol =
works
>> properly if the maximum window size is at most half of the sequence =
number
>> space available. The key is insure that once the receiver advances =
its
>> window, the new range of valid sequence numbers does not overlap with =
the
>> old one.
>>=20
>> In this case the maximum window size of TCP could be 2Gbyte instead =
of
>> 1Gbyte. Even with a window length of 2Gbyte it does not seem possible =
to
>> duplicate sequence numbers in the same window by overlap or =
segment/ack
>> losses. Duplicates are still possible due earlier packet =
incarnations,
>> however they must be handled by the Maximum Segment Lifetime or PAWS
>> extension.
>>=20
>> I would be grateful it someone could explain me or recommend a =
reference to
>> understand the reason why the TCP window max size is limited to =
1Gbyte
>> (2^30) instead 2Gb(2^31).
>>=20
>> Thank you for any help you can provide.
>>=20
>>        Vinicius M. Romao
>>=20
>>=20
>> References:
>> 1 - Tanenbaum, Andrew. Computer Networks. 4Ed. Prentice Hall Pub, =
2002.
>> Section: 3.4.3 A Protocol Using Selective Repeat.
>>=20
>> 2 - Peterson, Larry, and Davie, Bruce. Computer networks: A Systems
>> Approach. 3Ed. Boston: Morgan Kaufmann Pub, 2003. 104-105. Section: =
2.5.2
>> Sliding Window.
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From viniciusromao@gmail.com  Tue Aug 24 19:44:17 2010
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From: =?ISO-8859-1?Q?Vinicius_Rom=E3o?= <viniciusromao@gmail.com>
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Subject: Re: [tcpm] TCP window max size?
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   Borman,

Thank you very much for your clarifications.

After reading your explanation, I agree that 1Gbyte is the best choice to
the TCP window max size.

Thanks.

Vinicius M. Romao

2010/8/24 David Borman <david.borman@windriver.com>

>
> On Aug 24, 2010, at 4:22 PM, David Borman wrote:
>
> > This reasoning is explained in RFC 1323, section 2.3, at the bottom of
> > page 10 and the top of page 11:
> >
> >    TCP determines if a data segment is "old" or "new" by testing
> >    whether its sequence number is within 2**31 bytes of the left edge
> >    of the window, and if it is not, discarding the data as "old".  To
> >    insure that new data is never mistakenly considered old and vice-
> >    versa, the left edge of the sender's window has to be at most
> >    2**31 away from the right edge of the receiver's window.
> >    Similarly with the sender's right edge and receiver's left edge.
> >    Since the right and left edges of either the sender's or
> >    receiver's window differ by the window size, and since the sender
> >    and receiver windows can be out of phase by at most the window
> >    size, the above constraints imply that 2 * the max window size
> >    must be less than 2**31, or
> >
> >         max window < 2**30
> >
> >    Since the max window is 2**S (where S is the scaling shift count)
> >    times at most 2**16 - 1 (the maximum unscaled window), the maximum
> >    window is guaranteed to be < 2*30 if S <=3D 14.  Thus, the shift
> >    count must be limited to 14 (which allows windows of 2**30 =3D 1
> >    Gbyte)...
>
> And what I forgot to ask was whether you or someone else disagreed with
> this logic, but this is why it is 14.  However, even if the logic is
> faulty, the max value will remain at 14, the only way to change it would
> be via a new TCP option number.
>
> But mostly the place for trouble is at the boundary conditions, and if
> you had a 2^31 window and the sender/receiver windows were fully skewed,
> any outstanding packet from before left edge of the senders window could
> show up at the receiver and be viewed as new data at the right edge of
> the receivers window, which would be a bad thing.
>
> 1) Sender has a full 2^31 window of unacknowledged data,
>   receiver is a full 2^31 ahead of the sender.
> 2) Sender retransmits a packet at left edge of it's window.
> 3) In-transit Ack from Receiver arrives, acks all data, and
>   advances window by 2^31.
> 4) Sender sends a new packet.
> 5) New packet gets reordered in transit with the retransmitted packet.
> 6) New packet arrives, and receiver advances window.
> 7) Retransmitted packet arrives, and it now appears to be at the
>   right edge of the receivers window as new data.
>
> A 2^14 max window keeps things well away from those boundary conditions.
>
>
> >
> >                       -David Borman
> >
> > On Aug 24, 2010, at 9:30 AM, Vinicius Rom=E3o wrote:
> >
> >> Hi Fellows.
> >>
> >> (Apologies if this is not the correct forum to this e-mail, but the
> tcplw WG
> >> and list are already concluded.)
> >>
> >> The RFC-1323 specifies the TCP Window Scale Option. It states that the
> >> maximum window size is 2^30 (1Gbyte), please see the last paragraph in
> the
> >> page 10. It limits the shift.count <=3D 14, this way the requirement a=
bove
> is
> >> guaranteed. Note that the maximum window size is one-quarter of the TC=
P
> >> sequence number space (2^32 =3D 4Gbyte).
> >>
> >> However the references below indicate that a Sliding Window protocol
> works
> >> properly if the maximum window size is at most half of the sequence
> number
> >> space available. The key is insure that once the receiver advances its
> >> window, the new range of valid sequence numbers does not overlap with
> the
> >> old one.
> >>
> >> In this case the maximum window size of TCP could be 2Gbyte instead of
> >> 1Gbyte. Even with a window length of 2Gbyte it does not seem possible =
to
> >> duplicate sequence numbers in the same window by overlap or segment/ac=
k
> >> losses. Duplicates are still possible due earlier packet incarnations,
> >> however they must be handled by the Maximum Segment Lifetime or PAWS
> >> extension.
> >>
> >> I would be grateful it someone could explain me or recommend a referen=
ce
> to
> >> understand the reason why the TCP window max size is limited to 1Gbyte
> >> (2^30) instead 2Gb(2^31).
> >>
> >> Thank you for any help you can provide.
> >>
> >>        Vinicius M. Romao
> >>
> >>
> >> References:
> >> 1 - Tanenbaum, Andrew. Computer Networks. 4Ed. Prentice Hall Pub, 2002=
.
> >> Section: 3.4.3 A Protocol Using Selective Repeat.
> >>
> >> 2 - Peterson, Larry, and Davie, Bruce. Computer networks: A Systems
> >> Approach. 3Ed. Boston: Morgan Kaufmann Pub, 2003. 104-105. Section:
> 2.5.2
> >> Sliding Window.
> >> _______________________________________________
> >> 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
>
>

