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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions of the IETF.

        Title           : Transmission Control Protocol Specification
        Author          : Wesley M. Eddy
	Filename        : draft-ietf-tcpm-rfc793bis-03.txt
	Pages           : 90
	Date            : 2016-08-01

Abstract:
   This document specifies the Internet's Transmission Control Protocol
   (TCP).  TCP is an important transport layer protocol in the Internet
   stack, and has continuously evolved over decades of use and growth of
   the Internet.  Over this time, a number of changes have been made to
   TCP as it was specified in RFC 793, though these have only been
   documented in a piecemeal fashion.  This document collects and brings
   those changes together with the protocol specification from RFC 793.
   This document obsoletes RFC 793 and several other RFCs (TODO: list
   all actual RFCs when finished).

   RFC EDITOR NOTE: If approved for publication as an RFC, this should
   be marked additionally as "STD: 7" and replace RFC 793 in that role.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rfc793bis/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-03


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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From: Wesley Eddy <wes@mti-systems.com>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-03.txt
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There are some things in this draft that could use WG feedback. I'll 
start separate threads shortly rather than expect everyone to digest the 
diffs and figure it out!


On 8/1/2016 12:58 PM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions of the IETF.
>
>          Title           : Transmission Control Protocol Specification
>          Author          : Wesley M. Eddy
> 	Filename        : draft-ietf-tcpm-rfc793bis-03.txt
> 	Pages           : 90
> 	Date            : 2016-08-01
>
> Abstract:
>     This document specifies the Internet's Transmission Control Protocol
>     (TCP).  TCP is an important transport layer protocol in the Internet
>     stack, and has continuously evolved over decades of use and growth of
>     the Internet.  Over this time, a number of changes have been made to
>     TCP as it was specified in RFC 793, though these have only been
>     documented in a piecemeal fashion.  This document collects and brings
>     those changes together with the protocol specification from RFC 793.
>     This document obsoletes RFC 793 and several other RFCs (TODO: list
>     all actual RFCs when finished).
>
>     RFC EDITOR NOTE: If approved for publication as an RFC, this should
>     be marked additionally as "STD: 7" and replace RFC 793 in that role.
>
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rfc793bis/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-03
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon Aug  1 20:52:54 2016
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From: Gavin McCullagh <gmccullagh@gmail.com>
Date: Mon, 1 Aug 2016 20:52:43 -0700
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To: Derek Fawcus <dfawcus+lists-tcpcrypt@employees.org>
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Cc: tcpinc <tcpinc@ietf.org>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "Mirja Kuehlewind \(IETF\)" <ietf@kuehlewind.net>, Kyle Rose <krose@krose.org>, David Mazieres expires 2016-10-25 PDT <mazieres-5mehvxfqtr38mvjvf6tfwgcy62@temporary-address.scs.stanford.edu>
Subject: Re: [tcpm] [tcpinc] TCP's treatment of data in SYN packets
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--94eb2c05fb2036513805390ea6d1
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Just to satisfy my curiosity, how did this work (or did it?) with TCP SYN
Cookies?

I can imagine a tcp listener which was implementing SYN Cookies only ACKing
the initial SYN sequence number and not the data which it discarded, then
the application retransmitting that data either on the subsequent ACK or
the next data packet.  Does that generally just work with most/all tcp
implementations on the other side?  I suppose it ought to?

Gavin

On Wed, Jul 27, 2016 at 11:37 PM, Derek Fawcus <
dfawcus+lists-tcpcrypt@employees.org> wrote:

> On Thu, Jul 28, 2016 at 12:24:19am +0100, Derek Fawcus wrote:
> >
> > Implying that a passive opener which chose to ack such data bytes would
> > have to buffer such data until the handshake completes;  but I'm not
> > aware of any stack which actually works that way.
>
> Except the old KA9Q NOS,  which was tickling my memory...
>
> It did have the ability to send such SYN data.
>
> DF
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--94eb2c05fb2036513805390ea6d1
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<div dir=3D"ltr"><div><div>Just to satisfy my curiosity, how did this work =
(or did it?) with TCP SYN Cookies? <br><br></div>I can imagine a tcp listen=
er which was implementing SYN Cookies only ACKing the initial SYN sequence =
number and not the data which it discarded, then the application retransmit=
ting that data either on the subsequent ACK or the next data packet.=C2=A0 =
Does that generally just work with most/all tcp implementations on the othe=
r side?=C2=A0 I suppose it ought to?<br><br></div>Gavin<br></div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Jul 27, 2016 at 11:=
37 PM, Derek Fawcus <span dir=3D"ltr">&lt;<a href=3D"mailto:dfawcus+lists-t=
cpcrypt@employees.org" target=3D"_blank">dfawcus+lists-tcpcrypt@employees.o=
rg</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D""=
>On Thu, Jul 28, 2016 at 12:24:19am +0100, Derek Fawcus wrote:<br>
&gt;<br>
&gt; Implying that a passive opener which chose to ack such data bytes woul=
d<br>
&gt; have to buffer such data until the handshake completes;=C2=A0 but I&#3=
9;m not<br>
&gt; aware of any stack which actually works that way.<br>
<br>
</span>Except the old KA9Q NOS,=C2=A0 which was tickling my memory...<br>
<br>
It did have the ability to send such SYN data.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
DF<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div>

--94eb2c05fb2036513805390ea6d1--


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From: Derek Fawcus <dfawcus+lists-tcpcrypt@employees.org>
To: Gavin McCullagh <gmccullagh@gmail.com>
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Subject: Re: [tcpm] [tcpinc] TCP's treatment of data in SYN packets
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On Mon, Aug 01, 2016 at 08:52:43pm -0700, Gavin McCullagh wrote:
> Just to satisfy my curiosity, how did this work (or did it?) with TCP SYN Cookies?

The sender of the SYN would simply send the data again once the 3whs completed.

This was before SYN cookies were invented.  As I recall,  I'd stopped using KA9Q
before '96,  which a search indicates is when SYN cookies started.

The KA9Q configuration option was 'tcp syndata',  if you do searches for 'ka9q nos'
you'll find the code if you wish to play (or just read it).

DF


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] [tcpinc] TCP's treatment of data in SYN packets
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On 8/2/2016 1:24 AM, Derek Fawcus wrote:
> On Mon, Aug 01, 2016 at 08:52:43pm -0700, Gavin McCullagh wrote:
>> Just to satisfy my curiosity, how did this work (or did it?) with TCP SYN Cookies?
> The sender of the SYN would simply send the data again once the 3whs completed.
An important point - it would send exactly the same data again (or if it
sent more data, the overlapping portions never changed).

Joe



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From: Wesley Eddy <wes@mti-systems.com>
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Subject: [tcpm] 793bis: removing "Window Management Suggestions"
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In the latest revision of the 793bis draft, I added content from RFC 
1122 on the SWS avoidance algorithms for sender-side and receiver-side.  
These are laid out in 1122, but were not present in 793.

793 includes a section called "Window Management Suggestions", that I 
believe was a precursor to the more-developed content in 1122 on:

(1) SWS avoidance sender-side algorithm

(2) SWS avoidance receiver-side algorithm

(3) Nagle algorithm

(4) delayed ACKs

(5) zero-window probing


With all of that 1122 content worked into 793bis, I did not think it 
made sense to retain this section of the document, but it will be good 
to confirm that with the WG.

For convenience, the content from 793 that I believe is no longer needed 
is pasted below:

     Window Management Suggestions

       Allocating a very small window causes data to be transmitted in
       many small segments when better performance is achieved using
       fewer large segments.

       One suggestion for avoiding small windows is for the receiver to
       defer updating a window until the additional allocation is at
       least X percent of the maximum allocation possible for the
       connection (where X might be 20 to 40).

       Another suggestion is for the sender to avoid sending small
       segments by waiting until the window is large enough before
       sending data.  If the the user signals a push function then the
       data must be sent even if it is a small segment.

       Note that the acknowledgments should not be delayed or unnecessary
       retransmissions will result.  One strategy would be to send an
       acknowledgment when a small segment arrives (with out updating the
       window information), and then to send another acknowledgment with
       new window information when the window is larger.

       The segment sent to probe a zero window may also begin a break up
       of transmitted data into smaller and smaller segments.  If a
       segment containing a single data octet sent to probe a zero window
       is accepted, it consumes one octet of the window now available.
       If the sending TCP simply sends as much as it can whenever the
       window is non zero, the transmitted data will be broken into
       alternating big and small segments.  As time goes on, occasional
       pauses in the receiver making window allocation available will
       result in breaking the big segments into a small and not quite so
       big pair. And after a while the data transmission will be in
       mostly small segments.

       The suggestion here is that the TCP implementations need to
       actively attempt to combine small window allocations into larger
       windows, since the mechanisms for managing the window tend to lead
       to many small windows in the simplest minded implementations.


Any thoughts or feedback on removing this from 793bis will be 
appreciated, thanks!



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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com> wrote:
> In the latest revision of the 793bis draft, I added content from RFC 1122 on
> the SWS avoidance algorithms for sender-side and receiver-side.  These are
> laid out in 1122, but were not present in 793.
>
> 793 includes a section called "Window Management Suggestions", that I
> believe was a precursor to the more-developed content in 1122 on:
>
> (1) SWS avoidance sender-side algorithm
>
> (2) SWS avoidance receiver-side algorithm
>
> (3) Nagle algorithm
>
> (4) delayed ACKs
>
> (5) zero-window probing
>
>
> With all of that 1122 content worked into 793bis, I did not think it made
> sense to retain this section of the document, but it will be good to confirm
> that with the WG.
>
> For convenience, the content from 793 that I believe is no longer needed is
> pasted below:
+1

BTW on https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3
<quote>
   A TCP MAY keep its offered receive window closed indefinitely.  As
   long as the receiving TCP continues to send acknowledgments in
   response to the probe segments, the sending TCP MUST allow the
   connection to stay open.
</quote>

The second unconditional MUST opens up a simple and effective DoS
attack, which Linux no longer wants to be vulnerable of.
https://patchwork.ozlabs.org/patch/392217/


>
>     Window Management Suggestions
>
>       Allocating a very small window causes data to be transmitted in
>       many small segments when better performance is achieved using
>       fewer large segments.
>
>       One suggestion for avoiding small windows is for the receiver to
>       defer updating a window until the additional allocation is at
>       least X percent of the maximum allocation possible for the
>       connection (where X might be 20 to 40).
>
>       Another suggestion is for the sender to avoid sending small
>       segments by waiting until the window is large enough before
>       sending data.  If the the user signals a push function then the
>       data must be sent even if it is a small segment.
>
>       Note that the acknowledgments should not be delayed or unnecessary
>       retransmissions will result.  One strategy would be to send an
>       acknowledgment when a small segment arrives (with out updating the
>       window information), and then to send another acknowledgment with
>       new window information when the window is larger.
>
>       The segment sent to probe a zero window may also begin a break up
>       of transmitted data into smaller and smaller segments.  If a
>       segment containing a single data octet sent to probe a zero window
>       is accepted, it consumes one octet of the window now available.
>       If the sending TCP simply sends as much as it can whenever the
>       window is non zero, the transmitted data will be broken into
>       alternating big and small segments.  As time goes on, occasional
>       pauses in the receiver making window allocation available will
>       result in breaking the big segments into a small and not quite so
>       big pair. And after a while the data transmission will be in
>       mostly small segments.
>
>       The suggestion here is that the TCP implementations need to
>       actively attempt to combine small window allocations into larger
>       windows, since the mechanisms for managing the window tend to lead
>       to many small windows in the simplest minded implementations.
>
>
> Any thoughts or feedback on removing this from 793bis will be appreciated,
> thanks!
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com> wrote:
>> In the latest revision of the 793bis draft, I added content from RFC 1122 on
>> the SWS avoidance algorithms for sender-side and receiver-side.  These are
>> laid out in 1122, but were not present in 793.
>> ...
>>
>> For convenience, the content from 793 that I believe is no longer needed is
>> pasted below:
> +1
+1

> BTW on https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3
> <quote>
>    A TCP MAY keep its offered receive window closed indefinitely.  As
>    long as the receiving TCP continues to send acknowledgments in
>    response to the probe segments, the sending TCP MUST allow the
>    connection to stay open.
> </quote>
>
> The second unconditional MUST opens up a simple and effective DoS
> attack, which Linux no longer wants to be vulnerable of.
> https://patchwork.ozlabs.org/patch/392217/

I strongly discourage attempts to try to bulletproof unsecured protocols.

IMO, the ability of an endpoint to keep its window closed indefinitely
is its prerogative, just as it is the prerogative of the other end to
close such a connection. AFAICT, the Linux code is consistent with the
second half of this statement, but I don't think it should be required
or even mentioned.

It's always obvious that an endpoint OS can and should protect its
resources by whatever means it thinks necessary. These are not protocol
decisions.

Joe



>
>
>>     Window Management Suggestions
>>
>>       Allocating a very small window causes data to be transmitted in
>>       many small segments when better performance is achieved using
>>       fewer large segments.
>>
>>       One suggestion for avoiding small windows is for the receiver to
>>       defer updating a window until the additional allocation is at
>>       least X percent of the maximum allocation possible for the
>>       connection (where X might be 20 to 40).
>>
>>       Another suggestion is for the sender to avoid sending small
>>       segments by waiting until the window is large enough before
>>       sending data.  If the the user signals a push function then the
>>       data must be sent even if it is a small segment.
>>
>>       Note that the acknowledgments should not be delayed or unnecessary
>>       retransmissions will result.  One strategy would be to send an
>>       acknowledgment when a small segment arrives (with out updating the
>>       window information), and then to send another acknowledgment with
>>       new window information when the window is larger.
>>
>>       The segment sent to probe a zero window may also begin a break up
>>       of transmitted data into smaller and smaller segments.  If a
>>       segment containing a single data octet sent to probe a zero window
>>       is accepted, it consumes one octet of the window now available.
>>       If the sending TCP simply sends as much as it can whenever the
>>       window is non zero, the transmitted data will be broken into
>>       alternating big and small segments.  As time goes on, occasional
>>       pauses in the receiver making window allocation available will
>>       result in breaking the big segments into a small and not quite so
>>       big pair. And after a while the data transmission will be in
>>       mostly small segments.
>>
>>       The suggestion here is that the TCP implementations need to
>>       actively attempt to combine small window allocations into larger
>>       windows, since the mechanisms for managing the window tend to lead
>>       to many small windows in the simplest minded implementations.
>>
>>
>> Any thoughts or feedback on removing this from 793bis will be appreciated,
>> thanks!
>>
>>
>> _______________________________________________
>> 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


From nobody Tue Aug  2 13:14:55 2016
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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The point about keeping the connection open in the face of zero window =
probes has the classic example of sending a job to the printer just =
before heading off to lunch, and when you come back later discover that =
the printer ran out of paper and stopped accepting new TCP data.  The =
printer keeps responding to the probes with a zero window, and when =
paper is finally added to the printer, printing resumes, the TCP window =
is opened and the job completes.

Years ago there were examples of situations like this where TCP =
connections were being timed out relatively quickly in the face of a =
zero window, causing failures when there was no need for a failure.  =
It=E2=80=99d be nice if trying to address a potential DOS situation =
didn=E2=80=99t cause normal situations to regress back to those days; it =
then becomes a matter of balancing how long to wait before giving up on =
a zero window connection.

			-David Borman

> On Aug 2, 2016, at 2:52 PM, Joe Touch <touch@ISI.EDU> wrote:
>=20
>=20
>=20
> On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
>> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com> =
wrote:
>>> In the latest revision of the 793bis draft, I added content from RFC =
1122 on
>>> the SWS avoidance algorithms for sender-side and receiver-side.  =
These are
>>> laid out in 1122, but were not present in 793.
>>> ...
>>>=20
>>> For convenience, the content from 793 that I believe is no longer =
needed is
>>> pasted below:
>> +1
> +1
>=20
>> BTW on =
https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3
>> <quote>
>>   A TCP MAY keep its offered receive window closed indefinitely.  As
>>   long as the receiving TCP continues to send acknowledgments in
>>   response to the probe segments, the sending TCP MUST allow the
>>   connection to stay open.
>> </quote>
>>=20
>> The second unconditional MUST opens up a simple and effective DoS
>> attack, which Linux no longer wants to be vulnerable of.
>> https://patchwork.ozlabs.org/patch/392217/
>=20
> I strongly discourage attempts to try to bulletproof unsecured =
protocols.
>=20
> IMO, the ability of an endpoint to keep its window closed indefinitely
> is its prerogative, just as it is the prerogative of the other end to
> close such a connection. AFAICT, the Linux code is consistent with the
> second half of this statement, but I don't think it should be required
> or even mentioned.
>=20
> It's always obvious that an endpoint OS can and should protect its
> resources by whatever means it thinks necessary. These are not =
protocol
> decisions.
>=20
> Joe
>=20
>=20
>=20
>>=20
>>=20
>>>    Window Management Suggestions
>>>=20
>>>      Allocating a very small window causes data to be transmitted in
>>>      many small segments when better performance is achieved using
>>>      fewer large segments.
>>>=20
>>>      One suggestion for avoiding small windows is for the receiver =
to
>>>      defer updating a window until the additional allocation is at
>>>      least X percent of the maximum allocation possible for the
>>>      connection (where X might be 20 to 40).
>>>=20
>>>      Another suggestion is for the sender to avoid sending small
>>>      segments by waiting until the window is large enough before
>>>      sending data.  If the the user signals a push function then the
>>>      data must be sent even if it is a small segment.
>>>=20
>>>      Note that the acknowledgments should not be delayed or =
unnecessary
>>>      retransmissions will result.  One strategy would be to send an
>>>      acknowledgment when a small segment arrives (with out updating =
the
>>>      window information), and then to send another acknowledgment =
with
>>>      new window information when the window is larger.
>>>=20
>>>      The segment sent to probe a zero window may also begin a break =
up
>>>      of transmitted data into smaller and smaller segments.  If a
>>>      segment containing a single data octet sent to probe a zero =
window
>>>      is accepted, it consumes one octet of the window now available.
>>>      If the sending TCP simply sends as much as it can whenever the
>>>      window is non zero, the transmitted data will be broken into
>>>      alternating big and small segments.  As time goes on, =
occasional
>>>      pauses in the receiver making window allocation available will
>>>      result in breaking the big segments into a small and not quite =
so
>>>      big pair. And after a while the data transmission will be in
>>>      mostly small segments.
>>>=20
>>>      The suggestion here is that the TCP implementations need to
>>>      actively attempt to combine small window allocations into =
larger
>>>      windows, since the mechanisms for managing the window tend to =
lead
>>>      to many small windows in the simplest minded implementations.
>>>=20
>>>=20
>>> Any thoughts or feedback on removing this from 793bis will be =
appreciated,
>>> thanks!
>>>=20
>>>=20
>>> _______________________________________________
>>> 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
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Tue Aug  2 13:18:37 2016
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To: David Borman <dab@weston.borman.com>
References: <c40a6f5a-3b98-d74d-27fb-7dd6de59af00@mti-systems.com> <CAK6E8=fAR1fBmfr4V6Hd1mfWkYX7nwjAybSyonFwfhz8k7P_6g@mail.gmail.com> <57A0FA15.6090507@isi.edu> <DAEB54D1-2253-46D4-8641-0CC8C86C7F61@weston.borman.com>
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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Hi, David,

I agree. IMO, DOS situations should be handled by the OS by withdrawing
resources *as needed*. That includes closing down zero-window situations
as below, as well as closing idle connections when resources are limited.

I don't see this as something the protocol needs to decide.

Joe

On 8/2/2016 1:14 PM, David Borman wrote:
> The point about keeping the connection open in the face of zero window probes has the classic example of sending a job to the printer just before heading off to lunch, and when you come back later discover that the printer ran out of paper and stopped accepting new TCP data.  The printer keeps responding to the probes with a zero window, and when paper is finally added to the printer, printing resumes, the TCP window is opened and the job completes.
>
> Years ago there were examples of situations like this where TCP connections were being timed out relatively quickly in the face of a zero window, causing failures when there was no need for a failure.  It’d be nice if trying to address a potential DOS situation didn’t cause normal situations to regress back to those days; it then becomes a matter of balancing how long to wait before giving up on a zero window connection.
>
> 			-David Borman
>
>> On Aug 2, 2016, at 2:52 PM, Joe Touch <touch@ISI.EDU> wrote:
>>
>>
>>
>> On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
>>> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com> wrote:
>>>> In the latest revision of the 793bis draft, I added content from RFC 1122 on
>>>> the SWS avoidance algorithms for sender-side and receiver-side.  These are
>>>> laid out in 1122, but were not present in 793.
>>>> ...
>>>>
>>>> For convenience, the content from 793 that I believe is no longer needed is
>>>> pasted below:
>>> +1
>> +1
>>
>>> BTW on https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3
>>> <quote>
>>>   A TCP MAY keep its offered receive window closed indefinitely.  As
>>>   long as the receiving TCP continues to send acknowledgments in
>>>   response to the probe segments, the sending TCP MUST allow the
>>>   connection to stay open.
>>> </quote>
>>>
>>> The second unconditional MUST opens up a simple and effective DoS
>>> attack, which Linux no longer wants to be vulnerable of.
>>> https://patchwork.ozlabs.org/patch/392217/
>> I strongly discourage attempts to try to bulletproof unsecured protocols.
>>
>> IMO, the ability of an endpoint to keep its window closed indefinitely
>> is its prerogative, just as it is the prerogative of the other end to
>> close such a connection. AFAICT, the Linux code is consistent with the
>> second half of this statement, but I don't think it should be required
>> or even mentioned.
>>
>> It's always obvious that an endpoint OS can and should protect its
>> resources by whatever means it thinks necessary. These are not protocol
>> decisions.
>>
>> Joe
>>
>>
>>
>>>
>>>>    Window Management Suggestions
>>>>
>>>>      Allocating a very small window causes data to be transmitted in
>>>>      many small segments when better performance is achieved using
>>>>      fewer large segments.
>>>>
>>>>      One suggestion for avoiding small windows is for the receiver to
>>>>      defer updating a window until the additional allocation is at
>>>>      least X percent of the maximum allocation possible for the
>>>>      connection (where X might be 20 to 40).
>>>>
>>>>      Another suggestion is for the sender to avoid sending small
>>>>      segments by waiting until the window is large enough before
>>>>      sending data.  If the the user signals a push function then the
>>>>      data must be sent even if it is a small segment.
>>>>
>>>>      Note that the acknowledgments should not be delayed or unnecessary
>>>>      retransmissions will result.  One strategy would be to send an
>>>>      acknowledgment when a small segment arrives (with out updating the
>>>>      window information), and then to send another acknowledgment with
>>>>      new window information when the window is larger.
>>>>
>>>>      The segment sent to probe a zero window may also begin a break up
>>>>      of transmitted data into smaller and smaller segments.  If a
>>>>      segment containing a single data octet sent to probe a zero window
>>>>      is accepted, it consumes one octet of the window now available.
>>>>      If the sending TCP simply sends as much as it can whenever the
>>>>      window is non zero, the transmitted data will be broken into
>>>>      alternating big and small segments.  As time goes on, occasional
>>>>      pauses in the receiver making window allocation available will
>>>>      result in breaking the big segments into a small and not quite so
>>>>      big pair. And after a while the data transmission will be in
>>>>      mostly small segments.
>>>>
>>>>      The suggestion here is that the TCP implementations need to
>>>>      actively attempt to combine small window allocations into larger
>>>>      windows, since the mechanisms for managing the window tend to lead
>>>>      to many small windows in the simplest minded implementations.
>>>>
>>>>
>>>> Any thoughts or feedback on removing this from 793bis will be appreciated,
>>>> thanks!
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
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From: Yuchung Cheng <ycheng@google.com>
Date: Tue, 2 Aug 2016 13:22:25 -0700
Message-ID: <CAK6E8=cmHhNDtNc2VvU9_w9sGsxpsH3qAXN+BMuYSj3i-=URew@mail.gmail.com>
To: David Borman <dab@weston.borman.com>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On Tue, Aug 2, 2016 at 1:14 PM, David Borman <dab@weston.borman.com> wrote:
> The point about keeping the connection open in the face of zero window pr=
obes has the classic example of sending a job to the printer just before he=
ading off to lunch, and when you come back later discover that the printer =
ran out of paper and stopped accepting new TCP data.  The printer keeps res=
ponding to the probes with a zero window, and when paper is finally added t=
o the printer, printing resumes, the TCP window is opened and the job compl=
etes.
>
> Years ago there were examples of situations like this where TCP connectio=
ns were being timed out relatively quickly in the face of a zero window, ca=
using failures when there was no need for a failure.  It=E2=80=99d be nice =
if trying to address a potential DOS situation didn=E2=80=99t cause normal =
situations to regress back to those days; it then becomes a matter of balan=
cing how long to wait before giving up on a zero window connection.
>
>                         -David Borman
I am well aware of the printer story and which part in my original
email did I advocate to remove the MUST?

Linux only chops the ropes when these connections have no real owners
(i.e., orphaned socket in Linux) and the amount exceeds a configurable
system limit.  I am mentioning there is a real attack behind this
simple MUST.

>
>> On Aug 2, 2016, at 2:52 PM, Joe Touch <touch@ISI.EDU> wrote:
>>
>>
>>
>> On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
>>> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com> wrot=
e:
>>>> In the latest revision of the 793bis draft, I added content from RFC 1=
122 on
>>>> the SWS avoidance algorithms for sender-side and receiver-side.  These=
 are
>>>> laid out in 1122, but were not present in 793.
>>>> ...
>>>>
>>>> For convenience, the content from 793 that I believe is no longer need=
ed is
>>>> pasted below:
>>> +1
>> +1
>>
>>> BTW on https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section=
-3.8.3.3
>>> <quote>
>>>   A TCP MAY keep its offered receive window closed indefinitely.  As
>>>   long as the receiving TCP continues to send acknowledgments in
>>>   response to the probe segments, the sending TCP MUST allow the
>>>   connection to stay open.
>>> </quote>
>>>
>>> The second unconditional MUST opens up a simple and effective DoS
>>> attack, which Linux no longer wants to be vulnerable of.
>>> https://patchwork.ozlabs.org/patch/392217/
>>
>> I strongly discourage attempts to try to bulletproof unsecured protocols=
.
>>
>> IMO, the ability of an endpoint to keep its window closed indefinitely
>> is its prerogative, just as it is the prerogative of the other end to
>> close such a connection. AFAICT, the Linux code is consistent with the
>> second half of this statement, but I don't think it should be required
>> or even mentioned.
>>
>> It's always obvious that an endpoint OS can and should protect its
>> resources by whatever means it thinks necessary. These are not protocol
>> decisions.
>>
>> Joe
>>
>>
>>
>>>
>>>
>>>>    Window Management Suggestions
>>>>
>>>>      Allocating a very small window causes data to be transmitted in
>>>>      many small segments when better performance is achieved using
>>>>      fewer large segments.
>>>>
>>>>      One suggestion for avoiding small windows is for the receiver to
>>>>      defer updating a window until the additional allocation is at
>>>>      least X percent of the maximum allocation possible for the
>>>>      connection (where X might be 20 to 40).
>>>>
>>>>      Another suggestion is for the sender to avoid sending small
>>>>      segments by waiting until the window is large enough before
>>>>      sending data.  If the the user signals a push function then the
>>>>      data must be sent even if it is a small segment.
>>>>
>>>>      Note that the acknowledgments should not be delayed or unnecessar=
y
>>>>      retransmissions will result.  One strategy would be to send an
>>>>      acknowledgment when a small segment arrives (with out updating th=
e
>>>>      window information), and then to send another acknowledgment with
>>>>      new window information when the window is larger.
>>>>
>>>>      The segment sent to probe a zero window may also begin a break up
>>>>      of transmitted data into smaller and smaller segments.  If a
>>>>      segment containing a single data octet sent to probe a zero windo=
w
>>>>      is accepted, it consumes one octet of the window now available.
>>>>      If the sending TCP simply sends as much as it can whenever the
>>>>      window is non zero, the transmitted data will be broken into
>>>>      alternating big and small segments.  As time goes on, occasional
>>>>      pauses in the receiver making window allocation available will
>>>>      result in breaking the big segments into a small and not quite so
>>>>      big pair. And after a while the data transmission will be in
>>>>      mostly small segments.
>>>>
>>>>      The suggestion here is that the TCP implementations need to
>>>>      actively attempt to combine small window allocations into larger
>>>>      windows, since the mechanisms for managing the window tend to lea=
d
>>>>      to many small windows in the simplest minded implementations.
>>>>
>>>>
>>>> Any thoughts or feedback on removing this from 793bis will be apprecia=
ted,
>>>> thanks!
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: Christian Huitema <huitema@microsoft.com>
To: Joe Touch <touch@isi.edu>, David Borman <dab@weston.borman.com>
Thread-Topic: [tcpm] 793bis: removing "Window Management Suggestions"
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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From nobody Tue Aug  2 13:32:50 2016
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On 8/2/2016 1:25 PM, Christian Huitema wrote:
> At a minimum, that means switching the language from "MUST allow the connection to stay open" to "SHOULD allow the connection to stay open" maybe with a caveat like "unless server size resource management requires that it be closed."
I disagree.

The entirety of every protocol would turn into "SHOULD, unless resources
become restricted". That's OS design, not protocol design.

> And the printer-out-of-paper scenario really ought to be solved at the application layer. After all, if you are waiting for a printout, knowing why you are waiting is somewhat helpful...
It should, but that's another reason not to build a solution into the
protocol.

Joe

>
> -- Christian Huitema
>
>
>
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Joe Touch
>> Sent: Tuesday, August 2, 2016 1:18 PM
>> To: David Borman <dab@weston.borman.com>
>> Cc: tcpm@ietf.org Extensions <tcpm@ietf.org>
>> Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
>>
>> Hi, David,
>>
>> I agree. IMO, DOS situations should be handled by the OS by withdrawing
>> resources *as needed*. That includes closing down zero-window situations
>> as below, as well as closing idle connections when resources are limited.
>>
>> I don't see this as something the protocol needs to decide.
>>
>> Joe
>>
>> On 8/2/2016 1:14 PM, David Borman wrote:
>>> The point about keeping the connection open in the face of zero window
>> probes has the classic example of sending a job to the printer just before
>> heading off to lunch, and when you come back later discover that the printer
>> ran out of paper and stopped accepting new TCP data.  The printer keeps
>> responding to the probes with a zero window, and when paper is finally added
>> to the printer, printing resumes, the TCP window is opened and the job
>> completes.
>>> Years ago there were examples of situations like this where TCP connections
>> were being timed out relatively quickly in the face of a zero window, causing
>> failures when there was no need for a failure.  It’d be nice if trying to address a
>> potential DOS situation didn’t cause normal situations to regress back to those
>> days; it then becomes a matter of balancing how long to wait before giving up
>> on a zero window connection.
>>> 			-David Borman
>>>
>>>> On Aug 2, 2016, at 2:52 PM, Joe Touch <touch@ISI.EDU> wrote:
>>>>
>>>>
>>>>
>>>> On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
>>>>> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com>
>> wrote:
>>>>>> In the latest revision of the 793bis draft, I added content from RFC 1122
>> on
>>>>>> the SWS avoidance algorithms for sender-side and receiver-side.  These
>> are
>>>>>> laid out in 1122, but were not present in 793.
>>>>>> ...
>>>>>>
>>>>>> For convenience, the content from 793 that I believe is no longer needed
>> is
>>>>>> pasted below:
>>>>> +1
>>>> +1
>>>>
>>>>> BTW on https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-
>> 3.8.3.3
>>>>> <quote>
>>>>>   A TCP MAY keep its offered receive window closed indefinitely.  As
>>>>>   long as the receiving TCP continues to send acknowledgments in
>>>>>   response to the probe segments, the sending TCP MUST allow the
>>>>>   connection to stay open.
>>>>> </quote>
>>>>>
>>>>> The second unconditional MUST opens up a simple and effective DoS
>>>>> attack, which Linux no longer wants to be vulnerable of.
>>>>> https://patchwork.ozlabs.org/patch/392217/
>>>> I strongly discourage attempts to try to bulletproof unsecured protocols.
>>>>
>>>> IMO, the ability of an endpoint to keep its window closed indefinitely
>>>> is its prerogative, just as it is the prerogative of the other end to
>>>> close such a connection. AFAICT, the Linux code is consistent with the
>>>> second half of this statement, but I don't think it should be required
>>>> or even mentioned.
>>>>
>>>> It's always obvious that an endpoint OS can and should protect its
>>>> resources by whatever means it thinks necessary. These are not protocol
>>>> decisions.
>>>>
>>>> Joe
>>>>
>>>>
>>>>
>>>>>>    Window Management Suggestions
>>>>>>
>>>>>>      Allocating a very small window causes data to be transmitted in
>>>>>>      many small segments when better performance is achieved using
>>>>>>      fewer large segments.
>>>>>>
>>>>>>      One suggestion for avoiding small windows is for the receiver to
>>>>>>      defer updating a window until the additional allocation is at
>>>>>>      least X percent of the maximum allocation possible for the
>>>>>>      connection (where X might be 20 to 40).
>>>>>>
>>>>>>      Another suggestion is for the sender to avoid sending small
>>>>>>      segments by waiting until the window is large enough before
>>>>>>      sending data.  If the the user signals a push function then the
>>>>>>      data must be sent even if it is a small segment.
>>>>>>
>>>>>>      Note that the acknowledgments should not be delayed or unnecessary
>>>>>>      retransmissions will result.  One strategy would be to send an
>>>>>>      acknowledgment when a small segment arrives (with out updating the
>>>>>>      window information), and then to send another acknowledgment with
>>>>>>      new window information when the window is larger.
>>>>>>
>>>>>>      The segment sent to probe a zero window may also begin a break up
>>>>>>      of transmitted data into smaller and smaller segments.  If a
>>>>>>      segment containing a single data octet sent to probe a zero window
>>>>>>      is accepted, it consumes one octet of the window now available.
>>>>>>      If the sending TCP simply sends as much as it can whenever the
>>>>>>      window is non zero, the transmitted data will be broken into
>>>>>>      alternating big and small segments.  As time goes on, occasional
>>>>>>      pauses in the receiver making window allocation available will
>>>>>>      result in breaking the big segments into a small and not quite so
>>>>>>      big pair. And after a while the data transmission will be in
>>>>>>      mostly small segments.
>>>>>>
>>>>>>      The suggestion here is that the TCP implementations need to
>>>>>>      actively attempt to combine small window allocations into larger
>>>>>>      windows, since the mechanisms for managing the window tend to lead
>>>>>>      to many small windows in the simplest minded implementations.
>>>>>>
>>>>>>
>>>>>> Any thoughts or feedback on removing this from 793bis will be
>> appreciated,
>>>>>> thanks!
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>> _______________________________________________
>>>> 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
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On 8/2/2016 1:22 PM, Yuchung Cheng wrote:
> On Tue, Aug 2, 2016 at 1:14 PM, David Borman <dab@weston.borman.com> wrote:
>> > The point about keeping the connection open in the face of zero window probes has the classic example of sending a job to the printer just before heading off to lunch, and when you come back later discover that the printer ran out of paper and stopped accepting new TCP data.  The printer keeps responding to the probes with a zero window, and when paper is finally added to the printer, printing resumes, the TCP window is opened and the job completes.
>> >
>> > Years ago there were examples of situations like this where TCP connections were being timed out relatively quickly in the face of a zero window, causing failures when there was no need for a failure.  It’d be nice if trying to address a potential DOS situation didn’t cause normal situations to regress back to those days; it then becomes a matter of balancing how long to wait before giving up on a zero window connection.
>> >
>> >                         -David Borman
> I am well aware of the printer story and which part in my original
> email did I advocate to remove the MUST?
>
> Linux only chops the ropes when these connections have no real owners
> (i.e., orphaned socket in Linux) and the amount exceeds a configurable
> system limit.  I am mentioning there is a real attack behind this
> simple MUST.

There MIGHT BE an attack. There might not. Just because an endpoint does
something you didn't expect, does not mean it is malicious.

A variant of the Postel Principle applies here - IMO, anything a benign
endpoint CAN do MUST NOT be interpreted as an attack.

A node has the right to protect itself and its resources, but should not
infer malice where it isn't known to exist. It might be worth noting
somewhere the ways in which an *implementation* of TCP can help protect
resources, but that's an implementation suggestion, not protocol guidance.

Joe


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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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Sorry, my intention was to provide context on the mailing list for why =
that MUST is there, not necessarily aimed at you, but for anyone else =
that is reading this list and might not be familiar with the background.

I think it should remain a MUST, but implementations are always free to =
do what they need to do when under attack or run out of resources.
			-David

> On Aug 2, 2016, at 3:22 PM, Yuchung Cheng <ycheng@google.com> wrote:
>=20
> On Tue, Aug 2, 2016 at 1:14 PM, David Borman <dab@weston.borman.com> =
wrote:
>> The point about keeping the connection open in the face of zero =
window probes has the classic example of sending a job to the printer =
just before heading off to lunch, and when you come back later discover =
that the printer ran out of paper and stopped accepting new TCP data.  =
The printer keeps responding to the probes with a zero window, and when =
paper is finally added to the printer, printing resumes, the TCP window =
is opened and the job completes.
>>=20
>> Years ago there were examples of situations like this where TCP =
connections were being timed out relatively quickly in the face of a =
zero window, causing failures when there was no need for a failure.  =
It=E2=80=99d be nice if trying to address a potential DOS situation =
didn=E2=80=99t cause normal situations to regress back to those days; it =
then becomes a matter of balancing how long to wait before giving up on =
a zero window connection.
>>=20
>>                        -David Borman
> I am well aware of the printer story and which part in my original
> email did I advocate to remove the MUST?
>=20
> Linux only chops the ropes when these connections have no real owners
> (i.e., orphaned socket in Linux) and the amount exceeds a configurable
> system limit.  I am mentioning there is a real attack behind this
> simple MUST.
>=20
>>=20
>>> On Aug 2, 2016, at 2:52 PM, Joe Touch <touch@ISI.EDU> wrote:
>>>=20
>>>=20
>>>=20
>>> On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
>>>> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com> =
wrote:
>>>>> In the latest revision of the 793bis draft, I added content from =
RFC 1122 on
>>>>> the SWS avoidance algorithms for sender-side and receiver-side.  =
These are
>>>>> laid out in 1122, but were not present in 793.
>>>>> ...
>>>>>=20
>>>>> For convenience, the content from 793 that I believe is no longer =
needed is
>>>>> pasted below:
>>>> +1
>>> +1
>>>=20
>>>> BTW on =
https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3
>>>> <quote>
>>>>  A TCP MAY keep its offered receive window closed indefinitely.  As
>>>>  long as the receiving TCP continues to send acknowledgments in
>>>>  response to the probe segments, the sending TCP MUST allow the
>>>>  connection to stay open.
>>>> </quote>
>>>>=20
>>>> The second unconditional MUST opens up a simple and effective DoS
>>>> attack, which Linux no longer wants to be vulnerable of.
>>>> https://patchwork.ozlabs.org/patch/392217/
>>>=20
>>> I strongly discourage attempts to try to bulletproof unsecured =
protocols.
>>>=20
>>> IMO, the ability of an endpoint to keep its window closed =
indefinitely
>>> is its prerogative, just as it is the prerogative of the other end =
to
>>> close such a connection. AFAICT, the Linux code is consistent with =
the
>>> second half of this statement, but I don't think it should be =
required
>>> or even mentioned.
>>>=20
>>> It's always obvious that an endpoint OS can and should protect its
>>> resources by whatever means it thinks necessary. These are not =
protocol
>>> decisions.
>>>=20
>>> Joe
>>>=20
>>>=20
>>>=20
>>>>=20
>>>>=20
>>>>>   Window Management Suggestions
>>>>>=20
>>>>>     Allocating a very small window causes data to be transmitted =
in
>>>>>     many small segments when better performance is achieved using
>>>>>     fewer large segments.
>>>>>=20
>>>>>     One suggestion for avoiding small windows is for the receiver =
to
>>>>>     defer updating a window until the additional allocation is at
>>>>>     least X percent of the maximum allocation possible for the
>>>>>     connection (where X might be 20 to 40).
>>>>>=20
>>>>>     Another suggestion is for the sender to avoid sending small
>>>>>     segments by waiting until the window is large enough before
>>>>>     sending data.  If the the user signals a push function then =
the
>>>>>     data must be sent even if it is a small segment.
>>>>>=20
>>>>>     Note that the acknowledgments should not be delayed or =
unnecessary
>>>>>     retransmissions will result.  One strategy would be to send an
>>>>>     acknowledgment when a small segment arrives (with out updating =
the
>>>>>     window information), and then to send another acknowledgment =
with
>>>>>     new window information when the window is larger.
>>>>>=20
>>>>>     The segment sent to probe a zero window may also begin a break =
up
>>>>>     of transmitted data into smaller and smaller segments.  If a
>>>>>     segment containing a single data octet sent to probe a zero =
window
>>>>>     is accepted, it consumes one octet of the window now =
available.
>>>>>     If the sending TCP simply sends as much as it can whenever the
>>>>>     window is non zero, the transmitted data will be broken into
>>>>>     alternating big and small segments.  As time goes on, =
occasional
>>>>>     pauses in the receiver making window allocation available will
>>>>>     result in breaking the big segments into a small and not quite =
so
>>>>>     big pair. And after a while the data transmission will be in
>>>>>     mostly small segments.
>>>>>=20
>>>>>     The suggestion here is that the TCP implementations need to
>>>>>     actively attempt to combine small window allocations into =
larger
>>>>>     windows, since the mechanisms for managing the window tend to =
lead
>>>>>     to many small windows in the simplest minded implementations.
>>>>>=20
>>>>>=20
>>>>> Any thoughts or feedback on removing this from 793bis will be =
appreciated,
>>>>> thanks!
>>>>>=20
>>>>>=20
>>>>> _______________________________________________
>>>>> 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
>>>=20
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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References: <c40a6f5a-3b98-d74d-27fb-7dd6de59af00@mti-systems.com> <CAK6E8=fAR1fBmfr4V6Hd1mfWkYX7nwjAybSyonFwfhz8k7P_6g@mail.gmail.com> <57A0FA15.6090507@isi.edu> <DAEB54D1-2253-46D4-8641-0CC8C86C7F61@weston.borman.com> <CAK6E8=cmHhNDtNc2VvU9_w9sGsxpsH3qAXN+BMuYSj3i-=URew@mail.gmail.com> <A86AF09E-0CDC-4B8C-8475-C15DAAECE909@weston.borman.com>
From: John Heffner <johnwheffner@gmail.com>
Date: Tue, 2 Aug 2016 17:14:50 -0400
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To: David Borman <dab@weston.borman.com>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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This reminds me a lot of an old thread:
https://www.ietf.org/mail-archive/web/tcpm/current/msg04040.html

On Tue, Aug 2, 2016 at 4:56 PM, David Borman <dab@weston.borman.com> wrote:

> Sorry, my intention was to provide context on the mailing list for why
> that MUST is there, not necessarily aimed at you, but for anyone else tha=
t
> is reading this list and might not be familiar with the background.
>
> I think it should remain a MUST, but implementations are always free to d=
o
> what they need to do when under attack or run out of resources.
>                         -David
>
> > On Aug 2, 2016, at 3:22 PM, Yuchung Cheng <ycheng@google.com> wrote:
> >
> > On Tue, Aug 2, 2016 at 1:14 PM, David Borman <dab@weston.borman.com>
> wrote:
> >> The point about keeping the connection open in the face of zero window
> probes has the classic example of sending a job to the printer just befor=
e
> heading off to lunch, and when you come back later discover that the
> printer ran out of paper and stopped accepting new TCP data.  The printer
> keeps responding to the probes with a zero window, and when paper is
> finally added to the printer, printing resumes, the TCP window is opened
> and the job completes.
> >>
> >> Years ago there were examples of situations like this where TCP
> connections were being timed out relatively quickly in the face of a zero
> window, causing failures when there was no need for a failure.  It=E2=80=
=99d be
> nice if trying to address a potential DOS situation didn=E2=80=99t cause =
normal
> situations to regress back to those days; it then becomes a matter of
> balancing how long to wait before giving up on a zero window connection.
> >>
> >>                        -David Borman
> > I am well aware of the printer story and which part in my original
> > email did I advocate to remove the MUST?
> >
> > Linux only chops the ropes when these connections have no real owners
> > (i.e., orphaned socket in Linux) and the amount exceeds a configurable
> > system limit.  I am mentioning there is a real attack behind this
> > simple MUST.
> >
> >>
> >>> On Aug 2, 2016, at 2:52 PM, Joe Touch <touch@ISI.EDU> wrote:
> >>>
> >>>
> >>>
> >>> On 8/2/2016 12:42 PM, Yuchung Cheng wrote:
> >>>> On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy <wes@mti-systems.com>
> wrote:
> >>>>> In the latest revision of the 793bis draft, I added content from RF=
C
> 1122 on
> >>>>> the SWS avoidance algorithms for sender-side and receiver-side.
> These are
> >>>>> laid out in 1122, but were not present in 793.
> >>>>> ...
> >>>>>
> >>>>> For convenience, the content from 793 that I believe is no longer
> needed is
> >>>>> pasted below:
> >>>> +1
> >>> +1
> >>>
> >>>> BTW on
> https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3
> >>>> <quote>
> >>>>  A TCP MAY keep its offered receive window closed indefinitely.  As
> >>>>  long as the receiving TCP continues to send acknowledgments in
> >>>>  response to the probe segments, the sending TCP MUST allow the
> >>>>  connection to stay open.
> >>>> </quote>
> >>>>
> >>>> The second unconditional MUST opens up a simple and effective DoS
> >>>> attack, which Linux no longer wants to be vulnerable of.
> >>>> https://patchwork.ozlabs.org/patch/392217/
> >>>
> >>> I strongly discourage attempts to try to bulletproof unsecured
> protocols.
> >>>
> >>> IMO, the ability of an endpoint to keep its window closed indefinitel=
y
> >>> is its prerogative, just as it is the prerogative of the other end to
> >>> close such a connection. AFAICT, the Linux code is consistent with th=
e
> >>> second half of this statement, but I don't think it should be require=
d
> >>> or even mentioned.
> >>>
> >>> It's always obvious that an endpoint OS can and should protect its
> >>> resources by whatever means it thinks necessary. These are not protoc=
ol
> >>> decisions.
> >>>
> >>> Joe
> >>>
> >>>
> >>>
> >>>>
> >>>>
> >>>>>   Window Management Suggestions
> >>>>>
> >>>>>     Allocating a very small window causes data to be transmitted in
> >>>>>     many small segments when better performance is achieved using
> >>>>>     fewer large segments.
> >>>>>
> >>>>>     One suggestion for avoiding small windows is for the receiver t=
o
> >>>>>     defer updating a window until the additional allocation is at
> >>>>>     least X percent of the maximum allocation possible for the
> >>>>>     connection (where X might be 20 to 40).
> >>>>>
> >>>>>     Another suggestion is for the sender to avoid sending small
> >>>>>     segments by waiting until the window is large enough before
> >>>>>     sending data.  If the the user signals a push function then the
> >>>>>     data must be sent even if it is a small segment.
> >>>>>
> >>>>>     Note that the acknowledgments should not be delayed or
> unnecessary
> >>>>>     retransmissions will result.  One strategy would be to send an
> >>>>>     acknowledgment when a small segment arrives (with out updating
> the
> >>>>>     window information), and then to send another acknowledgment wi=
th
> >>>>>     new window information when the window is larger.
> >>>>>
> >>>>>     The segment sent to probe a zero window may also begin a break =
up
> >>>>>     of transmitted data into smaller and smaller segments.  If a
> >>>>>     segment containing a single data octet sent to probe a zero
> window
> >>>>>     is accepted, it consumes one octet of the window now available.
> >>>>>     If the sending TCP simply sends as much as it can whenever the
> >>>>>     window is non zero, the transmitted data will be broken into
> >>>>>     alternating big and small segments.  As time goes on, occasiona=
l
> >>>>>     pauses in the receiver making window allocation available will
> >>>>>     result in breaking the big segments into a small and not quite =
so
> >>>>>     big pair. And after a while the data transmission will be in
> >>>>>     mostly small segments.
> >>>>>
> >>>>>     The suggestion here is that the TCP implementations need to
> >>>>>     actively attempt to combine small window allocations into large=
r
> >>>>>     windows, since the mechanisms for managing the window tend to
> lead
> >>>>>     to many small windows in the simplest minded implementations.
> >>>>>
> >>>>>
> >>>>> Any thoughts or feedback on removing this from 793bis will be
> appreciated,
> >>>>> thanks!
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> 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
> >>>
> >>> _______________________________________________
> >>> 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
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr">This reminds me a lot of an old thread:=C2=A0<a href=3D"ht=
tps://www.ietf.org/mail-archive/web/tcpm/current/msg04040.html">https://www=
.ietf.org/mail-archive/web/tcpm/current/msg04040.html</a></div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Aug 2, 2016 at 4:56=
 PM, David Borman <span dir=3D"ltr">&lt;<a href=3D"mailto:dab@weston.borman=
.com" target=3D"_blank">dab@weston.borman.com</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">Sorry, my intention was to provide context on th=
e mailing list for why that MUST is there, not necessarily aimed at you, bu=
t for anyone else that is reading this list and might not be familiar with =
the background.<br>
<br>
I think it should remain a MUST, but implementations are always free to do =
what they need to do when under attack or run out of resources.<br>
<span class=3D"HOEnZb"><font color=3D"#888888">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 -David<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt; On Aug 2, 2016, at 3:22 PM, Yuchung Cheng &lt;<a href=3D"mailto:ycheng=
@google.com">ycheng@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On Tue, Aug 2, 2016 at 1:14 PM, David Borman &lt;<a href=3D"mailto:dab=
@weston.borman.com">dab@weston.borman.com</a>&gt; wrote:<br>
&gt;&gt; The point about keeping the connection open in the face of zero wi=
ndow probes has the classic example of sending a job to the printer just be=
fore heading off to lunch, and when you come back later discover that the p=
rinter ran out of paper and stopped accepting new TCP data.=C2=A0 The print=
er keeps responding to the probes with a zero window, and when paper is fin=
ally added to the printer, printing resumes, the TCP window is opened and t=
he job completes.<br>
&gt;&gt;<br>
&gt;&gt; Years ago there were examples of situations like this where TCP co=
nnections were being timed out relatively quickly in the face of a zero win=
dow, causing failures when there was no need for a failure.=C2=A0 It=E2=80=
=99d be nice if trying to address a potential DOS situation didn=E2=80=99t =
cause normal situations to regress back to those days; it then becomes a ma=
tter of balancing how long to wait before giving up on a zero window connec=
tion.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 -David Borman<br>
&gt; I am well aware of the printer story and which part in my original<br>
&gt; email did I advocate to remove the MUST?<br>
&gt;<br>
&gt; Linux only chops the ropes when these connections have no real owners<=
br>
&gt; (i.e., orphaned socket in Linux) and the amount exceeds a configurable=
<br>
&gt; system limit.=C2=A0 I am mentioning there is a real attack behind this=
<br>
&gt; simple MUST.<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt; On Aug 2, 2016, at 2:52 PM, Joe Touch &lt;<a href=3D"mailto:to=
uch@ISI.EDU">touch@ISI.EDU</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On 8/2/2016 12:42 PM, Yuchung Cheng wrote:<br>
&gt;&gt;&gt;&gt; On Tue, Aug 2, 2016 at 12:15 PM, Wesley Eddy &lt;<a href=
=3D"mailto:wes@mti-systems.com">wes@mti-systems.com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt; In the latest revision of the 793bis draft, I added co=
ntent from RFC 1122 on<br>
&gt;&gt;&gt;&gt;&gt; the SWS avoidance algorithms for sender-side and recei=
ver-side.=C2=A0 These are<br>
&gt;&gt;&gt;&gt;&gt; laid out in 1122, but were not present in 793.<br>
&gt;&gt;&gt;&gt;&gt; ...<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; For convenience, the content from 793 that I believe i=
s no longer needed is<br>
&gt;&gt;&gt;&gt;&gt; pasted below:<br>
&gt;&gt;&gt;&gt; +1<br>
&gt;&gt;&gt; +1<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; BTW on <a href=3D"https://tools.ietf.org/html/draft-ietf-t=
cpm-rfc793bis-03#section-3.8.3.3" rel=3D"noreferrer" target=3D"_blank">http=
s://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-03#section-3.8.3.3</a><br=
>
&gt;&gt;&gt;&gt; &lt;quote&gt;<br>
&gt;&gt;&gt;&gt;=C2=A0 A TCP MAY keep its offered receive window closed ind=
efinitely.=C2=A0 As<br>
&gt;&gt;&gt;&gt;=C2=A0 long as the receiving TCP continues to send acknowle=
dgments in<br>
&gt;&gt;&gt;&gt;=C2=A0 response to the probe segments, the sending TCP MUST=
 allow the<br>
&gt;&gt;&gt;&gt;=C2=A0 connection to stay open.<br>
&gt;&gt;&gt;&gt; &lt;/quote&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; The second unconditional MUST opens up a simple and effect=
ive DoS<br>
&gt;&gt;&gt;&gt; attack, which Linux no longer wants to be vulnerable of.<b=
r>
&gt;&gt;&gt;&gt; <a href=3D"https://patchwork.ozlabs.org/patch/392217/" rel=
=3D"noreferrer" target=3D"_blank">https://patchwork.ozlabs.org/patch/392217=
/</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I strongly discourage attempts to try to bulletproof unsecured=
 protocols.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; IMO, the ability of an endpoint to keep its window closed inde=
finitely<br>
&gt;&gt;&gt; is its prerogative, just as it is the prerogative of the other=
 end to<br>
&gt;&gt;&gt; close such a connection. AFAICT, the Linux code is consistent =
with the<br>
&gt;&gt;&gt; second half of this statement, but I don&#39;t think it should=
 be required<br>
&gt;&gt;&gt; or even mentioned.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; It&#39;s always obvious that an endpoint OS can and should pro=
tect its<br>
&gt;&gt;&gt; resources by whatever means it thinks necessary. These are not=
 protocol<br>
&gt;&gt;&gt; decisions.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Joe<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0Window Management Suggestions<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0Allocating a very small window caus=
es data to be transmitted in<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0many small segments when better per=
formance is achieved using<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0fewer large segments.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0One suggestion for avoiding small w=
indows is for the receiver to<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0defer updating a window until the a=
dditional allocation is at<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0least X percent of the maximum allo=
cation possible for the<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0connection (where X might be 20 to =
40).<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0Another suggestion is for the sende=
r to avoid sending small<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0segments by waiting until the windo=
w is large enough before<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0sending data.=C2=A0 If the the user=
 signals a push function then the<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0data must be sent even if it is a s=
mall segment.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0Note that the acknowledgments shoul=
d not be delayed or unnecessary<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0retransmissions will result.=C2=A0 =
One strategy would be to send an<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0acknowledgment when a small segment=
 arrives (with out updating the<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0window information), and then to se=
nd another acknowledgment with<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0new window information when the win=
dow is larger.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0The segment sent to probe a zero wi=
ndow may also begin a break up<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0of transmitted data into smaller an=
d smaller segments.=C2=A0 If a<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0segment containing a single data oc=
tet sent to probe a zero window<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0is accepted, it consumes one octet =
of the window now available.<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0If the sending TCP simply sends as =
much as it can whenever the<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0window is non zero, the transmitted=
 data will be broken into<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0alternating big and small segments.=
=C2=A0 As time goes on, occasional<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0pauses in the receiver making windo=
w allocation available will<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0result in breaking the big segments=
 into a small and not quite so<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0big pair. And after a while the dat=
a transmission will be in<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0mostly small segments.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0The suggestion here is that the TCP=
 implementations need to<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0actively attempt to combine small w=
indow allocations into larger<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0windows, since the mechanisms for m=
anaging the window tend to lead<br>
&gt;&gt;&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0to many small windows in the simple=
st minded implementations.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Any thoughts or feedback on removing this from 793bis =
will be appreciated,<br>
&gt;&gt;&gt;&gt;&gt; thanks!<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt;&gt;&gt; tcpm mailing list<br>
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From nobody Tue Aug  2 15:42:11 2016
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On 8/2/2016 2:14 PM, John Heffner wrote:
> This reminds me a lot of an old
> thread: https://www.ietf.org/mail-archive/web/tcpm/current/msg04040.html

Thanks for pointing that out.

RFC 6429 (the resulting doc) already says it's OK for the OS to
terminate ZWP connections if it runs out of resources.

It also very clearly states that this isn't a change to the standard and
that it is an implementation issue.

Thus I don't think this belongs in RFC793bis.

Joe





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On Tue, Aug 2, 2016 at 3:41 PM, Joe Touch <touch@isi.edu> wrote:
>
>
> On 8/2/2016 2:14 PM, John Heffner wrote:
>> This reminds me a lot of an old
>> thread: https://www.ietf.org/mail-archive/web/tcpm/current/msg04040.html
Thanks for the reference. I suggest a quick reference to RFC6429 at
the end of the section 3.8.3.3 in 793bis. RFC 793 is THE reference to
TCP, so making it informational on various caveats for new
implementors is really useful.

(and I expect Joe to disagree again, which he already did).

>
> Thanks for pointing that out.
>
> RFC 6429 (the resulting doc) already says it's OK for the OS to
> terminate ZWP connections if it runs out of resources.
>
> It also very clearly states that this isn't a change to the standard and
> that it is an implementation issue.
>
> Thus I don't think this belongs in RFC793bis.
>
> Joe
>
>
>
>


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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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-----BEGIN PGP SIGNED MESSAGE-----=0A=
Hash: SHA512=0A=
=0A=
On 08/02/2016 01:18 PM, Joe Touch wrote:=0A=
> Hi, David,=0A=
> =0A=
> I agree. IMO, DOS situations should be handled by the OS by=0A=
> withdrawing resources *as needed*. That includes closing down=0A=
> zero-window situations as below, as well as closing idle=0A=
> connections when resources are limited.=0A=
> =0A=
> I don't see this as something the protocol needs to decide.=0A=
=0A=
Right.  The relevant distinction is between OS and protocol.  The same=0A=
way the OS needn't inform open connections that the power has failed,=0A=
the OS needn't tell open connections that it's responding to a DDoS.=0A=
The possibility of catastrophic failure is assumed.=0A=
=0A=
(BTW, Joe, we're agreeing here. :-D)=0A=
=0A=
=0A=
- -- =0A=
Ted Faber <theodore.v.faber@aero.org>=0A=
Engineering Specialist=0A=
Computer Systems Research Department=0A=
310-336-7373=0A=
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From nobody Wed Aug  3 14:10:22 2016
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On 8/3/2016 12:49 PM, Yuchung Cheng wrote:
> On Tue, Aug 2, 2016 at 3:41 PM, Joe Touch <touch@isi.edu> wrote:
>>
>> On 8/2/2016 2:14 PM, John Heffner wrote:
>>> This reminds me a lot of an old
>>> thread: https://www.ietf.org/mail-archive/web/tcpm/current/msg04040.html
> Thanks for the reference. I suggest a quick reference to RFC6429 at
> the end of the section 3.8.3.3 in 793bis. RFC 793 is THE reference to
> TCP, so making it informational on various caveats for new
> implementors is really useful.
>
> (and I expect Joe to disagree again, which he already did).
You're right.

To clarify, my view is that RFC6429 is a recommendation to the OS, not
to TCP. That's the reason I don't think it belongs in a TCP doc. I think
it would simply waste space to put in every way in which the OS needs to
manage resources that TCP might either contribute to or be affected by.

793bis should not become a "junk drawer". The cleaner and simpler it is,
the more likely it will be supported correctly.

Joe


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From: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] 793bis: removing "Window Management Suggestions"
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On Wed, Aug 3, 2016 at 2:09 PM, Joe Touch <touch@isi.edu> wrote:
>
>
> On 8/3/2016 12:49 PM, Yuchung Cheng wrote:
>> On Tue, Aug 2, 2016 at 3:41 PM, Joe Touch <touch@isi.edu> wrote:
>>>
>>> On 8/2/2016 2:14 PM, John Heffner wrote:
>>>> This reminds me a lot of an old
>>>> thread: https://www.ietf.org/mail-archive/web/tcpm/current/msg04040.html
>> Thanks for the reference. I suggest a quick reference to RFC6429 at
>> the end of the section 3.8.3.3 in 793bis. RFC 793 is THE reference to
>> TCP, so making it informational on various caveats for new
>> implementors is really useful.
>>
>> (and I expect Joe to disagree again, which he already did).
> You're right.
>
> To clarify, my view is that RFC6429 is a recommendation to the OS, not
> to TCP. That's the reason I don't think it belongs in a TCP doc. I think

RFC6429 says
<quote>
Rather than making a change to the standard, this document clarifies
what has been until now a misinterpretation of the tandard as
specified in RFC 1122.
</quote>

RFC793-bis says
<quote>
1.  Purpose and Scope
[snip]
   For several decades, RFC 793 plus a number of other documents have
   combined to serve as the specification for TCP [17].  Over time, a
   number of errata have been identified on RFC 793, as well as
   deficiencies in security, performance, and other aspects.  A number
   of enhancements has grown and been documented separately.  These
were  never accumulated together into an update to the base
specification.
</quote>

> it would simply waste space to put in every way in which the OS needs to
> manage resources that TCP might either contribute to or be affected by.
>
> 793bis should not become a "junk drawer". The cleaner and simpler it is,
> the more likely it will be supported correctly.
I will agree with you if RFC6429 is a piece of junk. but I don't think so.

>
> Joe


From kobby.Carmona@qlogic.com  Thu Aug  4 02:09:09 2016
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From: Kobby Carmona <kobby.Carmona@qlogic.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: Possible deadlock scenario with retransmission on both sides at the same time
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Subject: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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Hi all,
While running a bidirectional scenario with random drops in a network simul=
ator of our (QLogic's NIC) TCP stack we found a case where it seems there i=
s deadlock in the TCP protocol (the connection will keep sending pure acks =
from both sides until RTO will expire multiple times and a RST will sent to=
 close the connection).
The scenario is as follows (there is an example with numbers for each stage=
 assuming the MSS and each packet is 1000B):
1. Both sides are transmitting data and a single packet is dropped on eithe=
r side and the next two packets are received properly
	Side A - SND.MAX=3D3000, SND.NXT=3D3000, SND.UNA=3D1000, RCV.NXT=3D11000, =
out-of-order block 12000-13000
	Side B - SND.MAX =3D13000, SND.NXT =3D13000, SND.UNA=3D11000, RCV.NXT=3D10=
00, out-of-order block 2000-3000
2. RTO timer expires on both sides
	Side A - SND.MAX=3D3000, SND.NXT=3D1000, SND.UNA=3D1000, RCV.NXT=3D11000, =
out-of-order block 12000-13000
	Side B - SND.MAX =3D13000, SND.NXT=3D11000, SND.UNA=3D11000, RCV.NXT=3D100=
0, out-of-order block 2000-3000
3. Both sides transmit a single packet to the peer:
	A->B - pkt.seq=3D1000, pkt.ack=3D11000, len=3D1000
	B->A - pkt.seq=3D11000, pkt.ack=3D1000, len=3D1000
3. Both sides receive the packets and update the receive context:
	Side A - SND.MAX=3D3000, SND.NXT=3D2000, SND.UNA=3D1000, RCV.NXT=3D13000
	Side B - SND.MAX=3D13000, SND.NXT=3D12000, SND.UNA=3D11000, RCV.NXT=3D3000
4. Both sides send another segment:
	A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D1000
	B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D1000
5. Both sides don't accept the packet (and don't update SND.UNA) since the =
sequence on the packet is less than RCV.NXT (sequence number check in page =
69 of RFC793) and send a pure ACK instead
	A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D0 (pure ACK)
	B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D0 (pure ACK)
6. This will continue forever (until the connection will be terminated by R=
ST) since every packet that ends before RCV.NXT (even a retransmit from SND=
.UNA) will be dropped.

Did anyone encountered this issue before? Is the anything we missed on this=
 sequence?
If this is indeed a real deadlock, there might be several solutions to this=
 which will require a modification in receive processing of RFC793. But I w=
ould like to know if you think this is a real issue before dealing with sol=
utions.
Thanks,

	Kobby



From nobody Thu Aug  4 12:36:43 2016
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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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After you pull back SND.NXT and do the retransmission, you should then =
restore SND.NXT back to where it was, not leave it at the backed off =
value; then the ACKs wouldn=E2=80=99t be dropped, since they wouldn=E2=80=99=
t have old seq values.  For example, in the 4.4BSD fast retransmit code =
it had:

			tcp_seq onxt =3D tp->snd_nxt;
			...
			tp->snd_nxt =3D th->th_ack;
			...
			(void) tcp_output(tp);
			...
			if (SEQ_GT(onxt, tp->snd_nxt)) =20
				tp->snd_nxt =3D onxt;


			-David Borman

> On Aug 4, 2016, at 4:08 AM, Kobby Carmona <kobby.Carmona@qlogic.com> =
wrote:
>=20
> Hi all,
> While running a bidirectional scenario with random drops in a network =
simulator of our (QLogic's NIC) TCP stack we found a case where it seems =
there is deadlock in the TCP protocol (the connection will keep sending =
pure acks from both sides until RTO will expire multiple times and a RST =
will sent to close the connection).
> The scenario is as follows (there is an example with numbers for each =
stage assuming the MSS and each packet is 1000B):
> 1. Both sides are transmitting data and a single packet is dropped on =
either side and the next two packets are received properly
> 	Side A - SND.MAX=3D3000, SND.NXT=3D3000, SND.UNA=3D1000, =
RCV.NXT=3D11000, out-of-order block 12000-13000
> 	Side B - SND.MAX =3D13000, SND.NXT =3D13000, SND.UNA=3D11000, =
RCV.NXT=3D1000, out-of-order block 2000-3000
> 2. RTO timer expires on both sides
> 	Side A - SND.MAX=3D3000, SND.NXT=3D1000, SND.UNA=3D1000, =
RCV.NXT=3D11000, out-of-order block 12000-13000
> 	Side B - SND.MAX =3D13000, SND.NXT=3D11000, SND.UNA=3D11000, =
RCV.NXT=3D1000, out-of-order block 2000-3000
> 3. Both sides transmit a single packet to the peer:
> 	A->B - pkt.seq=3D1000, pkt.ack=3D11000, len=3D1000
> 	B->A - pkt.seq=3D11000, pkt.ack=3D1000, len=3D1000
> 3. Both sides receive the packets and update the receive context:
> 	Side A - SND.MAX=3D3000, SND.NXT=3D2000, SND.UNA=3D1000, =
RCV.NXT=3D13000
> 	Side B - SND.MAX=3D13000, SND.NXT=3D12000, SND.UNA=3D11000, =
RCV.NXT=3D3000
> 4. Both sides send another segment:
> 	A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D1000
> 	B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D1000
> 5. Both sides don't accept the packet (and don't update SND.UNA) since =
the sequence on the packet is less than RCV.NXT (sequence number check =
in page 69 of RFC793) and send a pure ACK instead
> 	A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D0 (pure ACK)
> 	B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D0 (pure ACK)
> 6. This will continue forever (until the connection will be terminated =
by RST) since every packet that ends before RCV.NXT (even a retransmit =
from SND.UNA) will be dropped.
>=20
> Did anyone encountered this issue before? Is the anything we missed on =
this sequence?
> If this is indeed a real deadlock, there might be several solutions to =
this which will require a modification in receive processing of RFC793. =
But I would like to know if you think this is a real issue before =
dealing with solutions.
> Thanks,
>=20
> 	Kobby
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Sun Aug  7 07:51:46 2016
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-cubic-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions of the IETF.

        Title           : CUBIC for Fast Long-Distance Networks
        Authors         : Injong Rhee
                          Lisong Xu
                          Sangtae Ha
                          Alexander Zimmermann
                          Lars Eggert
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-cubic-02.txt
	Pages           : 15
	Date            : 2016-08-07

Abstract:
   CUBIC is an extension to the current TCP standards.  The protocol
   differs from the current TCP standards only in the congestion window
   adjustment function in the sender side.  In particular, it uses a
   cubic function instead of a linear window increase of the current TCP
   standards to improve scalability and stability under fast and long
   distance networks.  BIC-TCP, a predecessor of CUBIC, has been a
   default TCP adopted by Linux since year 2005 and has already been
   deployed globally and in use for several years by the Internet
   community at large.  CUBIC is using a similar window growth function
   as BIC-TCP and is designed to be less aggressive and fairer to TCP in
   bandwidth usage than BIC-TCP while maintaining the strengths of BIC-
   TCP such as stability, window scalability and RTT fairness.  Through
   extensive testing in various Internet scenarios, we believe that
   CUBIC is safe for deployment and testing in the global Internet.  The
   intent of this document is to provide the protocol specification of
   CUBIC for a third party implementation and solicit the community
   feedback through experimentation on the performance of CUBIC.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-cubic/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-cubic-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-cubic-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Sun Aug  7 07:58:41 2016
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Cc: draft-ietf-tcpm-cubic@ietf.org, tcpm-chairs@ietf.org
Subject: [tcpm]  A new Internet draft for TCP Cubic
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Hello,

We just submitted a new Internet draft for TCP Cubic.

https://tools.ietf.org/html/draft-ietf-tcpm-cubic-02

Compare to the previous version 01, this version has the following 
changes. Most changes are based on the email discussions with Karen and
Neal on draft-ietf-tcpm-cubic@ietf.org in March 2016. Thanks, Karen and 
Neal!

Section 3.1: to more clearly describe the behavior of CUBIC

* From "The protocol does not make any change to the fast recovery and
retransmit of TCP-NewReno [RFC6582] and TCP-SACK [RFC2018]"
    to "The protocol does not make any change to the fast recovery and
retransmit of TCP, such as TCP-NewReno [RFC6582] and TCP-SACK [RFC2018]"

* From "t is the elapsed time from the last window reduction"
    to "t is the elapsed time from the last window reduction (measured
right after fast recovery)"

* From "K is the time period that the above function takes to increase
the current window size to W_max when there is no further loss event"
    to "K is the time period that the above function takes to increase
the current window size to W_max if there is no further loss event"

Section 3.2: to more clearly explain TCP friendly region

* From " This is done as follows.
     We can analyze the window size of a TCP-friendly AIMD in terms of the
     elapsed time t.  Using a simple analysis in [FHP00], we can analyze
     the average window size of additive increase and multiplicative
     decrease (AIMD) with an additive factor alpha_aimd and a
     multiplicative factor beta_aimd with the following function:"
    to " This is done
     by estimating the average rate of the Standard TCP using a simple
     analysis described in <xref target="FHP00"/>. It considers the Standard
     TCP as a special case of an Additive Increase and Multiplicative
     Decrease algorithm (AIMD), which has an additive factor alpha_aimd
     and a multiplicative factor beta_aimd with the following function:"

New Subsection 4.8 to reflect the recent changes by Jana Iyengar in 2015

* Add "4.8. Behavior for Application-Limited Flows
    CUBIC does not raise its congestion window size if the
    flow is currently limited by the application instead of the
    congestion window.  In cases of idle periods, t in Eq. 1
    should not include the idle time; otherwise, W_cubic(t) might
    be very high after restarting from a long idle time."

Thanks
Lisong



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From: Kobby Carmona <kobby.Carmona@qlogic.com>
To: David Borman <dab@weston.borman.com>
Thread-Topic: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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From nobody Sun Aug  7 14:35:11 2016
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From: David Borman <dab@weston.borman.com>
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To: Kobby Carmona <kobby.Carmona@qlogic.com>
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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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> On Aug 7, 2016, at 3:37 AM, Kobby Carmona <kobby.Carmona@qlogic.com> =
wrote:
>=20
> Hi David,
> The code below is true for fast-retransmit.
> But in case of retransmit timer expiration (TCPT_REXMT) snd_nxt is set =
to snd_una. And in this case CQND will be set to 1MSS so in the example =
below the transmitters can send only a single segment from sequence =
2000/12000.

Your underlying problem is that the ACK-only packets are being sent with =
the wrong sequence number, and that is what is causing them to be =
dropped.  One way to fix that is to put SND.NXT back to its previous =
value after do the the retransmit.  But you are correct, in the BSD code =
it only does that in the fast retransmit code; when the timer based =
retransmit code fires, it just pulls back SND.NXT to SND.UNA.  However, =
in the BSD code that I=E2=80=99m looking at, it keeps track of the =
largest sequence number sent in SND.MAX, and in the tcp_output() path =
there is this bit of code:=20

       /*
        * If we are doing retransmissions, then snd_nxt will
        * not reflect the first unsent octet.  For ACK only
        * packets, we do not want the sequence number of the=20
        * retransmitted packet, we want the sequence number
        * of the next unsent octet.  So, if there is no data
        * (and no SYN or FIN), use snd_max instead of snd_nxt
        * when filling in th_seq.  But if we are in persist
        * state, snd_max might reflect one byte beyond the
        * right edge of the window, so use snd_nxt in that
        * case, since we know we aren't doing a retransmission.
        * (retransmit and persist are mutually exclusive...)
        */
       if (len || (flags & (TH_SYN|TH_FIN)) || =
tp->t_timer[TCPT_PERSIST])
               th->th_seq =3D htonl(tp->snd_nxt);
       else
               th->th_seq =3D htonl(tp->snd_max);

That is what your implementation appears to be missing, and what is =
causing your ACK storm.  So yes, some variant of your problem has been =
seen before, and this is how the BSD code fixed it.

			-David Borman

>=20
> 	Kobby
>=20
> -----Original Message-----
> From: David Borman [mailto:dab@weston.borman.com]=20
> Sent: Thursday, August 04, 2016 10:37 PM
> To: Kobby Carmona <kobby.Carmona@qlogic.com>
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Possible deadlock scenario with retransmission on =
both sides at the same time
>=20
> After you pull back SND.NXT and do the retransmission, you should then =
restore SND.NXT back to where it was, not leave it at the backed off =
value; then the ACKs wouldn=E2=80=99t be dropped, since they wouldn=E2=80=99=
t have old seq values.  For example, in the 4.4BSD fast retransmit code =
it had:
>=20
> 			tcp_seq onxt =3D tp->snd_nxt;
> 			...
> 			tp->snd_nxt =3D th->th_ack;
> 			...
> 			(void) tcp_output(tp);
> 			...
> 			if (SEQ_GT(onxt, tp->snd_nxt)) =20
> 				tp->snd_nxt =3D onxt;
>=20
>=20
> 			-David Borman
>=20
>> On Aug 4, 2016, at 4:08 AM, Kobby Carmona <kobby.Carmona@qlogic.com> =
wrote:
>>=20
>> Hi all,
>> While running a bidirectional scenario with random drops in a network =
simulator of our (QLogic's NIC) TCP stack we found a case where it seems =
there is deadlock in the TCP protocol (the connection will keep sending =
pure acks from both sides until RTO will expire multiple times and a RST =
will sent to close the connection).
>> The scenario is as follows (there is an example with numbers for each =
stage assuming the MSS and each packet is 1000B):
>> 1. Both sides are transmitting data and a single packet is dropped on =
either side and the next two packets are received properly
>> 	Side A - SND.MAX=3D3000, SND.NXT=3D3000, SND.UNA=3D1000, =
RCV.NXT=3D11000, out-of-order block 12000-13000
>> 	Side B - SND.MAX =3D13000, SND.NXT =3D13000, SND.UNA=3D11000, =
RCV.NXT=3D1000,=20
>> out-of-order block 2000-3000 2. RTO timer expires on both sides
>> 	Side A - SND.MAX=3D3000, SND.NXT=3D1000, SND.UNA=3D1000, =
RCV.NXT=3D11000, out-of-order block 12000-13000
>> 	Side B - SND.MAX =3D13000, SND.NXT=3D11000, SND.UNA=3D11000, =
RCV.NXT=3D1000,=20
>> out-of-order block 2000-3000 3. Both sides transmit a single packet =
to the peer:
>> 	A->B - pkt.seq=3D1000, pkt.ack=3D11000, len=3D1000
>> 	B->A - pkt.seq=3D11000, pkt.ack=3D1000, len=3D1000 3. Both sides =
receive=20
>> the packets and update the receive context:
>> 	Side A - SND.MAX=3D3000, SND.NXT=3D2000, SND.UNA=3D1000, =
RCV.NXT=3D13000
>> 	Side B - SND.MAX=3D13000, SND.NXT=3D12000, SND.UNA=3D11000, =
RCV.NXT=3D3000 4.=20
>> Both sides send another segment:
>> 	A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D1000
>> 	B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D1000 5. Both sides =
don't=20
>> accept the packet (and don't update SND.UNA) since the sequence on =
the packet is less than RCV.NXT (sequence number check in page 69 of =
RFC793) and send a pure ACK instead
>> 	A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D0 (pure ACK)
>> 	B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D0 (pure ACK) 6. =
This will=20
>> continue forever (until the connection will be terminated by RST) =
since every packet that ends before RCV.NXT (even a retransmit from =
SND.UNA) will be dropped.
>>=20
>> Did anyone encountered this issue before? Is the anything we missed =
on this sequence?
>> If this is indeed a real deadlock, there might be several solutions =
to this which will require a modification in receive processing of =
RFC793. But I would like to know if you think this is a real issue =
before dealing with solutions.
>> Thanks,
>>=20
>> 	Kobby
>>=20
>>=20
>> _______________________________________________
>> 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


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The following errata report has been submitted for RFC5961,
"Improving TCP's Robustness to Blind In-Window Attacks".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5961&eid=4772

--------------------------------------
Type: Technical
Reported by: Stéphane Bortzmeyer <bortzmeyer+ietf@nic.fr>

Section: 7

Original Text
-------------
[The entire section]

Corrected Text
--------------
No suggested text because it requires a much more serious analysis. 
May be adding that the rate-limit counter SHOULD be per-connection, 
in the spirit of RFC 6528?

Notes
-----
It appears the section does not specify that the counter for ACK throttling SHOULD be per-connection. In Linux, it is apparently global, which allowed its use as a side channel enabling nasty attacks (CVE-2016-5696 and the paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <http://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>)

Instructions:
-------------
This erratum is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC5961 (draft-ietf-tcpm-tcpsecure-13)
--------------------------------------
Title               : Improving TCP's Robustness to Blind In-Window Attacks
Publication Date    : August 2010
Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
Category            : PROPOSED STANDARD
Source              : TCP Maintenance and Minor Extensions
Area                : Transport
Stream              : IETF
Verifying Party     : IESG


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From: Loganaden Velvindron <loganaden@gmail.com>
Date: Thu, 11 Aug 2016 20:27:51 +0400
Message-ID: <CAOp4FwTogyBXLYdjHrrnM3-Uz2wpSX31eZg+GJwUP5LBnqu=sQ@mail.gmail.com>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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I submitted this draft, yesterday.:

https://tools.ietf.org/html/draft-lvelvindron-ack-throttling-02

It's a work-in-progress, but I welcome feedback.


On Wed, Aug 10, 2016 at 10:36 PM, RFC Errata System
<rfc-editor@rfc-editor.org> wrote:
> The following errata report has been submitted for RFC5961,
> "Improving TCP's Robustness to Blind In-Window Attacks".
>
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D5961&eid=3D4772
>
> --------------------------------------
> Type: Technical
> Reported by: St=C3=A9phane Bortzmeyer <bortzmeyer+ietf@nic.fr>
>
> Section: 7
>
> Original Text
> -------------
> [The entire section]
>
> Corrected Text
> --------------
> No suggested text because it requires a much more serious analysis.
> May be adding that the rate-limit counter SHOULD be per-connection,
> in the spirit of RFC 6528?
>
> Notes
> -----
> It appears the section does not specify that the counter for ACK throttli=
ng SHOULD be per-connection. In Linux, it is apparently global, which allow=
ed its use as a side channel enabling nasty attacks (CVE-2016-5696 and the =
paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <http=
://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>)
>
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>
> --------------------------------------
> RFC5961 (draft-ietf-tcpm-tcpsecure-13)
> --------------------------------------
> Title               : Improving TCP's Robustness to Blind In-Window Attac=
ks
> Publication Date    : August 2010
> Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
> Category            : PROPOSED STANDARD
> Source              : TCP Maintenance and Minor Extensions
> Area                : Transport
> Stream              : IETF
> Verifying Party     : IESG
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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From nobody Thu Aug 11 12:37:23 2016
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To: Loganaden Velvindron <loganaden@gmail.com>
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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FWIW, I see nothing in RFC5961 that implies that ACK throttling - or
anything in the doc - should be across connections.

This new doc can claim to clarify Sec 7 of that RFC, but I see no reason
to deprecate it.

Joe


On 8/11/2016 9:27 AM, Loganaden Velvindron wrote:
> I submitted this draft, yesterday.:
>
> https://tools.ietf.org/html/draft-lvelvindron-ack-throttling-02
>
> It's a work-in-progress, but I welcome feedback.
>
>
> On Wed, Aug 10, 2016 at 10:36 PM, RFC Errata System
> <rfc-editor@rfc-editor.org> wrote:
>> The following errata report has been submitted for RFC5961,
>> "Improving TCP's Robustness to Blind In-Window Attacks".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=5961&eid=4772
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Stéphane Bortzmeyer <bortzmeyer+ietf@nic.fr>
>>
>> Section: 7
>>
>> Original Text
>> -------------
>> [The entire section]
>>
>> Corrected Text
>> --------------
>> No suggested text because it requires a much more serious analysis.
>> May be adding that the rate-limit counter SHOULD be per-connection,
>> in the spirit of RFC 6528?
>>
>> Notes
>> -----
>> It appears the section does not specify that the counter for ACK throttling SHOULD be per-connection. In Linux, it is apparently global, which allowed its use as a side channel enabling nasty attacks (CVE-2016-5696 and the paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <http://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>)
>>
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC5961 (draft-ietf-tcpm-tcpsecure-13)
>> --------------------------------------
>> Title               : Improving TCP's Robustness to Blind In-Window Attacks
>> Publication Date    : August 2010
>> Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
>> Category            : PROPOSED STANDARD
>> Source              : TCP Maintenance and Minor Extensions
>> Area                : Transport
>> Stream              : IETF
>> Verifying Party     : IESG
>>
>>
>> _______________________________________________
>> 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


From nobody Thu Aug 11 12:41:18 2016
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To: Loganaden Velvindron <loganaden@gmail.com>
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From: Joe Touch <touch@isi.edu>
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FWIW, I don't really see the need for this new doc at all.

Everything in 5961 is within the context of a single connection. The
misreading of this doc doesn't warrant updating, clarifying, etc.

The behavior of Linux should be treated simply as the bug that it is and
corrected, but there's no reason to pollute the RFC stream or increase
the complexity of TCP requirements on the basis of a bug.

Joe


On 8/11/2016 12:36 PM, Joe Touch wrote:
> FWIW, I see nothing in RFC5961 that implies that ACK throttling - or
> anything in the doc - should be across connections.
>
> This new doc can claim to clarify Sec 7 of that RFC, but I see no reason
> to deprecate it.
>
> Joe
>
>
> On 8/11/2016 9:27 AM, Loganaden Velvindron wrote:
>> I submitted this draft, yesterday.:
>>
>> https://tools.ietf.org/html/draft-lvelvindron-ack-throttling-02
>>
>> It's a work-in-progress, but I welcome feedback.
>>
>>
>> On Wed, Aug 10, 2016 at 10:36 PM, RFC Errata System
>> <rfc-editor@rfc-editor.org> wrote:
>>> The following errata report has been submitted for RFC5961,
>>> "Improving TCP's Robustness to Blind In-Window Attacks".
>>>
>>> --------------------------------------
>>> You may review the report below and at:
>>> http://www.rfc-editor.org/errata_search.php?rfc=5961&eid=4772
>>>
>>> --------------------------------------
>>> Type: Technical
>>> Reported by: Stéphane Bortzmeyer <bortzmeyer+ietf@nic.fr>
>>>
>>> Section: 7
>>>
>>> Original Text
>>> -------------
>>> [The entire section]
>>>
>>> Corrected Text
>>> --------------
>>> No suggested text because it requires a much more serious analysis.
>>> May be adding that the rate-limit counter SHOULD be per-connection,
>>> in the spirit of RFC 6528?
>>>
>>> Notes
>>> -----
>>> It appears the section does not specify that the counter for ACK throttling SHOULD be per-connection. In Linux, it is apparently global, which allowed its use as a side channel enabling nasty attacks (CVE-2016-5696 and the paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <http://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>)
>>>
>>> Instructions:
>>> -------------
>>> This erratum is currently posted as "Reported". If necessary, please
>>> use "Reply All" to discuss whether it should be verified or
>>> rejected. When a decision is reached, the verifying party (IESG)
>>> can log in to change the status and edit the report, if necessary.
>>>
>>> --------------------------------------
>>> RFC5961 (draft-ietf-tcpm-tcpsecure-13)
>>> --------------------------------------
>>> Title               : Improving TCP's Robustness to Blind In-Window Attacks
>>> Publication Date    : August 2010
>>> Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
>>> Category            : PROPOSED STANDARD
>>> Source              : TCP Maintenance and Minor Extensions
>>> Area                : Transport
>>> Stream              : IETF
>>> Verifying Party     : IESG
>>>
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
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Date: Fri, 12 Aug 2016 13:30:38 +0200
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Hi Marcello,

I very much appreciate your draft.

Even more so as it addresses one of the things I find to be a bit
"disturbing"  in the l4s drafts - Namely the claim that the finer saw
tooth functionality of l4s/DCTCP alone solves the latency problems (while
DCTCP indeed also is tuning the RTO timeout aggressively).
E.g., the formulation in  section 1.1 of
https://datatracker.ietf.org/doc/draft-briscoe-tsvwg-ecn-l4s-id/?include_t
ext=1

It turns out that a TCP algorithm like DCTCP that solves TCP's
   scalability problem also solves the latency problem, because the
   finer sawteeth cause very little queuing delay.  A supporting paper
   [DCttH15] gives the full explanation of why the design solves both
   the latency and the scaling problems, both in plain English and in
   more precise mathematical form.  The explanation is summarised
   without the maths in [I-D.briscoe-aqm-dualq-coupled].

could benefit from a reference to your draft or the issue that you
addresses here (at least for comprehensiveness).

An additional comment is  that new approaches to retransmissions - like
TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
the moment) might fundamentally alter the picture. I.e., if
retransmissions are sent pro-actively in tail loss situations then more
conservative RTOs may be kept for situations where it is prudent to wait
longer. Don't know if TCP TLP is so widely deployed that it is something
that you should relate to even if it may be superseded by RACK. Just a
thought.

BR, Karen

> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo bagnulo
> braun
> Sent: 8. juli 2016 23:19
> To: tcpm IETF list <tcpm@ietf.org>
> Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>
> Hi,
>
> We just submitted this draft for consideration of the WG. Comments are
> appreciated.
>
> Regards, marcelo
>
>
>
>
> -------- Mensaje reenviado --------
> Asunto: 	I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> Fecha: 	Fri, 08 Jul 2016 14:16:35 -0700
> De: 	internet-drafts@ietf.org
> Responder a: 	internet-drafts@ietf.org
> Para: 	i-d-announce@ietf.org
>
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
directories.
>
>
>          Title           : Recommendations for increasing TCP
performance in low RTT
> networks.
>          Authors         : Marcelo Bagnulo
>                            Koen De Schepper
>                            Glenn Judd
> 	Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> 	Pages           : 7
> 	Date            : 2016-07-08
>
> Abstract:
>     This documents compiles a set of issues that negatively affect TCP
>     performance in low RTT networks as well as the recommendations to
>     overcome them.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
>
>
> Please note that it may take a couple of minutes from the time of
submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> _______________________________________________
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> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: Loganaden Velvindron <loganaden@gmail.com>
Date: Thu, 11 Aug 2016 20:24:36 +0400
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To: RFC Errata System <rfc-editor@rfc-editor.org>
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Cc: tcpm@ietf.org, bortzmeyer+ietf@nic.fr, ananth@cisco.com, ietf@kuehlewind.net, randall@lakerest.net, mdalal@cisco.com
Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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Hi folks,

I submitted this, independently:

https://tools.ietf.org/html/draft-lvelvindron-ack-throttling-02

On Wed, Aug 10, 2016 at 10:36 PM, RFC Errata System
<rfc-editor@rfc-editor.org> wrote:
> The following errata report has been submitted for RFC5961,
> "Improving TCP's Robustness to Blind In-Window Attacks".
>
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D5961&eid=3D4772
>
> --------------------------------------
> Type: Technical
> Reported by: St=C3=A9phane Bortzmeyer <bortzmeyer+ietf@nic.fr>
>
> Section: 7
>
> Original Text
> -------------
> [The entire section]
>
> Corrected Text
> --------------
> No suggested text because it requires a much more serious analysis.
> May be adding that the rate-limit counter SHOULD be per-connection,
> in the spirit of RFC 6528?
>
> Notes
> -----
> It appears the section does not specify that the counter for ACK throttli=
ng SHOULD be per-connection. In Linux, it is apparently global, which allow=
ed its use as a side channel enabling nasty attacks (CVE-2016-5696 and the =
paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <http=
://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>)
>
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>
> --------------------------------------
> RFC5961 (draft-ietf-tcpm-tcpsecure-13)
> --------------------------------------
> Title               : Improving TCP's Robustness to Blind In-Window Attac=
ks
> Publication Date    : August 2010
> Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
> Category            : PROPOSED STANDARD
> Source              : TCP Maintenance and Minor Extensions
> Area                : Transport
> Stream              : IETF
> Verifying Party     : IESG
>
>
> _______________________________________________
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>


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Thread-Topic: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 12 Aug 2016 09:54:07 -0700
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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On Thu, Aug 11, 2016 at 12:40 PM, Joe Touch <touch@isi.edu> wrote:
> FWIW, I don't really see the need for this new doc at all.
+1
>
> Everything in 5961 is within the context of a single connection. The
> misreading of this doc doesn't warrant updating, clarifying, etc.
I think an errata on recommending per-connection limit and cite the
research finding is worthy.