--000e0cd2281e735d5b048e9cdda7
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Content-Transfer-Encoding: quoted-printable

<br>=A0=A0 Borman, <br><br>Thank you very much for your clarifications. <br=
><br>After reading your explanation, I agree that 1Gbyte is the best choice=
 to the TCP window max size.<br><br>Thanks.<br><br>Vinicius M. Romao<br><br=
>
<div class=3D"gmail_quote">2010/8/24 David Borman <span dir=3D"ltr">&lt;<a =
href=3D"mailto:david.borman@windriver.com">david.borman@windriver.com</a>&g=
t;</span><br><blockquote class=3D"gmail_quote" style=3D"border-left: 1px so=
lid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class=3D"im"><br>
On Aug 24, 2010, at 4:22 PM, David Borman wrote:<br>
<br>
&gt; This reasoning is explained in RFC 1323, section 2.3, at the bottom of=
<br>
&gt; page 10 and the top of page 11:<br>
&gt;<br>
&gt; =A0 =A0TCP determines if a data segment is &quot;old&quot; or &quot;ne=
w&quot; by testing<br>
&gt; =A0 =A0whether its sequence number is within 2**31 bytes of the left e=
dge<br>
&gt; =A0 =A0of the window, and if it is not, discarding the data as &quot;o=
ld&quot;. =A0To<br>
&gt; =A0 =A0insure that new data is never mistakenly considered old and vic=
e-<br>
&gt; =A0 =A0versa, the left edge of the sender&#39;s window has to be at mo=
st<br>
&gt; =A0 =A02**31 away from the right edge of the receiver&#39;s window.<br=
>
&gt; =A0 =A0Similarly with the sender&#39;s right edge and receiver&#39;s l=
eft edge.<br>
&gt; =A0 =A0Since the right and left edges of either the sender&#39;s or<br=
>
&gt; =A0 =A0receiver&#39;s window differ by the window size, and since the =
sender<br>
&gt; =A0 =A0and receiver windows can be out of phase by at most the window<=
br>
&gt; =A0 =A0size, the above constraints imply that 2 * the max window size<=
br>
&gt; =A0 =A0must be less than 2**31, or<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 max window &lt; 2**30<br>
&gt;<br>
&gt; =A0 =A0Since the max window is 2**S (where S is the scaling shift coun=
t)<br>
&gt; =A0 =A0times at most 2**16 - 1 (the maximum unscaled window), the maxi=
mum<br>
&gt; =A0 =A0window is guaranteed to be &lt; 2*30 if S &lt;=3D 14. =A0Thus, =
the shift<br>
&gt; =A0 =A0count must be limited to 14 (which allows windows of 2**30 =3D =
1<br>
&gt; =A0 =A0Gbyte)...<br>
<br>
</div>And what I forgot to ask was whether you or someone else disagreed wi=
th<br>
this logic, but this is why it is 14. =A0However, even if the logic is<br>
faulty, the max value will remain at 14, the only way to change it would<br=
>
be via a new TCP option number.<br>
<br>
But mostly the place for trouble is at the boundary conditions, and if<br>
you had a 2^31 window and the sender/receiver windows were fully skewed,<br=
>
any outstanding packet from before left edge of the senders window could<br=
>
show up at the receiver and be viewed as new data at the right edge of<br>
the receivers window, which would be a bad thing.<br>
<br>
1) Sender has a full 2^31 window of unacknowledged data,<br>
 =A0 receiver is a full 2^31 ahead of the sender.<br>
2) Sender retransmits a packet at left edge of it&#39;s window.<br>