>
> The behavior of Linux should be treated simply as the bug that it is and
> corrected, but there's no reason to pollute the RFC stream or increase
> the complexity of TCP requirements on the basis of a bug.
>
> Joe
>
>
> On 8/11/2016 12:36 PM, Joe Touch wrote:
>> FWIW, I see nothing in RFC5961 that implies that ACK throttling - or
>> anything in the doc - should be across connections.
>>
>> This new doc can claim to clarify Sec 7 of that RFC, but I see no reason
>> to deprecate it.
>>
>> Joe
>>
>>
>> On 8/11/2016 9:27 AM, Loganaden Velvindron wrote:
>>> I submitted this draft, yesterday.:
>>>
>>> https://tools.ietf.org/html/draft-lvelvindron-ack-throttling-02
>>>
>>> It's a work-in-progress, but I welcome feedback.
>>>
>>>
>>> On Wed, Aug 10, 2016 at 10:36 PM, RFC Errata System
>>> <rfc-editor@rfc-editor.org> wrote:
>>>> The following errata report has been submitted for RFC5961,
>>>> "Improving TCP's Robustness to Blind In-Window Attacks".
>>>>
>>>> --------------------------------------
>>>> You may review the report below and at:
>>>> http://www.rfc-editor.org/errata_search.php?rfc=3D5961&eid=3D4772
>>>>
>>>> --------------------------------------
>>>> Type: Technical
>>>> Reported by: St=C3=A9phane Bortzmeyer <bortzmeyer+ietf@nic.fr>
>>>>
>>>> Section: 7
>>>>
>>>> Original Text
>>>> -------------
>>>> [The entire section]
>>>>
>>>> Corrected Text
>>>> --------------
>>>> No suggested text because it requires a much more serious analysis.
>>>> May be adding that the rate-limit counter SHOULD be per-connection,
>>>> in the spirit of RFC 6528?
>>>>
>>>> Notes
>>>> -----
>>>> It appears the section does not specify that the counter for ACK throt=
tling SHOULD be per-connection. In Linux, it is apparently global, which al=
lowed its use as a side channel enabling nasty attacks (CVE-2016-5696 and t=
he paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <h=
ttp://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>)
>>>>
>>>> Instructions:
>>>> -------------
>>>> This erratum is currently posted as "Reported". If necessary, please
>>>> use "Reply All" to discuss whether it should be verified or
>>>> rejected. When a decision is reached, the verifying party (IESG)
>>>> can log in to change the status and edit the report, if necessary.
>>>>
>>>> --------------------------------------
>>>> RFC5961 (draft-ietf-tcpm-tcpsecure-13)
>>>> --------------------------------------
>>>> Title               : Improving TCP's Robustness to Blind In-Window At=
tacks
>>>> Publication Date    : August 2010
>>>> Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
>>>> Category            : PROPOSED STANDARD
>>>> Source              : TCP Maintenance and Minor Extensions
>>>> Area                : Transport
>>>> Stream              : IETF
>>>> Verifying Party     : IESG
>>>>
>>>>
>>>> _______________________________________________
>>>> tcpm mailing list
>>>> tcpm@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>
> _______________________________________________
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> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 12 Aug 2016 12:16:50 -0700
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To: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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On Fri, Aug 12, 2016 at 4:30 AM, Karen Elisabeth Egede Nielsen
<karen.nielsen@tieto.com> wrote:
>
> Hi Marcello,
>
> I very much appreciate your draft.
>
> Even more so as it addresses one of the things I find to be a bit
> "disturbing"  in the l4s drafts - Namely the claim that the finer saw
> tooth functionality of l4s/DCTCP alone solves the latency problems (while
> DCTCP indeed also is tuning the RTO timeout aggressively).
> E.g., the formulation in  section 1.1 of
> https://datatracker.ietf.org/doc/draft-briscoe-tsvwg-ecn-l4s-id/?include_t
> ext=1
>
> It turns out that a TCP algorithm like DCTCP that solves TCP's
>    scalability problem also solves the latency problem, because the
>    finer sawteeth cause very little queuing delay.  A supporting paper
>    [DCttH15] gives the full explanation of why the design solves both
>    the latency and the scaling problems, both in plain English and in
>    more precise mathematical form.  The explanation is summarised
>    without the maths in [I-D.briscoe-aqm-dualq-coupled].
>
> could benefit from a reference to your draft or the issue that you
> addresses here (at least for comprehensiveness).
>
> An additional comment is  that new approaches to retransmissions - like
> TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
> the moment) might fundamentally alter the picture. I.e., if
> retransmissions are sent pro-actively in tail loss situations then more
> conservative RTOs may be kept for situations where it is prudent to wait
> longer. Don't know if TCP TLP is so widely deployed that it is something
> that you should relate to even if it may be superseded by RACK. Just a
> thought.
TLP and RACK help reduce timeout cases in DC environment for sure. But
still it can not completely avoid timeouts. So I agree we should keep
timeouts conservative, but the current RFCs are way too conservative:
min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.

The issue is, the "right" min-RTO depends on the actual DC and stacks:
they all have different RTTs and timer granularity. The draft cites
Glenn's paper, but Morgan Stanley's DC could be different than others.


Other comments on the draft:

1. min ssthresh or cwnd: with very large incast ((tens of) thousands
of senders into one receiver), cwnd of 0.1 pkt can still be too big. the
only way is to pace the packets over N*RTT intervals.


2. delayed acks:
as mentioned the actual time depends a lot on the DC implementation,
which is really "vender/owner"-specific. instead of perhaps an option
during setup to inform the sender about the max delay in the ack works
better.



>
> BR, Karen
>
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo bagnulo
> > braun
> > Sent: 8. juli 2016 23:19
> > To: tcpm IETF list <tcpm@ietf.org>
> > Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> >
> > Hi,
> >
> > We just submitted this draft for consideration of the WG. Comments are
> > appreciated.
> >
> > Regards, marcelo
> >
> >
> >
> >
> > -------- Mensaje reenviado --------
> > Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
> > De:   internet-drafts@ietf.org
> > Responder a:  internet-drafts@ietf.org
> > Para:         i-d-announce@ietf.org
> >
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> >
> >          Title           : Recommendations for increasing TCP
> performance in low RTT
> > networks.
> >          Authors         : Marcelo Bagnulo
> >                            Koen De Schepper
> >                            Glenn Judd
> >       Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> >       Pages           : 7
> >       Date            : 2016-07-08
> >
> > Abstract:
> >     This documents compiles a set of issues that negatively affect TCP
> >     performance in low RTT networks as well as the recommendations to
> >     overcome them.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
> >
> > There's also a htmlized version available at:
> > https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
> >
> >
> > Please note that it may take a couple of minutes from the time of
> submission
> > until the htmlized version and diff are available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Aug 12 12:17:56 2016
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From: Neal Cardwell <ncardwell@google.com>
Date: Fri, 12 Aug 2016 12:17:09 -0700
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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On Fri, Aug 12, 2016 at 9:54 AM, Yuchung Cheng <ycheng@google.com> wrote:
> I think an errata on recommending per-connection limit and cite the
> research finding is worthy.

I agree that it's worthwhile to have an errata to clarify that a
per-connection limit is best. At the very least there seems
some risky ambiguity here.

In fact, AFAICT RFC 5961, Section 7, page 13 --

  https://tools.ietf.org/html/rfc5961#section-7

-- has several passages that seem to implicitly suggest that
the rate-limiting mechanism it specifies as a SHOULD is not
per-connection but rather global.

First:

   ... an
   administrator who is more interested in faster cleanup of stale
   connections (i.e., concerned about excess TCP state) may decide to
   set a higher value thus allowing more RST's to be processed in any
   given time period.

If this is a single-connection rate limit, then what practical reason
would an administrator have to allow "more RST's to be processed in
any given time period"? I can't think of a case where a legit live
connection would send more than one RST. If this is about a single
connection, then allowing one mid-window RST (and responding to that
one with a single challenge ACK) should be enough, in most cases, to
send a challenge ACK that elicits a proper RST from the remote
endpoint. It's not clear to me what the motivation would be for
intentionally sending a challenge ACK in response to more than one
mid-window RST in a single connection.

Thus AFAICT "allowing more RST's to be processed in any given time
period" suggests that the rate limit is across connections.

Second:

   The time limit SHOULD be tunable to help timeout brute force attacks
   faster than a potential legitimate flood of RSTs.

I may be missing something, but if this rate-limiting mechanism is
only about a single connection, then what is a "legitimate flood of
RSTs"? Why would it be important for a single connection to allow a
"legitimate flood of RSTs" destinted for itself? A single legitimate
live connection should not send out a flood of RSTs; it should send
out one. Once there is one legitimate RST that arrives and is
processed, the receiver deletes the connection, and any other arriving
RSTs don't match any existing connection, so are ignored and
dropped, in which case is no challenge ACK to rate-limit.

By contrast, if this rate limit applies across connections then "a
potential legitimate flood of RSTs" makes sense, since a group of
legitimate live connections that all close in a burst may, in certain
cases, generate a legitimate flood of RSTs. So tuning the limit
upward would make sense.

Sorry if I'm missing something obvious. But it seems like there are at
least real ambiguities there, on a point significant enough to be worth
clearing up.

neal


From nobody Fri Aug 12 12:40:47 2016
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To: Yuchung Cheng <ycheng@google.com>, Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
References: <20160708211635.32095.77047.idtracker@ietfa.amsl.com> <2f56d1b8-7450-42de-4a24-53df9cf4c045@it.uc3m.es> <a1f456db6e656785da8e356b9f530717@mail.gmail.com> <CAK6E8=etaxFSgBvQt27s=rB03rRQxQ_u43qWRExTVXys9A82wg@mail.gmail.com>
From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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Karen, Yuchung,
Thanks for the comments.
Replies below.


El 12/08/16 a las 21:16, Yuchung Cheng escribió:
> On Fri, Aug 12, 2016 at 4:30 AM, Karen Elisabeth Egede Nielsen
> <karen.nielsen@tieto.com> wrote:
>> Hi Marcello,
>>
>> I very much appreciate your draft.
>>
>> Even more so as it addresses one of the things I find to be a bit
>> "disturbing"  in the l4s drafts - Namely the claim that the finer saw
>> tooth functionality of l4s/DCTCP alone solves the latency problems (while
>> DCTCP indeed also is tuning the RTO timeout aggressively).
>> E.g., the formulation in  section 1.1 of
>> https://datatracker.ietf.org/doc/draft-briscoe-tsvwg-ecn-l4s-id/?include_t
>> ext=1
>>
>> It turns out that a TCP algorithm like DCTCP that solves TCP's
>>     scalability problem also solves the latency problem, because the
>>     finer sawteeth cause very little queuing delay.  A supporting paper
>>     [DCttH15] gives the full explanation of why the design solves both
>>     the latency and the scaling problems, both in plain English and in
>>     more precise mathematical form.  The explanation is summarised
>>     without the maths in [I-D.briscoe-aqm-dualq-coupled].
>>
>> could benefit from a reference to your draft or the issue that you
>> addresses here (at least for comprehensiveness).
>>
>> An additional comment is  that new approaches to retransmissions - like
>> TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
>> the moment) might fundamentally alter the picture. I.e., if
>> retransmissions are sent pro-actively in tail loss situations then more
>> conservative RTOs may be kept for situations where it is prudent to wait
>> longer. Don't know if TCP TLP is so widely deployed that it is something
>> that you should relate to even if it may be superseded by RACK. Just a
>> thought.
> TLP and RACK help reduce timeout cases in DC environment for sure. But
> still it can not completely avoid timeouts. So I agree we should keep
> timeouts conservative, but the current RFCs are way too conservative:
> min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.
>
> The issue is, the "right" min-RTO depends on the actual DC and stacks:
> they all have different RTTs and timer granularity. The draft cites
> Glenn's paper, but Morgan Stanley's DC could be different than others.
>

Right.

I guess we should could say that the min RTO should be in the order of 
the RTT of the considered network when there is zero queueng delay.
But if the draft is aiming to be a reccomendation for imeplenters of DC 
network TCP stacks, then i guess it should provide a guidance that is 
independent of the DC network itself, since the implementation should 
work for a variety of DC networks, no?


I guess the more general question is whether we need this document i.e. 
a document describing reccoemndation about how to set TCP for low RTT 
networks. If people think it is a good idea to have such a document, we 
then figure how if this is

> Other comments on the draft:
>
> 1. min ssthresh or cwnd: with very large incast ((tens of) thousands
> of senders into one receiver), cwnd of 0.1 pkt can still be too big. the
> only way is to pace the packets over N*RTT intervals.
>
Right, If i understood correctly Bob's paper (cited in the draft) this 
is exactly what they are proposing.
I dont think this document is the right place to define the behaviour of 
CWND less than 1 MSS. I think this should be done in a separated 
document, which is then referred by the low-rtt draft.

Are you aware of other documents (draft or papers others than the one 
from Bob) describing mechanisms to achieve this?

> 2. delayed acks:
> as mentioned the actual time depends a lot on the DC implementation,
> which is really "vender/owner"-specific. instead of perhaps an option
> during setup to inform the sender about the max delay in the ack works
> better.
>
I am not sure if a follow.
What is the benefit of allowing dynamic negotiation of the delay ack time?
Thanks, marcelo



>
>> BR, Karen
>>
>>> -----Original Message-----
>>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo bagnulo
>>> braun
>>> Sent: 8. juli 2016 23:19
>>> To: tcpm IETF list <tcpm@ietf.org>
>>> Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>>>
>>> Hi,
>>>
>>> We just submitted this draft for consideration of the WG. Comments are
>>> appreciated.
>>>
>>> Regards, marcelo
>>>
>>>
>>>
>>>
>>> -------- Mensaje reenviado --------
>>> Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>>> Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
>>> De:   internet-drafts@ietf.org
>>> Responder a:  internet-drafts@ietf.org
>>> Para:         i-d-announce@ietf.org
>>>
>>>
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>>
>>>           Title           : Recommendations for increasing TCP
>> performance in low RTT
>>> networks.
>>>           Authors         : Marcelo Bagnulo
>>>                             Koen De Schepper
>>>                             Glenn Judd
>>>        Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>>>        Pages           : 7
>>>        Date            : 2016-07-08
>>>
>>> Abstract:
>>>      This documents compiles a set of issues that negatively affect TCP
>>>      performance in low RTT networks as well as the recommendations to
>>>      overcome them.
>>>
>>>
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
>>>
>>> There's also a htmlized version available at:
>>> https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
>>>
>>>
>>> Please note that it may take a couple of minutes from the time of
>> submission
>>> until the htmlized version and diff are available at tools.ietf.org.
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> _______________________________________________
>>> I-D-Announce mailing list
>>> I-D-Announce@ietf.org
>>> https://www.ietf.org/mailman/listinfo/i-d-announce
>>> Internet-Draft directories: http://www.ietf.org/shadow.html or
>>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 12 Aug 2016 12:50:59 -0700
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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On Fri, Aug 12, 2016 at 12:40 PM, marcelo bagnulo braun
<marcelo@it.uc3m.es> wrote:
> Karen, Yuchung,
> Thanks for the comments.
> Replies below.
>
>
> El 12/08/16 a las 21:16, Yuchung Cheng escribi=C3=B3:
>
>> On Fri, Aug 12, 2016 at 4:30 AM, Karen Elisabeth Egede Nielsen
>> <karen.nielsen@tieto.com> wrote:
>>>
>>> Hi Marcello,
>>>
>>> I very much appreciate your draft.
>>>
>>> Even more so as it addresses one of the things I find to be a bit
>>> "disturbing"  in the l4s drafts - Namely the claim that the finer saw
>>> tooth functionality of l4s/DCTCP alone solves the latency problems (whi=
le
>>> DCTCP indeed also is tuning the RTO timeout aggressively).
>>> E.g., the formulation in  section 1.1 of
>>>
>>> https://datatracker.ietf.org/doc/draft-briscoe-tsvwg-ecn-l4s-id/?includ=
e_t
>>> ext=3D1
>>>
>>> It turns out that a TCP algorithm like DCTCP that solves TCP's
>>>     scalability problem also solves the latency problem, because the
>>>     finer sawteeth cause very little queuing delay.  A supporting paper
>>>     [DCttH15] gives the full explanation of why the design solves both
>>>     the latency and the scaling problems, both in plain English and in
>>>     more precise mathematical form.  The explanation is summarised
>>>     without the maths in [I-D.briscoe-aqm-dualq-coupled].
>>>
>>> could benefit from a reference to your draft or the issue that you
>>> addresses here (at least for comprehensiveness).
>>>
>>> An additional comment is  that new approaches to retransmissions - like
>>> TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
>>> the moment) might fundamentally alter the picture. I.e., if
>>> retransmissions are sent pro-actively in tail loss situations then more
>>> conservative RTOs may be kept for situations where it is prudent to wai=
t
>>> longer. Don't know if TCP TLP is so widely deployed that it is somethin=
g
>>> that you should relate to even if it may be superseded by RACK. Just a
>>> thought.
>>
>> TLP and RACK help reduce timeout cases in DC environment for sure. But
>> still it can not completely avoid timeouts. So I agree we should keep
>> timeouts conservative, but the current RFCs are way too conservative:
>> min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.
>>
>> The issue is, the "right" min-RTO depends on the actual DC and stacks:
>> they all have different RTTs and timer granularity. The draft cites
>> Glenn's paper, but Morgan Stanley's DC could be different than others.
>>
>
> Right.
>
> I guess we should could say that the min RTO should be in the order of th=
e
> RTT of the considered network when there is zero queueng delay.
> But if the draft is aiming to be a reccomendation for imeplenters of DC
> network TCP stacks, then i guess it should provide a guidance that is
> independent of the DC network itself, since the implementation should wor=
k
> for a variety of DC networks, no?
>
>
> I guess the more general question is whether we need this document i.e. a
> document describing reccoemndation about how to set TCP for low RTT
> networks. If people think it is a good idea to have such a document, we t=
hen
> figure how if this is
my conundrum is that DC tcp stuff is usually not an inter-op issue for
the Internet so this draft is more of a "how to tune your DC tcp
stack". but I can be convinced either ways.

>
>> Other comments on the draft:
>>
>> 1. min ssthresh or cwnd: with very large incast ((tens of) thousands
>> of senders into one receiver), cwnd of 0.1 pkt can still be too big. the
>> only way is to pace the packets over N*RTT intervals.
>>
> Right, If i understood correctly Bob's paper (cited in the draft) this is
> exactly what they are proposing.
> I dont think this document is the right place to define the behaviour of
> CWND less than 1 MSS. I think this should be done in a separated document=
,
> which is then referred by the low-rtt draft.
>
> Are you aware of other documents (draft or papers others than the one fro=
m
> Bob) describing mechanisms to achieve this?
>
>> 2. delayed acks:
>> as mentioned the actual time depends a lot on the DC implementation,
>> which is really "vender/owner"-specific. instead of perhaps an option
>> during setup to inform the sender about the max delay in the ack works
>> better.
>>
> I am not sure if a follow.
> What is the benefit of allowing dynamic negotiation of the delay ack time=
?
> Thanks, marcelo
the receiver may decide to delay ack up to certain time. it could be
based on DC implementation or host limit. It'd be good for sender to
learn about that to adapt his RTO.

For example, say it's a cloud VM. VM of OS1 may use a max delay for
d-ack say 2ms. but VM of OS2 may use 40ms. The lack of such
information makes the sender to these VMs use something  conservative
of known max of all, otherwise he risks spurious RTOs.

>
>
>
>
>>
>>> BR, Karen
>>>
>>>> -----Original Message-----
>>>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo bagnulo
>>>> braun
>>>> Sent: 8. juli 2016 23:19
>>>> To: tcpm IETF list <tcpm@ietf.org>
>>>> Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>>>>
>>>> Hi,
>>>>
>>>> We just submitted this draft for consideration of the WG. Comments are
>>>> appreciated.
>>>>
>>>> Regards, marcelo
>>>>
>>>>
>>>>
>>>>
>>>> -------- Mensaje reenviado --------
>>>> Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>>>> Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
>>>> De:   internet-drafts@ietf.org
>>>> Responder a:  internet-drafts@ietf.org
>>>> Para:         i-d-announce@ietf.org
>>>>
>>>>
>>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>
>>> directories.
>>>>
>>>>
>>>>           Title           : Recommendations for increasing TCP
>>>
>>> performance in low RTT
>>>>
>>>> networks.
>>>>           Authors         : Marcelo Bagnulo
>>>>                             Koen De Schepper
>>>>                             Glenn Judd
>>>>        Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>>>>        Pages           : 7
>>>>        Date            : 2016-07-08
>>>>
>>>> Abstract:
>>>>      This documents compiles a set of issues that negatively affect TC=
P
>>>>      performance in low RTT networks as well as the recommendations to
>>>>      overcome them.
>>>>
>>>>
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
>>>>
>>>> There's also a htmlized version available at:
>>>> https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
>>>>
>>>>
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>>>>
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To: Neal Cardwell <ncardwell@google.com>, Yuchung Cheng <ycheng@google.com>
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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On 8/12/2016 12:17 PM, Neal Cardwell wrote:
> On Fri, Aug 12, 2016 at 9:54 AM, Yuchung Cheng <ycheng@google.com> wrote:
>> I think an errata on recommending per-connection limit and cite the
>> research finding is worthy.
> I agree that it's worthwhile to have an errata to clarify that a
> per-connection limit is best. At the very least there seems
> some risky ambiguity here.

I disagree, as per below...
>
> In fact, AFAICT RFC 5961, Section 7, page 13 --
>
>   https://tools.ietf.org/html/rfc5961#section-7
>
> -- has several passages that seem to implicitly suggest that
> the rate-limiting mechanism it specifies as a SHOULD is not
> per-connection but rather global.

There's plenty of benefit to setting a single configuration for an
entire system vs. setting per-connection limits. That's completely
different from whether the ACKs are tallied globally or per connection.

> First:
>
>    ... an
>    administrator who is more interested in faster cleanup of stale
>    connections (i.e., concerned about excess TCP state) may decide to
>    set a higher value thus allowing more RST's to be processed in any
>    given time period.
>
> If this is a single-connection rate limit, then what practical reason
> would an administrator have to allow "more RST's to be processed in
> any given time period"? I can't think of a case where a legit live
> connection would send more than one RST. 
A single connection might process many RSTs because some can be out of
window (they'd be dropped) or not at the end of the window (dropped
under some configurations). I.e., receiving and evaluating RSTs doesn't
always result in resetting the connection.

> If this is about a single
> connection, then allowing one mid-window RST (and responding to that
> one with a single challenge ACK) should be enough, 
It would be IF the other end issued a legitimate RST. If this is a
spoofing attack, the data can still keep coming which means that the
window might still be moving forward by the time the next attack RST
arrives. As a result, this sort of 'catch up' can go on until the
connection ends. The point of the text here is to avoid wasting
resources in such cases.
> in most cases, to
> send a challenge ACK that elicits a proper RST from the remote
> endpoint. It's not clear to me what the motivation would be for
> intentionally sending a challenge ACK in response to more than one
> mid-window RST in a single connection.
Protecting against that case is the motivation *of the entire document*.

> Thus AFAICT "allowing more RST's to be processed in any given time
> period" suggests that the rate limit is across connections.

There's no benefit to doing this across connections at all, unless you
think that there's some reason that legitimate connections shouldn't be
reset by valid RSTs (there's no benefit to that).

> Second:
>
>    The time limit SHOULD be tunable to help timeout brute force attacks
>    faster than a potential legitimate flood of RSTs.
>
> I may be missing something, but if this rate-limiting mechanism is
> only about a single connection, then what is a "legitimate flood of
> RSTs"?
A sequence of RSTs that pick different values, as described in RFC 4953.

Or a sequence of RSTs aimed at responses to the ACK-after-RST that is
chasing legitimate data, as described above.

>  Why would it be important for a single connection to allow a
> "legitimate flood of RSTs" destinted for itself? 
It wouldn't - that's the point of this text. You should treat a flood of
RSTs for a single connection as an attack and throttle them accordingly.

> A single legitimate
> live connection should not send out a flood of RSTs; it should send
> out one. Once there is one legitimate RST that arrives and is
> processed, the receiver deletes the connection, and any other arriving
> RSTs don't match any existing connection, so are ignored and
> dropped, in which case is no challenge ACK to rate-limit.
Again, the point of this document are attacks, not legitimate connections.
> By contrast, if this rate limit applies across connections then "a
> potential legitimate flood of RSTs" makes sense, since a group of
> legitimate live connections that all close in a burst may, in certain
> cases, generate a legitimate flood of RSTs. So tuning the limit
> upward would make sense.

There's no case I can imagine where a set of attacks on one connection
should be cause for concern on another necessarily.

Yes, there's no real need to set per-connection limits (i.e., a global
per-connection limit seems fine), but counting them as a group serves no
purpose against any threat, unless you think that "if I'm attacked on
one connection, I will be attacked on another" - but that's an incorrect
inference IMO.

>
> Sorry if I'm missing something obvious. But it seems like there are at
> least real ambiguities there, on a point significant enough to be worth
> clearing up.

I think you're missing the point of the document, but that's just my
interpretation of the exchange.

Joe


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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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>>2. delayed acks:
as mentioned the actual time depends a lot on the DC implementation, which =
is really "vender/owner"-specific. instead of perhaps an option during setu=
p to inform the sender about the max delay in the ack works better.

Yeah this is a real problem in IaaS environment today since TCP sender pick=
s minrto and receiver picks delayed ACK timeout independently and can cause=
 spurious RTOs if the minrto is too aggressive. Windows server tries to use=
 the connection handshake RTT (< 10 msec) to auto pick the DC values but th=
is is a heuristic that can fail and doesn't really work in the Linux intero=
p cases.=20

-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Yuchung Cheng
Sent: Friday, August 12, 2016 12:17 PM
To: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
Cc: tcpm IETF list <tcpm@ietf.org>; TCP Prague List <tcpPrague@ietf.org>
Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt

On Fri, Aug 12, 2016 at 4:30 AM, Karen Elisabeth Egede Nielsen <karen.niels=
en@tieto.com> wrote:
>
> Hi Marcello,
>
> I very much appreciate your draft.
>
> Even more so as it addresses one of the things I find to be a bit=20
> "disturbing"  in the l4s drafts - Namely the claim that the finer saw=20
> tooth functionality of l4s/DCTCP alone solves the latency problems=20
> (while DCTCP indeed also is tuning the RTO timeout aggressively).
> E.g., the formulation in  section 1.1 of=20
> https://datatracker.ietf.org/doc/draft-briscoe-tsvwg-ecn-l4s-id/?inclu
> de_t
> ext=3D1
>
> It turns out that a TCP algorithm like DCTCP that solves TCP's
>    scalability problem also solves the latency problem, because the
>    finer sawteeth cause very little queuing delay.  A supporting paper
>    [DCttH15] gives the full explanation of why the design solves both
>    the latency and the scaling problems, both in plain English and in
>    more precise mathematical form.  The explanation is summarised
>    without the maths in [I-D.briscoe-aqm-dualq-coupled].
>
> could benefit from a reference to your draft or the issue that you=20
> addresses here (at least for comprehensiveness).
>
> An additional comment is  that new approaches to retransmissions -=20
> like TCP TLP and TCP RACK (also SCTP TLR which however is not=20
> progressing at the moment) might fundamentally alter the picture.=20
> I.e., if retransmissions are sent pro-actively in tail loss situations=20
> then more conservative RTOs may be kept for situations where it is=20
> prudent to wait longer. Don't know if TCP TLP is so widely deployed=20
> that it is something that you should relate to even if it may be=20
> superseded by RACK. Just a thought.
TLP and RACK help reduce timeout cases in DC environment for sure. But stil=
l it can not completely avoid timeouts. So I agree we should keep timeouts =
conservative, but the current RFCs are way too conservative:
min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.

The issue is, the "right" min-RTO depends on the actual DC and stacks:
they all have different RTTs and timer granularity. The draft cites Glenn's=
 paper, but Morgan Stanley's DC could be different than others.


Other comments on the draft:

1. min ssthresh or cwnd: with very large incast ((tens of) thousands of sen=
ders into one receiver), cwnd of 0.1 pkt can still be too big. the only way=
 is to pace the packets over N*RTT intervals.


2. delayed acks:
as mentioned the actual time depends a lot on the DC implementation, which =
is really "vender/owner"-specific. instead of perhaps an option during setu=
p to inform the sender about the max delay in the ack works better.



>
> BR, Karen
>
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo=20
> > bagnulo braun
> > Sent: 8. juli 2016 23:19
> > To: tcpm IETF list <tcpm@ietf.org>
> > Subject: [tcpm] Fwd: I-D Action:=20
> > draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> >
> > Hi,
> >
> > We just submitted this draft for consideration of the WG. Comments=20
> > are appreciated.
> >
> > Regards, marcelo
> >
> >
> >
> >
> > -------- Mensaje reenviado --------
> > Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
> > De:   internet-drafts@ietf.org
> > Responder a:  internet-drafts@ietf.org
> > Para:         i-d-announce@ietf.org
> >
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> >
> >          Title           : Recommendations for increasing TCP
> performance in low RTT
> > networks.
> >          Authors         : Marcelo Bagnulo
> >                            Koen De Schepper
> >                            Glenn Judd
> >       Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> >       Pages           : 7
> >       Date            : 2016-07-08
> >
> > Abstract:
> >     This documents compiles a set of issues that negatively affect TCP
> >     performance in low RTT networks as well as the recommendations to
> >     overcome them.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
> >
> > There's also a htmlized version available at:
> > https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
> >
> >
> > Please note that it may take a couple of minutes from the time of
> submission
> > until the htmlized version and diff are available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or=20
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm

_______________________________________________
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From nobody Sat Aug 13 11:47:47 2016
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To: Yoav Nir <ynir.ietf@gmail.com>
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Subject: Re: [tcpm] [saag] Linux flaw from RFC5961 implementation
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Hi, all,

Moving this thread back to TCPM where it is ongong...

I'll address the note below.

Joe


On 8/13/2016 10:45 AM, Yoav Nir wrote:
>> On 13 Aug 2016, at 6:35 PM, Joe Touch <touch@isi.edu> wrote:
>>
>> Hi, all,
>>
>> This issue is already been discussed in TCPM. Please review the existing
>> thread and continue discussion there.
>>
>> A quick summary from my viewpoint:
>>
>> a) there already is a draft
>>
>> b) IMO, the Linux code is just (and yet another) bug based on an
>> incorrect read of RFC 5961, not worthy of even an errata
>>
>> All of this is in the thread on TCPM. Please discuss it there.
> Hi, Joe
>
> I’ve read the thread, but I disagree with your conclusion there ([1]) for several reasons.
>
> First, RFCs should be clear enough that reasonably competent implementers will get it right. I think it’s safe to say that Linux kernel developers are reasonably competent as a whole, so if they got it wrong the RFC is not clear enough and requires a clarification in one form or another. 
That is demonstrably false on many levels. My students just found a two
significant bugs in the Linux TCP code just this summer.

Then again, RFC5961 exists because Linux was all to happy to take
over-eager "security" fixes that could impact normal TCP operation.

> The fact that Microsoft got it right is not proof that any competent developer will get it right.
And the fact that Linux got it wrong is not proof that others will as well.

> Second, section 7 talks about rate-limiting to prevent DoS. It talks about protecting bandwidth and CPU utilization. It states that "An implementation may take an excessive number of invocations of the throttling mechanism as an indication that network conditions are unusual or hostile.” All of this is global to the system, not specific to one connection or another. As far as bandwidth is concerned, there is no difference between sending 5 Challenge Acks on each of 20 connections, or 100 Challenge Acks on a single connection. 
The entire doc is written in the context of a single connection except
where it indicates otherwise (per-machine values for per-connection
limits). Anyone who thinks that an attack on one TCP connection implies
an attack on another is already incorrect. The document is already
written to take that into account.

> Third, if a machine has enough CPU and bandwidth to allow X challenge acks per second, then dividing X by the number of concurrent connections sets a per-connection limit is too low for the mechanism to function well. 
If the CPU and bandwidth are that limited, then TCP won't function well
either.

> Of course, now that we know about the attack it seems obvious that the limit should be per-connection and that we shouldn’t let one connection starve other connections for challenge-acks. But it’s not so obvious that the RFC did not need to mention it.

The RFC doesn't mention not to put your credit card information in the
header too - hopefully we don't need to explain everything you shouldn't do.

Joe


From nobody Sun Aug 14 07:54:35 2016
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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: Joe Touch <touch@isi.edu>, Yoav Nir <ynir.ietf@gmail.com>
Thread-Topic: [tcpm] [saag] Linux flaw from RFC5961 implementation
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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
References: <20160708211635.32095.77047.idtracker@ietfa.amsl.com> <2f56d1b8-7450-42de-4a24-53df9cf4c045@it.uc3m.es> <a1f456db6e656785da8e356b9f530717@mail.gmail.com> <CAK6E8=etaxFSgBvQt27s=rB03rRQxQ_u43qWRExTVXys9A82wg@mail.gmail.com>
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Date: Mon, 15 Aug 2016 11:00:24 +0200
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To: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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Hi Yuchung,

> >
> > An additional comment is  that new approaches to retransmissions - like
> > TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
> > the moment) might fundamentally alter the picture. I.e., if
> > retransmissions are sent pro-actively in tail loss situations then more
> > conservative RTOs may be kept for situations where it is prudent to wait
> > longer. Don't know if TCP TLP is so widely deployed that it is something
> > that you should relate to even if it may be superseded by RACK. Just a
> > thought.
> TLP and RACK help reduce timeout cases in DC environment for sure. But
> still it can not completely avoid timeouts.

[Karen Elisabeth Egede Nielsen] Agreed.

For SCTP TLR we still see RTO-timeouts when also probes/retransmissions are
lost.
I assume that it would be something like this also for Rack/TLP -  though
potentially depending on how many consecutive probes that are allowed.
The role of RTO-timeouts are different when RACK/TLP is enabled and it might
be counterproductive to have RTO-timeouts be too aggressive as the function
indeed with TLP/RACK becomes something much closer to the original intend
(read: how I understand the original intend) - namely to introduce a pause
for the network to recover when things are really (consistently) bad.

If the RTO is lowered down to be comparable in order of magnitude with the
RTT (with some delay_ack considerations) we are narrowing the situation
where TLP/RACK probing is able to kick in before RTO-retransmissions starts
anyway.

I understand that RTOs should be adjusted to the network dynamics, but I do
think that it is important to understand - for the DC environment - if the
need for the low RTOs in DC's is
motivated mainly by having RTO-retransmissions fix the tail loss recovery
deficiencies of TCP, which RACK/TLP addresses, or more generally for a need
for more timely probe and reactivation for/after network recovery.

Or perhaps you see the TLP probe timeout and the RTO timeout eventually
being driven by the same timer, the reactions (e.g. CC reactions)  then
possibly depending on state (probe already sent or not).

BR, Karen

 So I agree we should keep
> timeouts conservative, but the current RFCs are way too conservative:
> min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.
>
> The issue is, the "right" min-RTO depends on the actual DC and stacks:
> they all have different RTTs and timer granularity. The draft cites
> Glenn's paper, but Morgan Stanley's DC could be different than others.
>
>
> Other comments on the draft:
>
> 1. min ssthresh or cwnd: with very large incast ((tens of) thousands
> of senders into one receiver), cwnd of 0.1 pkt can still be too big. the
> only way is to pace the packets over N*RTT intervals.
>
>
> 2. delayed acks:
> as mentioned the actual time depends a lot on the DC implementation,
> which is really "vender/owner"-specific. instead of perhaps an option
> during setup to inform the sender about the max delay in the ack works
> better.
>
>
>
> >
> > BR, Karen
> >
> > > -----Original Message-----
> > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo
> bagnulo
> > > braun
> > > Sent: 8. juli 2016 23:19
> > > To: tcpm IETF list <tcpm@ietf.org>
> > > Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > >
> > > Hi,
> > >
> > > We just submitted this draft for consideration of the WG. Comments are
> > > appreciated.
> > >
> > > Regards, marcelo
> > >
> > >
> > >
> > >
> > > -------- Mensaje reenviado --------
> > > Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > > Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
> > > De:   internet-drafts@ietf.org
> > > Responder a:  internet-drafts@ietf.org
> > > Para:         i-d-announce@ietf.org
> > >
> > >
> > >
> > > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> > >
> > >
> > >          Title           : Recommendations for increasing TCP
> > performance in low RTT
> > > networks.
> > >          Authors         : Marcelo Bagnulo
> > >                            Koen De Schepper
> > >                            Glenn Judd
> > >       Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > >       Pages           : 7
> > >       Date            : 2016-07-08
> > >
> > > Abstract:
> > >     This documents compiles a set of issues that negatively affect TCP
> > >     performance in low RTT networks as well as the recommendations to
> > >     overcome them.
> > >
> > >
> > > The IETF datatracker status page for this draft is:
> > > https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
> > >
> > > There's also a htmlized version available at:
> > > https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
> > >
> > >
> > > Please note that it may take a couple of minutes from the time of
> > submission
> > > until the htmlized version and diff are available at tools.ietf.org.
> > >
> > > Internet-Drafts are also available by anonymous FTP at:
> > > ftp://ftp.ietf.org/internet-drafts/
> > >
> > > _______________________________________________
> > > I-D-Announce mailing list
> > > I-D-Announce@ietf.org
> > > https://www.ietf.org/mailman/listinfo/i-d-announce
> > > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > >
> > >
> > > _______________________________________________
> > > tcpm mailing list
> > > tcpm@ietf.org
> > > https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon Aug 15 02:03:06 2016
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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
References: <20160708211635.32095.77047.idtracker@ietfa.amsl.com> <2f56d1b8-7450-42de-4a24-53df9cf4c045@it.uc3m.es> <a1f456db6e656785da8e356b9f530717@mail.gmail.com> <CAK6E8=etaxFSgBvQt27s=rB03rRQxQ_u43qWRExTVXys9A82wg@mail.gmail.com>
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Date: Mon, 15 Aug 2016 11:00:24 +0200
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To: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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Hi Yuchung,

> >
> > An additional comment is  that new approaches to retransmissions - like
> > TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
> > the moment) might fundamentally alter the picture. I.e., if
> > retransmissions are sent pro-actively in tail loss situations then more
> > conservative RTOs may be kept for situations where it is prudent to wait
> > longer. Don't know if TCP TLP is so widely deployed that it is something
> > that you should relate to even if it may be superseded by RACK. Just a
> > thought.
> TLP and RACK help reduce timeout cases in DC environment for sure. But
> still it can not completely avoid timeouts.

[Karen Elisabeth Egede Nielsen] Agreed.

For SCTP TLR we still see RTO-timeouts when also probes/retransmissions are
lost.
I assume that it would be something like this also for Rack/TLP -  though
potentially depending on how many consecutive probes that are allowed.
The role of RTO-timeouts are different when RACK/TLP is enabled and it might
be counterproductive to have RTO-timeouts be too aggressive as the function
indeed with TLP/RACK becomes something much closer to the original intend
(read: how I understand the original intend) - namely to introduce a pause
for the network to recover when things are really (consistently) bad.

If the RTO is lowered down to be comparable in order of magnitude with the
RTT (with some delay_ack considerations) we are narrowing the situation
where TLP/RACK probing is able to kick in before RTO-retransmissions starts
anyway.

I understand that RTOs should be adjusted to the network dynamics, but I do
think that it is important to understand - for the DC environment - if the
need for the low RTOs in DC's is
motivated mainly by having RTO-retransmissions fix the tail loss recovery
deficiencies of TCP, which RACK/TLP addresses, or more generally for a need
for more timely probe and reactivation for/after network recovery.

Or perhaps you see the TLP probe timeout and the RTO timeout eventually
being driven by the same timer, the reactions (e.g. CC reactions)  then
possibly depending on state (probe already sent or not).

BR, Karen

 So I agree we should keep
> timeouts conservative, but the current RFCs are way too conservative:
> min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.
>
> The issue is, the "right" min-RTO depends on the actual DC and stacks:
> they all have different RTTs and timer granularity. The draft cites
> Glenn's paper, but Morgan Stanley's DC could be different than others.
>
>
> Other comments on the draft:
>
> 1. min ssthresh or cwnd: with very large incast ((tens of) thousands
> of senders into one receiver), cwnd of 0.1 pkt can still be too big. the
> only way is to pace the packets over N*RTT intervals.
>
>
> 2. delayed acks:
> as mentioned the actual time depends a lot on the DC implementation,
> which is really "vender/owner"-specific. instead of perhaps an option
> during setup to inform the sender about the max delay in the ack works
> better.
>
>
>
> >
> > BR, Karen
> >
> > > -----Original Message-----
> > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo
> bagnulo
> > > braun
> > > Sent: 8. juli 2016 23:19
> > > To: tcpm IETF list <tcpm@ietf.org>
> > > Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > >
> > > Hi,
> > >
> > > We just submitted this draft for consideration of the WG. Comments are
> > > appreciated.
> > >
> > > Regards, marcelo
> > >
> > >
> > >
> > >
> > > -------- Mensaje reenviado --------
> > > Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > > Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
> > > De:   internet-drafts@ietf.org
> > > Responder a:  internet-drafts@ietf.org
> > > Para:         i-d-announce@ietf.org
> > >
> > >
> > >
> > > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> > >
> > >
> > >          Title           : Recommendations for increasing TCP
> > performance in low RTT
> > > networks.
> > >          Authors         : Marcelo Bagnulo
> > >                            Koen De Schepper
> > >                            Glenn Judd
> > >       Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
> > >       Pages           : 7
> > >       Date            : 2016-07-08
> > >
> > > Abstract:
> > >     This documents compiles a set of issues that negatively affect TCP
> > >     performance in low RTT networks as well as the recommendations to
> > >     overcome them.
> > >
> > >
> > > The IETF datatracker status page for this draft is:
> > > https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
> > >
> > > There's also a htmlized version available at:
> > > https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
> > >
> > >
> > > Please note that it may take a couple of minutes from the time of
> > submission
> > > until the htmlized version and diff are available at tools.ietf.org.
> > >
> > > Internet-Drafts are also available by anonymous FTP at:
> > > ftp://ftp.ietf.org/internet-drafts/
> > >
> > > _______________________________________________
> > > I-D-Announce mailing list
> > > I-D-Announce@ietf.org
> > > https://www.ietf.org/mailman/listinfo/i-d-announce
> > > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > >
> > >
> > > _______________________________________________
> > > tcpm mailing list
> > > tcpm@ietf.org
> > > https://www.ietf.org/mailman/listinfo/tcpm


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To: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>, Yoav Nir <ynir.ietf@gmail.com>
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] [saag] Linux flaw from RFC5961 implementation
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Hi, Michael,

The key question is whether there are actually *errors* in the RFC that
cause this problem.

AFAICT, this is just an incorrect misreading, not based on an error in
the text. I'm not sure why we want to address every one-off incorrect
misreading this way, but an errata seems like a better way to handle
this than a new RFC.

FWIW, the errata should be clear that this isn't based on an *error* in
the text.

Joe


On 8/14/2016 7:54 AM, Scharf, Michael (Nokia - DE) wrote:
> According to https://www.ietf.org/iesg/statement/errata-processing.html, the condition for "verified" errata is:
>
>   "Only errors that could cause implementation or deployment problems or significant confusion should be Verified."
>
> As an individual contributor to TCPM, in this case I believe there is empirical evidence that the wording of RFC 5961 can cause "significant confusion". To me, the conditions for a "verified" errata can thus be fulfilled in this case. BTW, the other two potential states for erratas are "hold for document update" or "rejected".
>
> Also, I don't believe that an errata with some additional explanation or implementation guidance would cause harm to the RFC series.
>
> Michael
>
>
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Joe Touch
>> Sent: Saturday, August 13, 2016 8:47 PM
>> To: Yoav Nir
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] [saag] Linux flaw from RFC5961 implementation
>>
>> Hi, all,
>>
>> Moving this thread back to TCPM where it is ongong...
>>
>> I'll address the note below.
>>
>> Joe
>>
>>
>> On 8/13/2016 10:45 AM, Yoav Nir wrote:
>>>> On 13 Aug 2016, at 6:35 PM, Joe Touch <touch@isi.edu> wrote:
>>>>
>>>> Hi, all,
>>>>
>>>> This issue is already been discussed in TCPM. Please review the
>> existing
>>>> thread and continue discussion there.
>>>>
>>>> A quick summary from my viewpoint:
>>>>
>>>> a) there already is a draft
>>>>
>>>> b) IMO, the Linux code is just (and yet another) bug based on an
>>>> incorrect read of RFC 5961, not worthy of even an errata
>>>>
>>>> All of this is in the thread on TCPM. Please discuss it there.
>>> Hi, Joe
>>>
>>> I’ve read the thread, but I disagree with your conclusion there ([1])
>> for several reasons.
>>> First, RFCs should be clear enough that reasonably competent
>> implementers will get it right. I think it’s safe to say that Linux
>> kernel developers are reasonably competent as a whole, so if they got
>> it wrong the RFC is not clear enough and requires a clarification in
>> one form or another.
>> That is demonstrably false on many levels. My students just found a two
>> significant bugs in the Linux TCP code just this summer.
>>
>> Then again, RFC5961 exists because Linux was all to happy to take
>> over-eager "security" fixes that could impact normal TCP operation.
>>
>>> The fact that Microsoft got it right is not proof that any competent
>> developer will get it right.
>> And the fact that Linux got it wrong is not proof that others will as
>> well.
>>
>>> Second, section 7 talks about rate-limiting to prevent DoS. It talks
>> about protecting bandwidth and CPU utilization. It states that "An
>> implementation may take an excessive number of invocations of the
>> throttling mechanism as an indication that network conditions are
>> unusual or hostile.” All of this is global to the system, not specific
>> to one connection or another. As far as bandwidth is concerned, there
>> is no difference between sending 5 Challenge Acks on each of 20
>> connections, or 100 Challenge Acks on a single connection.
>> The entire doc is written in the context of a single connection except
>> where it indicates otherwise (per-machine values for per-connection
>> limits). Anyone who thinks that an attack on one TCP connection implies
>> an attack on another is already incorrect. The document is already
>> written to take that into account.
>>
>>> Third, if a machine has enough CPU and bandwidth to allow X challenge
>> acks per second, then dividing X by the number of concurrent
>> connections sets a per-connection limit is too low for the mechanism to
>> function well.
>> If the CPU and bandwidth are that limited, then TCP won't function well
>> either.
>>
>>> Of course, now that we know about the attack it seems obvious that
>> the limit should be per-connection and that we shouldn’t let one
>> connection starve other connections for challenge-acks. But it’s not so
>> obvious that the RFC did not need to mention it.
>>
>> The RFC doesn't mention not to put your credit card information in the
>> header too - hopefully we don't need to explain everything you
>> shouldn't do.
>>
>> Joe
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 15 Aug 2016 09:49:12 -0700
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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On Mon, Aug 15, 2016 at 2:00 AM, Karen Elisabeth Egede Nielsen
<karen.nielsen@tieto.com> wrote:
> Hi Yuchung,
>
>> >
>> > An additional comment is  that new approaches to retransmissions - like
>> > TCP TLP and TCP RACK (also SCTP TLR which however is not progressing at
>> > the moment) might fundamentally alter the picture. I.e., if
>> > retransmissions are sent pro-actively in tail loss situations then more
>> > conservative RTOs may be kept for situations where it is prudent to wait
>> > longer. Don't know if TCP TLP is so widely deployed that it is something
>> > that you should relate to even if it may be superseded by RACK. Just a
>> > thought.
>> TLP and RACK help reduce timeout cases in DC environment for sure. But
>> still it can not completely avoid timeouts.
>
> [Karen Elisabeth Egede Nielsen] Agreed.
>
> For SCTP TLR we still see RTO-timeouts when also probes/retransmissions are
> lost.
> I assume that it would be something like this also for Rack/TLP -  though
> potentially depending on how many consecutive probes that are allowed.
> The role of RTO-timeouts are different when RACK/TLP is enabled and it might
> be counterproductive to have RTO-timeouts be too aggressive as the function
> indeed with TLP/RACK becomes something much closer to the original intend
> (read: how I understand the original intend) - namely to introduce a pause
> for the network to recover when things are really (consistently) bad.
>
> If the RTO is lowered down to be comparable in order of magnitude with the
> RTT (with some delay_ack considerations) we are narrowing the situation
> where TLP/RACK probing is able to kick in before RTO-retransmissions starts
> anyway.
>
> I understand that RTOs should be adjusted to the network dynamics, but I do
> think that it is important to understand - for the DC environment - if the
> need for the low RTOs in DC's is
> motivated mainly by having RTO-retransmissions fix the tail loss recovery
> deficiencies of TCP, which RACK/TLP addresses, or more generally for a need
> for more timely probe and reactivation for/after network recovery.
>
> Or perhaps you see the TLP probe timeout and the RTO timeout eventually
> being driven by the same timer, the reactions (e.g. CC reactions)  then
> possibly depending on state (probe already sent or not).
I agree with your thought here.