3) In-transit Ack from Receiver arrives, acks all data, and<br>
 =A0 advances window by 2^31.<br>
4) Sender sends a new packet.<br>
5) New packet gets reordered in transit with the retransmitted packet.<br>
6) New packet arrives, and receiver advances window.<br>
7) Retransmitted packet arrives, and it now appears to be at the<br>
 =A0 right edge of the receivers window as new data.<br>
<br>
A 2^14 max window keeps things well away from those boundary conditions.<br=
>
<div><div></div><div class=3D"h5"><br>
<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 -David Borman<br>
&gt;<br>
&gt; On Aug 24, 2010, at 9:30 AM, Vinicius Rom=E3o wrote:<br>
&gt;<br>
&gt;&gt; Hi Fellows.<br>
&gt;&gt;<br>
&gt;&gt; (Apologies if this is not the correct forum to this e-mail, but th=
e tcplw WG<br>
&gt;&gt; and list are already concluded.)<br>
&gt;&gt;<br>
&gt;&gt; The RFC-1323 specifies the TCP Window Scale Option. It states that=
 the<br>
&gt;&gt; maximum window size is 2^30 (1Gbyte), please see the last paragrap=
h in the<br>
&gt;&gt; page 10. It limits the shift.count &lt;=3D 14, this way the requir=
ement above is<br>
&gt;&gt; guaranteed. Note that the maximum window size is one-quarter of th=
e TCP<br>
&gt;&gt; sequence number space (2^32 =3D 4Gbyte).<br>
&gt;&gt;<br>
&gt;&gt; However the references below indicate that a Sliding Window protoc=
ol works<br>
&gt;&gt; properly if the maximum window size is at most half of the sequenc=
e number<br>
&gt;&gt; space available. The key is insure that once the receiver advances=
 its<br>
&gt;&gt; window, the new range of valid sequence numbers does not overlap w=
ith the<br>
&gt;&gt; old one.<br>
&gt;&gt;<br>
&gt;&gt; In this case the maximum window size of TCP could be 2Gbyte instea=
d of<br>
&gt;&gt; 1Gbyte. Even with a window length of 2Gbyte it does not seem possi=
ble to<br>
&gt;&gt; duplicate sequence numbers in the same window by overlap or segmen=
t/ack<br>
&gt;&gt; losses. Duplicates are still possible due earlier packet incarnati=
ons,<br>
&gt;&gt; however they must be handled by the Maximum Segment Lifetime or PA=
WS<br>
&gt;&gt; extension.<br>
&gt;&gt;<br>
&gt;&gt; I would be grateful it someone could explain me or recommend a ref=
erence to<br>
&gt;&gt; understand the reason why the TCP window max size is limited to 1G=
byte<br>
&gt;&gt; (2^30) instead 2Gb(2^31).<br>
&gt;&gt;<br>
&gt;&gt; Thank you for any help you can provide.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0Vinicius M. Romao<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; References:<br>
&gt;&gt; 1 - Tanenbaum, Andrew. Computer Networks. 4Ed. Prentice Hall Pub, =
2002.<br>
&gt;&gt; Section: 3.4.3 A Protocol Using Selective Repeat.<br>
&gt;&gt;<br>
&gt;&gt; 2 - Peterson, Larry, and Davie, Bruce. Computer networks: A System=
s<br>
&gt;&gt; Approach. 3Ed. Boston: Morgan Kaufmann Pub, 2003. 104-105. Section=
: 2.5.2<br>
&gt;&gt; Sliding Window.<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; tcpm mailing list<br>
&gt;&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br>
</div></div></blockquote></div><br>

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From lars.eggert@nokia.com  Wed Aug 25 00:26:42 2010
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Hi,


On 2010-8-24, at 23:37, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP] =
wrote:
> 1 - Would anyone in the working group be opposed to TCPM taking on
>    a revision to RFC 3782 as a working group item?

as an individual: no, good idea

> 2 - Does advancing NewReno to Draft Standard seem agreeable?

yes, but since the errata and Yoshifumi's proposal change the PS, the =
bis version needs to be a PS too, before it can move to DS at the =
earliest 6 months later.

Lars=

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From dwing@cisco.com  Wed Aug 25 15:09:58 2010
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From: "Dan Wing" <dwing@cisco.com>
To: <tcpm@ietf.org>, <behave@ietf.org>, <int-area@ietf.org>
Date: Wed, 25 Aug 2010 15:10:29 -0700
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Cc: Andrew Yourtchenko <ayourtch@cisco.com>
Subject: [tcpm] Revealing identity of TCP client connection when sharing IPv4 address
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Please send replies to int-area@ietf.org

Hi.  

Andrew and I have written two I-D's describing two approaches to solve a
problem that occurs when sharing an IPv4 address (NAPT44, NAPT64, HTTP
proxy, A+P, dual-IVI, etc.).  The abstract is common between the two
approaches and summarizes the problem:

   When an IP address is shared among several subscribers, it is
   impossible to  determine which subscriber has initiated that TCP
   connection.  This memo describes a technique to share the identity of
   a subscriber that initiated a TCP connection with the TCP server.
   The proposed method avoids altering the application-level payload
   and works well with SSL-protected connections.

approach 1, new TCP option:
  http://tools.ietf.org/html/draft-wing-nat-reveal-option

approach 2, overload existing TIMESTAMP TSVal option and IP ID:
  http://tools.ietf.org/html/draft-yourtchenko-nat-reveal-hash

We are leaning towards the new TCP option (approach 1).

The idea crosses BEHAVE (NAT), INTAREA (address sharing), and TCPM (TCP
option).  However, to reduce cross-posting and because INTAREA already has a
working group document related to address sharing, I propose we discuss the
document on the int-area@ietf.org mailing list.

Comments and feedback welcome.
-d



From wwwrun@rfc-editor.org  Thu Aug 26 10:01:11 2010
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Cc: tcpm@ietf.org, rfc-editor@rfc-editor.org
Subject: [tcpm] RFC 5961 on Improving TCP's Robustness to Blind In-Window Attacks
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 5961

        Title:      Improving TCP's Robustness to Blind 
                    In-Window Attacks 
        Author:     A. Ramaiah, R. Stewart,
                    M. Dalal
        Status:     Standards Track
        Stream:     IETF
        Date:       August 2010
        Mailbox:    ananth@cisco.com, 
                    rstewart@huawei.com, 
                    mdalal@cisco.com
        Pages:      19
        Characters: 44717
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-tcpm-tcpsecure-13.txt

        URL:        http://www.rfc-editor.org/rfc/rfc5961.txt

TCP has historically been considered to be protected against spoofed
off-path packet injection attacks by relying on the fact that it is
difficult to guess the 4-tuple (the source and destination IP
addresses and the source and destination ports) in combination with
the 32-bit sequence number(s).  A combination of increasing window
sizes and applications using longer-term connections (e.g., H-323 or
Border Gateway Protocol (BGP) [RFC4271]) have left modern TCP
implementations more vulnerable to these types of spoofed packet
injection attacks.