More generally, I'd like to evolve the state of art recovery with
these principles:
1. React with ack events as much as possible

2. Send tiny probes to trigger (1) after 2-3 RTTs

3. Have conservative  RTOs that (exp-backoff) expire at an order of
RTT to repair the head for reliability.


On thing about RTO: RTO current reset cwnd to 1 upon firing no matter
what. We should ONLY reset cwnd to 1 if no news (acks/data) for a long
duration. This is quite important for performance: today cwnd is reset
to 1 even if every packet but the first one has been recently
delivered. This unnecessarily conservative because the ack clock isn't
lost yet. But the collateral damage to performance is huge on large
BDP networks. This results work-arounds to shorten and avoid RTOs.


I'd like to address these points in the upcoming RACK/TLP draft.



>
> BR, Karen
>
>  So I agree we should keep
>> timeouts conservative, but the current RFCs are way too conservative:
>> min RTO of 1 second is 4-5 orders of magnitude of DC RTTs.
>>
>> The issue is, the "right" min-RTO depends on the actual DC and stacks:
>> they all have different RTTs and timer granularity. The draft cites
>> Glenn's paper, but Morgan Stanley's DC could be different than others.
>>
>>
>> Other comments on the draft:
>>
>> 1. min ssthresh or cwnd: with very large incast ((tens of) thousands
>> of senders into one receiver), cwnd of 0.1 pkt can still be too big. the
>> only way is to pace the packets over N*RTT intervals.
>>
>>
>> 2. delayed acks:
>> as mentioned the actual time depends a lot on the DC implementation,
>> which is really "vender/owner"-specific. instead of perhaps an option
>> during setup to inform the sender about the max delay in the ack works
>> better.
>>
>>
>>
>> >
>> > BR, Karen
>> >
>> > > -----Original Message-----
>> > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of marcelo
>> bagnulo
>> > > braun
>> > > Sent: 8. juli 2016 23:19
>> > > To: tcpm IETF list <tcpm@ietf.org>
>> > > Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>> > >
>> > > Hi,
>> > >
>> > > We just submitted this draft for consideration of the WG. Comments are
>> > > appreciated.
>> > >
>> > > Regards, marcelo
>> > >
>> > >
>> > >
>> > >
>> > > -------- Mensaje reenviado --------
>> > > Asunto:       I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>> > > Fecha:        Fri, 08 Jul 2016 14:16:35 -0700
>> > > De:   internet-drafts@ietf.org
>> > > Responder a:  internet-drafts@ietf.org
>> > > Para:         i-d-announce@ietf.org
>> > >
>> > >
>> > >
>> > > A New Internet-Draft is available from the on-line Internet-Drafts
>> > directories.
>> > >
>> > >
>> > >          Title           : Recommendations for increasing TCP
>> > performance in low RTT
>> > > networks.
>> > >          Authors         : Marcelo Bagnulo
>> > >                            Koen De Schepper
>> > >                            Glenn Judd
>> > >       Filename        : draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>> > >       Pages           : 7
>> > >       Date            : 2016-07-08
>> > >
>> > > Abstract:
>> > >     This documents compiles a set of issues that negatively affect TCP
>> > >     performance in low RTT networks as well as the recommendations to
>> > >     overcome them.
>> > >
>> > >
>> > > The IETF datatracker status page for this draft is:
>> > > https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-tcp-low-rtt/
>> > >
>> > > There's also a htmlized version available at:
>> > > https://tools.ietf.org/html/draft-bagnulo-tcpm-tcp-low-rtt-00
>> > >
>> > >
>> > > Please note that it may take a couple of minutes from the time of
>> > submission
>> > > until the htmlized version and diff are available at tools.ietf.org.
>> > >
>> > > Internet-Drafts are also available by anonymous FTP at:
>> > > ftp://ftp.ietf.org/internet-drafts/
>> > >
>> > > _______________________________________________
>> > > I-D-Announce mailing list
>> > > I-D-Announce@ietf.org
>> > > https://www.ietf.org/mailman/listinfo/i-d-announce
>> > > Internet-Draft directories: http://www.ietf.org/shadow.html or
>> > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> > >
>> > >
>> > > _______________________________________________
>> > > tcpm mailing list
>> > > tcpm@ietf.org
>> > > https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: Joe Touch <touch@isi.edu>, Yoav Nir <ynir.ietf@gmail.com>
Thread-Topic: [tcpm] [saag] Linux flaw from RFC5961 implementation
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Date: Mon, 15 Aug 2016 17:02:56 +0000
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Subject: Re: [tcpm] [saag] Linux flaw from RFC5961 implementation
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Date: Mon, 15 Aug 2016 15:39:15 -0700
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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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Hello,
I personally think this is an interesting corner case for discussion.
It looks a minor one, but I'm not not very sure if we can leave it for each
implementation.
I also guess a question would be if the BSD's fix is the best way for the
issue.

Thanks,
--
Yoshi

On Thu, Aug 11, 2016 at 1:46 PM, Kobby Carmona <kobby.Carmona@qlogic.com>
wrote:

> Hi David,
> This makes a lot of sense. We will fix our code.
> Thanks for your help on this,
>
> BTW,
> Is this issue of mentioned in any RFC? If not do you see a point in addin=
g
> explicit note on the SEQ of pure ACK in case of retransmission?
>
>         Kobby
>
> -----Original Message-----
> From: David Borman [mailto:dab@weston.borman.com]
> Sent: Monday, August 08, 2016 12:35 AM
> To: Kobby Carmona <kobby.Carmona@qlogic.com>
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Possible deadlock scenario with retransmission on bot=
h
> sides at the same time
>
> > On Aug 7, 2016, at 3:37 AM, Kobby Carmona <kobby.Carmona@qlogic.com>
> wrote:
> >
> > Hi David,
> > The code below is true for fast-retransmit.
> > But in case of retransmit timer expiration (TCPT_REXMT) snd_nxt is set
> to snd_una. And in this case CQND will be set to 1MSS so in the example
> below the transmitters can send only a single segment from sequence
> 2000/12000.
>
> Your underlying problem is that the ACK-only packets are being sent with
> the wrong sequence number, and that is what is causing them to be dropped=
.
> One way to fix that is to put SND.NXT back to its previous value after do
> the the retransmit.  But you are correct, in the BSD code it only does th=
at
> in the fast retransmit code; when the timer based retransmit code fires, =
it
> just pulls back SND.NXT to SND.UNA.  However, in the BSD code that I=E2=
=80=99m
> looking at, it keeps track of the largest sequence number sent in SND.MAX=
,
> and in the tcp_output() path there is this bit of code:
>
>        /*
>         * If we are doing retransmissions, then snd_nxt will
>         * not reflect the first unsent octet.  For ACK only
>         * packets, we do not want the sequence number of the
>         * retransmitted packet, we want the sequence number
>         * of the next unsent octet.  So, if there is no data
>         * (and no SYN or FIN), use snd_max instead of snd_nxt
>         * when filling in th_seq.  But if we are in persist
>         * state, snd_max might reflect one byte beyond the
>         * right edge of the window, so use snd_nxt in that
>         * case, since we know we aren't doing a retransmission.
>         * (retransmit and persist are mutually exclusive...)
>         */
>        if (len || (flags & (TH_SYN|TH_FIN)) || tp->t_timer[TCPT_PERSIST])
>                th->th_seq =3D htonl(tp->snd_nxt);
>        else
>                th->th_seq =3D htonl(tp->snd_max);
>
> That is what your implementation appears to be missing, and what is
> causing your ACK storm.  So yes, some variant of your problem has been se=
en
> before, and this is how the BSD code fixed it.
>
>                         -David Borman
>
> >
> >       Kobby
> >
> > -----Original Message-----
> > From: David Borman [mailto:dab@weston.borman.com]
> > Sent: Thursday, August 04, 2016 10:37 PM
> > To: Kobby Carmona <kobby.Carmona@qlogic.com>
> > Cc: tcpm@ietf.org
> > Subject: Re: [tcpm] Possible deadlock scenario with retransmission on
> > both sides at the same time
> >
> > After you pull back SND.NXT and do the retransmission, you should then
> restore SND.NXT back to where it was, not leave it at the backed off valu=
e;
> then the ACKs wouldn=E2=80=99t be dropped, since they wouldn=E2=80=99t ha=
ve old seq
> values.  For example, in the 4.4BSD fast retransmit code it had:
> >
> >                       tcp_seq onxt =3D tp->snd_nxt;
> >                       ...
> >                       tp->snd_nxt =3D th->th_ack;
> >                       ...
> >                       (void) tcp_output(tp);
> >                       ...
> >                       if (SEQ_GT(onxt, tp->snd_nxt))
> >                               tp->snd_nxt =3D onxt;
> >
> >
> >                       -David Borman
> >
> >> On Aug 4, 2016, at 4:08 AM, Kobby Carmona <kobby.Carmona@qlogic.com>
> wrote:
> >>
> >> Hi all,
> >> While running a bidirectional scenario with random drops in a network
> simulator of our (QLogic's NIC) TCP stack we found a case where it seems
> there is deadlock in the TCP protocol (the connection will keep sending
> pure acks from both sides until RTO will expire multiple times and a RST
> will sent to close the connection).
> >> The scenario is as follows (there is an example with numbers for each
> stage assuming the MSS and each packet is 1000B):
> >> 1. Both sides are transmitting data and a single packet is dropped on
> either side and the next two packets are received properly
> >>      Side A - SND.MAX=3D3000, SND.NXT=3D3000, SND.UNA=3D1000, RCV.NXT=
=3D11000,
> out-of-order block 12000-13000
> >>      Side B - SND.MAX =3D13000, SND.NXT =3D13000, SND.UNA=3D11000,
> >> RCV.NXT=3D1000, out-of-order block 2000-3000 2. RTO timer expires on b=
oth
> sides
> >>      Side A - SND.MAX=3D3000, SND.NXT=3D1000, SND.UNA=3D1000, RCV.NXT=
=3D11000,
> out-of-order block 12000-13000
> >>      Side B - SND.MAX =3D13000, SND.NXT=3D11000, SND.UNA=3D11000,
> RCV.NXT=3D1000,
> >> out-of-order block 2000-3000 3. Both sides transmit a single packet to
> the peer:
> >>      A->B - pkt.seq=3D1000, pkt.ack=3D11000, len=3D1000
> >>      B->A - pkt.seq=3D11000, pkt.ack=3D1000, len=3D1000 3. Both sides =
receive
> >> the packets and update the receive context:
> >>      Side A - SND.MAX=3D3000, SND.NXT=3D2000, SND.UNA=3D1000, RCV.NXT=
=3D13000
> >>      Side B - SND.MAX=3D13000, SND.NXT=3D12000, SND.UNA=3D11000, RCV.N=
XT=3D3000
> 4.
> >> Both sides send another segment:
> >>      A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D1000
> >>      B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D1000 5. Both sides =
don't
> >> accept the packet (and don't update SND.UNA) since the sequence on the
> packet is less than RCV.NXT (sequence number check in page 69 of RFC793)
> and send a pure ACK instead
> >>      A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D0 (pure ACK)
> >>      B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D0 (pure ACK) 6. Thi=
s will
> >> continue forever (until the connection will be terminated by RST) sinc=
e
> every packet that ends before RCV.NXT (even a retransmit from SND.UNA) wi=
ll
> be dropped.
> >>
> >> Did anyone encountered this issue before? Is the anything we missed on
> this sequence?
> >> If this is indeed a real deadlock, there might be several solutions to
> this which will require a modification in receive processing of RFC793. B=
ut
> I would like to know if you think this is a real issue before dealing wit=
h
> solutions.
> >> Thanks,
> >>
> >>      Kobby
> >>
> >>
> >> _______________________________________________
> >> 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
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr"><div>Hello,</div>I personally think this is an interesting=
 corner case for discussion.=C2=A0<div>It looks a minor one, but I&#39;m no=
t not very sure if we can leave it for each implementation.=C2=A0<div><div>=
<div><div>I also guess a question would be if the BSD&#39;s fix is the best=
 way for the issue.<div><br><div>Thanks,<br><div>--</div><div>Yoshi<br><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Aug 11, 2016 =
at 1:46 PM, Kobby Carmona <span dir=3D"ltr">&lt;<a href=3D"mailto:kobby.Car=
mona@qlogic.com" target=3D"_blank">kobby.Carmona@qlogic.com</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">Hi David,<br>
This makes a lot of sense. We will fix our code.<br>
Thanks for your help on this,<br>
<br>
BTW,<br>
Is this issue of mentioned in any RFC? If not do you see a point in adding =
explicit note on the SEQ of pure ACK in case of retransmission?<br>
<span><br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Kobby<br>
<br>
-----Original Message-----<br>
From: David Borman [mailto:<a href=3D"mailto:dab@weston.borman.com" target=
=3D"_blank">dab@weston.borman.com</a>]<br>
</span><div><div>Sent: Monday, August 08, 2016 12:35 AM<br>
To: Kobby Carmona &lt;<a href=3D"mailto:kobby.Carmona@qlogic.com" target=3D=
"_blank">kobby.Carmona@qlogic.com</a>&gt;<br>
Cc: <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br=
>
Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both =
sides at the same time<br>
<br>
&gt; On Aug 7, 2016, at 3:37 AM, Kobby Carmona &lt;<a href=3D"mailto:kobby.=
Carmona@qlogic.com" target=3D"_blank">kobby.Carmona@qlogic.com</a>&gt; wrot=
e:<br>
&gt;<br>
&gt; Hi David,<br>
&gt; The code below is true for fast-retransmit.<br>
&gt; But in case of retransmit timer expiration (TCPT_REXMT) snd_nxt is set=
 to snd_una. And in this case CQND will be set to 1MSS so in the example be=
low the transmitters can send only a single segment from sequence 2000/1200=
0.<br>
<br>
Your underlying problem is that the ACK-only packets are being sent with th=
e wrong sequence number, and that is what is causing them to be dropped.=C2=
=A0 One way to fix that is to put SND.NXT back to its previous value after =
do the the retransmit.=C2=A0 But you are correct, in the BSD code it only d=
oes that in the fast retransmit code; when the timer based retransmit code =
fires, it just pulls back SND.NXT to SND.UNA.=C2=A0 However, in the BSD cod=
e that I=E2=80=99m looking at, it keeps track of the largest sequence numbe=
r sent in SND.MAX, and in the tcp_output() path there is this bit of code:<=
br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0/*<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * If we are doing retransmissions, then snd_nxt=
 will<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * not reflect the first unsent octet.=C2=A0 For=
 ACK only<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * packets, we do not want the sequence number o=
f the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * retransmitted packet, we want the sequence nu=
mber<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * of the next unsent octet.=C2=A0 So, if there =
is no data<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * (and no SYN or FIN), use snd_max instead of s=
nd_nxt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * when filling in th_seq.=C2=A0 But if we are i=
n persist<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * state, snd_max might reflect one byte beyond =
the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * right edge of the window, so use snd_nxt in t=
hat<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * case, since we know we aren&#39;t doing a ret=
ransmission.<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 * (retransmit and persist are mutually exclusiv=
e...)<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 */<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0if (len || (flags &amp; (TH_SYN|TH_FIN)) || tp-&=
gt;t_timer[TCPT_PERSIST])<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0th-&gt;th_seq =3D ht=
onl(tp-&gt;snd_nxt);<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0else<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0th-&gt;th_seq =3D ht=
onl(tp-&gt;snd_max);<br>
<br>
That is what your implementation appears to be missing, and what is causing=
 your ACK storm.=C2=A0 So yes, some variant of your problem has been seen b=
efore, and this is how the BSD code fixed it.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 -David Borman<br>
<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Kobby<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: David Borman [mailto:<a href=3D"mailto:dab@weston.borman.com" ta=
rget=3D"_blank">dab@weston.borman.com</a>]<br>
&gt; Sent: Thursday, August 04, 2016 10:37 PM<br>
&gt; To: Kobby Carmona &lt;<a href=3D"mailto:kobby.Carmona@qlogic.com" targ=
et=3D"_blank">kobby.Carmona@qlogic.com</a>&gt;<br>
&gt; Cc: <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</=
a><br>
&gt; Subject: Re: [tcpm] Possible deadlock scenario with retransmission on<=
br>
&gt; both sides at the same time<br>
&gt;<br>
&gt; After you pull back SND.NXT and do the retransmission, you should then=
 restore SND.NXT back to where it was, not leave it at the backed off value=
; then the ACKs wouldn=E2=80=99t be dropped, since they wouldn=E2=80=99t ha=
ve old seq values.=C2=A0 For example, in the 4.4BSD fast retransmit code it=
 had:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0tcp_seq onxt =3D tp-&gt;snd_nxt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0...<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0tp-&gt;snd_nxt =3D th-&gt;th_ack;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0...<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0(void) tcp_output(tp);<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0...<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0if (SEQ_GT(onxt, tp-&gt;snd_nxt))<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0tp-&gt;snd_nxt =3D onxt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0-David Borman<br>
&gt;<br>
&gt;&gt; On Aug 4, 2016, at 4:08 AM, Kobby Carmona &lt;<a href=3D"mailto:ko=
bby.Carmona@qlogic.com" target=3D"_blank">kobby.Carmona@qlogic.com</a>&gt; =
wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi all,<br>
&gt;&gt; While running a bidirectional scenario with random drops in a netw=
ork simulator of our (QLogic&#39;s NIC) TCP stack we found a case where it =
seems there is deadlock in the TCP protocol (the connection will keep sendi=
ng pure acks from both sides until RTO will expire multiple times and a RST=
 will sent to close the connection).<br>
&gt;&gt; The scenario is as follows (there is an example with numbers for e=
ach stage assuming the MSS and each packet is 1000B):<br>
&gt;&gt; 1. Both sides are transmitting data and a single packet is dropped=
 on either side and the next two packets are received properly<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Side A - SND.MAX=3D3000, SND.NXT=3D3000, SND.U=
NA=3D1000, RCV.NXT=3D11000, out-of-order block 12000-13000<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Side B - SND.MAX =3D13000, SND.NXT =3D13000, S=
ND.UNA=3D11000,<br>
&gt;&gt; RCV.NXT=3D1000, out-of-order block 2000-3000 2. RTO timer expires =
on both sides<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Side A - SND.MAX=3D3000, SND.NXT=3D1000, SND.U=
NA=3D1000, RCV.NXT=3D11000, out-of-order block 12000-13000<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Side B - SND.MAX =3D13000, SND.NXT=3D11000, SN=
D.UNA=3D11000, RCV.NXT=3D1000,<br>
&gt;&gt; out-of-order block 2000-3000 3. Both sides transmit a single packe=
t to the peer:<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 A-&gt;B - pkt.seq=3D1000, pkt.ack=3D11000, len=
=3D1000<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 B-&gt;A - pkt.seq=3D11000, pkt.ack=3D1000, len=
=3D1000 3. Both sides receive<br>
&gt;&gt; the packets and update the receive context:<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Side A - SND.MAX=3D3000, SND.NXT=3D2000, SND.U=
NA=3D1000, RCV.NXT=3D13000<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Side B - SND.MAX=3D13000, SND.NXT=3D12000, SND=
.UNA=3D11000, RCV.NXT=3D3000 4.<br>
&gt;&gt; Both sides send another segment:<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 A-&gt;B - pkt.seq=3D2000, pkt.ack=3D13000, len=
=3D1000<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 B-&gt;A - pkt.seq=3D12000, pkt.ack=3D3000, len=
=3D1000 5. Both sides don&#39;t<br>
&gt;&gt; accept the packet (and don&#39;t update SND.UNA) since the sequenc=
e on the packet is less than RCV.NXT (sequence number check in page 69 of R=
FC793) and send a pure ACK instead<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 A-&gt;B - pkt.seq=3D2000, pkt.ack=3D13000, len=
=3D0 (pure ACK)<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 B-&gt;A - pkt.seq=3D12000, pkt.ack=3D3000, len=
=3D0 (pure ACK) 6. This will<br>
&gt;&gt; continue forever (until the connection will be terminated by RST) =
since every packet that ends before RCV.NXT (even a retransmit from SND.UNA=
) will be dropped.<br>
&gt;&gt;<br>
&gt;&gt; Did anyone encountered this issue before? Is the anything we misse=
d on this sequence?<br>
&gt;&gt; If this is indeed a real deadlock, there might be several solution=
s to this which will require a modification in receive processing of RFC793=
. But I would like to know if you think this is a real issue before dealing=
 with solutions.<br>
&gt;&gt; Thanks,<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 Kobby<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; ______________________________<wbr>_________________<br>
&gt;&gt; tcpm mailing list<br>
&gt;&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</=
a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</=
a><br>
&gt;<br>
&gt; ______________________________<wbr>_________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><b=
r>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><b=
r>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
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arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div></div></div></div></div></di=
v></div></div></div>

--94eb2c04c796ebf35b053a23e6fc--


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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida
<nishida@sfc.wide.ad.jp> wrote:
> Hello,
> I personally think this is an interesting corner case for discussion.
> It looks a minor one, but I'm not not very sure if we can leave it for each
> implementation.
> I also guess a question would be if the BSD's fix is the best way for the
> issue.

Yes, I agree this is an interesting case for discussion.

FWIW, as a point of comparison for discussion, Linux's approach is a
little different: in Linux, the sender does not rewind SND.NXT on
retransmissions (RTO or Fast Recovery). Then the sender usually uses
SND.NXT for the seq field of outgoing pure ACKs. I say "usually"
because the Linux code has some code to deal with the case where the
receiver has withdrawn the receive window, so that SND.NXT is now
beyond the receive window. The code tcp_send_ack() uses to pick a seq
for outgoing pure ACKs looks like:

/* SND.NXT, if window was not shrunk.
 * If window has been shrunk, what should we make? It is not clear at
all.
 * Using SND.UNA we will fail to open window, SND.NXT is out of
window. :-(
 * Anything in between SND.UNA...SND.UNA+SND.WND also can be already
 * invalid. OK, let's make this for now:
 */
static inline __u32 tcp_acceptable_seq(const struct sock *sk)
{
        const struct tcp_sock *tp = tcp_sk(sk);

        if (!before(tcp_wnd_end(tp), tp->snd_nxt))
                return tp->snd_nxt;
        else
                return tcp_wnd_end(tp);
}

neal


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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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Probably the more correct way of stating it is that ACK-only packets =
should be sent with the largest in-window sequence number that has ever =
been sent.  It is the only way to guarantee that the sequence number of =
the ACK will be within the window at the receiving side.

That being said, probe packets can be out of window.  And if the window =
is truly closed, you don=E2=80=99t want to send and ACK-only packet with =
that sequence number, as it will also be out of window.  Hence, send =
ACK-only packets with the largest in window sequence number.  In the BSD =
code probes are only one byte, and in persist state SND.NXT will be the =
largest in-window that has ever been sent.  But if we are in persist =
state because the ACK was lost or the window just opened and the probe =
data is accepted, the ACK will be to the left of the window by the size =
of the probe packet.  Hence the receiver needs to accept ACK packets one =
byte to the left of the window to prevent an ACK war.  The more general =
case is that you need to accept ACKs of at least N bytes to the left of =
the window, where N represents the largest size of probe that you will =
ever send.  That, along with sending ACKs with the largest in-window =
sequence number, will guarantee that you don=E2=80=99t get into an ACK =
war, even the two sides have different values for N  (only one side =
needs to accept the ACK to stop the ACK-war, and that will happen on the =
side with the larger value for N).

			-David Borman

> On Aug 15, 2016, at 5:39 PM, Yoshifumi Nishida =
<nishida@sfc.wide.ad.jp> wrote:
>=20
> Hello,
> I personally think this is an interesting corner case for discussion.=20=

> It looks a minor one, but I'm not not very sure if we can leave it for =
each implementation.=20
> I also guess a question would be if the BSD's fix is the best way for =
the issue.
>=20
> Thanks,
> --
> Yoshi
>=20
> On Thu, Aug 11, 2016 at 1:46 PM, Kobby Carmona =
<kobby.Carmona@qlogic.com> wrote:
> Hi David,
> This makes a lot of sense. We will fix our code.
> Thanks for your help on this,
>=20
> BTW,
> Is this issue of mentioned in any RFC? If not do you see a point in =
adding explicit note on the SEQ of pure ACK in case of retransmission?
>=20
>        Kobby
>=20
> -----Original Message-----
> From: David Borman [mailto:dab@weston.borman.com]
> Sent: Monday, August 08, 2016 12:35 AM
> To: Kobby Carmona <kobby.Carmona@qlogic.com>
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Possible deadlock scenario with retransmission on =
both sides at the same time
>=20
>> On Aug 7, 2016, at 3:37 AM, Kobby Carmona <kobby.Carmona@qlogic.com> =
wrote:
>>=20
>> Hi David,
>> The code below is true for fast-retransmit.
>> But in case of retransmit timer expiration (TCPT_REXMT) snd_nxt is =
set to snd_una. And in this case CQND will be set to 1MSS so in the =
example below the transmitters can send only a single segment from =
sequence 2000/12000.
>=20
> Your underlying problem is that the ACK-only packets are being sent =
with the wrong sequence number, and that is what is causing them to be =
dropped.  One way to fix that is to put SND.NXT back to its previous =
value after do the the retransmit.  But you are correct, in the BSD code =
it only does that in the fast retransmit code; when the timer based =
retransmit code fires, it just pulls back SND.NXT to SND.UNA.  However, =
in the BSD code that I=E2=80=99m looking at, it keeps track of the =
largest sequence number sent in SND.MAX, and in the tcp_output() path =
there is this bit of code:
>=20
>       /*
>        * If we are doing retransmissions, then snd_nxt will
>        * not reflect the first unsent octet.  For ACK only
>        * packets, we do not want the sequence number of the
>        * retransmitted packet, we want the sequence number
>        * of the next unsent octet.  So, if there is no data
>        * (and no SYN or FIN), use snd_max instead of snd_nxt
>        * when filling in th_seq.  But if we are in persist
>        * state, snd_max might reflect one byte beyond the
>        * right edge of the window, so use snd_nxt in that
>        * case, since we know we aren't doing a retransmission.
>        * (retransmit and persist are mutually exclusive...)
>        */
>       if (len || (flags & (TH_SYN|TH_FIN)) || =
tp->t_timer[TCPT_PERSIST])
>               th->th_seq =3D htonl(tp->snd_nxt);
>       else
>               th->th_seq =3D htonl(tp->snd_max);
>=20
> That is what your implementation appears to be missing, and what is =
causing your ACK storm.  So yes, some variant of your problem has been =
seen before, and this is how the BSD code fixed it.
>=20
>                        -David Borman
>=20
>>=20
>>      Kobby
>>=20
>> -----Original Message-----
>> From: David Borman [mailto:dab@weston.borman.com]
>> Sent: Thursday, August 04, 2016 10:37 PM
>> To: Kobby Carmona <kobby.Carmona@qlogic.com>
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] Possible deadlock scenario with retransmission on
>> both sides at the same time
>>=20
>> After you pull back SND.NXT and do the retransmission, you should =
then restore SND.NXT back to where it was, not leave it at the backed =
off value; then the ACKs wouldn=E2=80=99t be dropped, since they =
wouldn=E2=80=99t have old seq values.  For example, in the 4.4BSD fast =
retransmit code it had:
>>=20
>>                      tcp_seq onxt =3D tp->snd_nxt;
>>                      ...
>>                      tp->snd_nxt =3D th->th_ack;
>>                      ...
>>                      (void) tcp_output(tp);
>>                      ...
>>                      if (SEQ_GT(onxt, tp->snd_nxt))
>>                              tp->snd_nxt =3D onxt;
>>=20
>>=20
>>                      -David Borman
>>=20
>>> On Aug 4, 2016, at 4:08 AM, Kobby Carmona <kobby.Carmona@qlogic.com> =
wrote:
>>>=20
>>> Hi all,
>>> While running a bidirectional scenario with random drops in a =
network simulator of our (QLogic's NIC) TCP stack we found a case where =
it seems there is deadlock in the TCP protocol (the connection will keep =
sending pure acks from both sides until RTO will expire multiple times =
and a RST will sent to close the connection).
>>> The scenario is as follows (there is an example with numbers for =
each stage assuming the MSS and each packet is 1000B):
>>> 1. Both sides are transmitting data and a single packet is dropped =
on either side and the next two packets are received properly
>>>     Side A - SND.MAX=3D3000, SND.NXT=3D3000, SND.UNA=3D1000, =
RCV.NXT=3D11000, out-of-order block 12000-13000
>>>     Side B - SND.MAX =3D13000, SND.NXT =3D13000, SND.UNA=3D11000,
>>> RCV.NXT=3D1000, out-of-order block 2000-3000 2. RTO timer expires on =
both sides
>>>     Side A - SND.MAX=3D3000, SND.NXT=3D1000, SND.UNA=3D1000, =
RCV.NXT=3D11000, out-of-order block 12000-13000
>>>     Side B - SND.MAX =3D13000, SND.NXT=3D11000, SND.UNA=3D11000, =
RCV.NXT=3D1000,
>>> out-of-order block 2000-3000 3. Both sides transmit a single packet =
to the peer:
>>>     A->B - pkt.seq=3D1000, pkt.ack=3D11000, len=3D1000
>>>     B->A - pkt.seq=3D11000, pkt.ack=3D1000, len=3D1000 3. Both sides =
receive
>>> the packets and update the receive context:
>>>     Side A - SND.MAX=3D3000, SND.NXT=3D2000, SND.UNA=3D1000, =
RCV.NXT=3D13000
>>>     Side B - SND.MAX=3D13000, SND.NXT=3D12000, SND.UNA=3D11000, =
RCV.NXT=3D3000 4.
>>> Both sides send another segment:
>>>     A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D1000
>>>     B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D1000 5. Both sides =
don't
>>> accept the packet (and don't update SND.UNA) since the sequence on =
the packet is less than RCV.NXT (sequence number check in page 69 of =
RFC793) and send a pure ACK instead
>>>     A->B - pkt.seq=3D2000, pkt.ack=3D13000, len=3D0 (pure ACK)
>>>     B->A - pkt.seq=3D12000, pkt.ack=3D3000, len=3D0 (pure ACK) 6. =
This will
>>> continue forever (until the connection will be terminated by RST) =
since every packet that ends before RCV.NXT (even a retransmit from =
SND.UNA) will be dropped.
>>>=20
>>> Did anyone encountered this issue before? Is the anything we missed =
on this sequence?
>>> If this is indeed a real deadlock, there might be several solutions =
to this which will require a modification in receive processing of =
RFC793. But I would like to know if you think this is a real issue =
before dealing with solutions.
>>> Thanks,
>>>=20
>>>     Kobby
>>>=20
>>>=20
>>> _______________________________________________
>>> 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
>=20
> _______________________________________________
> tcpm mailing list
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>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
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To: Joe Touch <touch@isi.edu>, Neal Cardwell <ncardwell@google.com>, Yuchung Cheng <ycheng@google.com>
References: <20160810183654.05358B80C3A@rfc-editor.org> <CAOp4FwTogyBXLYdjHrrnM3-Uz2wpSX31eZg+GJwUP5LBnqu=sQ@mail.gmail.com> <7ac89d58-fc3a-9e12-9d22-0602944f8677@isi.edu> <e8d9584f-a069-ec4f-d6f1-f8e199fdb071@isi.edu> <CAK6E8=ewOCUBu7Ko9aK78_e2rmR70G_yvUpZKLm3n0tYdLs=Yw@mail.gmail.com> <CADVnQy=-6RFns-CDLwJkueA2+d9Ov7SbSXmFgkTvrrNMAsUo6g@mail.gmail.com> <bc8861bd-b9a8-153b-7bd8-ffbda9aa85ad@isi.edu>
From: =?UTF-8?Q?Mirja_K=c3=bchlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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Hi all,

let me try to wrap this up:

To me this does not seem to be a case for an errata. For sure some 
clarification seems to be helpful here but that's not what erratas are for.

Therefore, I would suggest to simply set this into "Held for Document Update" 
state such that it is noticed in case we decide to update the document at 
some point. Does that make sense?

Thanks,
Mirja



On 12.08.2016 22:19, Joe Touch wrote:
>
>
> On 8/12/2016 12:17 PM, Neal Cardwell wrote:
>> On Fri, Aug 12, 2016 at 9:54 AM, Yuchung Cheng <ycheng@google.com> wrote:
>>> I think an errata on recommending per-connection limit and cite the
>>> research finding is worthy.
>> I agree that it's worthwhile to have an errata to clarify that a
>> per-connection limit is best. At the very least there seems
>> some risky ambiguity here.
>
> I disagree, as per below...
>>
>> In fact, AFAICT RFC 5961, Section 7, page 13 --
>>
>>    https://tools.ietf.org/html/rfc5961#section-7
>>
>> -- has several passages that seem to implicitly suggest that
>> the rate-limiting mechanism it specifies as a SHOULD is not
>> per-connection but rather global.
>
> There's plenty of benefit to setting a single configuration for an
> entire system vs. setting per-connection limits. That's completely
> different from whether the ACKs are tallied globally or per connection.
>
>> First:
>>
>>     ... an
>>     administrator who is more interested in faster cleanup of stale
>>     connections (i.e., concerned about excess TCP state) may decide to
>>     set a higher value thus allowing more RST's to be processed in any
>>     given time period.
>>
>> If this is a single-connection rate limit, then what practical reason
>> would an administrator have to allow "more RST's to be processed in
>> any given time period"? I can't think of a case where a legit live
>> connection would send more than one RST.
> A single connection might process many RSTs because some can be out of
> window (they'd be dropped) or not at the end of the window (dropped
> under some configurations). I.e., receiving and evaluating RSTs doesn't
> always result in resetting the connection.
>
>> If this is about a single
>> connection, then allowing one mid-window RST (and responding to that
>> one with a single challenge ACK) should be enough,
> It would be IF the other end issued a legitimate RST. If this is a
> spoofing attack, the data can still keep coming which means that the
> window might still be moving forward by the time the next attack RST
> arrives. As a result, this sort of 'catch up' can go on until the
> connection ends. The point of the text here is to avoid wasting
> resources in such cases.
>> in most cases, to
>> send a challenge ACK that elicits a proper RST from the remote
>> endpoint. It's not clear to me what the motivation would be for
>> intentionally sending a challenge ACK in response to more than one
>> mid-window RST in a single connection.
> Protecting against that case is the motivation *of the entire document*.
>
>> Thus AFAICT "allowing more RST's to be processed in any given time
>> period" suggests that the rate limit is across connections.
>
> There's no benefit to doing this across connections at all, unless you
> think that there's some reason that legitimate connections shouldn't be
> reset by valid RSTs (there's no benefit to that).
>
>> Second:
>>
>>     The time limit SHOULD be tunable to help timeout brute force attacks
>>     faster than a potential legitimate flood of RSTs.
>>
>> I may be missing something, but if this rate-limiting mechanism is
>> only about a single connection, then what is a "legitimate flood of
>> RSTs"?
> A sequence of RSTs that pick different values, as described in RFC 4953.
>
> Or a sequence of RSTs aimed at responses to the ACK-after-RST that is
> chasing legitimate data, as described above.
>
>>   Why would it be important for a single connection to allow a
>> "legitimate flood of RSTs" destinted for itself?
> It wouldn't - that's the point of this text. You should treat a flood of
> RSTs for a single connection as an attack and throttle them accordingly.
>
>> A single legitimate
>> live connection should not send out a flood of RSTs; it should send
>> out one. Once there is one legitimate RST that arrives and is
>> processed, the receiver deletes the connection, and any other arriving
>> RSTs don't match any existing connection, so are ignored and
>> dropped, in which case is no challenge ACK to rate-limit.
> Again, the point of this document are attacks, not legitimate connections.
>> By contrast, if this rate limit applies across connections then "a
>> potential legitimate flood of RSTs" makes sense, since a group of
>> legitimate live connections that all close in a burst may, in certain
>> cases, generate a legitimate flood of RSTs. So tuning the limit
>> upward would make sense.
>
> There's no case I can imagine where a set of attacks on one connection
> should be cause for concern on another necessarily.
>
> Yes, there's no real need to set per-connection limits (i.e., a global
> per-connection limit seems fine), but counting them as a group serves no
> purpose against any threat, unless you think that "if I'm attacked on
> one connection, I will be attacked on another" - but that's an incorrect
> inference IMO.
>
>>
>> Sorry if I'm missing something obvious. But it seems like there are at
>> least real ambiguities there, on a point significant enough to be worth
>> clearing up.
>
> I think you're missing the point of the document, but that's just my
> interpretation of the exchange.
>
> Joe
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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References: <20160810183654.05358B80C3A@rfc-editor.org> <CAOp4FwTogyBXLYdjHrrnM3-Uz2wpSX31eZg+GJwUP5LBnqu=sQ@mail.gmail.com> <7ac89d58-fc3a-9e12-9d22-0602944f8677@isi.edu> <e8d9584f-a069-ec4f-d6f1-f8e199fdb071@isi.edu> <CAK6E8=ewOCUBu7Ko9aK78_e2rmR70G_yvUpZKLm3n0tYdLs=Yw@mail.gmail.com> <CADVnQy=-6RFns-CDLwJkueA2+d9Ov7SbSXmFgkTvrrNMAsUo6g@mail.gmail.com> <bc8861bd-b9a8-153b-7bd8-ffbda9aa85ad@isi.edu> <57B31020.60506@tik.ee.ethz.ch>
From: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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On Tue, Aug 16, 2016 at 6:07 AM, Mirja K=C3=BChlewind <
mirja.kuehlewind@tik.ee.ethz.ch> wrote:

> Hi all,
>
> let me try to wrap this up:
>
> To me this does not seem to be a case for an errata. For sure some
> clarification seems to be helpful here but that's not what erratas are fo=
r.
>
> Therefore, I would suggest to simply set this into "Held for Document
> Update" state such that it is noticed in case we decide to update the
> document at some point. Does that make sense?
>
sounds fine with me but do documents really get updated?


>
> Thanks,
> Mirja
>
>
>
>
> On 12.08.2016 22:19, Joe Touch wrote:
>
>>
>>
>> On 8/12/2016 12:17 PM, Neal Cardwell wrote:
>>
>>> On Fri, Aug 12, 2016 at 9:54 AM, Yuchung Cheng <ycheng@google.com>
>>> wrote:
>>>
>>>> I think an errata on recommending per-connection limit and cite the
>>>> research finding is worthy.
>>>>
>>> I agree that it's worthwhile to have an errata to clarify that a
>>> per-connection limit is best. At the very least there seems
>>> some risky ambiguity here.
>>>
>>
>> I disagree, as per below...
>>
>>>
>>> In fact, AFAICT RFC 5961, Section 7, page 13 --
>>>
>>>    https://tools.ietf.org/html/rfc5961#section-7
>>>
>>> -- has several passages that seem to implicitly suggest that
>>> the rate-limiting mechanism it specifies as a SHOULD is not
>>> per-connection but rather global.
>>>
>>
>> There's plenty of benefit to setting a single configuration for an
>> entire system vs. setting per-connection limits. That's completely
>> different from whether the ACKs are tallied globally or per connection.
>>
>> First:
>>>
>>>     ... an
>>>     administrator who is more interested in faster cleanup of stale
>>>     connections (i.e., concerned about excess TCP state) may decide to
>>>     set a higher value thus allowing more RST's to be processed in any
>>>     given time period.
>>>
>>> If this is a single-connection rate limit, then what practical reason
>>> would an administrator have to allow "more RST's to be processed in
>>> any given time period"? I can't think of a case where a legit live
>>> connection would send more than one RST.
>>>
>> A single connection might process many RSTs because some can be out of
>> window (they'd be dropped) or not at the end of the window (dropped
>> under some configurations). I.e., receiving and evaluating RSTs doesn't
>> always result in resetting the connection.
>>
>> If this is about a single
>>> connection, then allowing one mid-window RST (and responding to that
>>> one with a single challenge ACK) should be enough,
>>>
>> It would be IF the other end issued a legitimate RST. If this is a
>> spoofing attack, the data can still keep coming which means that the
>> window might still be moving forward by the time the next attack RST
>> arrives. As a result, this sort of 'catch up' can go on until the
>> connection ends. The point of the text here is to avoid wasting
>> resources in such cases.
>>
>>> in most cases, to
>>> send a challenge ACK that elicits a proper RST from the remote
>>> endpoint. It's not clear to me what the motivation would be for
>>> intentionally sending a challenge ACK in response to more than one
>>> mid-window RST in a single connection.
>>>
>> Protecting against that case is the motivation *of the entire document*.
>>
>> Thus AFAICT "allowing more RST's to be processed in any given time
>>> period" suggests that the rate limit is across connections.
>>>
>>
>> There's no benefit to doing this across connections at all, unless you
>> think that there's some reason that legitimate connections shouldn't be
>> reset by valid RSTs (there's no benefit to that).
>>
>> Second:
>>>
>>>     The time limit SHOULD be tunable to help timeout brute force attack=
s
>>>     faster than a potential legitimate flood of RSTs.
>>>
>>> I may be missing something, but if this rate-limiting mechanism is
>>> only about a single connection, then what is a "legitimate flood of
>>> RSTs"?
>>>
>> A sequence of RSTs that pick different values, as described in RFC 4953.
>>
>> Or a sequence of RSTs aimed at responses to the ACK-after-RST that is
>> chasing legitimate data, as described above.
>>
>>   Why would it be important for a single connection to allow a
>>> "legitimate flood of RSTs" destinted for itself?
>>>
>> It wouldn't - that's the point of this text. You should treat a flood of
>> RSTs for a single connection as an attack and throttle them accordingly.
>>
>> A single legitimate
>>> live connection should not send out a flood of RSTs; it should send
>>> out one. Once there is one legitimate RST that arrives and is
>>> processed, the receiver deletes the connection, and any other arriving
>>> RSTs don't match any existing connection, so are ignored and
>>> dropped, in which case is no challenge ACK to rate-limit.
>>>
>> Again, the point of this document are attacks, not legitimate connection=
s.
>>
>>> By contrast, if this rate limit applies across connections then "a
>>> potential legitimate flood of RSTs" makes sense, since a group of
>>> legitimate live connections that all close in a burst may, in certain
>>> cases, generate a legitimate flood of RSTs. So tuning the limit
>>> upward would make sense.
>>>
>>
>> There's no case I can imagine where a set of attacks on one connection
>> should be cause for concern on another necessarily.
>>
>> Yes, there's no real need to set per-connection limits (i.e., a global
>> per-connection limit seems fine), but counting them as a group serves no
>> purpose against any threat, unless you think that "if I'm attacked on
>> one connection, I will be attacked on another" - but that's an incorrect
>> inference IMO.
>>
>>
>>> Sorry if I'm missing something obvious. But it seems like there are at
>>> least real ambiguities there, on a point significant enough to be worth
>>> clearing up.
>>>
>>
>> I think you're missing the point of the document, but that's just my
>> interpretation of the exchange.
>>
>> Joe
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>>

--001a113f6d5e04a744053a33e374
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Aug 16, 2016 at 6:07 AM, Mirja K=C3=BChlewind <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:mirja.kuehlewind@tik.ee.ethz.ch" target=3D"_blank">m=
irja.kuehlewind@tik.ee.ethz.<wbr>ch</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex">Hi all,<br>
<br>
let me try to wrap this up:<br>
<br>
To me this does not seem to be a case for an errata. For sure some clarific=
ation seems to be helpful here but that&#39;s not what erratas are for.<br>
<br>
Therefore, I would suggest to simply set this into &quot;Held for Document =
Update&quot; state such that it is noticed in case we decide to update the =
document at some point. Does that make sense?<br></blockquote><div>sounds f=
ine with me but do documents really get updated?</div><div>=C2=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">
<br>
Thanks,<br>
Mirja<div><div class=3D"m_-7226869345942298187h5"><br>
<br>
<br>
<br>
On 12.08.2016 22:19, Joe Touch wrote:<br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"m_-7226869345=
942298187h5">
<br>
<br>
On 8/12/2016 12:17 PM, Neal Cardwell wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Fri, Aug 12, 2016 at 9:54 AM, Yuchung Cheng &lt;<a href=3D"mailto:ycheng=
@google.com" target=3D"_blank">ycheng@google.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think an errata on recommending per-connection limit and cite the<br>
research finding is worthy.<br>
</blockquote>
I agree that it&#39;s worthwhile to have an errata to clarify that a<br>
per-connection limit is best. At the very least there seems<br>
some risky ambiguity here.<br>
</blockquote>
<br>
I disagree, as per below...<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
In fact, AFAICT RFC 5961, Section 7, page 13 --<br>
<br>
=C2=A0 =C2=A0<a href=3D"https://tools.ietf.org/html/rfc5961#section-7" rel=
=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/r<wbr>fc5961#=
section-7</a><br>
<br>
-- has several passages that seem to implicitly suggest that<br>
the rate-limiting mechanism it specifies as a SHOULD is not<br>
per-connection but rather global.<br>
</blockquote>
<br>
There&#39;s plenty of benefit to setting a single configuration for an<br>
entire system vs. setting per-connection limits. That&#39;s completely<br>
different from whether the ACKs are tallied globally or per connection.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
First:<br>
<br>
=C2=A0 =C2=A0 ... an<br>
=C2=A0 =C2=A0 administrator who is more interested in faster cleanup of sta=
le<br>
=C2=A0 =C2=A0 connections (i.e., concerned about excess TCP state) may deci=
de to<br>
=C2=A0 =C2=A0 set a higher value thus allowing more RST&#39;s to be process=
ed in any<br>
=C2=A0 =C2=A0 given time period.<br>
<br>
If this is a single-connection rate limit, then what practical reason<br>
would an administrator have to allow &quot;more RST&#39;s to be processed i=
n<br>
any given time period&quot;? I can&#39;t think of a case where a legit live=
<br>
connection would send more than one RST.<br>
</blockquote>
A single connection might process many RSTs because some can be out of<br>
window (they&#39;d be dropped) or not at the end of the window (dropped<br>
under some configurations). I.e., receiving and evaluating RSTs doesn&#39;t=
<br>
always result in resetting the connection.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
If this is about a single<br>
connection, then allowing one mid-window RST (and responding to that<br>
one with a single challenge ACK) should be enough,<br>
</blockquote>
It would be IF the other end issued a legitimate RST. If this is a<br>
spoofing attack, the data can still keep coming which means that the<br>
window might still be moving forward by the time the next attack RST<br>
arrives. As a result, this sort of &#39;catch up&#39; can go on until the<b=
r>
connection ends. The point of the text here is to avoid wasting<br>
resources in such cases.<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
in most cases, to<br>
send a challenge ACK that elicits a proper RST from the remote<br>
endpoint. It&#39;s not clear to me what the motivation would be for<br>
intentionally sending a challenge ACK in response to more than one<br>
mid-window RST in a single connection.<br>
</blockquote>
Protecting against that case is the motivation *of the entire document*.<br=
>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Thus AFAICT &quot;allowing more RST&#39;s to be processed in any given time=
<br>
period&quot; suggests that the rate limit is across connections.<br>
</blockquote>
<br>
There&#39;s no benefit to doing this across connections at all, unless you<=
br>
think that there&#39;s some reason that legitimate connections shouldn&#39;=
t be<br>
reset by valid RSTs (there&#39;s no benefit to that).<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Second:<br>
<br>
=C2=A0 =C2=A0 The time limit SHOULD be tunable to help timeout brute force =
attacks<br>
=C2=A0 =C2=A0 faster than a potential legitimate flood of RSTs.<br>
<br>
I may be missing something, but if this rate-limiting mechanism is<br>
only about a single connection, then what is a &quot;legitimate flood of<br=
>
RSTs&quot;?<br>
</blockquote>
A sequence of RSTs that pick different values, as described in RFC 4953.<br=
>
<br>
Or a sequence of RSTs aimed at responses to the ACK-after-RST that is<br>
chasing legitimate data, as described above.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=C2=A0 Why would it be important for a single connection to allow a<br>
&quot;legitimate flood of RSTs&quot; destinted for itself?<br>
</blockquote>
It wouldn&#39;t - that&#39;s the point of this text. You should treat a flo=
od of<br>
RSTs for a single connection as an attack and throttle them accordingly.<br=
>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
A single legitimate<br>
live connection should not send out a flood of RSTs; it should send<br>
out one. Once there is one legitimate RST that arrives and is<br>
processed, the receiver deletes the connection, and any other arriving<br>
RSTs don&#39;t match any existing connection, so are ignored and<br>
dropped, in which case is no challenge ACK to rate-limit.<br>
</blockquote>
Again, the point of this document are attacks, not legitimate connections.<=
br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
By contrast, if this rate limit applies across connections then &quot;a<br>
potential legitimate flood of RSTs&quot; makes sense, since a group of<br>
legitimate live connections that all close in a burst may, in certain<br>
cases, generate a legitimate flood of RSTs. So tuning the limit<br>
upward would make sense.<br>
</blockquote>
<br>
There&#39;s no case I can imagine where a set of attacks on one connection<=
br>
should be cause for concern on another necessarily.<br>
<br>
Yes, there&#39;s no real need to set per-connection limits (i.e., a global<=
br>
per-connection limit seems fine), but counting them as a group serves no<br=
>
purpose against any threat, unless you think that &quot;if I&#39;m attacked=
 on<br>
one connection, I will be attacked on another&quot; - but that&#39;s an inc=
orrect<br>
inference IMO.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Sorry if I&#39;m missing something obvious. But it seems like there are at<=
br>
least real ambiguities there, on a point significant enough to be worth<br>
clearing up.<br>
</blockquote>
<br>
I think you&#39;re missing the point of the document, but that&#39;s just m=
y<br>
interpretation of the exchange.<br>
<br>
Joe<br>
<br></div></div><span>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
<br>
</span></blockquote>
</blockquote></div><br></div></div>

--001a113f6d5e04a744053a33e374--


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Hi TCPM,

Just a quick note; Daniel and Tim have made an update to the TCP Tuning =
for HTTP draft:
  https://tools.ietf.org/html/draft-stenberg-httpbis-tcp

We've had a Call for Adoption open for this draft for a while, and will =
likely adopt it soon. However, we'd like to get feedback from this =
community first -- both about the latest version of the input document, =
and to see if there's interest in helping out.

You can give feedback on the HTTP WG mailing list =
<https://lists.w3.org/Archives/Public/ietf-http-wg/>, or  by responding =
to this e-mail (Please leave the CC line; Patrick and I will try to =
summarise the feedback to the WG).

Cheers,

--
Mark Nottingham   https://www.mnot.net/





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From: Joe Touch <touch@isi.edu>
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Hi, Mark, et al.,

I posted a review of this document to both to TCPM and HTTP WGs.

This update fails to address the issues I raised - notably that many of
the issues therein are known *and published*.

So first, can we discuss the issue of PLAGIARISM?

Not only of two of my works, but of many others that pointed out most of
the information summarized in this doc.

Second, the step of "adoption" needs to wait until there's something new
here that wasn't known 20 years ago and the issue of plagiarism is
addressed.

Joe

On 8/16/2016 7:02 PM, Mark Nottingham wrote:
> Hi TCPM,
>
> Just a quick note; Daniel and Tim have made an update to the TCP Tuning for HTTP draft:
>   https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>
> We've had a Call for Adoption open for this draft for a while, and will likely adopt it soon. However, we'd like to get feedback from this community first -- both about the latest version of the input document, and to see if there's interest in helping out.
>
> You can give feedback on the HTTP WG mailing list <https://lists.w3.org/Archives/Public/ietf-http-wg/>, or  by responding to this e-mail (Please leave the CC line; Patrick and I will try to summarise the feedback to the WG).
>
> Cheers,
>
> --
> Mark Nottingham   https://www.mnot.net/
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Fwd: Re: [tcpm] FW: Call for Adoption: TCP Tuning for HTTP
Cc: touch@isi.edu
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Hi, all,

This doc was noted on the TCPM list.

See my observations below.

Joe


-------- Forwarded Message --------
Subject: Re: [tcpm] FW: Call for Adoption: TCP Tuning for HTTP
Date: Wed, 2 Mar 2016 12:25:07 -0800
From: Joe Touch <touch@isi.edu>
To: Scharf, Michael (Nokia - DE) <michael.scharf@nokia.com>,
tcpm@ietf.org Extensions <tcpm@ietf.org>
CC: touch@isi.edu



On 3/2/2016 1:39 AM, Scharf, Michael (Nokia - DE) wrote:
> I assume this could be of interest to the TCPM community.

I have doubts:

- it reads like a Linux manual page

	All Linux-specific references and commands would need to be
	moved to an appendix to be useful as an RFC.

- this repeats (sometimes correctly, sometimes in error) existing advice

J. Heidemann, K. Obraczka, J. Touch, Modeling the Performance of HTTP
Over Several Transport Protocols, IEEE/ACM Transactions on Networking,
V5, N5, Oct. 1997, pp.616-630.

T. Faber, J. Touch, and W. Yue, The TIME-WAIT state in TCP and Its
Effect on Busy Servers, in Proc. IEEE Infocom, 1999, pp. 1573-1583.

- it has significant errors

	TIME-WAIT issues apply to servers, not clients.

	Nagle has been known to perform poorly for multibyte
	interactive traffic for a very long time, including not
	only web traffic but also multi-byte character or keyboard
	signals.

	Disabling slow-start after idle is safe only with pacing.
	Without pacing, the resulting traffic can generate a burst
	that was never experienced and result in both poor performance
	for the current connection and potential impact to competing
	traffic.

(those are just a few)

Overall, I think a man page might be useful, but this summary isn't
useful for the IETF.

Joe

> -----Original Message-----
> From: Scharf, Michael (Nokia - DE) 
> Sent: Wednesday, March 02, 2016 10:37 AM
> To: 'Mark Nottingham'; HTTP WG
> Cc: amankin@verisign.com; Daniel Stenberg
> Subject: RE: Call for Adoption: TCP Tuning for HTTP
> 
> The document refers to several TCPM RFCs with experimental status, e.g., in Section 3. That may have to be taken into account when heading towards BCP status.
> 
> Michael
> (TCPM co-chair)
> 
> 
> -----Original Message-----
> From: Mark Nottingham [mailto:mnot@mnot.net] 
> Sent: Wednesday, March 02, 2016 6:47 AM
> To: HTTP WG
> Cc: amankin@verisign.com; Daniel Stenberg
> Subject: Call for Adoption: TCP Tuning for HTTP
> 
> [ copying Alison as our Transport Tech Advisor ]
> 
> Daniel has kindly started a document about how HTTP uses TCP, both for /1 and /2:
>   <https://tools.ietf.org/html/draft-stenberg-httpbis-tcp>
> 
> We haven't explicitly discussed this at a meeting, but I have heard interest in this topic from a variety of folks.
> 
> What do people think about adopting this with a target of Best Current Practice?
> 
> Please comment on-list.
> 
> Regards,
> 
> --
> Mark Nottingham   https://www.mnot.net/
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 


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Subject: Re: [tcpm] FW: Call for Adoption: TCP Tuning for HTTP
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On 3/2/2016 1:39 AM, Scharf, Michael (Nokia - DE) wrote:
> I assume this could be of interest to the TCPM community.

I have doubts:

- it reads like a Linux manual page

	All Linux-specific references and commands would need to be
	moved to an appendix to be useful as an RFC.

- this repeats (sometimes correctly, sometimes in error) existing advice

J. Heidemann, K. Obraczka, J. Touch, Modeling the Performance of HTTP
Over Several Transport Protocols, IEEE/ACM Transactions on Networking,
V5, N5, Oct. 1997, pp.616-630.

T. Faber, J. Touch, and W. Yue, The TIME-WAIT state in TCP and Its
Effect on Busy Servers, in Proc. IEEE Infocom, 1999, pp. 1573-1583.

- it has significant errors

	TIME-WAIT issues apply to servers, not clients.

	Nagle has been known to perform poorly for multibyte
	interactive traffic for a very long time, including not
	only web traffic but also multi-byte character or keyboard
	signals.

	Disabling slow-start after idle is safe only with pacing.
	Without pacing, the resulting traffic can generate a burst
	that was never experienced and result in both poor performance
	for the current connection and potential impact to competing
	traffic.

(those are just a few)

Overall, I think a man page might be useful, but this summary isn't
useful for the IETF.

Joe

> -----Original Message-----
> From: Scharf, Michael (Nokia - DE) 
> Sent: Wednesday, March 02, 2016 10:37 AM
> To: 'Mark Nottingham'; HTTP WG
> Cc: amankin@verisign.com; Daniel Stenberg
> Subject: RE: Call for Adoption: TCP Tuning for HTTP
> 
> The document refers to several TCPM RFCs with experimental status, e.g., in Section 3. That may have to be taken into account when heading towards BCP status.
> 
> Michael
> (TCPM co-chair)
> 
> 
> -----Original Message-----
> From: Mark Nottingham [mailto:mnot@mnot.net] 
> Sent: Wednesday, March 02, 2016 6:47 AM
> To: HTTP WG
> Cc: amankin@verisign.com; Daniel Stenberg
> Subject: Call for Adoption: TCP Tuning for HTTP
> 
> [ copying Alison as our Transport Tech Advisor ]
> 
> Daniel has kindly started a document about how HTTP uses TCP, both for /1 and /2:
>   <https://tools.ietf.org/html/draft-stenberg-httpbis-tcp>
> 
> We haven't explicitly discussed this at a meeting, but I have heard interest in this topic from a variety of folks.
> 
> What do people think about adopting this with a target of Best Current Practice?
> 
> Please comment on-list.
> 
> Regards,
> 
> --
> Mark Nottingham   https://www.mnot.net/
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 

--------------B84BA9D3A06B2BC51706E71F--


From nobody Tue Aug 16 21:43:29 2016
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Joe,

> On 17 Aug 2016, at 2:21 PM, Joe Touch <touch@isi.edu> wrote:
>=20
> Hi, Mark, et al.,
>=20
> I posted a review of this document to both to TCPM and HTTP WGs.
>=20
> This update fails to address the issues I raised - notably that many =
of
> the issues therein are known *and published*.

I'm assuming you're talking about =
<https://lists.w3.org/Archives/Public/ietf-http-wg/2016JanMar/0314.html>.


> So first, can we discuss the issue of PLAGIARISM?

Sure, let's discuss it. That's a very serious accusation. Are you saying =
that your material has been intentionally used without proper =
acknowledgement?

Personally, I doubt that. What may have happened is that the text =
brushes up against things that you've written about in the past, and you =
feel that you're not adequately acknowledged.=20

If that's the case, I'd observe that the IETF isn't an academic =
publisher, and acknowledging all prior work in an area is neither =
practical, nor required, nor current practice.

On the other hand, if it turns out that directing readers toward other =
documents (including yours) adds value, a reference might make sense.


> Not only of two of my works, but of many others that pointed out most =
of
> the information summarized in this doc.
>=20
> Second, the step of "adoption" needs to wait until there's something =
new
> here that wasn't known 20 years ago and the issue of plagiarism is
> addressed.

Other people in the HTTP *and* TCP communities have commented that such =
a document would be very useful, whether or not it's something "new that =
wasn't known 20 years ago".=20

Cheers,


>=20
> Joe
>=20
> On 8/16/2016 7:02 PM, Mark Nottingham wrote:
>> Hi TCPM,
>>=20
>> Just a quick note; Daniel and Tim have made an update to the TCP =
Tuning for HTTP draft:
>>  https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>>=20
>> We've had a Call for Adoption open for this draft for a while, and =
will likely adopt it soon. However, we'd like to get feedback from this =
community first -- both about the latest version of the input document, =
and to see if there's interest in helping out.
>>=20
>> You can give feedback on the HTTP WG mailing list =
<https://lists.w3.org/Archives/Public/ietf-http-wg/>, or  by responding =
to this e-mail (Please leave the CC line; Patrick and I will try to =
summarise the feedback to the WG).
>>=20
>> Cheers,
>>=20
>> --
>> Mark Nottingham   https://www.mnot.net/
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> <Attached Message.eml><Attached Message.eml>

--
Mark Nottingham   https://www.mnot.net/





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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi, Mark,


On 8/16/2016 9:43 PM, Mark Nottingham wrote:
> Joe,
>
>> On 17 Aug 2016, at 2:21 PM, Joe Touch <touch@isi.edu> wrote:
>>
>> Hi, Mark, et al.,
>>
>> I posted a review of this document to both to TCPM and HTTP WGs.
>>
>> This update fails to address the issues I raised - notably that many of
>> the issues therein are known *and published*.
> I'm assuming you're talking about <https://lists.w3.org/Archives/Public/ietf-http-wg/2016JanMar/0314.html>.

Yes.
>
>
>> So first, can we discuss the issue of PLAGIARISM?
> Sure, let's discuss it. That's a very serious accusation. Are you saying that your material has been intentionally used without proper acknowledgement?

Yes.

> Personally, I doubt that. What may have happened is that the text brushes up against things that you've written about in the past, and you feel that you're not adequately acknowledged. 
Plagiarism requires only that the material was published elsewhere
before. Intent has no bearing.

In addition, I informed the author - and both lists - about this over 5
months ago. You might claim that the first two versions were issued out
of ignorance, but you cannot claim that of the update.

> If that's the case, I'd observe that the IETF isn't an academic publisher, and acknowledging all prior work in an area is neither practical, nor required, nor current practice.
Plagiarism isn't an issue limited to academic environments. Publication
of a document on the web is still publication.

> On the other hand, if it turns out that directing readers toward other documents (including yours) adds value, a reference might make sense.

Those docs explain the issues more correctly and in more detail. That
should add enough value.

The real question is whether this draft adds value to those - which are
*already published*.


>
>> Not only of two of my works, but of many others that pointed out most of
>> the information summarized in this doc.
>>
>> Second, the step of "adoption" needs to wait until there's something new
>> here that wasn't known 20 years ago and the issue of plagiarism is
>> addressed.
> Other people in the HTTP *and* TCP communities have commented that such a document would be very useful, whether or not it's something "new that wasn't known 20 years ago". 

We don't need to issue new documents for people who don't read old ones.

Joe


>
> Cheers,
>
>
>> Joe
>>
>> On 8/16/2016 7:02 PM, Mark Nottingham wrote:
>>> Hi TCPM,
>>>
>>> Just a quick note; Daniel and Tim have made an update to the TCP Tuning for HTTP draft:
>>>  https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>>>
>>> We've had a Call for Adoption open for this draft for a while, and will likely adopt it soon. However, we'd like to get feedback from this community first -- both about the latest version of the input document, and to see if there's interest in helping out.
>>>
>>> You can give feedback on the HTTP WG mailing list <https://lists.w3.org/Archives/Public/ietf-http-wg/>, or  by responding to this e-mail (Please leave the CC line; Patrick and I will try to summarise the feedback to the WG).
>>>
>>> Cheers,
>>>
>>> --
>>> Mark Nottingham   https://www.mnot.net/
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>> <Attached Message.eml><Attached Message.eml>
> --
> Mark Nottingham   https://www.mnot.net/
>
>
>
>


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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Joe,

I've pinged some ADs to weigh in on this.

In the meantime -

> On 17 Aug 2016, at 3:23 PM, Joe Touch <touch@isi.edu> wrote:
>=20
> Plagiarism requires only that the material was published elsewhere
> before. Intent has no bearing.

So to paraphrase, you're accusing Daniel of failure to cite a relevant =
source out of ignorance in the first case...=20


> In addition, I informed the author - and both lists - about this over =
5
> months ago. You might claim that the first two versions were issued =
out
> of ignorance, but you cannot claim that of the update.

... and then continuing to do so after being notified.

Correct?

Looking over <http://www.isi.edu/touch/pubs/ton97.pdf> and =
<http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/>, I'm having =
trouble identifying passages that are plagiarised; can you show us?

In particular, the latter paper proposes TIME-WAIT negotiation using a =
TCP option, sending a RST from clients, and a HTTP extension, whereas =
Daniel's draft only says this about TIME_WAIT:

"""
2.5.  Lower the TCP FIN timeout


   High connection completion rates will consume ephemeral ports
   quickly.  Lower the time during which connections are in FIN-WAIT-2/
   TIME_WAIT states so that they can be purged faster and thus maintain
   a maximal number of available sockets.  The primitives for the
   assignment of these values were described in [RFC0793], however
   significantly lower values are commonly used.


2.6.  Reuse sockets in TIME_WAIT state


   When running backend servers on a managed, low latency network you
   might allow the reuse of sockets in TIME_WAIT state for new
   connections when a protocol complete termination has occurred.  There
   is no RFC that covers this behaviour.
"""

So, where is the plagiarism?

Also, how do you reconcile that with the statement in your original =
e-mail that the draft repeats your material "sometimes correctly, =
sometimes in error"? Have you considered that what you consider to be an =
error in his plagiarism might in fact be a different idea (no matter =
whose is ultimately correct)?


>> If that's the case, I'd observe that the IETF isn't an academic =
publisher, and acknowledging all prior work in an area is neither =
practical, nor required, nor current practice.
> Plagiarism isn't an issue limited to academic environments. =
Publication
> of a document on the web is still publication.

Sure. It also isn't a legal issue in this form (unless you're asserting =
copyright?). Effectively, it's a cultural norm. Again, I will point out =
that in the culture of the IETF, we historically have not cited the =
complete provenance of every idea, both because it's impractical and =
because it doesn't benefit the reader.=20

As far as I know, the IETF does not have a stated position about what =
you regard as PLAGIARISM. Hopefully we can get some clarity about that =
from the ADs, as well as some definitive evidence of what you're =
asserting.


>> On the other hand, if it turns out that directing readers toward =
other documents (including yours) adds value, a reference might make =
sense.
>=20
> Those docs explain the issues more correctly and in more detail. That
> should add enough value.
>=20
> The real question is whether this draft adds value to those - which =
are
> *already published*.
>=20
>=20
>>=20
>>> Not only of two of my works, but of many others that pointed out =
most of
>>> the information summarized in this doc.
>>>=20
>>> Second, the step of "adoption" needs to wait until there's something =
new
>>> here that wasn't known 20 years ago and the issue of plagiarism is
>>> addressed.
>> Other people in the HTTP *and* TCP communities have commented that =
such a document would be very useful, whether or not it's something "new =
that wasn't known 20 years ago".=20
>=20
> We don't need to issue new documents for people who don't read old =
ones.



--
Mark Nottingham   https://www.mnot.net/


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Date: Wed, 17 Aug 2016 08:45:45 +0200
From: Willy Tarreau <w@1wt.eu>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi Joe,

On Tue, Aug 16, 2016 at 10:23:01PM -0700, Joe Touch wrote:
> > Other people in the HTTP *and* TCP communities have commented that such a
> > document would be very useful, whether or not it's something "new that
> > wasn't known 20 years ago". 
> 
> We don't need to issue new documents for people who don't read old ones.

I've just checked the two documents you referenced. They seem to be very
well detailed but they are *scientific* research. What Daniel created are
configuration advises for people who need to configure their servers. Yes
as you mentionned they look like man pages precisely because the purpose
is to ensure they're easily understood by people who are just seeking some
help to improve their configuration. You cannot expect a server admin to
read scientific papers explaining some TCP models with some formulas to
know what to do on their servers.

Also, I don't know if there have been any update, but these documents use
SunOS 4.1.3 running on a sparc 20 as a reference. While I used to love
working on such systems 20 years ago, they predate the web era and systems
have evolved a lot since to deal with high traffic. I remember by then I
used to be impressed when seeing a server sustain 50 connections per second.
Now when someone tells me he's limited to 100000 connections per second on
a single server, I ask how they bound their interrupts to see if there's
some headroom to go further.

So you need to expect that only researchers and maybe TCP stack developers
will find your work useful these days, server admins can hardly use this
anymore. However it is very possible that some TCP stacks have taken benefit
of your work to reach the level of performance they achieve right now, I
don't know. Thus I think that Daniel's work completes quite well what you've
done in that it directly addresses people's concerns without requiring the
scientific background.

Regards,
Willy


From nobody Wed Aug 17 08:08:26 2016
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/16/2016 11:42 PM, Mark Nottingham wrote:
> Joe,
>
> I've pinged some ADs to weigh in on this.
>
> In the meantime -
>
>> On 17 Aug 2016, at 3:23 PM, Joe Touch <touch@isi.edu> wrote:
>>
>> Plagiarism requires only that the material was published elsewhere
>> before. Intent has no bearing.
> So to paraphrase, you're accusing Daniel of failure to cite a relevant source out of ignorance in the first case... 
I'm being generous in assuming ignorance, but yes.

>
>> In addition, I informed the author - and both lists - about this over 5
>> months ago. You might claim that the first two versions were issued out
>> of ignorance, but you cannot claim that of the update.
> ... and then continuing to do so after being notified.
>
> Correct?
Yes.

> Looking over <http://www.isi.edu/touch/pubs/ton97.pdf> and <http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/>, I'm having trouble identifying passages that are plagiarised; can you show us?
It's the content - turning off Nagle, slow-start restart after idle, etc.

> In particular, the latter paper proposes TIME-WAIT negotiation using a TCP option, sending a RST from clients, and a HTTP extension,

It also outlines the whole problem for the first time and explains the
choice and impact of lowering TW timeout.

>  ...
>
> Also, how do you reconcile that with the statement in your original e-mail that the draft repeats your material "sometimes correctly, sometimes in error"? Have you considered that what you consider to be an error in his plagiarism might in fact be a different idea (no matter whose is ultimately correct)?
That might be a reasonable conclusion if there were discussion
explaining why the "different idea" was correct in comparison.

But that's missing. Saying to lower the timer for TW without discussing
the potential impact isn't a new idea. It's an incomplete discussion of
an old idea already discussed in detail elsewhere.

>
>>> If that's the case, I'd observe that the IETF isn't an academic publisher, and acknowledging all prior work in an area is neither practical, nor required, nor current practice.
>> Plagiarism isn't an issue limited to academic environments. Publication
>> of a document on the web is still publication.
> Sure. It also isn't a legal issue in this form (unless you're asserting copyright?). Effectively, it's a cultural norm. Again, I will point out that in the culture of the IETF, we historically have not cited the complete provenance of every idea, both because it's impractical and because it doesn't benefit the reader. 
Although that's true in the smallest cases, the IETF does have two
concepts that support this norm: an author list and a set of references.

Can you explain how it helps the reader to not cite two documents that
are both squarely in the same area as this doc (interaction between HTTP
and TCP and the impact of running many small connections closed at the
client as for HTTP)?

> As far as I know, the IETF does not have a stated position about what you regard as PLAGIARISM. Hopefully we can get some clarity about that from the ADs, as well as some definitive evidence of what you're asserting.

You can if you want, but my primary point here is to have this work
corrected - and to stop the myth that "it doesn't matter" whether
*reasonable* citations are included.

Joe


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To: Willy Tarreau <w@1wt.eu>
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/16/2016 11:45 PM, Willy Tarreau wrote:
> Hi Joe,
>
> On Tue, Aug 16, 2016 at 10:23:01PM -0700, Joe Touch wrote:
>>> Other people in the HTTP *and* TCP communities have commented that such a
>>> document would be very useful, whether or not it's something "new that
>>> wasn't known 20 years ago". 
>> We don't need to issue new documents for people who don't read old ones.
> I've just checked the two documents you referenced. They seem to be very
> well detailed but they are *scientific* research. What Daniel created are
> configuration advises for people who need to configure their servers. Yes
> as you mentionned they look like man pages precisely because the purpose
> is to ensure they're easily understood by people who are just seeking some
> help to improve their configuration. You cannot expect a server admin to
> read scientific papers explaining some TCP models with some formulas to
> know what to do on their servers.
Actually, my docs are the tip of the iceberg, addressing the reasons why
some of these configuration recommendations make sense.

There are many other sites - and books - that already indicate how to
configure systems efficiently.

So if your argument is that a man page summary is needed, sure - but
again, is a new one needed? And why is this then needed as an RFC?

> Also, I don't know if there have been any update, but these documents use
> SunOS 4.1.3 running on a sparc 20 as a reference. While I used to love
> working on such systems 20 years ago, they predate the web era and systems
> have evolved a lot since to deal with high traffic. ...
Yes, and discussing those issues would be useful - but not in this
document either.

> ...
> So you need to expect that only researchers and maybe TCP stack developers
> will find your work useful these days, server admins can hardly use this
> anymore. However it is very possible that some TCP stacks have taken benefit
> of your work to reach the level of performance they achieve right now, I
> don't know. Thus I think that Daniel's work completes quite well what you've
> done in that it directly addresses people's concerns without requiring the
> scientific background.
Let me see if I get your complete argument:

    - the appropriate refs are 20 years old
    - server admins need a doc

What exactly do server admins need regarding Nagle (which is configured
inside the app already), socket sizing (configured inside the app), etc?

I.e., at the most this is a man page (specific to an OS). At the least,
this isn't useful at all.

Joe


From nobody Wed Aug 17 08:29:10 2016
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To: Joe Touch <touch@isi.edu>, Mark Nottingham <mnot@mnot.net>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Joe,

On 17/08/2016 16:08, Joe Touch wrote:
> On 8/16/2016 11:42 PM, Mark Nottingham wrote:
>>> On 17 Aug 2016, at 3:23 PM, Joe Touch <touch@isi.edu> wrote:
>>>
  [snip]
>>>> If that's the case, I'd observe that the IETF isn't an academic publisher, and acknowledging all prior work in an area is neither practical, nor required, nor current practice.
>>> Plagiarism isn't an issue limited to academic environments. Publication
>>> of a document on the web is still publication.
>> Sure. It also isn't a legal issue in this form (unless you're asserting copyright?). Effectively, it's a cultural norm. Again, I will point out that in the culture of the IETF, we historically have not cited the complete provenance of every idea, both because it's impractical and because it doesn't benefit the reader.
> Although that's true in the smallest cases, the IETF does have two
> concepts that support this norm: an author list and a set of references.
>
> Can you explain how it helps the reader to not cite two documents that
> are both squarely in the same area as this doc (interaction between HTTP
> and TCP and the impact of running many small connections closed at the
> client as for HTTP)?
Instead of starting your discussion with words like "plagiarism", you 
could have just asked for information to be clarified and a 
citation/acknowledgement added? With your current introduction you 
pissed off lots of people.
>> As far as I know, the IETF does not have a stated position about what you regard as PLAGIARISM. Hopefully we can get some clarity about that from the ADs, as well as some definitive evidence of what you're asserting.
> You can if you want, but my primary point here is to have this work
> corrected - and to stop the myth that "it doesn't matter" whether
> *reasonable* citations are included.
Noted.


From nobody Wed Aug 17 08:30:06 2016
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Subject: [tcpm] Personal change in chairing
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Hi all,

I have to announce a personal change in the tcpm chair position:

Pasi Sarolahti is stepping down due to time restrictions. Pasi, thanks a lot 
for your work and engagement as chair! Your did a great job and we hope to 
still see you as often as possible as contributor in the working group!

Michael Tüxen is starting as a new tcpm chair. Micheal, great to have you on 
broad! We look forward to work with you!

Mirja, as responsible AD


From nobody Wed Aug 17 08:55:05 2016
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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> On Aug 17, 2016, at 8:28 AM, Alexey Melnikov <alexey.melnikov@isode.com> w=
rote:
>=20
> Joe,
>=20
>> On 17/08/2016 16:08, Joe Touch wrote:
>> ...
>> Can you explain how it helps the reader to not cite two documents that
>> are both squarely in the same area as this doc (interaction between HTTP
>> and TCP and the impact of running many small connections closed at the
>> client as for HTTP)?
> Instead of starting your discussion with words like "plagiarism", you coul=
d have just asked for information to be clarified and a citation/acknowledge=
ment added?
...

I did that 5 months ago, as noted in the attachments I included yesterday.

Joe=


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To: Alexey Melnikov <alexey.melnikov@isode.com>, Joe Touch <touch@isi.edu>, Mark Nottingham <mnot@mnot.net>
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From: Eliot Lear <lear@cisco.com>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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From: Eliot Lear <lear@cisco.com>
To: Alexey Melnikov <alexey.melnikov@isode.com>, Joe Touch <touch@isi.edu>,
 Mark Nottingham <mnot@mnot.net>
Cc: tcpm@ietf.org, HTTP Working Group <ietf-http-wg@w3.org>,
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
References: <0CC24FC1-37E1-4125-9627-05726A9D9406@mnot.net>
 <7fa95741-ac58-3183-1b92-238bd4b4dae6@isi.edu>
 <5CD67877-19E3-4E79-BBF2-3E270343A378@mnot.net>
 <2197232f-10d7-28cb-fcc9-05bd495e3c22@isi.edu>
 <87D56B16-B091-4E9C-9B88-D136ED4BC819@mnot.net>
 <b1fe8ceb-c48c-8d89-1026-1fbed8d31d62@isi.edu>
 <2f276fc8-45d9-50e3-08bf-d712bc94302d@isode.com>
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Perhaps we can agree that the reasonable course of action here is for
Joe to (re)-recommend a compact set of citations to the authors, perhaps
even in some easily consumable form to them (kramdown-2629 or XML)?

Eliot


On 8/17/16 5:28 PM, Alexey Melnikov wrote:
> Joe,
>
> On 17/08/2016 16:08, Joe Touch wrote:
>> On 8/16/2016 11:42 PM, Mark Nottingham wrote:
>>>> On 17 Aug 2016, at 3:23 PM, Joe Touch <touch@isi.edu> wrote:
>>>>
>  [snip]
>>>>> If that's the case, I'd observe that the IETF isn't an academic
>>>>> publisher, and acknowledging all prior work in an area is neither
>>>>> practical, nor required, nor current practice.
>>>> Plagiarism isn't an issue limited to academic environments.
>>>> Publication
>>>> of a document on the web is still publication.
>>> Sure. It also isn't a legal issue in this form (unless you're
>>> asserting copyright?). Effectively, it's a cultural norm. Again, I
>>> will point out that in the culture of the IETF, we historically have
>>> not cited the complete provenance of every idea, both because it's
>>> impractical and because it doesn't benefit the reader.
>> Although that's true in the smallest cases, the IETF does have two
>> concepts that support this norm: an author list and a set of reference=
s.
>>
>> Can you explain how it helps the reader to not cite two documents that=

>> are both squarely in the same area as this doc (interaction between HT=
TP
>> and TCP and the impact of running many small connections closed at the=

>> client as for HTTP)?
> Instead of starting your discussion with words like "plagiarism", you
> could have just asked for information to be clarified and a
> citation/acknowledgement added? With your current introduction you
> pissed off lots of people.
>>> As far as I know, the IETF does not have a stated position about
>>> what you regard as PLAGIARISM. Hopefully we can get some clarity
>>> about that from the ADs, as well as some definitive evidence of what
>>> you're asserting.
>> You can if you want, but my primary point here is to have this work
>> corrected - and to stop the myth that "it doesn't matter" whether
>> *reasonable* citations are included.
> Noted.
>
>
>



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To: Eliot Lear <lear@cisco.com>, Alexey Melnikov <alexey.melnikov@isode.com>,  Joe Touch <touch@isi.edu>, Mark Nottingham <mnot@mnot.net>
References: <0CC24FC1-37E1-4125-9627-05726A9D9406@mnot.net> <7fa95741-ac58-3183-1b92-238bd4b4dae6@isi.edu> <5CD67877-19E3-4E79-BBF2-3E270343A378@mnot.net> <2197232f-10d7-28cb-fcc9-05bd495e3c22@isi.edu> <87D56B16-B091-4E9C-9B88-D136ED4BC819@mnot.net> <b1fe8ceb-c48c-8d89-1026-1fbed8d31d62@isi.edu> <2f276fc8-45d9-50e3-08bf-d712bc94302d@isode.com> <ebb0ae0e-8590-b02e-b8c8-1caa4f1435a0@cisco.com>
From: Tim Wicinski <tjw.ietf@gmail.com>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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That sounds like a fine idea.  I'll be glad to go through those.

tim

On 8/17/16 12:26 PM, Eliot Lear wrote:
> Perhaps we can agree that the reasonable course of action here is for
> Joe to (re)-recommend a compact set of citations to the authors, perhaps
> even in some easily consumable form to them (kramdown-2629 or XML)?
>
> Eliot
>
>
> On 8/17/16 5:28 PM, Alexey Melnikov wrote:
>> Joe,
>>
>> On 17/08/2016 16:08, Joe Touch wrote:
>>> On 8/16/2016 11:42 PM, Mark Nottingham wrote:
>>>>> On 17 Aug 2016, at 3:23 PM, Joe Touch <touch@isi.edu> wrote:
>>>>>
>>  [snip]
>>>>>> If that's the case, I'd observe that the IETF isn't an academic
>>>>>> publisher, and acknowledging all prior work in an area is neither
>>>>>> practical, nor required, nor current practice.
>>>>> Plagiarism isn't an issue limited to academic environments.
>>>>> Publication
>>>>> of a document on the web is still publication.
>>>> Sure. It also isn't a legal issue in this form (unless you're
>>>> asserting copyright?). Effectively, it's a cultural norm. Again, I
>>>> will point out that in the culture of the IETF, we historically have
>>>> not cited the complete provenance of every idea, both because it's
>>>> impractical and because it doesn't benefit the reader.
>>> Although that's true in the smallest cases, the IETF does have two
>>> concepts that support this norm: an author list and a set of references.
>>>
>>> Can you explain how it helps the reader to not cite two documents that
>>> are both squarely in the same area as this doc (interaction between HTTP
>>> and TCP and the impact of running many small connections closed at the
>>> client as for HTTP)?
>> Instead of starting your discussion with words like "plagiarism", you
>> could have just asked for information to be clarified and a
>> citation/acknowledgement added? With your current introduction you
>> pissed off lots of people.
>>>> As far as I know, the IETF does not have a stated position about
>>>> what you regard as PLAGIARISM. Hopefully we can get some clarity
>>>> about that from the ADs, as well as some definitive evidence of what
>>>> you're asserting.
>>> You can if you want, but my primary point here is to have this work
>>> corrected - and to stop the myth that "it doesn't matter" whether
>>> *reasonable* citations are included.
>> Noted.
>>
>>
>>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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All,

It sounds like the job of an author or a search engine (or both).

I've given you a place to start.

Joe


On 8/17/2016 10:00 AM, Tim Wicinski wrote:
>
> That sounds like a fine idea.  I'll be glad to go through those.
>
> tim
>
> On 8/17/16 12:26 PM, Eliot Lear wrote:
>> Perhaps we can agree that the reasonable course of action here is for
>> Joe to (re)-recommend a compact set of citations to the authors, perhaps
>> even in some easily consumable form to them (kramdown-2629 or XML)?
>>
>> Eliot
>>
>>
>> On 8/17/16 5:28 PM, Alexey Melnikov wrote:
>>> Joe,
>>>
>>> On 17/08/2016 16:08, Joe Touch wrote:
>>>> On 8/16/2016 11:42 PM, Mark Nottingham wrote:
>>>>>> On 17 Aug 2016, at 3:23 PM, Joe Touch <touch@isi.edu> wrote:
>>>>>>
>>>  [snip]
>>>>>>> If that's the case, I'd observe that the IETF isn't an academic
>>>>>>> publisher, and acknowledging all prior work in an area is neither
>>>>>>> practical, nor required, nor current practice.
>>>>>> Plagiarism isn't an issue limited to academic environments.
>>>>>> Publication
>>>>>> of a document on the web is still publication.
>>>>> Sure. It also isn't a legal issue in this form (unless you're
>>>>> asserting copyright?). Effectively, it's a cultural norm. Again, I
>>>>> will point out that in the culture of the IETF, we historically have
>>>>> not cited the complete provenance of every idea, both because it's
>>>>> impractical and because it doesn't benefit the reader.
>>>> Although that's true in the smallest cases, the IETF does have two
>>>> concepts that support this norm: an author list and a set of
>>>> references.
>>>>
>>>> Can you explain how it helps the reader to not cite two documents that
>>>> are both squarely in the same area as this doc (interaction between
>>>> HTTP
>>>> and TCP and the impact of running many small connections closed at the
>>>> client as for HTTP)?
>>> Instead of starting your discussion with words like "plagiarism", you
>>> could have just asked for information to be clarified and a
>>> citation/acknowledgement added? With your current introduction you
>>> pissed off lots of people.
>>>>> As far as I know, the IETF does not have a stated position about
>>>>> what you regard as PLAGIARISM. Hopefully we can get some clarity
>>>>> about that from the ADs, as well as some definitive evidence of what
>>>>> you're asserting.
>>>> You can if you want, but my primary point here is to have this work
>>>> corrected - and to stop the myth that "it doesn't matter" whether
>>>> *reasonable* citations are included.
>>> Noted.
>>>
>>>
>>>
>>
>>
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>


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From: Willy Tarreau <w@1wt.eu>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On Wed, Aug 17, 2016 at 08:14:08AM -0700, Joe Touch wrote:
> There are many other sites - and books - that already indicate how to
> configure systems efficiently.
> 
> So if your argument is that a man page summary is needed, sure - but
> again, is a new one needed? And why is this then needed as an RFC?

The difference is that a man page is OS-specific while an RFC gets a
unique number and serves as a reference. It can be cited in new RFCs
to justify certain choices. It can receive errata. You would probably
say that the current document is very linux-centric for now but I
understood it as a beginning of something more generic where advices
are given based on principles before sysctls and that the resulting
sysctls are just examples of applications of the principles in the
document.

> > Also, I don't know if there have been any update, but these documents use
> > SunOS 4.1.3 running on a sparc 20 as a reference. While I used to love
> > working on such systems 20 years ago, they predate the web era and systems
> > have evolved a lot since to deal with high traffic. ...
> Yes, and discussing those issues would be useful - but not in this
> document either.

Why ? Lots of admins don't understand why the time_wait timeout remains
at 240 seconds on Solaris with people saying "if you want to be conservative
don't touch it but if you want to be modern simply shrink it to 30 seconds
or so". People need to understand why advices have changed over 3 decades.

> > So you need to expect that only researchers and maybe TCP stack developers
> > will find your work useful these days, server admins can hardly use this
> > anymore. However it is very possible that some TCP stacks have taken benefit
> > of your work to reach the level of performance they achieve right now, I
> > don't know. Thus I think that Daniel's work completes quite well what you've
> > done in that it directly addresses people's concerns without requiring the
> > scientific background.
> Let me see if I get your complete argument:
> 
>     - the appropriate refs are 20 years old
>     - server admins need a doc
> 
> What exactly do server admins need regarding Nagle (which is configured
> inside the app already), socket sizing (configured inside the app), etc?

Lots of things : 
  - time_wait tuning (which everybody gives different advices on, I've
    even seen firewall vendors recommend to shrink it to one second because
    it allowed their product to perform better in benchmarks)

  - TCP timestamps: what they provide, what are the risks (some people in
    banking environments refuse to enable them so that they cannot be used
    as an oracle to help in timing attacks).

  - window scaling : how much is needed.

  - socket sizing : contrary to what you write, there's a lot of tuning
    on the web where people set the default buffer sizes to 16MB without
    understanding the impacts when dealing with many sockets

  - SACK : why it's better. DSACK what it adds on top of SACK.

  - ECN : is it needed ? does it really work ? where does it cause issues ?

  - SYN cookies : benefits, risks

  - TCP reuse/recycling : benefits, risks

  - dealing with buffer bloat : tradeoffs between NIC-based acceleration
    and pacing

  - what are orphans and why you should care about them in HTTP close mode

  - TCP fastopen : how does it work, what type of workload is improved,
    what are the risks (ie: do not enable socket creation without cookie
    by default just because you find it reduces your server load)

  - whether to choose a short or a large SYN backlog depending on your
    workload (ie: do you prefer to process everything even if the dequeuing
    is expensive or to drop early in order to recover fast).

... and probably many other that don't immediately come to my mind. None
of these ones was a real issue 20 years ago. All of them became issues for
many web server admins who just copy-paste random settings from various
blogs found on the net who just copy the same stupidities over and over
resulting in the same trouble being caused to each of their reader.

> I.e., at the most this is a man page (specific to an OS). At the least,
> this isn't useful at all.

As you can see above, nothing I cited was OS-specific but only workload
specific. That's why I think that an informational RFC is much more suited
to this than an OS-specific man page. The OS man page may rely on the RFC
to propose various tuning profiles for different workloads however.

Regards,
Willy


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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/17/2016 11:08 AM, Willy Tarreau wrote:
> On Wed, Aug 17, 2016 at 08:14:08AM -0700, Joe Touch wrote:
>> There are many other sites - and books - that already indicate how to
>> configure systems efficiently.
>>
>> So if your argument is that a man page summary is needed, sure - but
>> again, is a new one needed? And why is this then needed as an RFC?
> The difference is that a man page is OS-specific while an RFC gets a
> unique number and serves as a reference. 
That means the OS-specific stuff here needs to be removed...

> It can be cited in new RFCs
> to justify certain choices. 
Hmm. Like the refs I gave could be cited in this doc to justify *its*
choices? :-)
...
>>> Also, I don't know if there have been any update, but these documents use
>>> SunOS 4.1.3 running on a sparc 20 as a reference. While I used to love
>>> working on such systems 20 years ago, they predate the web era and systems
>>> have evolved a lot since to deal with high traffic. ...
>> Yes, and discussing those issues would be useful - but not in this
>> document either.
> Why ? Lots of admins don't understand why the time_wait timeout remains
> at 240 seconds on Solaris with people saying "if you want to be conservative
> don't touch it but if you want to be modern simply shrink it to 30 seconds
> or so". People need to understand why advices have changed over 3 decades.

The advice hasn't really changed - the advice was given in the 99 ref,
which includes some cases where it can still be appropriate to decrease
that timer.

Again, though - the 99 ref explains the implications of both decisions -
why it's 240 seconds and what happens when it's lowered.

>
>>> So you need to expect that only researchers and maybe TCP stack developers
>>> will find your work useful these days, server admins can hardly use this
>>> anymore. However it is very possible that some TCP stacks have taken benefit
>>> of your work to reach the level of performance they achieve right now, I
>>> don't know. Thus I think that Daniel's work completes quite well what you've
>>> done in that it directly addresses people's concerns without requiring the
>>> scientific background.
>> Let me see if I get your complete argument:
>>
>>     - the appropriate refs are 20 years old
>>     - server admins need a doc
>>
>> What exactly do server admins need regarding Nagle (which is configured
>> inside the app already), socket sizing (configured inside the app), etc?
> Lots of things : 
>   - time_wait tuning (which everybody gives different advices on, I've
>     even seen firewall vendors recommend to shrink it to one second because
>     it allowed their product to perform better in benchmarks)
Again, the 99 ref gives that detail.
>   - TCP timestamps: what they provide, what are the risks (some people in
>     banking environments refuse to enable them so that they cannot be used
>     as an oracle to help in timing attacks).
That's already covered in the security considerations of RFC 7323. How
is HTTP different, if at all, from any other app?
>   - window scaling : how much is needed.
Same issue here, same ref - how is HTTP different?

>   - socket sizing : contrary to what you write, there's a lot of tuning
>     on the web where people set the default buffer sizes to 16MB without
>     understanding the impacts when dealing with many sockets
There's a whole book that encompasses that and some related issues:
http://www.onlamp.com/pub/a/onlamp/2005/11/17/tcp_tuning.html

Some advice is also given in Sec 6.3.3 of this:
J. Sterbenz, J. Touch, /High Speed Networking: A Systematic Approach to
High-Bandwidth Low-Latency Communication/
<http://www.wiley.com/WileyCDA/WileyTitle/productCd-0471330361.html>,
Wiley, May 2001.