Many of these long-term TCP applications tend to have predictable IP
addresses and ports that makes it far easier for the 4-tuple (4-tuple
is the same as the socket pair mentioned in RFC 793) to be guessed.
Having guessed the 4-tuple correctly, an attacker can inject a TCP
segment with the RST bit set, the SYN bit set or data into a TCP
connection by systematically guessing the sequence number of the
spoofed segment to be in the current receive window.  This can cause
the connection to abort or cause data corruption.  This document
specifies small modifications to the way TCP handles inbound segments
that can reduce the chances of a successful attack.  [STANDARDS TRACK]

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

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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The RFC Editor Team
Association Management Solutions, LLC



From Xiangsong.Cui@huawei.com  Sun Aug 29 20:28:57 2010
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Date: Mon, 30 Aug 2010 11:29:16 +0800
From: Xiangsong Cui <Xiangsong.Cui@huawei.com>
To: tcpm@ietf.org
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Subject: [tcpm] Checksum of TCP
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Hi Guys,

I'm reading RFC793, and in section 3.1, it reads:

  Checksum:  16 bits

    The checksum field is the 16 bit one's complement of the one's
    complement sum of all 16 bit words in the header and text.  If a
    segment contains an odd number of header and text octets to be
    checksummed, the last octet is padded on the right with zeros to
    form a 16 bit word for checksum purposes.  The pad is not
    transmitted as part of the segment.  While computing the checksum,
    the checksum field itself is replaced with zeros.

    The checksum also covers a *96* bit pseudo header conceptually
    prefixed to the TCP header.  This pseudo header contains the Source
    Address, the Destination Address, the Protocol, and TCP length.
    This gives the TCP protection against misrouted segments.  This
    information is carried in the Internet Protocol and is transferred
    across the TCP/Network interface in the arguments or results of
    calls by the TCP on the IP.

                     +--------+--------+--------+--------+
                     |           Source Address          |
                     +--------+--------+--------+--------+
                     |         Destination Address       |
                     +--------+--------+--------+--------+
                     |  zero  |  PTCL  |    TCP Length   |
                     +--------+--------+--------+--------+

      The TCP Length is the TCP header length plus the data length in
      octets (this is not an explicitly transmitted quantity, but is
      computed), and it does not count the *12* octets of the pseudo
      header.


In my impresion, TCP is an IP agnostic protocol, but in the second
paragraph, It specifies that "a *96* bit pseudo header"(the first sentence)
and "the *12* octets of the pseudo header."(the last sentence), which seems
to be coupled with IPv4.

Considering TCP over IPv6, should we remove the accurate number from the
statement? Such as " The checksum also covers a pseudo header ..." and "it
does not count the octets of the pseudo header."  Adding a note to this?

PS, I believe TCP over IPv6 has been implemented, but I havn't see any
standard fix on this issue, if it has been addressed, please tell me where
to get the new document, thanks!


Regards,
Xiangsong




From anil.agarwal@viasat.com  Mon Aug 30 04:59:14 2010
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From: "Agarwal, Anil" <Anil.Agarwal@viasat.com>
To: "Xiangsong Cui" <Xiangsong.Cui@huawei.com>, <tcpm@ietf.org>
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Subject: Re: [tcpm] Checksum of TCP
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Xiangsong,
=20
RFC 2460, section 8.1 covers this issue.
=20
Anil

________________________________

From: tcpm-bounces@ietf.org on behalf of Xiangsong Cui
Sent: Sun 8/29/2010 11:29 PM
To: tcpm@ietf.org
Subject: [tcpm] Checksum of TCP



Hi Guys,

I'm reading RFC793, and in section 3.1, it reads:

  Checksum:  16 bits

    The checksum field is the 16 bit one's complement of the one's
    complement sum of all 16 bit words in the header and text.  If a
    segment contains an odd number of header and text octets to be
    checksummed, the last octet is padded on the right with zeros to
    form a 16 bit word for checksum purposes.  The pad is not
    transmitted as part of the segment.  While computing the checksum,
    the checksum field itself is replaced with zeros.

    The checksum also covers a *96* bit pseudo header conceptually
    prefixed to the TCP header.  This pseudo header contains the Source
    Address, the Destination Address, the Protocol, and TCP length.
    This gives the TCP protection against misrouted segments.  This
    information is carried in the Internet Protocol and is transferred
    across the TCP/Network interface in the arguments or results of
    calls by the TCP on the IP.

                     +--------+--------+--------+--------+
                     |           Source Address          |
                     +--------+--------+--------+--------+
                     |         Destination Address       |
                     +--------+--------+--------+--------+
                     |  zero  |  PTCL  |    TCP Length   |
                     +--------+--------+--------+--------+

      The TCP Length is the TCP header length plus the data length in
      octets (this is not an explicitly transmitted quantity, but is
      computed), and it does not count the *12* octets of the pseudo
      header.