>   - SACK : why it's better. DSACK what it adds on top of SACK.
That's in the SACK docs, which aren't cited. Again, how is HTTP
different from any app?
>   - ECN : is it needed ? does it really work ? where does it cause issues ?
That's in the ECN docs, which aren't cited. Again, how is HTTP different
from any app?
>   - SYN cookies : benefits, risks
That's in the RFC 4987, which at least IS cited. Again, how is HTTP
different from any app?

>   - TCP reuse/recycling : benefits, risks
Not sure what you mean here. There are a lot of docs on the issues with
persistent-HTTP vs per-connection HTTP.

>   - dealing with buffer bloat : tradeoffs between NIC-based acceleration
>     and pacing
Bufferbloat typically involves large *uplink* transfers and how they
interact with other uplink connections. Neither TCP nor HTTP is involved
in this really.

>   - what are orphans and why you should care about them in HTTP close mode
Orphaned TCP connections or orphaned HTTP processes?

>   - TCP fastopen : how does it work, what type of workload is improved,
>     what are the risks (ie: do not enable socket creation without cookie
>     by default just because you find it reduces your server load)

Another doc that exists.

>   - whether to choose a short or a large SYN backlog depending on your
>     workload (ie: do you prefer to process everything even if the dequeuing
>     is expensive or to drop early in order to recover fast).
>
> ... and probably many other that don't immediately come to my mind. None
> of these ones was a real issue 20 years ago.

See above. Many were known around that time, but weren't documented in
detail (it took a while for a proper ref to SYN cookies, and the book I
wrote with Sterbenz came about because we'd seen wheels being
rediscovered for 15 years).

>  All of them became issues for
> many web server admins who just copy-paste random settings from various
> blogs found on the net who just copy the same stupidities over and over
> resulting in the same trouble being caused to each of their reader.

This doc is all over the place.

If you want a doc to advise web admins, do so. But most of the items
above aren't under admin control; they're buried in app and OS
implementations, and most have already evolved do to the right thing.

I agree that a summary of a *focused set* of these might be useful *as a
review* (note that review discussions usually include lots of refs). The
key question is "what is the focus":

    - HTTP/TCP interactions
    - server administration advice
    - ?

IMO, RFCs should focus on issues specific to the protocols and their
deployment - not general knowledge that exists in courses and textbooks.

>> I.e., at the most this is a man page (specific to an OS). At the least,
>> this isn't useful at all.
> As you can see above, nothing I cited was OS-specific but only workload
> specific. That's why I think that an informational RFC is much more suited
> to this than an OS-specific man page. The OS man page may rely on the RFC
> to propose various tuning profiles for different workloads however.

You have a good point that this is general info, but OS issues are not
in the scope of the IETF and there are courses and books that already
provide good advice on efficiently running web (and other) servers.

Joe


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    <br>
    <div class="moz-cite-prefix">On 8/17/2016 11:08 AM, Willy Tarreau
      wrote:<br>
    </div>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">On Wed, Aug 17, 2016 at 08:14:08AM -0700, Joe Touch wrote:
</pre>
      <blockquote type="cite">
        <pre wrap="">There are many other sites - and books - that already indicate how to
configure systems efficiently.

So if your argument is that a man page summary is needed, sure - but
again, is a new one needed? And why is this then needed as an RFC?
</pre>
      </blockquote>
      <pre wrap="">
The difference is that a man page is OS-specific while an RFC gets a
unique number and serves as a reference. </pre>
    </blockquote>
    That means the OS-specific stuff here needs to be removed...<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">It can be cited in new RFCs
to justify certain choices. </pre>
    </blockquote>
    Hmm. Like the refs I gave could be cited in this doc to justify
    *its* choices? :-)<br>
    ...<br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">
</pre>
      <blockquote type="cite">
        <blockquote type="cite">
          <pre wrap="">Also, I don't know if there have been any update, but these documents use
SunOS 4.1.3 running on a sparc 20 as a reference. While I used to love
working on such systems 20 years ago, they predate the web era and systems
have evolved a lot since to deal with high traffic. ...
</pre>
        </blockquote>
        <pre wrap="">Yes, and discussing those issues would be useful - but not in this
document either.
</pre>
      </blockquote>
      <pre wrap="">
Why ? Lots of admins don't understand why the time_wait timeout remains
at 240 seconds on Solaris with people saying "if you want to be conservative
don't touch it but if you want to be modern simply shrink it to 30 seconds
or so". People need to understand why advices have changed over 3 decades.</pre>
    </blockquote>
    <br>
    The advice hasn't really changed - the advice was given in the 99
    ref, which includes some cases where it can still be appropriate to
    decrease that timer.<br>
    <br>
    Again, though - the 99 ref explains the implications of both
    decisions - why it's 240 seconds and what happens when it's lowered.<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite"><br>
      <blockquote type="cite">
        <blockquote type="cite">
          <pre wrap="">So you need to expect that only researchers and maybe TCP stack developers
will find your work useful these days, server admins can hardly use this
anymore. However it is very possible that some TCP stacks have taken benefit
of your work to reach the level of performance they achieve right now, I
don't know. Thus I think that Daniel's work completes quite well what you've
done in that it directly addresses people's concerns without requiring the
scientific background.
</pre>
        </blockquote>
        <pre wrap="">Let me see if I get your complete argument:

    - the appropriate refs are 20 years old
    - server admins need a doc

What exactly do server admins need regarding Nagle (which is configured
inside the app already), socket sizing (configured inside the app), etc?
</pre>
      </blockquote>
      <pre wrap="">
Lots of things : 
  - time_wait tuning (which everybody gives different advices on, I've
    even seen firewall vendors recommend to shrink it to one second because
    it allowed their product to perform better in benchmarks)</pre>
    </blockquote>
    Again, the 99 ref gives that detail.<br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - TCP timestamps: what they provide, what are the risks (some people in
    banking environments refuse to enable them so that they cannot be used
    as an oracle to help in timing attacks).</pre>
    </blockquote>
    That's already covered in the security considerations of RFC 7323.
    How is HTTP different, if at all, from any other app?<br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - window scaling : how much is needed.</pre>
    </blockquote>
    Same issue here, same ref - how is HTTP different?<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - socket sizing : contrary to what you write, there's a lot of tuning
    on the web where people set the default buffer sizes to 16MB without
    understanding the impacts when dealing with many sockets</pre>
    </blockquote>
    There's a whole book that encompasses that and some related issues:<br>
    <a class="moz-txt-link-freetext" href="http://www.onlamp.com/pub/a/onlamp/2005/11/17/tcp_tuning.html">http://www.onlamp.com/pub/a/onlamp/2005/11/17/tcp_tuning.html</a><br>
    <br>
    Some advice is also given in Sec 6.3.3 of this:<br>
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mso-ansi-language:EN-US;mso-fareast-language:EN-US;mso-bidi-language:AR-SA;
      mso-no-proof:yes">J. Sterbenz, J. Touch, </span><span
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    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - SACK : why it's better. DSACK what it adds on top of SACK.</pre>
    </blockquote>
    That's in the SACK docs, which aren't cited. Again, how is HTTP
    different from any app?<br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - ECN : is it needed ? does it really work ? where does it cause issues ?</pre>
    </blockquote>
    That's in the ECN docs, which aren't cited. Again, how is HTTP
    different from any app?
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - SYN cookies : benefits, risks</pre>
    </blockquote>
    That's in the RFC 4987, which at least IS cited. Again, how is HTTP
    different from any app?<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - TCP reuse/recycling : benefits, risks</pre>
    </blockquote>
    Not sure what you mean here. There are a lot of docs on the issues
    with persistent-HTTP vs per-connection HTTP.<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - dealing with buffer bloat : tradeoffs between NIC-based acceleration
    and pacing</pre>
    </blockquote>
    Bufferbloat typically involves large *uplink* transfers and how they
    interact with other uplink connections. Neither TCP nor HTTP is
    involved in this really.<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - what are orphans and why you should care about them in HTTP close mode</pre>
    </blockquote>
    Orphaned TCP connections or orphaned HTTP processes?<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - TCP fastopen : how does it work, what type of workload is improved,
    what are the risks (ie: do not enable socket creation without cookie
    by default just because you find it reduces your server load)</pre>
    </blockquote>
    <br>
    Another doc that exists.<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">  - whether to choose a short or a large SYN backlog depending on your
    workload (ie: do you prefer to process everything even if the dequeuing
    is expensive or to drop early in order to recover fast).

... and probably many other that don't immediately come to my mind. None
of these ones was a real issue 20 years ago.</pre>
    </blockquote>
    <br>
    See above. Many were known around that time, but weren't documented
    in detail (it took a while for a proper ref to SYN cookies, and the
    book I wrote with Sterbenz came about because we'd seen wheels being
    rediscovered for 15 years).<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap=""> All of them became issues for
many web server admins who just copy-paste random settings from various
blogs found on the net who just copy the same stupidities over and over
resulting in the same trouble being caused to each of their reader.</pre>
    </blockquote>
    <br>
    This doc is all over the place.<br>
    <br>
    If you want a doc to advise web admins, do so. But most of the items
    above aren't under admin control; they're buried in app and OS
    implementations, and most have already evolved do to the right
    thing.<br>
    <br>
    I agree that a summary of a *focused set* of these might be useful
    *as a review* (note that review discussions usually include lots of
    refs). The key question is "what is the focus":<br>
    <br>
     - HTTP/TCP interactions<br>
     - server administration advice<br>
     - ?<br>
    <br>
    IMO, RFCs should focus on issues specific to the protocols and their
    deployment - not general knowledge that exists in courses and
    textbooks.<br>
    <br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <blockquote type="cite">
        <pre wrap="">I.e., at the most this is a man page (specific to an OS). At the least,
this isn't useful at all.
</pre>
      </blockquote>
      <pre wrap="">
As you can see above, nothing I cited was OS-specific but only workload
specific. That's why I think that an informational RFC is much more suited
to this than an OS-specific man page. The OS man page may rely on the RFC
to propose various tuning profiles for different workloads however.</pre>
    </blockquote>
    <br>
    You have a good point that this is general info, but OS issues are
    not in the scope of the IETF and there are courses and books that
    already provide good advice on efficiently running web (and other)
    servers.<br>
    <br>
    Joe<br>
    <blockquote cite="mid:20160817180802.GA16773@1wt.eu" type="cite">
      <pre wrap="">
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------99199A629C86782671B44500--


From nobody Wed Aug 17 11:44:41 2016
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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: Mark Nottingham <mnot@mnot.net>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] TCP Tuning for HTTP - update
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Cc: Patrick McManus <pmcmanus@mozilla.com>, Daniel Stenberg <daniel@haxx.se>
Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Speaking as individual contributor...

* I support an *informational* document in httpbis under the constraint tha=
t the document is finally also last called in tcpm. However, I have some do=
ubts that this document has to be a BCP. Apart from the formal aspects that=
 a BCP would have to be carefully reviewed regarding consistency with TCPM =
documents, I also believe that some of the content can easily get outdated =
as TCP and operating systems evolve, so "current" seems relative. See below=
 for one example for how fast TCP and its implementations can evolve.

* There is a related discussion on a document currently discussed in LWIP d=
raft-gomez-core-tcp-constrained-node-networks, which focuses quite on the o=
pposite type of TCP deployment. I believe the IETF should handle both docum=
ents consistently (but this is a topic for our ADs). In case of LWIP, the c=
urrent AD guidance is to have a brief readouts in TCPM from time to time (m=
aybe every second meeting or so). Given that a lot if TCP implementers foll=
ow TCPM, I believe such a readout would be a good opportunity to get feedba=
ck.

* I believe TCP is not only implemented on a single operating system only ;=
-) I'd actually be interested in the roadmap for adding a more comprehensiv=
e coverage of TCP stacks, if stack-specific recommendations should be inclu=
ded e.g. in the appendix. I do not object to OS-specific examples in genera=
l. To me this seems doable in an informational document.

* I suggest to reference RFC 7414 for a more complete overview of TCP featu=
res=20

* There has recently been quite a bit of TCPM discussion on RACK, which is =
still under design but if the document wants to cover TLP it may be reasona=
ble to consider RACK. Some context can be found e.g. in https://www.ietf.or=
g/proceedings/96/slides/slides-96-tcpm-3.pdf and https://www.ietf.org/proce=
edings/96/slides/slides-96-tcpm-4.pdf . Others may know more than me.

* I offer to more review the technical content once the document has mature=
d a bit; right now I still see quite a bit of work to be completed

Michael




-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Mark Nottingham
Sent: Wednesday, August 17, 2016 4:03 AM
To: tcpm@ietf.org
Cc: Patrick McManus; Daniel Stenberg
Subject: [tcpm] TCP Tuning for HTTP - update

Hi TCPM,

Just a quick note; Daniel and Tim have made an update to the TCP Tuning for=
 HTTP draft:
  https://tools.ietf.org/html/draft-stenberg-httpbis-tcp

We've had a Call for Adoption open for this draft for a while, and will lik=
ely adopt it soon. However, we'd like to get feedback from this community f=
irst -- both about the latest version of the input document, and to see if =
there's interest in helping out.

You can give feedback on the HTTP WG mailing list <https://lists.w3.org/Arc=
hives/Public/ietf-http-wg/>, or  by responding to this e-mail (Please leave=
 the CC line; Patrick and I will try to summarise the feedback to the WG).

Cheers,

--
Mark Nottingham   https://www.mnot.net/




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


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From: Willy Tarreau <w@1wt.eu>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On Wed, Aug 17, 2016 at 11:31:33AM -0700, Joe Touch wrote:
> > It can be cited in new RFCs
> > to justify certain choices. 
> Hmm. Like the refs I gave could be cited in this doc to justify *its*
> choices? :-)

I think it would be nice that this is cited, but to be clear on one
point, I've never heard about your papers before you advertised them
here in this thread, and yet I've been dealing with timewait issues
for 15 years like many people facing moderate to large web sites
nowadays. It just happens that you probably identified these issues
very early at low connection rates, but today anyone dealing with
more than 500-1000 connections per second on a server or worse on
a gateway quickly discovers that he has to make a choice.

> >> Yes, and discussing those issues would be useful - but not in this
> >> document either.
> > Why ? Lots of admins don't understand why the time_wait timeout remains
> > at 240 seconds on Solaris with people saying "if you want to be conservative
> > don't touch it but if you want to be modern simply shrink it to 30 seconds
> > or so". People need to understand why advices have changed over 3 decades.
> 
> The advice hasn't really changed - the advice was given in the 99 ref,
> which includes some cases where it can still be appropriate to decrease
> that timer.

Most people see it the other way around : they see no valid case to *increase*
it beyond a few seconds, because for them the default value should be extremely
low (ie this firewall vendor several years ago trying to insist on one second).
Yes that's really sad but that's reality. And you can tell them to read 6191
they won't care.

> >   - TCP timestamps: what they provide, what are the risks (some people in
> >     banking environments refuse to enable them so that they cannot be used
> >     as an oracle to help in timing attacks).
> That's already covered in the security considerations of RFC 7323. How
> is HTTP different, if at all, from any other app?

HTTP is special in that it is fairly common to have to deal with tens of
thousands of connections per second between one client and one server when
you are on the server side, because you place a number of gateways (also
called reverse-proxies) which combine all of the possible issues you can
think of at a single place. Timestamps are one way to improve fast connection
recycling between the client and the server without having to cheat on
timewaits. But since they consume 12 bytes per packet, it's often advised
to disable them in benchmarks to get the highest throughput...

> >   - window scaling : how much is needed.
> Same issue here, same ref - how is HTTP different?

Same as above.

> >   - socket sizing : contrary to what you write, there's a lot of tuning
> >     on the web where people set the default buffer sizes to 16MB without
> >     understanding the impacts when dealing with many sockets
> There's a whole book that encompasses that and some related issues:
> http://www.onlamp.com/pub/a/onlamp/2005/11/17/tcp_tuning.html

Looks fine, could be added to the list of references.

> Some advice is also given in Sec 6.3.3 of this:
> J. Sterbenz, J. Touch, /High Speed Networking: A Systematic Approach to
> High-Bandwidth Low-Latency Communication/
> <http://www.wiley.com/WileyCDA/WileyTitle/productCd-0471330361.html>,
> Wiley, May 2001.
> 
> >   - SACK : why it's better. DSACK what it adds on top of SACK.
> That's in the SACK docs, which aren't cited. Again, how is HTTP
> different from any app?
> >   - ECN : is it needed ? does it really work ? where does it cause issues ?
> That's in the ECN docs, which aren't cited. Again, how is HTTP different
> from any app?
> >   - SYN cookies : benefits, risks
> That's in the RFC 4987, which at least IS cited. Again, how is HTTP
> different from any app?

So probably you're starting to see the benefit of having a single doc
to concentrate all this. You provided at least 3 different articles
to read and 2 or 3 different RFCs in addition to the original ones,
of course. A hosting provider whose web sites are down due to a lack
of tuning doesn't start to read many very long articles and even less
the most scientific ones, they need to find quick responses that they
can apply immediately (matter of minutes). So they launch google, they
type "web site dead, time-wait overflow" and they get plenty of
responses on stackoverflow and serverfault, many from people having
done the same in the past and repeating the same mistakes over and over.

A document validated by several people and giving links for further
reading can help improve this situation.

> >   - TCP reuse/recycling : benefits, risks
> Not sure what you mean here. There are a lot of docs on the issues with
> persistent-HTTP vs per-connection HTTP.

Often on a gateway you cannot completely chose. You have a mix of both.

> >   - dealing with buffer bloat : tradeoffs between NIC-based acceleration
> >     and pacing
> Bufferbloat typically involves large *uplink* transfers and how they
> interact with other uplink connections. Neither TCP nor HTTP is involved
> in this really.

Maybe in the end Daniel is not the only one not to read all articles
published on the web :-)

     http://www.cringely.com/2012/03/25/linux-3-3-finally-a-little-good-news-for-bufferbloat/
     https://www.ietf.org/proceedings/86/slides/slides-86-iccrg-0.pdf
     https://lwn.net/Articles/564978/

In short, by sending 64kB segments to your NIC and counting on it to
cut them into small pieces for you and sending the resulting packets
very close together, you increase the risk of losses on many network
equipments which run with not-that-large buffers. It's easy to observe
even on some 10Gbps switches. When you switch mixes 10G and 40G, it's
horrible, really.

> >   - what are orphans and why you should care about them in HTTP close mode
> Orphaned TCP connections or orphaned HTTP processes?

TCP connections. The server sends a response, performs a close() on the
socket, the data remain in the kernel buffers for the time it takes to
deliver these data to the client and to get them ACKed. From this point
the socket is called orphaned on the system because it doesn't belong to
any process anymore. But immediately after this, a process which runs with
a limited connection count can accept a new connection, which will in turn
be fed with large chunks of data and closed. And this runs over and over,
eating a huge amount of socket memory despite a small limit imposed on the
server's connection concurrency. At some point the kernel doesn't have
enough socket buffers anymore (especially after admins have set them to
16MB as instructed on $random tuning guide) and starts to kill some orphaned
connections to get some memory back. The not-so-nice effect that the admin
cannot detect is that the client gets truncated responses. Only a small part
of the tail is missing but in the logs, it's said that everything was sent.
But sent by the process means sent to the system only.

> >   - TCP fastopen : how does it work, what type of workload is improved,
> >     what are the risks (ie: do not enable socket creation without cookie
> >     by default just because you find it reduces your server load)
> 
> Another doc that exists.
> 
> >   - whether to choose a short or a large SYN backlog depending on your
> >     workload (ie: do you prefer to process everything even if the dequeuing
> >     is expensive or to drop early in order to recover fast).
> >
> > ... and probably many other that don't immediately come to my mind. None
> > of these ones was a real issue 20 years ago.
> 
> See above. Many were known around that time, but weren't documented in
> detail (it took a while for a proper ref to SYN cookies, and the book I
> wrote with Sterbenz came about because we'd seen wheels being
> rediscovered for 15 years).

People rediscover wheels because it's hard to find simple and accurate
information on the net. Basically you have the choice :
  - either uneducated blog posts saying "how I saved my web site using 2
    sysctls"
  - or academic papers which are only understandable by scientific people
    having enough time

At least the first ones have the merit of being easy to test, and since
they appear to work they are viral.

> >  All of them became issues for
> > many web server admins who just copy-paste random settings from various
> > blogs found on the net who just copy the same stupidities over and over
> > resulting in the same trouble being caused to each of their reader.
> 
> This doc is all over the place.
> 
> If you want a doc to advise web admins, do so.

That's *exactly* what Daniel started to do when you told him he shouldn't
do it.

> But most of the items
> above aren't under admin control; they're buried in app and OS
> implementations, and most have already evolved do to the right thing.

That's not true anymore. There was a time where Solaris had one of the
most configurable TCP stack, you could do ndd /dev/tcp with a lot of
things. Nowadays all modern operating systems let you fiddle with a ton
of stuff. And even for the missing parts you can very easily find patches
all over the web because opensource has changed this considerably.

> I agree that a summary of a *focused set* of these might be useful *as a
> review* (note that review discussions usually include lots of refs).

The purpose as I understood it was precisely to gather knowledge from
people here operating such systems and adding some elements found from
various sources. I also expressed my interest in sharing such experience
which some will find valid and others not, which is perfect because it
means there's a choice to make depending on a use case.

> The key question is "what is the focus":
> 
>     - HTTP/TCP interactions
>     - server administration advice
>     - ?

Those two go hand-in-hand nowadays. You probably know that the difficulty
for HTTP implementors to find a properly tuned TCP stack is what makes them
consider UDP-based alternatives, so that they can totally control the
behaviour from user-space and provide quick updates for their protocol.
Want an example ?

    https://www.chromium.org/quic

If it was possible to always have a perfectly tuned TCP stack I'm pretty
sure we wouldn't even hear about such initiatives. And this is not something
new, I happened to discuss about this subject with some people at IETF83 in
2012 already. By then I thought "they have no idea what they're trying to
reinvent" and now I'm thinking "well, they have their reasons and they
might be right after all given all the resistance they may be facing on
the TCP side to get some default timers changed".

> IMO, RFCs should focus on issues specific to the protocols and their
> deployment - not general knowledge that exists in courses and textbooks.

Courses and textbooks are totally outdated when they're not simply wrong.
I've first been tought that it was normal to fork a process after performing
an accept(), resulting in my very first proxy working this way. Then I've
been taught that it was mandatory to send an empty ACK after a SYN-ACK to
validate a connection (which is totally wrong otherwise it would not allow
this ACK to be lost). I've been taught that in order to accept an incoming
connection you had to have your socket in listen mode exclusively. This is
false as well, two clients can connect together during their connect()
phase, it even has some security implications that are often overlooked.
Like it or not, the HTTP protocol has brought TCP to an area it was not
initially intended for and I find it fantastic to see that this protocol
still scales so well. But we need to consider modern usages of this protocol
for the web, and not just academic research and e-mail.

> >> I.e., at the most this is a man page (specific to an OS). At the least,
> >> this isn't useful at all.
> > As you can see above, nothing I cited was OS-specific but only workload
> > specific. That's why I think that an informational RFC is much more suited
> > to this than an OS-specific man page. The OS man page may rely on the RFC
> > to propose various tuning profiles for different workloads however.
> 
> You have a good point that this is general info, but OS issues are not
> in the scope of the IETF and there are courses and books that already
> provide good advice on efficiently running web (and other) servers.

Well, you want to run a quick test ? Google "tcp tuning for http server".
Skip the first two responses which are Daniel's document, take the next one :

    https://gist.github.com/kgriffs/4027835

Read a bit... Not bad, could have been much worse. OK found : 5 seconds
time_wait timeout for conntrack. Dig a little bit more... suggests 2M
time_wait sockets. At 5s per socket, that's an expected 400k connections
per second limit. This with buffers that can be as large as 16MB on both
sides (one could ask why to put 16MB buffers on the Rx path for a web
server). So basically the guy probably thinks that there's a need to
fill up to 6.4 TB/s and despite this the max-orphans was not set according
to the time-wait value, so users of his doc will start to cause some data
truncation before they run into connection failures, thus only their visitors
will know there are problems.

And this is the highest ranked doc on google after Daniel's. And overall
it's really not bad, much better than what we easily find everywhere.

The fact that Daniel's doc found before is a good hope that this sad state
of affairs can reach an end. That's why I think we should encourage him to
continue and give him all the reference he needs to have an undiscutably
good reference doc.

Regards,
Willy


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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/17/2016 2:13 PM, Willy Tarreau wrote:
> On Wed, Aug 17, 2016 at 11:31:33AM -0700, Joe Touch wrote:
>>> It can be cited in new RFCs
>>> to justify certain choices. 
>> Hmm. Like the refs I gave could be cited in this doc to justify *its*
>> choices? :-)
> I think it would be nice that this is cited, but to be clear on one
> point, I've never heard about your papers before you advertised them
> here in this thread, 

A search engine on the terms "TCP HTTP interaction" would have popped
them up rather quickly.

> and yet I've been dealing with timewait issues
> for 15 years like many people facing moderate to large web sites
> nowadays. 

"timewait issues" and we're the 5th hit in Google.


> It just happens that you probably identified these issues
> very early at low connection rates, but today anyone dealing with
> more than 500-1000 connections per second on a server or worse on
> a gateway quickly discovers that he has to make a choice.

The issue was very common when the doc was written in 99, when even at
that time there were two issues - running out of the number space and
running out of kernel memory.

The number space issue of running out of ports was the basis of the IETF
port names doc in 2006
(https://tools.ietf.org/html/draft-touch-tcp-portnames-00) that became
the current proposal for a TCP "service number option" in 2013 (which
has been discussed at various IETFs in TCPM since then).

>>>> Yes, and discussing those issues would be useful - but not in this
>>>> document either.
>>> Why ? Lots of admins don't understand why the time_wait timeout remains
>>> at 240 seconds on Solaris with people saying "if you want to be conservative
>>> don't touch it but if you want to be modern simply shrink it to 30 seconds
>>> or so". People need to understand why advices have changed over 3 decades.
>> The advice hasn't really changed - the advice was given in the 99 ref,
>> which includes some cases where it can still be appropriate to decrease
>> that timer.
> Most people see it the other way around : they see no valid case to *increase*
> it beyond a few seconds, because for them the default value should be extremely
> low (ie this firewall vendor several years ago trying to insist on one second).
> Yes that's really sad but that's reality. And you can tell them to read 6191
> they won't care.

Most people's servers don't need to run fast enough to care (note that
nearly everyone runs some sort of web server on nearly every device,
whether for control or configuration). The only issue are high-volume
servers (the kind sysadmins deal with), and those people tend to already
know what the tradeoffs are and accept the risks.

>
>>>   - TCP timestamps: what they provide, what are the risks (some people in
>>>     banking environments refuse to enable them so that they cannot be used
>>>     as an oracle to help in timing attacks).
>> That's already covered in the security considerations of RFC 7323. How
>> is HTTP different, if at all, from any other app?
> HTTP is special in that it is fairly common to have to deal with tens of
> thousands of connections per second between one client and one server when
> you are on the server side, because you place a number of gateways (also
> called reverse-proxies) which combine all of the possible issues you can
> think of at a single place.

There are lots of services that have that many transactions - DNS
servers (even local ones), remote databases, etc.

The point is that HTTP doesn't make the problem different, so this isn't
an HTTP issue. It's a high rate server issue.


>  Timestamps are one way to improve fast connection
> recycling between the client and the server without having to cheat on
> timewaits. But since they consume 12 bytes per packet, it's often advised
> to disable them in benchmarks to get the highest throughput...
>
>>>   - window scaling : how much is needed.
>> Same issue here, same ref - how is HTTP different?
> Same as above.
>
>>>   - socket sizing : contrary to what you write, there's a lot of tuning
>>>     on the web where people set the default buffer sizes to 16MB without
>>>     understanding the impacts when dealing with many sockets
>> There's a whole book that encompasses that and some related issues:
>> http://www.onlamp.com/pub/a/onlamp/2005/11/17/tcp_tuning.html
> Looks fine, could be added to the list of references.
>
>> Some advice is also given in Sec 6.3.3 of this:
>> J. Sterbenz, J. Touch, /High Speed Networking: A Systematic Approach to
>> High-Bandwidth Low-Latency Communication/
>> <http://www.wiley.com/WileyCDA/WileyTitle/productCd-0471330361.html>,
>> Wiley, May 2001.
>>
>>>   - SACK : why it's better. DSACK what it adds on top of SACK.
>> That's in the SACK docs, which aren't cited. Again, how is HTTP
>> different from any app?
>>>   - ECN : is it needed ? does it really work ? where does it cause issues ?
>> That's in the ECN docs, which aren't cited. Again, how is HTTP different
>> from any app?
>>>   - SYN cookies : benefits, risks
>> That's in the RFC 4987, which at least IS cited. Again, how is HTTP
>> different from any app?
> So probably you're starting to see the benefit of having a single doc
> to concentrate all this.
The same reason it's useful to have this all in one place is the reason
we already do - there are books and courses on this.

>  You provided at least 3 different articles
> to read and 2 or 3 different RFCs in addition to the original ones,
> of course. A hosting provider whose web sites are down due to a lack
> of tuning doesn't start to read many very long articles and even less
> the most scientific ones, they need to find quick responses that they
> can apply immediately (matter of minutes). So they launch google, they
> type "web site dead, time-wait overflow" and they get plenty of
> responses on stackoverflow and serverfault, many from people having
> done the same in the past and repeating the same mistakes over and over.

These people don't read RFCs to fix problems. They take online courses
or read "how to" books - which do already exist in this space.

> A document validated by several people and giving links for further
> reading can help improve this situation.
Those are the books and courses I'm talking about already.

>
>>>   - TCP reuse/recycling : benefits, risks
>> Not sure what you mean here. There are a lot of docs on the issues with
>> persistent-HTTP vs per-connection HTTP.
> Often on a gateway you cannot completely chose. You have a mix of both.

Sure... but again, that's not in this doc either.

>
>>>   - dealing with buffer bloat : tradeoffs between NIC-based acceleration
>>>     and pacing
>> Bufferbloat typically involves large *uplink* transfers and how they
>> interact with other uplink connections. Neither TCP nor HTTP is involved
>> in this really.
> Maybe in the end Daniel is not the only one not to read all articles
> published on the web :-)
>
>      http://www.cringely.com/2012/03/25/linux-3-3-finally-a-little-good-news-for-bufferbloat/
>      https://www.ietf.org/proceedings/86/slides/slides-86-iccrg-0.pdf
>      https://lwn.net/Articles/564978/
>
> In short, by sending 64kB segments to your NIC and counting on it to
> cut them into small pieces for you and sending the resulting packets
> very close together, you increase the risk of losses on many network
> equipments which run with not-that-large buffers.

Bufferbloat describes what happens when the buffers are too large, not
too small.

The problem you're describing is the interaction between burstiness and
tail-drop, which is addressed by ECN.

>  It's easy to observe
> even on some 10Gbps switches. When you switch mixes 10G and 40G, it's
> horrible, really.
RFCs aren't typically used as context in switch design...


>
>>>   - what are orphans and why you should care about them in HTTP close mode
>> Orphaned TCP connections or orphaned HTTP processes?
> TCP connections. The server sends a response, performs a close() on the
> socket, the data remain in the kernel buffers for the time it takes to
> deliver these data to the client and to get them ACKed. From this point
> the socket is called orphaned on the system because it doesn't belong to
> any process anymore. But immediately after this, a process which runs with
> a limited connection count can accept a new connection, which will in turn
> be fed with large chunks of data and closed. And this runs over and over,
> eating a huge amount of socket memory despite a small limit imposed on the
> server's connection concurrency. At some point the kernel doesn't have
> enough socket buffers anymore (especially after admins have set them to
> 16MB as instructed on $random tuning guide) and starts to kill some orphaned
> connections to get some memory back. The not-so-nice effect that the admin
> cannot detect is that the client gets truncated responses. Only a small part
> of the tail is missing but in the logs, it's said that everything was sent.
> But sent by the process means sent to the system only.

That's an implementation issue of the OS (or a bug, depending on whether
you consider TCP a reliable transport or not, IMO).

>
>>>   - TCP fastopen : how does it work, what type of workload is improved,
>>>     what are the risks (ie: do not enable socket creation without cookie
>>>     by default just because you find it reduces your server load)
>> Another doc that exists.
>>
>>>   - whether to choose a short or a large SYN backlog depending on your
>>>     workload (ie: do you prefer to process everything even if the dequeuing
>>>     is expensive or to drop early in order to recover fast).
>>>
>>> ... and probably many other that don't immediately come to my mind. None
>>> of these ones was a real issue 20 years ago.
>> See above. Many were known around that time, but weren't documented in
>> detail (it took a while for a proper ref to SYN cookies, and the book I
>> wrote with Sterbenz came about because we'd seen wheels being
>> rediscovered for 15 years).
> People rediscover wheels because it's hard to find simple and accurate
> information on the net.
Nobody looks to RFCs to solve that problem...

>  Basically you have the choice :
>   - either uneducated blog posts saying "how I saved my web site using 2
>     sysctls"
>   - or academic papers which are only understandable by scientific people
>     having enough time

... that's what net FAQs are for, as well as courses and books.

> At least the first ones have the merit of being easy to test, and since
> they appear to work they are viral.
>
>>>  All of them became issues for
>>> many web server admins who just copy-paste random settings from various
>>> blogs found on the net who just copy the same stupidities over and over
>>> resulting in the same trouble being caused to each of their reader.
>> This doc is all over the place.
>>
>> If you want a doc to advise web admins, do so.
> That's *exactly* what Daniel started to do when you told him he shouldn't
> do it.

I didn't say a doc to advise web admins wasn't useful. I said it wasn't
an RFC.

It's a web FAQ, a book, etc.


>
>> But most of the items
>> above aren't under admin control; they're buried in app and OS
>> implementations, and most have already evolved do to the right thing.
> That's not true anymore. There was a time where Solaris had one of the
> most configurable TCP stack, you could do ndd /dev/tcp with a lot of
> things. Nowadays all modern operating systems let you fiddle with a ton
> of stuff. And even for the missing parts you can very easily find patches
> all over the web because opensource has changed this considerably.
There are no kernel configs to tell apps to open more than one
connection at a time. You don't need a kernel config to tell Firefox to
disable Nagle, use a reasonable socket size, etc - yes, there are OS
defaults, but most web servers and clients build in the correct
overrides already.

>
>> I agree that a summary of a *focused set* of these might be useful *as a
>> review* (note that review discussions usually include lots of refs).
> The purpose as I understood it was precisely to gather knowledge from
> people here operating such systems and adding some elements found from
> various sources. I also expressed my interest in sharing such experience
> which some will find valid and others not, which is perfect because it
> means there's a choice to make depending on a use case.
>
>> The key question is "what is the focus":
>>
>>     - HTTP/TCP interactions
>>     - server administration advice
>>     - ?
> Those two go hand-in-hand nowadays. You probably know that the difficulty
> for HTTP implementors to find a properly tuned TCP stack is what makes them
> consider UDP-based alternatives, 
They want something different for a variety of reasons - the same kind
of airtight logic by which TBL developed HTTP instead of using FTP (he
said that you'd only typically need one file from a location, so why
open 2 connections? now we're stuck trying to mux control and data
rather than having a proper solution that already existed at the time -
it took nearly a decade for HTTP servers to catch up to the performance
of FTP).

> so that they can totally control the
> behaviour from user-space and provide quick updates for their protocol.
> Want an example ?
>
>     https://www.chromium.org/quic

There are many thousands of monkeys typing everywhere - look at the
Linux source if you want even better examples.

> If it was possible to always have a perfectly tuned TCP stack I'm pretty
> sure we wouldn't even hear about such initiatives. And this is not something
> new, I happened to discuss about this subject with some people at IETF83 in
> 2012 already. By then I thought "they have no idea what they're trying to
> reinvent" and now I'm thinking "well, they have their reasons and they
> might be right after all given all the resistance they may be facing on
> the TCP side to get some default timers changed".
>
>> IMO, RFCs should focus on issues specific to the protocols and their
>> deployment - not general knowledge that exists in courses and textbooks.
> Courses and textbooks are totally outdated when they're not simply wrong.
I'm now confused.

You don't want sysadmins to read books or take courses because they're
outdated and thus wrong, but you want to issue an immutable RFC (which
will likely be outdated by the time it's issued)?
> I've first been tought that it was normal to fork a process after performing
> an accept(), resulting in my very first proxy working this way. Then I've
> been taught that it was mandatory to send an empty ACK after a SYN-ACK to
> validate a connection (which is totally wrong otherwise it would not allow
> this ACK to be lost). I've been taught that in order to accept an incoming
> connection you had to have your socket in listen mode exclusively. This is
> false as well, two clients can connect together during their connect()
> phase, it even has some security implications that are often overlooked.
I'm the first to admit there are bad courses, certainly.

> Like it or not, the HTTP protocol has brought TCP to an area it was not
> initially intended for 
HTTP makes mistakes that people blame on TCP (like HOL blocking), and
TCP is based on assumptions that are no longer true (not just for HTTP,
but for many other app protocols, e.g., the issue of burst after idle is
based on the outdated assumption that most transfers are roughly symmetric).

> and I find it fantastic to see that this protocol
> still scales so well. 
> But we need to consider modern usages of this protocol
> for the web, and not just academic research and e-mail.
You might consider that TCPM and TSVWG don't exist for just "academic
research and e-mail". What do you think we've been doing for the past 40
years?

>
>>>> I.e., at the most this is a man page (specific to an OS). At the least,
>>>> this isn't useful at all.
>>> As you can see above, nothing I cited was OS-specific but only workload
>>> specific. That's why I think that an informational RFC is much more suited
>>> to this than an OS-specific man page. The OS man page may rely on the RFC
>>> to propose various tuning profiles for different workloads however.
>> You have a good point that this is general info, but OS issues are not
>> in the scope of the IETF and there are courses and books that already
>> provide good advice on efficiently running web (and other) servers.
> Well, you want to run a quick test ? Google "tcp tuning for http server".
> Skip the first two responses which are Daniel's document, take the next one :
>
>     https://gist.github.com/kgriffs/4027835

Like I said about Linux... ;-)

> Read a bit... Not bad, could have been much worse. OK found : 5 seconds
> time_wait timeout for conntrack. Dig a little bit more... suggests 2M
> time_wait sockets. At 5s per socket, that's an expected 400k connections
> per second limit. This with buffers that can be as large as 16MB on both
> sides (one could ask why to put 16MB buffers on the Rx path for a web
> server). So basically the guy probably thinks that there's a need to
> fill up to 6.4 TB/s and despite this the max-orphans was not set according
> to the time-wait value, so users of his doc will start to cause some data
> truncation before they run into connection failures, thus only their visitors
> will know there are problems.
>
> And this is the highest ranked doc on google after Daniel's. And overall
> it's really not bad, much better than what we easily find everywhere.
>
> The fact that Daniel's doc found before is a good hope that this sad state
> of affairs can reach an end. That's why I think we should encourage him to
> continue and give him all the reference he needs to have an undiscutably
> good reference doc.

You might try some other terms in your searches. Like just "TCP tuning",
or "TCP HTTP interactions".

Joe


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From: "Adrien de Croy" <adrien@qbik.com>
To: "Joe Touch" <touch@isi.edu>, "Willy Tarreau" <w@1wt.eu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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------ Original Message ------
From: "Joe Touch" <touch@isi.edu>

>They want something different for a variety of reasons - the same kind
>of airtight logic by which TBL developed HTTP instead of using FTP (he
>said that you'd only typically need one file from a location, so why
>open 2 connections? now we're stuck trying to mux control and data
>rather than having a proper solution that already existed at the time -
>it took nearly a decade for HTTP servers to catch up to the performance
>of FTP).
>
Whilst I've been finding this discussion very informative and=20
interesting, I have to raise an objection on this point.

FTP was never going to be suitable for the web, and a very simple RTT=20
analysis shows that.

Apart from initial 3 way TCP handshake and close, which is the same for=20
both, with http you have a request and a response, whereas FTP requires=20
you to wait for the server welcome, log in, negotiate another port, set=20
up a data connection in addition to retrieving the file

So it's at minimum 5 round trips more.

Then try adding all the firewall issues due to transmitting data=20
connection endpoint information over the control connection and it's no=20
surprise FTP is not favoured for downloads.
So FTP was never going to be a "proper solution" for the web without a=20
complete re-architecture.

Adrien


>


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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This is a bit of a side track, but...

On 8/17/2016 3:51 PM, Adrien de Croy wrote:
>
>
> ------ Original Message ------
> From: "Joe Touch" <touch@isi.edu>
>
>> They want something different for a variety of reasons - the same kind
>> of airtight logic by which TBL developed HTTP instead of using FTP (he
>> said that you'd only typically need one file from a location, so why
>> open 2 connections? now we're stuck trying to mux control and data
>> rather than having a proper solution that already existed at the time -
>> it took nearly a decade for HTTP servers to catch up to the performance
>> of FTP).
>>
> Whilst I've been finding this discussion very informative and
> interesting, I have to raise an objection on this point.
>
> FTP was never going to be suitable for the web, and a very simple RTT
> analysis shows that.
>
> Apart from initial 3 way TCP handshake and close, which is the same
> for both, with http you have a request and a response, whereas FTP
> requires you to wait for the server welcome, log in, negotiate another
> port, set up a data connection in addition to retrieving the file

That's only the first time you go somewhere new. You don't need to close
both ports so quickly; the control channel can stay open and you thus
avoid HOL blocking between data and control (and thus the need to
chunk-and-mux within persistent HTTP), which increases other delays for
HTTP.

Neither protocol matches exactly what is really needed for a true
transaction-oriented protocol.

> ...
> Then try adding all the firewall issues due to transmitting data
> connection endpoint information over the control connection and it's
> no surprise FTP is not favoured for downloads.

FTP had a passive mode even back then that avoids this issue. It also
had suspend/resume, compression, and format conversion.

Joe


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From: "Adrien de Croy" <adrien@qbik.com>
To: "Joe Touch" <touch@isi.edu>, "Willy Tarreau" <w@1wt.eu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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------ Original Message ------
From: "Joe Touch" <touch@isi.edu>
To: "Adrien de Croy" <adrien@qbik.com>; "Willy Tarreau" <w@1wt.eu>
Cc: "Mark Nottingham" <mnot@mnot.net>; "tcpm@ietf.org" <tcpm@ietf.org>;=20
"HTTP Working Group" <ietf-http-wg@w3.org>; "Patrick McManus"=20
<pmcmanus@mozilla.com>; "Daniel Stenberg" <daniel@haxx.se>
Sent: 18/08/2016 11:01:57 AM
Subject: Re: [tcpm] TCP Tuning for HTTP - update

>This is a bit of a side track, but...
>
>On 8/17/2016 3:51 PM, Adrien de Croy wrote:
>>
>>
>>  ------ Original Message ------
>>  From: "Joe Touch" <touch@isi.edu>
>>
>>>  They want something different for a variety of reasons - the same=20
>>>kind
>>>  of airtight logic by which TBL developed HTTP instead of using FTP=20
>>>(he
>>>  said that you'd only typically need one file from a location, so why
>>>  open 2 connections? now we're stuck trying to mux control and data
>>>  rather than having a proper solution that already existed at the=20
>>>time -
>>>  it took nearly a decade for HTTP servers to catch up to the=20
>>>performance
>>>  of FTP).
>>>
>>  Whilst I've been finding this discussion very informative and
>>  interesting, I have to raise an objection on this point.
>>
>>  FTP was never going to be suitable for the web, and a very simple RTT
>>  analysis shows that.
>>
>>  Apart from initial 3 way TCP handshake and close, which is the same
>>  for both, with http you have a request and a response, whereas FTP
>>  requires you to wait for the server welcome, log in, negotiate=20
>>another
>>  port, set up a data connection in addition to retrieving the file
>
>That's only the first time you go somewhere new. You don't need to=20
>close
>both ports so quickly; the control channel can stay open and you thus
>avoid HOL blocking between data and control (and thus the need to
>chunk-and-mux within persistent HTTP), which increases other delays for
>HTTP.
You still need to send another PORT/PASV and wait for the response=20
before making another TCP connection for data, since you can't re-use=20
this one.

So subsequent requests to the same server will be quicker, still at=20
least 3 round-trips more than a subsequent http request on a persistent=20
connection.

>
>Neither protocol matches exactly what is really needed for a true
>transaction-oriented protocol.
We are probably in agreement here.

>
>>  ...
>>  Then try adding all the firewall issues due to transmitting data
>>  connection endpoint information over the control connection and it's
>>  no surprise FTP is not favoured for downloads.
>
>FTP had a passive mode even back then that avoids this issue. It also
>had suspend/resume, compression, and format conversion.

Those all take more round trips to negotiate.  As for format=20
conversion..... bad idea, never should have been the server's job.

We've seen a lot of server-side firewall problems with PASV as well.

Adrien


>
>Joe


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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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This is a multi-part message in MIME format.
--------------C37E2CC18571EAC4F9C39822
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On 8/17/2016 4:17 PM, Adrien de Croy wrote:
>> FTP had a passive mode even back then that avoids this issue. It also
>> had suspend/resume, compression, and format conversion.
>
> Those all take more round trips to negotiate.  As for format
> conversion..... bad idea, never should have been the server's job.
Sure, but they can be negotiated once for a connection.

FTP also supports wildcard retrieval (e.g., "get *"), which can help
with anticipation.

My point is that FTP could have been much better place to start towards
a transaction protocol.