In my impresion, TCP is an IP agnostic protocol, but in the second
paragraph, It specifies that "a *96* bit pseudo header"(the first =
sentence)
and "the *12* octets of the pseudo header."(the last sentence), which =
seems
to be coupled with IPv4.

Considering TCP over IPv6, should we remove the accurate number from the
statement? Such as " The checksum also covers a pseudo header ..." and =
"it
does not count the octets of the pseudo header."  Adding a note to this?

PS, I believe TCP over IPv6 has been implemented, but I havn't see any
standard fix on this issue, if it has been addressed, please tell me =
where
to get the new document, thanks!


Regards,
Xiangsong



_______________________________________________
tcpm mailing list
tcpm@ietf.org
https://www.ietf.org/mailman/listinfo/tcpm =
<https://www.ietf.org/mailman/listinfo/tcpm>=20



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<HTML dir=3Dltr><HEAD><TITLE>[tcpm] Checksum of TCP</TITLE>=0A=
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dunicode">=0A=
<META content=3D"MSHTML 6.00.2900.5921" name=3DGENERATOR></HEAD>=0A=
<BODY>=0A=
<DIV id=3DidOWAReplyText59969 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3D"Courier New" color=3D#000000 =
size=3D2>Xiangsong,</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3D"Courier New" size=3D2>RFC 2460, section 8.1 =
covers this issue.</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3D"Courier New" =
size=3D2>Anil</FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR><FONT face=3D"Courier New">=0A=
<HR tabIndex=3D-1>=0A=
</FONT><FONT face=3D"Courier New"><FONT size=3D2><B>From:</B> =
tcpm-bounces@ietf.org on behalf of Xiangsong Cui<BR><B>Sent:</B> Sun =
8/29/2010 11:29 PM<BR><B>To:</B> tcpm@ietf.org<BR><B>Subject:</B> [tcpm] =
Checksum of TCP<BR></FONT><BR></FONT></DIV>=0A=
<DIV>=0A=
<P><FONT size=3D2><FONT face=3D"Courier New">Hi Guys,<BR><BR>I'm reading =
RFC793, and in section 3.1, it reads:<BR><BR>&nbsp; Checksum:&nbsp; 16 =
bits<BR><BR>&nbsp;&nbsp;&nbsp; The checksum field is the 16 bit one's =
complement of the one's<BR>&nbsp;&nbsp;&nbsp; complement sum of all 16 =
bit words in the header and text.&nbsp; If a<BR>&nbsp;&nbsp;&nbsp; =
segment contains an odd number of header and text octets to =
be<BR>&nbsp;&nbsp;&nbsp; checksummed, the last octet is padded on the =
right with zeros to<BR>&nbsp;&nbsp;&nbsp; form a 16 bit word for =
checksum purposes.&nbsp; The pad is not<BR>&nbsp;&nbsp;&nbsp; =
transmitted as part of the segment.&nbsp; While computing the =
checksum,<BR>&nbsp;&nbsp;&nbsp; the checksum field itself is replaced =
with zeros.<BR><BR>&nbsp;&nbsp;&nbsp; The checksum also covers a *96* =
bit pseudo header conceptually<BR>&nbsp;&nbsp;&nbsp; prefixed to the TCP =
header.&nbsp; This pseudo header contains the =
Source<BR>&nbsp;&nbsp;&nbsp; Address, the Destination Address, the =
Protocol, and TCP length.<BR>&nbsp;&nbsp;&nbsp; This gives the TCP =
protection against misrouted segments.&nbsp; This<BR>&nbsp;&nbsp;&nbsp; =
information is carried in the Internet Protocol and is =
transferred<BR>&nbsp;&nbsp;&nbsp; across the TCP/Network interface in =
the arguments or results of<BR>&nbsp;&nbsp;&nbsp; calls by the TCP on =
the =
IP.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
+--------+--------+--------+--------+<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Source =
Address&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
+--------+--------+--------+--------+<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Destination Address&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
+--------+--------+--------+--------+<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; |&nbsp; zero&nbsp; |&nbsp; PTCL&nbsp; =
|&nbsp;&nbsp;&nbsp; TCP Length&nbsp;&nbsp; =
|<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
+--------+--------+--------+--------+<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; The TCP Length is the TCP header length plus the data length =
in<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; octets (this is not an explicitly =
transmitted quantity, but is<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
computed), and it does not count the *12* octets of the =
pseudo<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; header.<BR><BR><BR>In my =
impresion, TCP is an IP agnostic protocol, but in the =
second<BR>paragraph, It specifies that "a *96* bit pseudo header"(the =
first sentence)<BR>and "the *12* octets of the pseudo header."(the last =
sentence), which seems<BR>to be coupled with IPv4.<BR><BR>Considering =
TCP over IPv6, should we remove the accurate number from =
the<BR>statement? Such as " The checksum also covers a pseudo header =
..." and "it<BR>does not count the octets of the pseudo header."&nbsp; =
Adding a note to this?<BR><BR>PS, I believe TCP over IPv6 has been =
implemented, but I havn't see any<BR>standard fix on this issue, if it =
has been addressed, please tell me where<BR>to get the new document, =
thanks!<BR><BR><BR>Regards,<BR>Xiangsong<BR><BR><BR><BR>_________________=
______________________________<BR>tcpm mailing =
list<BR>tcpm@ietf.org<BR></FONT><A =
href=3D"https://www.ietf.org/mailman/listinfo/tcpm"><FONT =
face=3D"Courier =
New">https://www.ietf.org/mailman/listinfo/tcpm</FONT></A><BR></FONT></P>=
</DIV></BODY></HTML>
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From touch@isi.edu  Mon Aug 30 08:55:05 2010
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Date: Mon, 30 Aug 2010 08:54:49 -0700
From: Joe Touch <touch@isi.edu>
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On 8/29/2010 8:29 PM, Xiangsong Cui wrote:
> Hi Guys,
>
> I'm reading RFC793, and in section 3.1, it reads:
...
> In my impresion, TCP is an IP agnostic protocol,