Joe

--------------C37E2CC18571EAC4F9C39822
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: 8bit

<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <p><br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 8/17/2016 4:17 PM, Adrien de Croy
      wrote:<br>
    </div>
    <blockquote
      cite="mid:ema6771211-5d5d-4106-b3f0-87616b83f9f6@bodybag"
      type="cite">
      <blockquote type="cite" style="color: #000000;">FTP had a passive
        mode even back then that avoids this issue. It also
        <br>
        had suspend/resume, compression, and format conversion.
        <br>
      </blockquote>
      <br>
      Those all take more round trips to negotiate.  As for format
      conversion..... bad idea, never should have been the server's job.
      <br>
    </blockquote>
    Sure, but they can be negotiated once for a connection.<br>
    <br>
    FTP also supports wildcard retrieval (e.g., "get *"), which can help
    with anticipation.<br>
    <br>
    My point is that FTP could have been much better place to start
    towards a transaction protocol.<br>
    <br>
    Joe<br>
  </body>
</html>

--------------C37E2CC18571EAC4F9C39822--


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From: Matthew Kerwin <matthew@kerwin.net.au>
Date: Thu, 18 Aug 2016 09:35:13 +1000
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Cc: tcpm@ietf.org, Daniel Stenberg <daniel@haxx.se>, Mark Nottingham <mnot@mnot.net>, Patrick McManus <pmcmanus@mozilla.com>, Willy Tarreau <w@1wt.eu>, HTTP Working Group <ietf-http-wg@w3.org>
Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi folks, I'm stepping in here on just a couple of points. I'll snip the
bits I can't or won't talk to.

On 18 August 2016 at 08:23, Joe Touch <touch@isi.edu> wrote:

>
>
> On 8/17/2016 2:13 PM, Willy Tarreau wrote:
> > On Wed, Aug 17, 2016 at 11:31:33AM -0700, Joe Touch wrote:
> >>> It can be cited in new RFCs
> >>> to justify certain choices.
> >> Hmm. Like the refs I gave could be cited in this doc to justify *its*
> >> choices? :-)
> > I think it would be nice that this is cited, but to be clear on one
> > point, I've never heard about your papers before you advertised them
> > here in this thread,
>
> A search engine on the terms "TCP HTTP interaction" would have popped
> them up rather quickly.
>
> > and yet I've been dealing with timewait issues
> > for 15 years like many people facing moderate to large web sites
> > nowadays.
>
> "
> =E2=80=8B=E2=80=8B
> timewait issues" and we're the 5th hit in Google.
>
> =E2=80=8B
Google, unless it's changed again recently, tailors search results for the
user. My first page of hits for that query are all serverfault.com,
superuser.com, serverframework.com, stackoverflow.com, etc.

Guess where I get most of my advice.
=E2=80=8B


>
> >>>> Yes, and discussing those issues would be useful - but not in this
> >>>> document either.
> >>> Why ? Lots of admins don't understand why the time_wait timeout remai=
ns
> >>> at 240 seconds on Solaris with people saying "if you want to be
> conservative
> >>> don't touch it but if you want to be modern simply shrink it to 30
> seconds
> >>> or so". People need to understand why advices have changed over 3
> decades.
> >> The advice hasn't really changed - the advice was given in the 99 ref,
> >> which includes some cases where it can still be appropriate to decreas=
e
> >> that timer.
> > Most people see it the other way around : they see no valid case to
> *increase*
> > it beyond a few seconds, because for them the default value should be
> extremely
> > low (ie this firewall vendor several years ago trying to insist on one
> second).
> > Yes that's really sad but that's reality. And you can tell them to read
> 6191
> > they won't care.
>
> Most people's servers don't need to run fast enough to care (note that
> nearly everyone runs some sort of web server on nearly every device,
> whether for control or configuration). The only issue are high-volume
> servers (the kind sysadmins deal with), and those people tend to already
> know what the tradeoffs are and accept the risks.
>
>
=E2=80=8BYour sysadmins are not like my sysadmins. But these are generalisa=
tions
and anecdotes.=E2=80=8B



> >
> >>>   - TCP timestamps: what they provide, what are the risks (some peopl=
e
> in
> >>>     banking environments refuse to enable them so that they cannot be
> used
> >>>     as an oracle to help in timing attacks).
> >> That's already covered in the security considerations of RFC 7323. How
> >> is HTTP different, if at all, from any other app?
> > HTTP is special in that it is fairly common to have to deal with tens o=
f
> > thousands of connections per second between one client and one server
> when
> > you are on the server side, because you place a number of gateways (als=
o
> > called reverse-proxies) which combine all of the possible issues you ca=
n
> > think of at a single place.
>
> There are lots of services that have that many transactions - DNS
> servers (even local ones), remote databases, etc.
>
> The point is that HTTP doesn't make the problem different, so this isn't
> an HTTP issue. It's a high rate server issue.
>
>
=E2=80=8BWhat makes HTTP different is that I expect most high-rate applicat=
ions
would exist in a context where the people running the servers and
applications have some amount of specialist experience and knowledge, or at
least an expectation that they're doing something that requires such
knowledge, in high-rate throughput.

HTTP is ubiquitous, and resides all along the traffic scale, from my
website (~no bits per second) to google.com (all the bits); and the slide
-- or sudden jump -- up that scale doesn't always correspond with acquiring
expertise in TCP stack tuning.=E2=80=8B



>
> > So probably you're starting to see the benefit of having a single doc
> > to concentrate all this.
> The same reason it's useful to have this all in one place is the reason
> we already do - there are books and courses on this.
>
>
=E2=80=8BMy users aren't getting my content right at the time my site is bo=
oming.
I've just been slashdotted/hackernewsed/whatever. Do I enroll in a course?
Buy a textbook and swot up (which I haven't done since I finished my IT
degree 15+ years ago)? Or do I hit up serverfault.com and bash the keyboard
until the fires go out?

Summary information is really important. Having it published by the same
people who published the protocol spec adds some serious cred, at least in
my eyes.=E2=80=8B



> >  You provided at least 3 different articles
> > to read and 2 or 3 different RFCs in addition to the original ones,
> > of course. A hosting provider whose web sites are down due to a lack
> > of tuning doesn't start to read many very long articles and even less
> > the most scientific ones, they need to find quick responses that they
> > can apply immediately (matter of minutes). So they launch google, they
> > type "web site dead, time-wait overflow" and they get plenty of
> > responses on stackoverflow and serverfault, many from people having
> > done the same in the past and repeating the same mistakes over and over=
.
>
> These people don't read RFCs to fix problems. They take online courses
> or read "how to" books - which do already exist in this space.
>
>
=E2=80=8BWhich people? I tend to google the error and see if there's some s=
ort of
consensus on stackoverflow. And then, as often as not, I have to advise my
sysadmins of a course of action because they know as much as me, or they
don't want to deal with my situation, or some other reason.=E2=80=8B



> > A document validated by several people and giving links for further
> > reading can help improve this situation.
> Those are the books and courses I'm talking about already.
> =E2=80=8B=E2=80=8B
>
> > People rediscover wheels because it's hard to find simple and accurate
> > information on the net.
> Nobody looks to RFCs to solve that problem...
>
> >  Basically you have the choice :
> >   - either uneducated blog posts saying "how I saved my web site using =
2
> >     sysctls"
> >   - or academic papers which are only understandable by scientific peop=
le
> >     having enough time
>
> ... that's what net FAQs are for, as well as courses and books.
>
> > At least the first ones have the merit of being easy to test, and since
> > they appear to work they are viral.
> >
> >>>  All of them became issues for
> >>> many web server admins who just copy-paste random settings from vario=
us
> >>> blogs found on the net who just copy the same stupidities over and ov=
er
> >>> resulting in the same trouble being caused to each of their reader.
> >> This doc is all over the place.
> >>
> >> If you want a doc to advise web admins, do so.
> > That's *exactly* what Daniel started to do when you told him he shouldn=
't
> > do it.
>
> I didn't say a doc to advise web admins wasn't useful. I said it wasn't
> an RFC.
>
> It's a web FAQ, a book, etc.
>
>
=E2=80=8BHere's the crux of the issue. What do you think an RFC is, that we
(apparently) don't? Why is an informational RFC not allowed to present the
same sort of information as an FAQ? (Isn't that what a BCP is?)

Personally I'd be happy if it was written up exactly like an FAQ, and
published as an informational RFC; because that tells me that this FAQ was
published by the IETF. It's not some random dude's unreliable blog full of
cargo-cult advice and anecdotes; it met the consensus of the organisation
that published the HTTP protocol spec. It's legitimate and reliable. And
one day, when it's out of date, it'll be updated or obsoleted by the new
consensus wisdom of the IETF.

That, as far as I know, doesn't defy the official definition of what an RFC
is*, nor does it devalue any existing (or future) RFCs.

And Google indexes RFCs. If this ends up being a really useful document
(which I imagine it would), with lots of inbound links and references,
people won't need to "look to RFCs to solve [their] problem", they'll do
what they already do -- look to Google, and Google will point them to this
RFC.


* heh=E2=80=8B



>
> > and I find it fantastic to see that this protocol
> > still scales so well.
> > But we need to consider modern usages of this protocol
> > for the web, and not just academic research and e-mail.
> You might consider that TCPM and TSVWG don't exist for just "academic
> research and e-mail". What do you think we've been doing for the past 40
> years?
>
>
=E2=80=8BDunno, I was only born 35-odd years ago. ;) Shall we talk about ge=
neration
gaps? Greybeards vs millennials? (Of which I'm neither, BTW. Not yet, at
least.)

Where we come from, how we find information to solve our problems, the way
we view the IETF and its RFCs are all different. If this argument is just
that this stuff doesn't belong in an RFC, that's a cultural issue and not
one I think we can resolve in this one technical working group.

Cheers
--=20
  Matthew Kerwin
  http://matthew.kerwin.net.au/

--001a113f8aaec5a037053a4cea10
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif;color:#073763">Hi folks, I&#39;m stepping in here on just a couple of=
 points. I&#39;ll snip the bits I can&#39;t or won&#39;t talk to.</div><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On 18 August 2016 at =
08:23, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" tar=
get=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><span class=3D""><br>
<br>
On 8/17/2016 2:13 PM, Willy Tarreau wrote:<br>
&gt; On Wed, Aug 17, 2016 at 11:31:33AM -0700, Joe Touch wrote:<br>
&gt;&gt;&gt; It can be cited in new RFCs<br>
&gt;&gt;&gt; to justify certain choices.<br>
&gt;&gt; Hmm. Like the refs I gave could be cited in this doc to justify *i=
ts*<br>
&gt;&gt; choices? :-)<br>
&gt; I think it would be nice that this is cited, but to be clear on one<br=
>
&gt; point, I&#39;ve never heard about your papers before you advertised th=
em<br>
&gt; here in this thread,<br>
<br>
</span>A search engine on the terms &quot;TCP HTTP interaction&quot; would =
have popped<br>
them up rather quickly.<br>
<span class=3D""><br>
&gt; and yet I&#39;ve been dealing with timewait issues<br>
&gt; for 15 years like many people facing moderate to large web sites<br>
&gt; nowadays.<br>
<br>
</span>&quot;<div class=3D"gmail_default" style=3D"font-family:georgia,seri=
f;color:rgb(7,55,99);display:inline">=E2=80=8B=E2=80=8B</div>timewait issue=
s&quot; and we&#39;re the 5th hit in Google.<br>
<span class=3D""><br></span></blockquote><div><div class=3D"gmail_default" =
style=3D"font-family:georgia,serif;color:rgb(7,55,99);display:inline">=E2=
=80=8B</div></div><div><div class=3D"gmail_default" style=3D"font-family:ge=
orgia,serif;color:rgb(7,55,99);display:inline">Google, unless it&#39;s chan=
ged again recently, tailors search results for the user. My first page of h=
its for that query are all <a href=3D"http://serverfault.com">serverfault.c=
om</a>, <a href=3D"http://superuser.com">superuser.com</a>, <a href=3D"http=
://serverframework.com">serverframework.com</a>, <a href=3D"http://stackove=
rflow.com">stackoverflow.com</a>, etc.</div></div><div><div class=3D"gmail_=
default" style=3D"font-family:georgia,serif;color:rgb(7,55,99);display:inli=
ne"><br></div></div><div><div class=3D"gmail_default" style=3D"font-family:=
georgia,serif;color:rgb(7,55,99);display:inline">Guess where I get most of =
my advice.</div></div><div><div class=3D"gmail_default" style=3D"font-famil=
y:georgia,serif;color:rgb(7,55,99);display:inline">=E2=80=8B</div>=C2=A0</d=
iv><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex"><span class=3D""><br>
&gt;&gt;&gt;&gt; Yes, and discussing those issues would be useful - but not=
 in this<br>
&gt;&gt;&gt;&gt; document either.<br>
&gt;&gt;&gt; Why ? Lots of admins don&#39;t understand why the time_wait ti=
meout remains<br>
&gt;&gt;&gt; at 240 seconds on Solaris with people saying &quot;if you want=
 to be conservative<br>
&gt;&gt;&gt; don&#39;t touch it but if you want to be modern simply shrink =
it to 30 seconds<br>
&gt;&gt;&gt; or so&quot;. People need to understand why advices have change=
d over 3 decades.<br>
&gt;&gt; The advice hasn&#39;t really changed - the advice was given in the=
 99 ref,<br>
&gt;&gt; which includes some cases where it can still be appropriate to dec=
rease<br>
&gt;&gt; that timer.<br>
&gt; Most people see it the other way around : they see no valid case to *i=
ncrease*<br>
&gt; it beyond a few seconds, because for them the default value should be =
extremely<br>
&gt; low (ie this firewall vendor several years ago trying to insist on one=
 second).<br>
&gt; Yes that&#39;s really sad but that&#39;s reality. And you can tell the=
m to read 6191<br>
&gt; they won&#39;t care.<br>
<br>
</span>Most people&#39;s servers don&#39;t need to run fast enough to care =
(note that<br>
nearly everyone runs some sort of web server on nearly every device,<br>
whether for control or configuration). The only issue are high-volume<br>
servers (the kind sysadmins deal with), and those people tend to already<br=
>
know what the tradeoffs are and accept the risks.<br>
<span class=3D""><br></span></blockquote><div><br></div><div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)">=E2=
=80=8BYour sysadmins are not like my sysadmins. But these are generalisatio=
ns and anecdotes.=E2=80=8B</div><br></div><div>=C2=A0</div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex"><span class=3D"">
&gt;<br>
&gt;&gt;&gt;=C2=A0 =C2=A0- TCP timestamps: what they provide, what are the =
risks (some people in<br>
&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0banking environments refuse to enable them =
so that they cannot be used<br>
&gt;&gt;&gt;=C2=A0 =C2=A0 =C2=A0as an oracle to help in timing attacks).<br=
>
&gt;&gt; That&#39;s already covered in the security considerations of RFC 7=
323. How<br>
&gt;&gt; is HTTP different, if at all, from any other app?<br>
&gt; HTTP is special in that it is fairly common to have to deal with tens =
of<br>
&gt; thousands of connections per second between one client and one server =
when<br>
&gt; you are on the server side, because you place a number of gateways (al=
so<br>
&gt; called reverse-proxies) which combine all of the possible issues you c=
an<br>
&gt; think of at a single place.<br>
<br>
</span>There are lots of services that have that many transactions - DNS<br=
>
servers (even local ones), remote databases, etc.<br>
<br>
The point is that HTTP doesn&#39;t make the problem different, so this isn&=
#39;t<br>
an HTTP issue. It&#39;s a high rate server issue.<br>
<div><div class=3D"h5"><br></div></div></blockquote><div><br></div><div><di=
v class=3D"gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55=
,99)">=E2=80=8BWhat makes HTTP different is that I expect most high-rate ap=
plications would exist in a context where the people running the servers an=
d applications have some amount of specialist experience and knowledge, or =
at least an expectation that they&#39;re doing something that requires such=
 knowledge, in high-rate throughput.</div><div class=3D"gmail_default" styl=
e=3D"font-family:georgia,serif;color:rgb(7,55,99)"><br></div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)">HTTP =
is ubiquitous, and resides all along the traffic scale, from my website (~n=
o bits per second) to <a href=3D"http://google.com">google.com</a> (all the=
 bits); and the slide -- or sudden jump -- up that scale doesn&#39;t always=
 correspond with acquiring expertise in TCP stack tuning.=E2=80=8B</div><br=
></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h=
5">
<br>&gt; So probably you&#39;re starting to see the benefit of having a sin=
gle doc<br>
&gt; to concentrate all this.<br>
</div></div>The same reason it&#39;s useful to have this all in one place i=
s the reason<br>
we already do - there are books and courses on this.<br>
<span class=3D""><br></span></blockquote><div><br></div><div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)">=E2=
=80=8BMy users aren&#39;t getting my content right at the time my site is b=
ooming. I&#39;ve just been slashdotted/hackernewsed/whatever. Do I enroll i=
n a course? Buy a textbook and swot up (which I haven&#39;t done since I fi=
nished my IT degree 15+ years ago)? Or do I hit up <a href=3D"http://server=
fault.com">serverfault.com</a> and bash the keyboard until the fires go out=
?</div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;colo=
r:rgb(7,55,99)"><br></div><div class=3D"gmail_default" style=3D"font-family=
:georgia,serif;color:rgb(7,55,99)">Summary information is really important.=
 Having it published by the same people who published the protocol spec add=
s some serious cred, at least in my eyes.=E2=80=8B</div><br></div><div>=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><span class=3D"">
&gt;=C2=A0 You provided at least 3 different articles<br>
&gt; to read and 2 or 3 different RFCs in addition to the original ones,<br=
>
&gt; of course. A hosting provider whose web sites are down due to a lack<b=
r>
&gt; of tuning doesn&#39;t start to read many very long articles and even l=
ess<br>
&gt; the most scientific ones, they need to find quick responses that they<=
br>
&gt; can apply immediately (matter of minutes). So they launch google, they=
<br>
&gt; type &quot;web site dead, time-wait overflow&quot; and they get plenty=
 of<br>
&gt; responses on stackoverflow and serverfault, many from people having<br=
>
&gt; done the same in the past and repeating the same mistakes over and ove=
r.<br>
<br>
</span>These people don&#39;t read RFCs to fix problems. They take online c=
ourses<br>
or read &quot;how to&quot; books - which do already exist in this space.<br=
>
<span class=3D""><br></span></blockquote><div><br></div><div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)">=E2=
=80=8BWhich people? I tend to google the error and see if there&#39;s some =
sort of consensus on stackoverflow. And then, as often as not, I have to ad=
vise my sysadmins of a course of action because they know as much as me, or=
 they don&#39;t want to deal with my situation, or some other reason.=E2=80=
=8B</div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span cl=
ass=3D"">
&gt; A document validated by several people and giving links for further<br=
>
&gt; reading can help improve this situation.<br>
</span>Those are the books and courses I&#39;m talking about already.<br><s=
pan class=3D""><div class=3D"gmail_default" style=3D"font-family:georgia,se=
rif;color:rgb(7,55,99);display:inline">=E2=80=8B=E2=80=8B</div><br>
&gt; People rediscover wheels because it&#39;s hard to find simple and accu=
rate<br>
&gt; information on the net.<br>
</span>Nobody looks to RFCs to solve that problem...<br>
<span class=3D""><br>
&gt;=C2=A0 Basically you have the choice :<br>
&gt;=C2=A0 =C2=A0- either uneducated blog posts saying &quot;how I saved my=
 web site using 2<br>
&gt;=C2=A0 =C2=A0 =C2=A0sysctls&quot;<br>
&gt;=C2=A0 =C2=A0- or academic papers which are only understandable by scie=
ntific people<br>
&gt;=C2=A0 =C2=A0 =C2=A0having enough time<br>
<br>
</span>... that&#39;s what net FAQs are for, as well as courses and books.<=
br>
<span class=3D""><br>
&gt; At least the first ones have the merit of being easy to test, and sinc=
e<br>
&gt; they appear to work they are viral.<br>
&gt;<br>
&gt;&gt;&gt;=C2=A0 All of them became issues for<br>
&gt;&gt;&gt; many web server admins who just copy-paste random settings fro=
m various<br>
&gt;&gt;&gt; blogs found on the net who just copy the same stupidities over=
 and over<br>
&gt;&gt;&gt; resulting in the same trouble being caused to each of their re=
ader.<br>
&gt;&gt; This doc is all over the place.<br>
&gt;&gt;<br>
&gt;&gt; If you want a doc to advise web admins, do so.<br>
&gt; That&#39;s *exactly* what Daniel started to do when you told him he sh=
ouldn&#39;t<br>
&gt; do it.<br>
<br>
</span>I didn&#39;t say a doc to advise web admins wasn&#39;t useful. I sai=
d it wasn&#39;t<br>
an RFC.<br>
<br>
It&#39;s a web FAQ, a book, etc.<br>
<span class=3D""><br></span></blockquote><div><br></div><div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)">=E2=
=80=8BHere&#39;s the crux of the issue. What do you think an RFC is, that w=
e (apparently) don&#39;t? Why is an informational RFC not allowed to presen=
t the same sort of information as an FAQ? (Isn&#39;t that what a BCP is?)</=
div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;color:r=
gb(7,55,99)"><br></div><div class=3D"gmail_default" style=3D"font-family:ge=
orgia,serif;color:rgb(7,55,99)">Personally I&#39;d be happy if it was writt=
en up exactly like an FAQ, and published as an informational RFC; because t=
hat tells me that this FAQ was published by the IETF. It&#39;s not some ran=
dom dude&#39;s unreliable blog full of cargo-cult advice and anecdotes; it =
met the consensus of the organisation that published the HTTP protocol spec=
. It&#39;s legitimate and reliable. And one day, when it&#39;s out of date,=
 it&#39;ll be updated or obsoleted by the new consensus wisdom of the IETF.=
</div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;color=
:rgb(7,55,99)"><br></div><div class=3D"gmail_default" style=3D"font-family:=
georgia,serif;color:rgb(7,55,99)">That, as far as I know, doesn&#39;t defy =
the official definition of what an RFC is*, nor does it devalue any existin=
g (or future) RFCs.</div><div class=3D"gmail_default" style=3D"font-family:=
georgia,serif;color:rgb(7,55,99)"><br></div><div class=3D"gmail_default" st=
yle=3D"font-family:georgia,serif;color:rgb(7,55,99)">And Google indexes RFC=
s. If this ends up being a really useful document (which I imagine it would=
), with lots of inbound links and references, people won&#39;t need to &quo=
t;look to RFCs to solve [their] problem&quot;, they&#39;ll do what they alr=
eady do -- look to Google, and Google will point them to this RFC.</div><di=
v class=3D"gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55=
,99)"><br></div><div class=3D"gmail_default" style=3D"font-family:georgia,s=
erif;color:rgb(7,55,99)"><br></div><div class=3D"gmail_default" style=3D"fo=
nt-family:georgia,serif;color:rgb(7,55,99)">* heh=E2=80=8B</div><br></div><=
div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><span class=3D""><br>
&gt; and I find it fantastic to see that this protocol<br>
&gt; still scales so well.<br>
&gt; But we need to consider modern usages of this protocol<br>
&gt; for the web, and not just academic research and e-mail.<br>
</span>You might consider that TCPM and TSVWG don&#39;t exist for just &quo=
t;academic<br>
research and e-mail&quot;. What do you think we&#39;ve been doing for the p=
ast 40<br>
years?<br>
<span class=3D""><br></span></blockquote><div><br></div><div><div class=3D"=
gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)">=E2=
=80=8BDunno, I was only born 35-odd years ago. ;) Shall we talk about gener=
ation gaps? Greybeards vs millennials? (Of which I&#39;m neither, BTW. Not =
yet, at least.)</div></div><div class=3D"gmail_default" style=3D"font-famil=
y:georgia,serif;color:rgb(7,55,99)"><br></div><div class=3D"gmail_default" =
style=3D"font-family:georgia,serif;color:rgb(7,55,99)">Where we come from, =
how we find information to solve our problems, the way we view the IETF and=
 its RFCs are all different. If this argument is just that this stuff doesn=
&#39;t belong in an RFC, that&#39;s a cultural issue and not one I think we=
 can resolve in this one technical working group.</div><div class=3D"gmail_=
default" style=3D"font-family:georgia,serif;color:rgb(7,55,99)"><br></div><=
div class=3D"gmail_default" style=3D"font-family:georgia,serif;color:rgb(7,=
55,99)">Cheers</div></div>-- <br><div class=3D"gmail_signature" data-smartm=
ail=3D"gmail_signature"><div dir=3D"ltr">=C2=A0 Matthew Kerwin<br>=C2=A0 <a=
 href=3D"http://matthew.kerwin.net.au/" target=3D"_blank">http://matthew.ke=
rwin.net.au/</a></div></div>
</div></div>

--001a113f8aaec5a037053a4cea10--


From nobody Wed Aug 17 17:22:27 2016
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To: Matthew Kerwin <matthew@kerwin.net.au>
References: <0CC24FC1-37E1-4125-9627-05726A9D9406@mnot.net> <7fa95741-ac58-3183-1b92-238bd4b4dae6@isi.edu> <5CD67877-19E3-4E79-BBF2-3E270343A378@mnot.net> <2197232f-10d7-28cb-fcc9-05bd495e3c22@isi.edu> <20160817064545.GD16017@1wt.eu> <7f7b129c-f156-d067-bef8-4a2213f461ac@isi.edu> <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com>
From: Joe Touch <touch@isi.edu>
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Cc: tcpm@ietf.org, Daniel Stenberg <daniel@haxx.se>, Mark Nottingham <mnot@mnot.net>, Patrick McManus <pmcmanus@mozilla.com>, Willy Tarreau <w@1wt.eu>, HTTP Working Group <ietf-http-wg@w3.org>
Subject: Re: [tcpm] TCP Tuning for HTTP - update
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This is a multi-part message in MIME format.
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Popping up to the key point...

On 8/17/2016 4:35 PM, Matthew Kerwin wrote:
> Hi folks, I'm stepping in here on just a couple of points. I'll snip
> the bits I can't or won't talk to.
...
> Where we come from, how we find information to solve our problems, the
> way we view the IETF and its RFCs are all different. If this argument
> is just that this stuff doesn't belong in an RFC, that's a cultural
> issue and not one I think we can resolve in this one technical working
> group.

IMO, RFCs deal with *protocols* and protocol deployment issues.

Here that would mean for HTTP in specific
    - turn Nagle off
    - watch out for ACK compression effects (turn it off in favor of ABC
if you can)
    - deal with slow-start restart (there are a variety of approaches here)

You also want to be generally efficient with TCP, which isn't related to
HTTP but still good advice for all apps:
    - be generally efficient and robust (use SACK, ECN, good window
scaling, etc.)
    - watch out for resource overloads (use SYN cookies/fast open,
time-wait buildup, etc.)

That advice is useful for all TCP for transactions, too.

Here's what it definitely does not mean:
    - setting socket buffer sizes (that's an OS-ism, and may be useful
for app designers to know, but isn''t directly part of TCP or HTTP)
    - running an efficient server for small connections (that's a pure
implementation performance issue, and depends on OS support for things
like threads, process/thread pools, etc.)
 
I.e., OS-specific issues in running servers are not in scope IMO.

Keep in mind who reads RFCs - mostly protocol people. Managers take
training courses to get certification, and that's where they learn their
operational issues.

Joe

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<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <p>Popping up to the key point...<br>
    </p>
    On 8/17/2016 4:35 PM, Matthew Kerwin wrote:<br>
    <blockquote
cite="mid:CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_quote">Hi folks, I'm stepping in here on just
          a couple of points. I'll snip the bits I can't or won't talk
          to.</div>
      </div>
    </blockquote>
    ...
    <blockquote
cite="mid:CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_quote">
          <div class="gmail_default"
            style="font-family:georgia,serif;color:rgb(7,55,99)">Where
            we come from, how we find information to solve our problems,
            the way we view the IETF and its RFCs are all different. If
            this argument is just that this stuff doesn't belong in an
            RFC, that's a cultural issue and not one I think we can
            resolve in this one technical working group.<br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    IMO, RFCs deal with *protocols* and protocol deployment issues.<br>
    <br>
    Here that would mean for HTTP in specific<br>
        - turn Nagle off<br>
        - watch out for ACK compression effects (turn it off in favor of
    ABC if you can)<br>
        - deal with slow-start restart (there are a variety of
    approaches here)<br>
    <br>
    You also want to be generally efficient with TCP, which isn't
    related to HTTP but still good advice for all apps:<br>
        - be generally efficient and robust (use SACK, ECN, good window
    scaling, etc.)<br>
        - watch out for resource overloads (use SYN cookies/fast open,
    time-wait buildup, etc.)<br>
    <br>
    That advice is useful for all TCP for transactions, too.<br>
    <br>
    Here's what it definitely does not mean:<br>
        - setting socket buffer sizes (that's an OS-ism, and may be
    useful for app designers to know, but isn''t directly part of TCP or
    HTTP)<br>
        - running an efficient server for small connections (that's a
    pure implementation performance issue, and depends on OS support for
    things like threads, process/thread pools, etc.)<br>
      <br>
    I.e., OS-specific issues in running servers are not in scope IMO.<br>
    <br>
    Keep in mind who reads RFCs - mostly protocol people. Managers take
    training courses to get certification, and that's where they learn
    their operational issues.<br>
    <br>
    Joe<br>
  </body>
</html>

--------------F10D1AF5F07272054D777B16--


From nobody Wed Aug 17 17:51:59 2016
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From: Mark Nottingham <mnot@mnot.net>
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To: Joe Touch <touch@ISI.EDU>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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The AD has spoken, and I think we're rapidly approaching the point of =
being off-topic here. I'll respond to your question below, and of course =
if you change your mind and do want to follow up with a helpful list of =
suggested references, you're most welcome.


> On 18 Aug 2016, at 1:08 AM, Joe Touch <touch@ISI.EDU> wrote:
>=20
>>>> If that's the case, I'd observe that the IETF isn't an academic =
publisher, and acknowledging all prior work in an area is neither =
practical, nor required, nor current practice.
>>> Plagiarism isn't an issue limited to academic environments. =
Publication
>>> of a document on the web is still publication.
>> Sure. It also isn't a legal issue in this form (unless you're =
asserting copyright?). Effectively, it's a cultural norm. Again, I will =
point out that in the culture of the IETF, we historically have not =
cited the complete provenance of every idea, both because it's =
impractical and because it doesn't benefit the reader.=20
>=20
> Although that's true in the smallest cases, the IETF does have two
> concepts that support this norm: an author list and a set of =
references.
>=20
> Can you explain how it helps the reader to not cite two documents that
> are both squarely in the same area as this doc (interaction between =
HTTP
> and TCP and the impact of running many small connections closed at the
> client as for HTTP)?

Oh, it can be very helpful; I don't think anyone disputes that =
appropriate references are helpful.=20

Whether or not specific documents -- such as yours -- should be =
referenced is a matter for editorial discretion and eventually WG =
consensus. =20

I note that no one else has supported your claims, but others have said =
(e.g., [1]) that your work probably isn't appropriate for the intended =
audience (HTTP implementers and administrators).

Directing Daniel to add citations to mollify you at this point would be =
entirely inappropriate, because it would give an incentive to others who =
want their works cited without full review to make similar claims -- =
whether that be due to a genuine desire to help, a desire to increase =
their h-index, or to bolster an IPR claim against a technology.

Or, of course, Daniel could use his editorial discretion to decide to =
cite your work before the WG adopts it; however, I'd very much =
understand if he didn't want to do so at this point.

Regards,


1. http://www.w3.org/mid/20160817064545.GD16017@1wt.eu

--
Mark Nottingham   https://www.mnot.net/


From nobody Wed Aug 17 18:00:10 2016
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Mark,

My point is simple: this work is useless if it fails to cite the
relevant prior work that explains the rationale behind its recommendations.

Mine isn't the only work not being cited here - there's all the work in
TCP tuning from PSC, many web workshops from the 90s, and numerous RFCs
and books. I've pointed out just a few.

I don't care to do the author's homework, but as AD I'm disappointed you
don't seem to care about that either.

Joe


On 8/17/2016 5:51 PM, Mark Nottingham wrote:
> The AD has spoken, and I think we're rapidly approaching the point of being off-topic here. I'll respond to your question below, and of course if you change your mind and do want to follow up with a helpful list of suggested references, you're most welcome.
>
>
>> On 18 Aug 2016, at 1:08 AM, Joe Touch <touch@ISI.EDU> wrote:
>>
>>>>> If that's the case, I'd observe that the IETF isn't an academic publisher, and acknowledging all prior work in an area is neither practical, nor required, nor current practice.
>>>> Plagiarism isn't an issue limited to academic environments. Publication
>>>> of a document on the web is still publication.
>>> Sure. It also isn't a legal issue in this form (unless you're asserting copyright?). Effectively, it's a cultural norm. Again, I will point out that in the culture of the IETF, we historically have not cited the complete provenance of every idea, both because it's impractical and because it doesn't benefit the reader. 
>> Although that's true in the smallest cases, the IETF does have two
>> concepts that support this norm: an author list and a set of references.
>>
>> Can you explain how it helps the reader to not cite two documents that
>> are both squarely in the same area as this doc (interaction between HTTP
>> and TCP and the impact of running many small connections closed at the
>> client as for HTTP)?
> Oh, it can be very helpful; I don't think anyone disputes that appropriate references are helpful. 
>
> Whether or not specific documents -- such as yours -- should be referenced is a matter for editorial discretion and eventually WG consensus.  
>
> I note that no one else has supported your claims, but others have said (e.g., [1]) that your work probably isn't appropriate for the intended audience (HTTP implementers and administrators).
>
> Directing Daniel to add citations to mollify you at this point would be entirely inappropriate, because it would give an incentive to others who want their works cited without full review to make similar claims -- whether that be due to a genuine desire to help, a desire to increase their h-index, or to bolster an IPR claim against a technology.
>
> Or, of course, Daniel could use his editorial discretion to decide to cite your work before the WG adopts it; however, I'd very much understand if he didn't want to do so at this point.
>
> Regards,
>
>
> 1. http://www.w3.org/mid/20160817064545.GD16017@1wt.eu
>
> --
> Mark Nottingham   https://www.mnot.net/
>


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On 8/17/2016 5:59 PM, Joe Touch wrote:
> Mark,
>
> ...
> I don't care to do the author's homework, but as AD ...
WG chair.

Sorry for the continued confusion on that.

Joe



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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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One other point...

On 8/17/2016 5:51 PM, Mark Nottingham wrote:
> The AD has spoken,
The only post from the AD so far was to ask why I didn't just point out
the relevant citations before.

Which, as I already replied, I had - many months ago.

Joe


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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On the point of references:


On 8/17/2016 5:51 PM, Mark Nottingham wrote:
> ...
> Whether or not specific documents -- such as yours -- should be referenced is a matter for editorial discretion and eventually WG consensus.  
>
> I note that no one else has supported your claims, but others have said (e.g., [1]) that your work probably isn't appropriate for the intended audience (HTTP implementers and administrators).
Others have claimed that the papers I cited are not a sufficient
replacement for this document, but the only claim made to not cite them
at all is based on "RFCs don't need to cite sources or detail" - which
nearly ever RFC published disproves. Citations serve many purposes, esp.
informational ones - to provide detailed background information and
context among them.

Note that if this doc is limited to *protocol* recommendations (i.e.,
the scope of the IETF), I agree that it remains useful. My primary
objection to the document itself is it includes far too many OS
operational considerations that are out of scope for the IETF.

> Directing Daniel to add citations to mollify you at this point would be entirely inappropriate, because it would give an incentive to others who want their works cited without full review to make similar claims -- whether that be due to a genuine desire to help, a desire to increase their h-index, or to bolster an IPR claim against a technology.
I agree that directing anyone to cite someone for mollification would be
inappropriate; I never asked for that.

There are plenty of ways that citation considerations avoid the issues
you raise:
    - does the document provide informational background?
        are you claiming that the docs I cite do not?
    - is it the original or most complete reference (it can be useful to
cite surveys rather than original literature where the surveys cite the
originals)
        have you found an earlier or more complete reference?

These considerations prevent people from merely claiming that their work
should be gratuitously cited.

I made my claim in the original post back in March - the bulk of the
actual TCP interactions are discussed in more detail with rationale in
the one document, and the same is true for the TIMEWAIT issue for the
second document.

The only argument I've seen put forth is that "RFCs don't need to cite
things", which is false by nearly every RFC published.

Joe


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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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> On 18 Aug 2016, at 12:29 PM, Joe Touch <touch@isi.edu> wrote:

> There are plenty of ways that citation considerations avoid the issues
> you raise:
>    - does the document provide informational background?
>        are you claiming that the docs I cite do not?
>    - is it the original or most complete reference (it can be useful =
to
> cite surveys rather than original literature where the surveys cite =
the
> originals)
>        have you found an earlier or more complete reference?
>=20
> These considerations prevent people from merely claiming that their =
work
> should be gratuitously cited.

Sure, and if the WG adopts the document, we can have those discussions. =
Adopting the document does not mean that we're going to rubber-stamp its =
content; it's just being taken as a starting point.

In the meantime, let's discuss the scope of work being proposed and =
whether it's appropriate, what modifications might be needed, etc., so =
as to inform the decision to adopt.

I'd especially like to hear from other people in the TCP community; =
we've heard from Joe and Michael; do others share their opinions, or =
have another view?


> I made my claim in the original post back in March - the bulk of the
> actual TCP interactions are discussed in more detail with rationale in
> the one document, and the same is true for the TIMEWAIT issue for the
> second document.
>=20
> The only argument I've seen put forth is that "RFCs don't need to cite
> things", which is false by nearly every RFC published.

Now you're misrepresenting what others have said.  Please stop.

--
Mark Nottingham   https://www.mnot.net/


From nobody Wed Aug 17 20:20:08 2016
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/17/2016 7:43 PM, Mark Nottingham wrote:

>> > The only argument I've seen put forth is that "RFCs don't need to cite
>> > things", which is false by nearly every RFC published.
> Now you're misrepresenting what others have said.  Please stop.
Here's a list of everyone I can find from the list who has commented on
the issue of citation.

I agree that the quote above misrepresents your position, and I
apologize for that.

I leave the sum of everyone's words - hopefully quoted with enough
context to remain accurate - as context for how to proceed.

Joe

--------

You:

Again, I will point out that in the culture of the IETF, we historically have not cited the complete provenance of every idea, both because it's impractical and because it doesn't benefit the reader.

{and in a later post:)

Whether or not specific documents -- such as yours -- should be referenced is a matter for editorial discretion and eventually WG consensus.  

I note that no one else has supported your claims, but others have said (e.g., [1]) that your work probably isn't appropriate for the intended audience (HTTP implementers and administrators).

{NB: you cite Willy, below)

---

Eliot Lear:

Perhaps we can agree that the reasonable course of action here is for
Joe to (re)-recommend a compact set of citations to the authors, perhaps
even in some easily consumable form to them (kramdown-2629 or XML)?

--

Tim Wicinsk (in ref to Eliot's post):

That sounds like a fine idea.  I'll be glad to go through those.


--
Alexy Melnikov:

Instead of starting your discussion with words like "plagiarism", you 
could have just asked for information to be clarified and a 
citation/acknowledgement added?

--


Willy Tarreau:

I've just checked the two documents you referenced. They seem to be very
well detailed but they are *scientific* research. ...
...

... Thus I think that Daniel's work completes quite well what you've
done in that it directly addresses people's concerns without requiring the
scientific background.

{and later, addressing the references I asked to be cited)

I think it would be nice that this is cited, ...

...
A document validated by several people and giving links for further
reading can help improve this situation.
...
That's why I think we should encourage him to
continue and give him all the reference he needs to have an undisputably
good reference doc.


--


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<html>
  <head>
    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <p><br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 8/17/2016 7:43 PM, Mark Nottingham
      wrote:<br>
      <br>
    </div>
    <blockquote cite="mid:781AFF22-3532-44A0-AB14-628FF86BE866@mnot.net"
      type="cite">
      <blockquote type="cite" style="color: #000000;">
        <pre wrap=""><span class="moz-txt-citetags">&gt; </span>The only argument I've seen put forth is that "RFCs don't need to cite
<span class="moz-txt-citetags">&gt; </span>things", which is false by nearly every RFC published.
</pre>
      </blockquote>
      <pre wrap="">Now you're misrepresenting what others have said.  Please stop.</pre>
    </blockquote>
    Here's a list of everyone I can find from the list who has commented
    on the issue of citation. <br>
    <br>
    I agree that the quote above misrepresents your position, and I
    apologize for that.<br>
    <br>
    I leave the sum of everyone's words - hopefully quoted with enough
    context to remain accurate - as context for how to proceed.<br>
    <br>
    Joe<br>
    <br>
    --------<br>
    <br>
    You:<br>
    <pre wrap="">Again, I will point out that in the culture of the IETF, we historically have not cited the complete provenance of every idea, both because it's impractical and because it doesn't benefit the reader.

{and in a later post:)

Whether or not specific documents -- such as yours -- should be referenced is a matter for editorial discretion and eventually WG consensus.  

I note that no one else has supported your claims, but others have said (e.g., [1]) that your work probably isn't appropriate for the intended audience (HTTP implementers and administrators).

{NB: you cite Willy, below)
</pre>
    ---<br>
    <br>
    Eliot Lear:<br>
    <pre wrap="">Perhaps we can agree that the reasonable course of action here is for
Joe to (re)-recommend a compact set of citations to the authors, perhaps
even in some easily consumable form to them (kramdown-2629 or XML)?

--

Tim Wicinsk (in ref to Eliot's post):

That sounds like a fine idea. I'll be glad to go through those.


--
Alexy Melnikov:

Instead of starting your discussion with words like "plagiarism", you 
could have just asked for information to be clarified and a 
citation/acknowledgement added?

--


Willy Tarreau:

I've just checked the two documents you referenced. They seem to be very
well detailed but they are <b class="moz-txt-star"><span class="moz-txt-tag">*</span>scientific<span class="moz-txt-tag">*</span></b> research. ...
...

... Thus I think that Daniel's work completes quite well what you've
done in that it directly addresses people's concerns without requiring the
scientific background.

{and later, addressing the references I asked to be cited)

I think it would be nice that this is cited, ...

...
A document validated by several people and giving links for further
reading can help improve this situation.
...
That's why I think we should encourage him to
continue and give him all the reference he needs to have an undisputably
good reference doc.


--

</pre>
  </body>
</html>

--------------3DF91FCEB386408A0C7120EA--


From nobody Wed Aug 17 22:38:50 2016
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On Wed, Aug 17, 2016 at 05:21:21PM -0700, Joe Touch wrote:
> Popping up to the key point...
> 
> On 8/17/2016 4:35 PM, Matthew Kerwin wrote:
> > Hi folks, I'm stepping in here on just a couple of points. I'll snip
> > the bits I can't or won't talk to.
> ...
> > Where we come from, how we find information to solve our problems, the
> > way we view the IETF and its RFCs are all different. If this argument
> > is just that this stuff doesn't belong in an RFC, that's a cultural
> > issue and not one I think we can resolve in this one technical working
> > group.
> 
> IMO, RFCs deal with *protocols* and protocol deployment issues.
> 
> Here that would mean for HTTP in specific
>     - turn Nagle off

All applications have been doing this for almost two decades.

>     - watch out for ACK compression effects (turn it off in favor of ABC
> if you can)

It does not happen that much with HTTP. Many connections on the server side
see only one, sometimes two requests, and most responses are small (about
20kB on average, with favicon fitting in a single segment). Note, I'm talking
about observations on average web sites.

>     - deal with slow-start restart (there are a variety of approaches here)

Due to the above this doesn't even happen most of the time, though with H/2
it will be more common.

> You also want to be generally efficient with TCP, which isn't related to
> HTTP but still good advice for all apps:
>     - be generally efficient and robust (use SACK, ECN, good window
> scaling, etc.)
>     - watch out for resource overloads (use SYN cookies/fast open,
> time-wait buildup, etc.)
> 
> That advice is useful for all TCP for transactions, too.

Absolutely, it's just that HTTP tends to magnify those effects because you
can get a lot of traffic concentrated on one or a few machines, and there's
a huge pressure on the admins as they know that there are tens of thousands
of visitors witnessing their performance issues. Performance is constantly
monitored by customers as well, trying to shrink the last millisecond to rank
better on search engines, which further adds to the pressure.

> Here's what it definitely does not mean:
>     - setting socket buffer sizes (that's an OS-ism, and may be useful
> for app designers to know, but isn''t directly part of TCP or HTTP)
>     - running an efficient server for small connections (that's a pure
> implementation performance issue, and depends on OS support for things
> like threads, process/thread pools, etc.)
>  
> I.e., OS-specific issues in running servers are not in scope IMO.

So what you have cited above are implementation details which are either
basic to any implementer or covered by the application itself. Admins
have already dealt with this in the default tuning and the application
has already done what is supposed to be done. Yet the admin is left with
a web site that is down under load, and they need to understand what
sysctls to change and what their effects can be on the short term (ie:
put the site back online) and the mid term (ie: limit the amount of
problems caused to random visitors). Lots of people for example enable
time-wait recycling because this is done by some people in benchmarks,
it's a single sysctl and they don't realize that they cause a lot of
issues even under moderate load.

> Keep in mind who reads RFCs - mostly protocol people. Managers take
> training courses to get certification, and that's where they learn their
> operational issues.

As Matthew said, admins tend to trust better the people who write the
protocols. Having an RFC authored by various protocol people will
definitely help make admins do the right thing.