FWIW, it really is not. TCP uses a "socket pair" as a connection 
identifier, i.e.:
	source IP address
	destination IP address
	source TCP port
	destination IP port

As such, the IP addresses are an instrinsic part of a TCP connection, so 
TCP cannot exist in the absence of IP - or an emulation thereof. The 
other two fields of the pseudoheader are:
	PCTL (transport protocol number - for TCP, always "6")
	TCP segment length (computed from IP length - IP options)

These are already contained in the IP header (explicitly or implicitly), 
it are not repeated in the TCP header - so instead they are included in 
the pseudoheader.

...
> PS, I believe TCP over IPv6 has been implemented, but I havn't see any
> standard fix on this issue, if it has been addressed,

As noted, this was defined in RFC 2460, in which IPv6 is defined.

Joe

From Xiangsong.Cui@huawei.com  Mon Aug 30 17:54:54 2010
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Date: Tue, 31 Aug 2010 08:55:10 +0800
From: Xiangsong Cui <Xiangsong.Cui@huawei.com>
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Subject: Re: [tcpm] Checksum of TCP
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I see, thank you Joe and Anil!

Regards,
Xiangsong



> -----Original Message-----
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Monday, August 30, 2010 11:55 PM
> To: Xiangsong Cui
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Checksum of TCP
> 
> 
> 
> On 8/29/2010 8:29 PM, Xiangsong Cui wrote:
> > Hi Guys,
> >
> > I'm reading RFC793, and in section 3.1, it reads:
> ...
> > In my impresion, TCP is an IP agnostic protocol,
> 
> FWIW, it really is not. TCP uses a "socket pair" as a connection
> identifier, i.e.:
> 	source IP address
> 	destination IP address
> 	source TCP port
> 	destination IP port
> 
> As such, the IP addresses are an instrinsic part of a TCP connection, so
> TCP cannot exist in the absence of IP - or an emulation thereof. The
> other two fields of the pseudoheader are:
> 	PCTL (transport protocol number - for TCP, always "6")
> 	TCP segment length (computed from IP length - IP options)
> 
> These are already contained in the IP header (explicitly or implicitly),
> it are not repeated in the TCP header - so instead they are included in
> the pseudoheader.
> 
> ...
> > PS, I believe TCP over IPv6 has been implemented, but I havn't see any
> > standard fix on this issue, if it has been addressed,
> 
> As noted, this was defined in RFC 2460, in which IPv6 is defined.
> 
> Joe