Willy


From nobody Thu Aug 18 00:26:23 2016
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Subject: Re: [tcpm] Personal change in chairing
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s/personal change/personnel change/

> Am 17.08.2016 um 17:28 schrieb Mirja K=C3=BChlewind =
<ietf@kuehlewind.net>:
>=20
> Hi all,
>=20
> I have to announce a personal change in the tcpm chair position:
>=20
> Pasi Sarolahti is stepping down due to time restrictions. Pasi, thanks =
a lot for your work and engagement as chair! Your did a great job and we =
hope to still see you as often as possible as contributor in the working =
group!
>=20
> Michael T=C3=BCxen is starting as a new tcpm chair. Micheal, great to =
have you on broad! We look forward to work with you!
>=20
> Mirja, as responsible AD
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Thu Aug 18 01:05:53 2016
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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
References: <20160708211635.32095.77047.idtracker@ietfa.amsl.com> <2f56d1b8-7450-42de-4a24-53df9cf4c045@it.uc3m.es> <a1f456db6e656785da8e356b9f530717@mail.gmail.com> <CAK6E8=etaxFSgBvQt27s=rB03rRQxQ_u43qWRExTVXys9A82wg@mail.gmail.com> <623f46dedcc661ae74563f6bf08f46a5@mail.gmail.com> <CAK6E8=cYtP_KdvK9TEDov3j523OfcLfeBMgNSvaSueqzqwTC7Q@mail.gmail.com>
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Date: Thu, 18 Aug 2016 10:05:45 +0200
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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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HI Yuchung, Marcello

> -----Original Message-----
> From: Yuchung Cheng [mailto:ycheng@google.com]
> Sent: 15. august 2016 18:49
> To: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
> Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>; TCP Prague List
> <tcpPrague@ietf.org>; tcpm IETF list <tcpm@ietf.org>
> Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
>
> On Mon, Aug 15, 2016 at 2:00 AM, Karen Elisabeth Egede Nielsen
> <karen.nielsen@tieto.com> wrote:
> > Hi Yuchung,
> >
> >> >
> >> > An additional comment is  that new approaches to retransmissions -
> >> > like TCP TLP and TCP RACK (also SCTP TLR which however is not
> >> > progressing at the moment) might fundamentally alter the picture.
> >> > I.e., if retransmissions are sent pro-actively in tail loss
> >> > situations then more conservative RTOs may be kept for situations
> >> > where it is prudent to wait longer. Don't know if TCP TLP is so
> >> > widely deployed that it is something that you should relate to even
> >> > if it may be superseded by RACK. Just a thought.
> >> TLP and RACK help reduce timeout cases in DC environment for sure.
> >> But still it can not completely avoid timeouts.
> >
> > [Karen Elisabeth Egede Nielsen] Agreed.
> >
> > For SCTP TLR we still see RTO-timeouts when also
> > probes/retransmissions are lost.
> > I assume that it would be something like this also for Rack/TLP -
> > though potentially depending on how many consecutive probes that are
> allowed.
> > The role of RTO-timeouts are different when RACK/TLP is enabled and it
> > might be counterproductive to have RTO-timeouts be too aggressive as
> > the function indeed with TLP/RACK becomes something much closer to the
> > original intend
> > (read: how I understand the original intend) - namely to introduce a
> > pause for the network to recover when things are really (consistently)
> > bad.
> >
> > If the RTO is lowered down to be comparable in order of magnitude with
> > the RTT (with some delay_ack considerations) we are narrowing the
> > situation where TLP/RACK probing is able to kick in before
> > RTO-retransmissions starts anyway.
> >
> > I understand that RTOs should be adjusted to the network dynamics, but
> > I do think that it is important to understand - for the DC environment
> > - if the need for the low RTOs in DC's is motivated mainly by having
> > RTO-retransmissions fix the tail loss recovery deficiencies of TCP,
> > which RACK/TLP addresses, or more generally for a need for more timely
> > probe and reactivation for/after network recovery.
> >
> > Or perhaps you see the TLP probe timeout and the RTO timeout
> > eventually being driven by the same timer, the reactions (e.g. CC
> > reactions)  then possibly depending on state (probe already sent or
> > not).
> I agree with your thought here.
>
> More generally, I'd like to evolve the state of art recovery with these
> principles:
> 1. React with ack events as much as possible
>
> 2. Send tiny probes to trigger (1) after 2-3 RTTs
>
> 3. Have conservative  RTOs that (exp-backoff) expire at an order of RTT to
> repair the head for reliability.
>
>
> On thing about RTO: RTO current reset cwnd to 1 upon firing no matter
> what.
> We should ONLY reset cwnd to 1 if no news (acks/data) for a long duration.
> This is quite important for performance: today cwnd is reset to 1 even if
> every packet but the first one has been recently delivered. This
> unnecessarily conservative because the ack clock isn't lost yet. But the
> collateral damage to performance is huge on large BDP networks. This
> results
> work-arounds to shorten and avoid RTOs.
>
>
> I'd like to address these points in the upcoming RACK/TLP draft.
>
>
[Karen Elisabeth Egede Nielsen] I agree completely with your view and
approach.

In respect to the draft draft-bagnulo-tcpm-tcp-low-rtt-00.txt and the given
recommendations for low RTO
is that they do not (likely) take 2. Into account - meaning that they are
looking for the low RTO to solve 2 for tail recovery.
The bad thing is that they then also get the CWND reductions (as you also
say).


I would like for that this aspect is taken into consideration/given thoughts
in the draft draft-bagnulo-tcpm-tcp-low-rtt-00.txt and the general 4LS
effort.
If we end with very low recommendations for RTO in order to handle 2. we
risk undermining the RTO-retransmission function as well as the TLP function
for now and forever.

It might be that one for TCPs, that do not support TLP(/RACK), need to
operate with very low RTOs to handle tail loss's "fast enough", but the
recommendation for
"future standard ?" TCP with TLP always enabled might be different. In this
context lowering the RTO to handle 2. should be considered as a "temporary
hack" ?

BR, Karen


From nobody Thu Aug 18 08:08:21 2016
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To: Willy Tarreau <w@1wt.eu>
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/17/2016 10:38 PM, Willy Tarreau wrote:
>> Keep in mind who reads RFCs - mostly protocol people. Managers take
>> > training courses to get certification, and that's where they learn their
>> > operational issues.
> As Matthew said, admins tend to trust better the people who write the
> protocols. Having an RFC authored by various protocol people will
> definitely help make admins do the right thing.
Admins don't know who protocol people are.

I do think that this doc needs to figure out whom it is speaking to,
what advice they actually need, etc.

If the result is a set of recommendations that involve the word
"sysctl", I remain skeptical it is appropriate as an RFC.

Joe

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    <br>
    <div class="moz-cite-prefix">On 8/17/2016 10:38 PM, Willy Tarreau
      wrote:<br>
    </div>
    <blockquote cite="mid:20160818053837.GC16773@1wt.eu" type="cite">
      <blockquote type="cite" style="color: #000000;">
        <pre wrap="">Keep in mind who reads RFCs - mostly protocol people. Managers take
<span class="moz-txt-citetags">&gt; </span>training courses to get certification, and that's where they learn their
<span class="moz-txt-citetags">&gt; </span>operational issues.
</pre>
      </blockquote>
      <pre wrap="">As Matthew said, admins tend to trust better the people who write the
protocols. Having an RFC authored by various protocol people will
definitely help make admins do the right thing.</pre>
    </blockquote>
    Admins don't know who protocol people are.<br>
    <br>
    I do think that this doc needs to figure out whom it is speaking to,
    what advice they actually need, etc.<br>
    <br>
    If the result is a set of recommendations that involve the word
    "sysctl", I remain skeptical it is appropriate as an RFC.<br>
    <br>
    Joe<br>
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From nobody Thu Aug 18 09:35:55 2016
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To: Praveen Balasubramanian <pravb@microsoft.com>
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Subject: Re: [tcpm] Review of draft-ietf-tcpm-dctcp-01
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Praveen,


On 17/07/16 21:25, Praveen Balasubramanian wrote:
>
> Thanks Bob. Sorry for the <ridiculous> delay.
>
[BB] And now sorry for my delay. I tried to finish this email a number 
of times, but kept getting interrupted.
>
> Comments inline prefixed by <Praveen>. Most of these edits will be in 
> the next revision.
>
[BB] I've tried very hard to only comment where I really do think there 
is still a problem (even if minor), and suggested specific text to help 
you...

> Thanks
>
> *From:*Bob Briscoe [mailto:ietf@bobbriscoe.net]
> *Sent:* Friday, November 6, 2015 10:27 AM
> *To:* Praveen Balasubramanian <pravb@microsoft.com>; EGGERT, Lars 
> <lars@netapp.com>; Stephen Bensley <sbens@microsoft.com>; Dave Thaler 
> <dthaler@microsoft.com>; glenn.judd@morganstanley.com
> *Cc:* tcpm IETF list <tcpm@ietf.org>
> *Subject:* Review of draft-ietf-tcpm-dctcp-01
>
> Praveen & co-authors,
>
> As promised, here's my review of draft-ietf-tcpm-dctcp-01
>
> *_General Comments
> _*
> In general, in the comments below I have tried to reduce the multiple 
> assumptions about controlled environments down to the one assumption 
> that they are controlled by a single authority. I believe DCTCP can 
> work without the other assumptions, so the following do not apply in 
> all controlled environments:
>   * not always low RTT (e.g. inter-DC scenarios, or global private 
> networks)
>   * not always no risk of traffic attacks (e.g. internally arranged 
> attacks)
> I may have missed some other occurrences of these assumptions, so pls 
> feel free to seek out them all.
>
> <Praveen> DCTCP can be extended to other environments but this draft 
> does not cover any of that. It is explicitly for deployment in a 
> datacenter where the RTT is low and there is no risk of internal 
> attack. We cannot cover other scenarios that have not been tested or 
> deployed. All the known deployments are in controlled datacenters. The 
> TCP Prague effort should result in a draft that covers additional 
> requirements that will make DCTCP work well on high latency links.
>
>
> Many of the comments argue with your choices of MUST, SHOULD, MAY etc. 
> This might seem nit-picking, given the primary purpose is to document 
> the algo. However, it would be wrong to require things that aren't 
> required or to allow exceptions when exceptions would not be 
> interoperable. An alternative approach would be to just remove all 
> capitalised language.
>
> <Praveen> I am fine with removing such capitalization except for when 
> removing it will lead to interop problems between different 
> implementations of DCTCP. I am responding to each comment below and 
> adding whether the new revision will fix it not.
>
>
> *_Section-by-section review.
> _*
> *Abstract
> *
> I think it needs to have an applicability statement in the abstract 
> (and introduction) that refers to the deployment and implementation 
> status sections at the end. Something like:
>
> "This is an informational specification of the implementation of DCTCP 
> in Microsoft Windows Server 2012. It is applicable to deployments in 
> controlled environments like data centres but it MUST NOT be deployed 
> over the public Internet without additional measures, as detailed in 
> sections 4-6. "
>
> <Praveen> This is already addressed in the introduction. I don’t see 
> the value of repeating it multiple times. An implementer will read the 
> introduction. If you still feel strongly, we can add it to the 
> abstract and remove it from the introduction section. This draft will 
> not cover any additional measures that are intended for deployments 
> outside the datacenter – that’s not the intended purpose.
>
[BB] It is very common for the IETF to require even much less minor 
applicability statements to be highlighted in the abstract (e.g. see TFO 
RFC). This is because nowadays implementers / readers often skip 
introductory text assuming they understand the problem, etc. and they 
just scan for the normative parts. Anyone reading an IETF RFCs assumes 
the context is the public Internet, so any RFC that doesn't have that 
context needs to flag that. I know the name says Data Centre, but the 
abstract needs to explain that this is not an IETF RFC that is 
standardizing DCTCP for the public Internet.

If you still don't think so, we should put this to the WG (which will 
slow it down).
>
>
> *1. Intro
> *s/its many servers/
>  /their many servers/
>
> <Praveen> Fixed in next revision.
>
>
> "...limited queue capacities..."
> I understand that the motivation has been abbreviated, but I think 
> this has lost too much. Perhaps mention memory shared between 
> interfaces, so either bloated buffers or too little, making traffic 
> susceptible to packet losses?
>
>    [RFC3168] describes a mechanism for using Explicit Congestion
>    Notification (ECN) from the switches for early detection of
>    congestion, rather than waiting for packet loss to occur.
>
> This isn't correct. 3168 requires ECN marking to occur no earlier than 
> drop. It is precisely that 3168 does not allow early marking unless 
> there is also early drop that is the problem.
>
> <Praveen> Fixed in next revision.
>
>
>    It is recommended that DCTCP be deployed...
>
> "recommended" is too weak. I suggest you use something like the 
> applicability text given above, both here and in the abstract.
>
> <Praveen> Added the word “only” to make this stronger per Michael’s 
> review. Fixed in next revision.
>
[BB] I suggest the applicability para (here in the intro) refers forward 
to section 5 for details (e.g. so you don't have to explain why it 
shouldn't be deployed in the intro).
>
>
>
> *2. Terminology
> *
> Suggest you explain that capitalised words are not quite the same as 
> in most RFCs:
> "Normative language is used to describe how necessary the various 
> aspects of the Microsoft implementation are for interoperability, but 
> even compliant implementations without the measures in sections 4-6 
> would still only be safe to deploy in controlled environments."
>
> <Praveen> Fair enough. Added text in next revision.
>
>
> *3. DCTCP Algo
> *
> The three bullets say no more than "this is normal stuff". Would it be 
> better to supplement each sentence with a brief phrase saying what is 
> different about each aspect compared to RFC3168?
>
> <Praveen> The goal here it to describe the changes needed on top of 
> 3168 so I don’t think further explanation is necessary.
>
>
> *3.1 Marking
> *
> This needs to say something about how you mark the IP header from L2. 
> I suggest you refer to the up-and-forward mode of 
> draft-ietf-tsvwg-ecn-encap.
>
> <Praveen> Why is that relevant to this document? It is not intended as 
> a specification for operation of L2 switches. Won’t fix.
>
[BB] Well, you specify a lot about the marking algo, e.g.
K > (RTT *  C)/7

A simple solution would be: s/switch/L3 switch/

I just thought that a reader would think, "You say switches (i.e. L2 
devices) mark the IP header - how can that work?" unless you explain.

>
> s/sending rate/
>  /link rate/
>
> <Praveen> Fixed in next revision
>
>
> *3.2
> *
> CURRENT:
>
>     1.  If the CE codepoint is set and DCTCP.CE is false, send an ACK for
>         any previously unacknowledged packets and set DCTCP.CE to true.
>     2.  If the CE codepoint is not set and DCTCP.CE is true, send an ACK
>         for any previously unacknowledged packets and set DCTCP.CE to
>         false.
>
> SUGGESTED:
>
>     1.  If the CE codepoint is set and DCTCP.CE is false, set DCTCP.CE to
>         true and send an ACK for any previously unacknowledged packets.
>     2.  If the CE codepoint is not set and DCTCP.CE is true, set DCTCP.CE
>         to false and send an ACK for any previously unacknowledged packets.
>
>     The figure would have to be changed to be consistent too.
> RATIONALE:
> If the receiver changes DCTCP.CE /after/ sending the ACK like this, it 
> delays each signal until the following ACK is sent. That could add a 
> delay of anything from 1 to m packets. I've never understood why the 
> order of these operations was specified in this way. If there is no 
> reason, then I suggest that it is specified in the opposite order, as 
> I describe above.
>
>
> A note could be added to say the Windows Server 12 implementation does 
> these steps in reverse order, but the order specified is preferred 
> because it reduces signalling delay and has no interoperability issues 
> with the original Windows implementation.
>
> <Praveen> Won’t fix. This is how the Windows implementation behaves so 
> the ordering is required. Both the paper and the draft are consistent 
> in this.
>
[BB] The paper and the draft are not consistent. In the draft, the 
labels on the top and bottom arrows in the diagram have been switched 
round relative to those in the paper.

However, the text in the draft has not (yet?) been changed to reflect this.
>
> There is no delay being introduced here. This ACK is sent immediately 
> so the order does not matter.
>

[BB] I agree that the ACK itself is not delayed. However, even tho a CE 
has been received, feedback of this event is held back until the next ACK.

This is a minor 'unnecessary delay' bug.
* I believe it was fixed in the FreeBSD implementation after I pointed 
it out.
* I've just checked the Linux code, and it's not been fixed there.
* Obviously I cannot see the Windows source code.

If this bug is still in the Windows implementation, it's up to you if 
you also want the spec to require that the bugs are implemented exactly 
as they are in Windows ;)

>        
>     The handling of the "Congestion Window Reduced" (CWR) bit is also
>     exactly as per [RFC3168 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d>] including [RFC3168-ERRATA3639 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-RFC3168-ERRATA3639&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=wDJv%2bcypXj5layfzvQGhXYqB6kwl8lW%2fpdQ4iYg2cN8%3d>].  That is, on
>     receipt of a segment with both the CE and CWR bits set, CWR is
>     processed first and then ECE is processed.
>
> Even tho this is the receiver section, I suggest:
> s/The handling/Receiver handling/
>
> <Praveen> Isn’t this assumed by a reader familiar with 3168? But this 
> is a minor editorial fix so fixed in the next revision.
>
>
> Whatever, this cannot be correct. A DCTCP receiver surely does nothing 
> on receipt of CWR, whereas an RFC3168 receiver does something. A DCTCP 
> receiver certainly cannot do exactly what RFC3168 says, because that 
> says stop setting ECE on receipt of CWR until the next CE arrives, 
> which would surely break this feedback protocol.
>
> <Praveen> I don’t see why this would break the feedback protocol. The 
> receiver will start echoing again as soon as it sees the first CE 
> arrives again. If packet has both CWR and CE set then CE takes 
> precedence and the echoing is done.
>
[BB] I'm not saying it breaks the feedback protocol. I'm just saying 
it's confusing for the reader.

Of course, when the receiver is already not doing something, ignoring a 
signal to stop has the same outcome as stopping. However, describing 
this as "handling CWR ... is also exactly as per RFC3168" is misleading 
for the reader/implementer.

I guess changing to "handling CWR ... is as per RFC3168" would be a 
compromise.
>
>
> *3.3 *
>
>     DCTCP.Alpha, which is initialized to 1 and MUST be updated
>     as follows:
>
> s/MUST/SHOULD/
> And add "An alternative congestion avoidance algorithm MAY be used, 
> but it MUST lead to flow rate inversely proportional to 1/M".
>
> Rationale: There are many ways to implement interoperable 1/p cc. This 
> is just one.
>
> <Praveen> Alternate algorithms are out of scope of this draft. We 
> cannot say without interop testing if a different algorithm will 
> perform well in datacenters or interop well with existing 
> implementations. If there are other ways to implement 1/p they should 
> be covered by other drafts. This draft will only describe the 
> algorithm that has been tested and deployed. That said I am willing to 
> make it a SHOULD so fixed in the next revision.
>
>     DCTCP.BytesSent
>
> This variable actually counts bytes received. it's pretty confusing to 
> call it BytesSent.
>
> <Praveen> No, this is tracking bytes that have been sent.
>
[BB] I meant it counts bytes now acknowledged as received by the other 
end (which is not the same as bytes sent by this end).
DCTCP.BytesAcked would have been more meaningful. I'm not insisting on 
this - the original comment was just to improve comprehensibility.
>
>
> s/TCP SACK options [RFC2018] are ignored./
>  /The sender MUST ignore TCP SACK options [RFC2018] for this computation./
> RATIONALE:
>   a) I think the idea is to ignore CE marks after a gap because, if 
> the gap becomes considered a loss, the response to loss will override 
> any need to count incoming CE marks for the following round trip. 
> However, if the gap is filled before it is considered a loss {Note 1}, 
> then CE marks that were ignored because they arrived after a gap will 
> need to be "unignored".
>   b) We need to make clear that SACK is not ignored altogether, only 
> for this computation.
>
> {Note 1}: I can think of two cases:
> * Less than three dupacks then the gap is filled
> * The gap is filled by a delayed segment so that an earlier 
> retransmission was spurious
>
> <Praveen> Ok I will update the text.
>
> the sender MUST update cwnd as follows:
>        cwnd = cwnd * (1 - DCTCP.Alpha / 2)
>
> s/MUST/SHOULD/
> Rationale: Alternative interoperable response functions are possible. 
> For instance an algorithm like Relentless TCP that simply reduces cwnd 
> by half the size of any CE-marked segment. Then there is no Alpha and 
> no g variable.
>
> <Praveen> Fixed
>
>
> CURRENT:
>
>     Just as specified in [RFC3168 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d>], TCP should not react to congestion
>     indications more than once for every window of data.
>
> SUGGESTED:
>
>     Just as specified in [RFC3168 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d>], DCTCP does not react to congestion
>     indications more than once for every window of data.
>
> I don't know whether you intended this to be an upper-case SHOULD NOT. 
> If so, I would argue against this being normative, because it isn't 
> necessary for interop; therefore I've suggested avoiding he 'should' word.
>
> <Praveen> Fixed
>
>     The setting of
>     the "Congestion Window Reduced" (CWR) bit is also exactly as per
>     [RFC3168 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d>].
>   
>
> It is wrong to say "exactly". The sender certainly sets CWR when it 
> does a congestion response. But in DCTCP the congestion response 
> occurs every RTT, triggered by its own measurement-period logic, 
> whereas in RFC3168 it is triggered by the arrival of an ECE after the 
> ACK of any previous segment it sent with CWR set.
>
> I think it is worth saying that setting CWR is necessary for safety, 
> otherwise in the case where there has been a configuration management 
> error and the receiver complies with RFC 3168 not DCTCP, it will 
> continue sending ECE for ever and starve the session.
>
> It might help to explicitly confirm that:
> "Other aspects of TCP congestion avoidance and control such as the 
> slow start phase are no different to those in RFC5681."
>
> <Praveen> Fixed that CWR is needed for safety. Not adding the text 
> around 5681, it is presumed.
>
>
> *3.4 SYN, SYN-ACK, RST
> *
>    [RFC3168] requires that a compliant TCP MUST NOT set ECT on SYN or
>    SYN-ACK packets.
>
> s/MUST NOT/'MUST NOT'/
> Rationale: Best to make it clear this is quoting a normative statement 
> in another RFC, because it is going to be contradicted.
>
[BB] I don't know whether you saw this one - it's not been changed in 
draft-02.

I suggested adding quotes around 'MUST NOT'. This is good practice when 
quoting another RFC. Otherwise, to a careless reader (or non-native 
English speaker), it can look like it is also a requirement of this RFC.
>
>
> CURRENT
>
>     These RFCs, however, are
>     intended for general Internet use, and do not directly apply to a
>     controlled datacenter environment.
> ...
>     For DCTCP connections, the sender
>     SHOULD set ECT for SYN, SYN-ACK and RST packets.
>
> SUGGESTED:
>
>    Also, a SYN is sent before the ECN-capability has been
>    negotiated, so if it is CE-marked, a non-ECN TCP server would not
>    recognise the marking. RFC 3168 does not specify the ECN field on
>    a RST. The security concerns addressed by these RFCs
>    might not apply in controlled environments like data centres, and
>    it might not be necessary to cater for the presence of non-ECN
>    servers.
> ...
>
>     Therefore, the sender MUST set ECT for SYN/ACK and RST packets and a
>     configuration option SHOULD be provided to set ECT for SYNs.
>
> RATIONALE:
>    The draft should avoid assuming that security concerns do not apply 
> in all controlled environments.
> NOTE:
>    This might not reflect the way your implementation is coded, so you 
> might want to make that clear.
>
> <Praveen> Adding some text around security concerns.
>
[BB] The security sentence you have added in draft-02 doesn't make sense 
where it is. It ought to have been added once sentence earlier (and it 
would be better to start a new para). Specifically:
CURRENT:
                 be dropped with high probability.  For DCTCP 
connections, the sender
                 SHOULD set ECT for SYN, SYN-ACK and RST packets.  The 
security
                 concerns addressed by both these RFCs might not apply 
in controlled
                 environments like datacenters, and it might not be 
necessary to cater
                 to both the presence of non-ECN servers.
SUGGESTED:
                 be dropped with high probability.
<NEW PARA>
The security concerns addressed by both these RFCs might not apply in 
controlled
                 environments like datacenters, and it might not be 
necessary to cater
for the presence of non-ECN servers. For DCTCP connections, the sender
                 SHOULD set ECT for SYN, SYN-ACK and RST packets.

(Note I also fixed a glitch after "cater".)


>
> *4. Implementation Issues
> *
> CURRENT:
>
>     the implementation MUST choose a suitable
>     estimation gain.  [DCTCP10 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-DCTCP10&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=33dRUUxyNa2tcX0xJtMl12nWwzIdHZa6j3kgAL1dxIM%3d>] provides a theoretical basis
>
> SUGGEST:
>
>     the implementation will need to choose a suitable
>     estimation gain.  [ADCTCP10 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-DCTCP10&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=33dRUUxyNa2tcX0xJtMl12nWwzIdHZa6j3kgAL1dxIM%3d>] provides a theoretical basis ...
>
> RATIONALE:
>    a) As above, this is only one way to implement the cc, and others 
> could well prove to be interoperable when tested.
>    b) More particularly, it would be better for an implementation to 
> adapt the estimation gain to the RTT, so the spec ought not to imply 
> that one value has to be chosen.
>    c) The later paper [ADCTCP10] provides a corrected way to determine 
> the gain. From memory (I'm offline at the mo) it at least makes 
> throughput inversely proportional to 1/RTT, whereas the approach in 
> [DCTCP10] made it inversely proportional to 1/(RTT^2). I believe the 
> Linux implementation uses the [ADCTCP10] approach.
>
> <Praveen> Fair enough. Updated.
>
>     The implementation must also decide when to use DCTCP.
> ...
>     likely in the same datacenter network.
>
> The 2nd & 3rd paras are a mix of implementation and deployment issues, 
> but mostly deployment, so they might be better moved to the deployment 
> issues section.
>
> <Praveen> I think they are mainly implementation issues – the fact 
> that a lack of negotiation mechanism places a burden on the 
> implementation to provide knobs or heuristics to use DCTCP.
>
>
> CURRENT:
>
>     It is RECOMMENDED that an implementation deal with loss episodes in
>     the same way as conventional TCP.
> ...
>     DCTCP implementation MAY also allow configuration of resetting the
>     value of DCTCP.Alpha as part of processing any loss episodes.
>
> SUGGESTED: Move this whole para to 3.3 Sender Behaviour section, and:
>
>     An implementation MUST deal with loss episodes in
>     the same way as conventional TCP.
> ...
>     DCTCP implementation SHOULD also reset the
>     value of DCTCP.Alpha as part of processing any loss episodes.
>     This aspect of the behaviour MAY be configurable.
>
> RATIONALE:
>    a) There would never be a case for not falling back to conventional 
> TCP  (or do you have one?), so "RECOMMENDED" is too weak.
>    b) I have tried to interpret what you might have meant about 
> resetting Alpha. I'm not sure why you think it might need to be 
> configurable though?
>
> <Praveen> Yes this is a useful suggestion. Making loss handling a must 
> in sender section. Reset of alpha was discussed in tcpm list, and 
> Linux implementation supports it.
>
>     ...it is also RECOMMENDED that an implementation allow
>     configuration of restarting the congestion window (cwnd) of idle
>     DCTCP connections
>
> Where special implementation measures are necessary for low RTT, these 
> should be related to the measured RTT, not hard-coded. There is an 
> assumption in a number of places  that "data centre" implies low RTT. 
> Of course this is true in a single DC, but I believe the the single 
> admin aspect of a DC is what characterises DCTCP, not the low RTT 
> aspect. With the modifications in [ADCTCP], DCTCP generalises to a 
> network with larger RTTs as long as it is still  operated by a single 
> admin.
>
> <Praveen> this paragraph has now been removed per Michaels’ feedback.
>
>
> CURRENT:
>    [RFC3168] forbids the ECN-marking of pure ACK packets, because of the
>    inability of TCP to mitigate ACK-path congestion and protocol-wise
>    preferential treatment by routers.  However, dropping pure ACKs -
>    rather than ECN marking them - has disadvantages for typical
>    datacenter traffic patterns.  Because of the prevalence of bursty
>    traffic patterns that feature transient congestion, dropping of ACKs
>    causes subsequent retransmissions. It is RECOMMENDED that an
>    implementation provide a configuration knob that will cause ECT to 
> be set
>    on such control packes, which can be used in environments where such
>    concerns do not apply.
> SUGGESTED:
>    [RFC3168] forbids the ECN-marking of pure ACK packets, because of the
>    inability of TCP to mitigate ACK-path congestion and the extra
>    advantage to injection attackers that ECN is perceived to offer.
>    For the latter reason RFC 3168 also forbids ECN-marking of
>    retransmissions, window probes and RSTs.
>
>    However, dropping all these control packets -
>    rather than ECN marking them - has considerable performance
>    disadvantages.  It is RECOMMENDED that an
>    implementation provide a configuration knob that will cause ECT to 
> be set
>    on such control packes, which can be used in environments where such
>    concerns do not apply.
> NOTE:
>    This might not reflect the way your implementation is coded, so you 
> might want to make that clear.
>
> <Praveen> Fixed
>
[BB] A new comment...

As pointed out in draft-bagulo-tsvwg-generalized-ecn-00, RFC3168 only 
talks about injection attacks with retransmissions (not ACKs etc). But 
you imply injection attacks were the concern for other control packets 
as well.

I'm not insisting you change this. However, unwarranted accusations of 
security vulnerabilities tend to stick, whether or not they are correct. 
So it would be better not to throw mud around onto even more control 
packets than RFC3168 did.

Cheers


Bob

>
>
> *5. Deployment Issues*
>
>     If
>     the traffic in the datacenter is a mix of conventional TCP and DCTCP,
>     it is RECOMMENDED that DCTCP traffic be segregated from conventional
>     TCP traffic.
>
> In a data centre (or anywhere), is there any case where not 
> segregating them would make sense? If not, this needs to be a MUST, 
> not just a RECOMMENDED. I can imagine cases where there is no 
> long-running conventional TCP or no long-running DCTCP, but that's 
> covered by the "if" at the start of the sentence.
>
> You might want to refer to draft-briscoe-aqm-dualq-coupled as another 
> segregation approach here too.
>
>     Since DCTCP relies on congestion marking by the switches, DCTCP can
>     only be deployed in datacenters where the entire network
>     infrastructure supports ECN.
>
> Surely, the "fall-back to conventional TCP on loss" rule means you can 
> delete this constraint.
>
> <Praveen> Reworded to imply that DCTCP will only work as expected in 
> such cases.
>
>     A
>     variant of DCTCP that can be deployed unilaterally and only requires
>     standard ECN behavior has been described in [ODCTCP 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-ODCTCP&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=ZwfJ0LgHQ3vk%2bMD0wiia5yLrcMMSUBcGgPYXmeRu8hE%3d>][BSDCAN],
>
>
> Which part of "standard ECN behaviour" do you mean? The host part? The 
> network part? The receiver part? And "can be deployed unilaterally" 
> ought to be in the active voice not the passive, which would help 
> resolve the ambiguity too.
>
> I haven't read [ODCTCP] and [BSDCAN] (I'm offline on a plane at the 
> mo). Are they similar to "Instant ECN" in [Wu12] listed below? If not, 
> that could be referred to as well for a technique that uses the 
> regular ECN host behaviour.
>
> <Praveen> This edit predates me and  I don’t have the full context, I 
> am leaving this as is.
>
>
> *6. Known Issues
> *
> First para could refer to appendix A of RFC7560, which describes the 
> problem precisely.
>
> A method for improving the fairness of DCTCP has been proposed
>     in [ADCTCP 
> <https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-ADCTCP&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=edU2TPGpLvyFGvgocDRJiYRQyjCT%2fWEOgS2MwmwzYuU%3d>], but requires additional experimental evaluation.
>
> s/fairness/RTT fairness/
>
> (In our experiments, it's much better than without.)
>
> <Praveen> Goal is to cite other work that can address potential 
> issues. It is not a recommendation. s/fairness/RTT fairness/ Fixed in 
> next revision.
>
>
> *11. References
> *
> I think the following are cited normatively, not just informatively:
> RFC5681, RFC5562.
>
> Additional informational Reference:
>
> [Wu12] Wu, H., Ju, J., Lu, G., Guo, C., Xiong, Y. & Zhang, Y., "Tuning 
> ECN for Data Center Networks," In: Proceedings of the 8th 
> International Conference on Emerging Networking Experiments and 
> Technologies CoNEXT '12 pp.25-36 ACM (2012)
>
> <Praveen> Citation to RFCs updated as normative. Not including the new 
> reference since even though it may be relevant – it was not used as a 
> reference in writing the draft.
>
>
> That's it. HTH.
>
>
> Bob
>
> -- 
> ________________________________________________________________
> Bob Briscoehttp://bobbriscoe.net/ 
> <https://na01.safelinks.protection.outlook.com/?url=http%3a%2f%2fbobbriscoe.net%2f&data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&sdata=iEvhWj9FCPozWAp8rnxk5VZdNF%2fv7W7aD1Cp4MxOiSs%3d>

-- 
________________________________________________________________
Bob Briscoehttp://bobbriscoe.net/

-- 
________________________________________________________________
Bob Briscoehttp://bobbriscoe.net/


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    Praveen,<br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 17/07/16 21:25, Praveen
      Balasubramanian wrote:<br>
    </div>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">Thanks
            Bob. Sorry for the &lt;ridiculous&gt; delay. </span></p>
      </div>
    </blockquote>
    [BB] And now sorry for my delay. I tried to finish this email a
    number of times, but kept getting interrupted.<br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">Comments
            inline prefixed by &lt;Praveen&gt;. Most of these edits will
            be in the next revision.<o:p></o:p></span></p>
      </div>
    </blockquote>
    [BB] I've tried very hard to only comment where I really do think
    there is still a problem (even if minor), and suggested specific
    text to help you...<br>
    <br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">Thanks<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #E1E1E1
            1.0pt;padding:3.0pt 0in 0in 0in">
            <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">From:</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">
                Bob Briscoe [<a class="moz-txt-link-freetext"
                  href="mailto:ietf@bobbriscoe.net">mailto:ietf@bobbriscoe.net</a>]
                <br>
                <b>Sent:</b> Friday, November 6, 2015 10:27 AM<br>
                <b>To:</b> Praveen Balasubramanian <a
                  class="moz-txt-link-rfc2396E"
                  href="mailto:pravb@microsoft.com">&lt;pravb@microsoft.com&gt;</a>;
                EGGERT, Lars <a class="moz-txt-link-rfc2396E"
                  href="mailto:lars@netapp.com">&lt;lars@netapp.com&gt;</a>;
                Stephen Bensley <a class="moz-txt-link-rfc2396E"
                  href="mailto:sbens@microsoft.com">&lt;sbens@microsoft.com&gt;</a>;
                Dave Thaler <a class="moz-txt-link-rfc2396E"
                  href="mailto:dthaler@microsoft.com">&lt;dthaler@microsoft.com&gt;</a>;
                <a class="moz-txt-link-abbreviated"
                  href="mailto:glenn.judd@morganstanley.com">glenn.judd@morganstanley.com</a><br>
                <b>Cc:</b> tcpm IETF list <a
                  class="moz-txt-link-rfc2396E"
                  href="mailto:tcpm@ietf.org">&lt;tcpm@ietf.org&gt;</a><br>
                <b>Subject:</b> Review of draft-ietf-tcpm-dctcp-01<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p> </o:p></p>
        <p class="MsoNormal">Praveen &amp; co-authors,<br>
          <br>
          As promised, here's my review of draft-ietf-tcpm-dctcp-01<br>
          <br>
          <b><u>General Comments<br>
            </u></b><br>
          In general, in the comments below I have tried to reduce the
          multiple assumptions about controlled environments down to the
          one assumption that they are controlled by a single authority.
          I believe DCTCP can work without the other assumptions, so the
          following do not apply in all controlled environments:<br>
            * not always low RTT (e.g. inter-DC scenarios, or global
          private networks)<br>
            * not always no risk of traffic attacks (e.g. internally
          arranged attacks)<br>
          I may have missed some other occurrences of these assumptions,
          so pls feel free to seek out them all.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            DCTCP can be extended to other environments but this draft
            does not cover any of that. It is explicitly for deployment
            in a datacenter where the RTT is low and there is no risk of
            internal attack. We cannot cover other scenarios that have
            not been tested or deployed. All the known deployments are
            in controlled datacenters. The TCP Prague effort should
            result in a draft that covers additional requirements that
            will make DCTCP work well on high latency links. <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          Many of the comments argue with your choices of MUST, SHOULD,
          MAY etc. This might seem nit-picking, given the primary
          purpose is to document the algo. However, it would be wrong to
          require things that aren't required or to allow exceptions
          when exceptions would not be interoperable. An alternative
          approach would be to just remove all capitalised language.<br>
          <br>
          <o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            I am fine with removing such capitalization except for when
            removing it will lead to interop problems between different
            implementations of DCTCP. I am responding to each comment
            below and adding whether the new revision will fix it not.</span><o:p></o:p></p>
        <p class="MsoNormal"><br>
          <b><u>Section-by-section review.<br>
            </u></b><br>
          <b>Abstract<br>
          </b><br>
          I think it needs to have an applicability statement in the
          abstract (and introduction) that refers to the deployment and
          implementation status sections at the end. Something like:<br>
          <br>
          "This is an informational specification of the implementation
          of DCTCP in Microsoft Windows Server 2012. It is applicable to
          deployments in controlled environments like data centres but
          it MUST NOT be deployed over the public Internet without
          additional measures, as detailed in sections 4-6. "<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            This is already addressed in the introduction. I don’t see
            the value of repeating it multiple times. An implementer
            will read the introduction. If you still feel strongly, we
            can add it to the abstract and remove it from the
            introduction section. This draft will not cover any
            additional measures that are intended for deployments
            outside the datacenter – that’s not the intended purpose. </span></p>
      </div>
    </blockquote>
    [BB] It is very common for the IETF to require even much less minor
    applicability statements to be highlighted in the abstract (e.g. see
    TFO RFC). This is because nowadays implementers / readers often skip
    introductory text assuming they understand the problem, etc. and
    they just scan for the normative parts. Anyone reading an IETF RFCs
    assumes the context is the public Internet, so any RFC that doesn't
    have that context needs to flag that. I know the name says Data
    Centre, but the abstract needs to explain that this is not an IETF
    RFC that is standardizing DCTCP for the public Internet. <br>
    <br>
    If you still don't think so, we should put this to the WG (which
    will slow it down).<br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>1. Intro<br>
          </b>s/its many servers/<br>
           /their many servers/<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fixed in next revision.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          "...limited queue capacities..." <br>
          I understand that the motivation has been abbreviated, but I
          think this has lost too much. Perhaps mention memory shared
          between interfaces, so either bloated buffers or too little,
          making traffic susceptible to packet losses?<br>
          <br>
          <tt><span style="font-size:10.0pt">   [RFC3168] describes a
              mechanism for using Explicit Congestion</span></tt><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;"><br>
            <tt>   Notification (ECN) from the switches for early
              detection of</tt><br>
            <tt>   congestion, rather than waiting for packet loss to
              occur.</tt><br>
          </span><br>
          This isn't correct. 3168 requires ECN marking to occur no
          earlier than drop. It is precisely that 3168 does not allow
          early marking unless there is also early drop that is the
          problem.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fixed in next revision.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <tt><span style="font-size:10.0pt">   It is recommended that
              DCTCP be deployed...</span></tt><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;"><br>
          </span><br>
          "recommended" is too weak. I suggest you use something like
          the applicability text given above, both here and in the
          abstract.<span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Added the word “only” to make this stronger per Michael’s
            review. Fixed in next revision.</span></p>
      </div>
    </blockquote>
    [BB] I suggest the applicability para (here in the intro) refers
    forward to section 5 for details (e.g. so you don't have to explain
    why it shouldn't be deployed in the intro).<br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <br>
          <b>2. Terminology<br>
          </b><br>
          Suggest you explain that capitalised words are not quite the
          same as in most RFCs:<br>
          "Normative language is used to describe how necessary the
          various aspects of the Microsoft implementation are for
          interoperability, but even compliant implementations without
          the measures in sections 4-6 would still only be safe to
          deploy in controlled environments."<br>
          <br>
          <span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fair enough. Added text in next revision.</span><o:p></o:p></p>
        <p class="MsoNormal"><o:p> </o:p></p>
        <p class="MsoNormal"><br>
          <b>3. DCTCP Algo<br>
          </b><br>
          The three bullets say no more than "this is normal stuff".
          Would it be better to supplement each sentence with a brief
          phrase saying what is different about each aspect compared to
          RFC3168?<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            The goal here it to describe the changes needed on top of
            3168 so I don’t think further explanation is necessary. <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>3.1 Marking<br>
          </b><br>
          This needs to say something about how you mark the IP header
          from L2. I suggest you refer to the up-and-forward mode of
          draft-ietf-tsvwg-ecn-encap.<br>
          <br>
          <o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Why is that relevant to this document? It is not intended as
            a specification for operation of L2 switches. Won’t fix.</span></p>
      </div>
    </blockquote>
    [BB] Well, you specify a lot about the marking algo, e.g. <br>
    K &gt; (RTT *  C)/7<br>
    <br>
    A simple solution would be: s/switch/L3 switch/ <br>
    <br>
    I just thought that a reader would think, "You say switches (i.e. L2
    devices) mark the IP header - how can that work?" unless you
    explain.<br>
    <br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><o:p></o:p></p>
        <p class="MsoNormal"><br>
          s/sending rate/<br>
           /link rate/<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fixed in next revision<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>3.2 <br>
          </b><br>
          CURRENT:<o:p></o:p></p>
        <pre>   1.  If the CE codepoint is set and DCTCP.CE is false, send an ACK for<o:p></o:p></pre>
        <pre>       any previously unacknowledged packets and set DCTCP.CE to true.<o:p></o:p></pre>
        <pre><o:p> </o:p></pre>
        <pre>   2.  If the CE codepoint is not set and DCTCP.CE is true, send an ACK<o:p></o:p></pre>
        <pre>       for any previously unacknowledged packets and set DCTCP.CE to<o:p></o:p></pre>
        <pre>       false.<o:p></o:p></pre>
        <p class="MsoNormal">SUGGESTED:<o:p></o:p></p>
        <pre>   1.  If the CE codepoint is set and DCTCP.CE is false, set DCTCP.CE to <o:p></o:p></pre>
        <pre>       true and send an ACK for any previously unacknowledged packets.<o:p></o:p></pre>
        <pre><o:p> </o:p></pre>
        <pre>   2.  If the CE codepoint is not set and DCTCP.CE is true, set DCTCP.CE<o:p></o:p></pre>
        <pre>       to false and send an ACK for any previously unacknowledged packets.<o:p></o:p></pre>
        <p class="MsoNormal" style="margin-bottom:12.0pt">    The figure
          would have to be changed to be consistent too.<br>
          RATIONALE:<br>
          If the receiver changes DCTCP.CE /after/ sending the ACK like
          this, it delays each signal until the following ACK is sent.
          That could add a delay of anything from 1 to m packets. I've
          never understood why the order of these operations was
          specified in this way. If there is no reason, then I suggest
          that it is specified in the opposite order, as I describe
          above.<span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><br>
          A note could be added to say the Windows Server 12
          implementation does these steps in reverse order, but the
          order specified is preferred because it reduces signalling
          delay and has no interoperability issues with the original
          Windows implementation. <o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Won’t fix. This is how the Windows implementation behaves so
            the ordering is required. Both the paper and the draft are
            consistent in this. </span></p>
      </div>
    </blockquote>
    [BB] The paper and the draft are not consistent. In the draft, the
    labels on the top and bottom arrows in the diagram have been
    switched round relative to those in the paper. <br>
    <br>
    However, the text in the draft has not (yet?) been changed to
    reflect this. <br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">There
            is no delay being introduced here. This ACK is sent
            immediately so the order does not matter. </span></p>
      </div>
    </blockquote>
    <br>
    [BB] I agree that the ACK itself is not delayed. However, even tho a
    CE has been received, feedback of this event is held back until the
    next ACK. <br>
    <br>
    This is a minor 'unnecessary delay' bug. <br>
    * I believe it was fixed in the FreeBSD implementation after I
    pointed it out. <br>
    * I've just checked the Linux code, and it's not been fixed there. <br>
    * Obviously I cannot see the Windows source code. <br>
    <br>
    If this bug is still in the Windows implementation, it's up to you
    if you also want the spec to require that the bugs are implemented
    exactly as they are in Windows ;)<br>
    <br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <pre><span style="color:#1F497D"><o:p> </o:p></span><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span>      </pre>
        <pre>   The handling of the "Congestion Window Reduced" (CWR) bit is also<o:p></o:p></pre>
        <pre>   exactly as per [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>] including [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-RFC3168-ERRATA3639&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=wDJv%2bcypXj5layfzvQGhXYqB6kwl8lW%2fpdQ4iYg2cN8%3d">RFC3168-ERRATA3639</a>].  That is, on<o:p></o:p></pre>
        <pre>   receipt of a segment with both the CE and CWR bits set, CWR is<o:p></o:p></pre>
        <pre>   processed first and then ECE is processed.<o:p></o:p></pre>
        <p class="MsoNormal">Even tho this is the receiver section, I
          suggest:<br>
          s/The handling/Receiver handling/<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Isn’t this assumed by a reader familiar with 3168? But this
            is a minor editorial fix so fixed in the next revision.<o:p></o:p></span></p>
        <p class="MsoNormal"><br>
          Whatever, this cannot be correct. A DCTCP receiver surely does
          nothing on receipt of CWR, whereas an RFC3168 receiver does
          something. A DCTCP receiver certainly cannot do exactly what
          RFC3168 says, because that says stop setting ECE on receipt of
          CWR until the next CE arrives, which would surely break this
          feedback protocol.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            I don’t see why this would break the feedback protocol. The
            receiver will start echoing again as soon as it sees the
            first CE arrives again. If packet has both CWR and CE set
            then CE takes precedence and the echoing is done.</span></p>
      </div>
    </blockquote>
    [BB] I'm not saying it breaks the feedback protocol. I'm just saying
    it's confusing for the reader.<br>
    <br>
    Of course, when the receiver is already not doing something,
    ignoring a signal to stop has the same outcome as stopping. However,
    describing this as "handling CWR ... is also exactly as per RFC3168"
    is misleading for the reader/implementer.<br>
    <br>
    I guess changing to "handling CWR ... is as per RFC3168" would be a
    compromise.<br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>3.3 </b><o:p></o:p></p>
        <pre>   DCTCP.Alpha, which is initialized to 1 and MUST be updated<o:p></o:p></pre>
        <pre>   as follows:<o:p></o:p></pre>
        <p class="MsoNormal">s/MUST/SHOULD/<br>
          And add "An alternative congestion avoidance algorithm MAY be
          used, but it MUST lead to flow rate inversely proportional to
          1/M".<br>
          <br>
          Rationale: There are many ways to implement interoperable 1/p
          cc. This is just one. <o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Alternate algorithms are out of scope of this draft. We
            cannot say without interop testing if a different algorithm
            will perform well in datacenters or interop well with
            existing implementations. If there are other ways to
            implement 1/p they should be covered by other drafts. This
            draft will only describe the algorithm that has been tested
            and deployed. That said I am willing to make it a SHOULD so
            fixed in the next revision.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <pre>   DCTCP.BytesSent<o:p></o:p></pre>
        <p class="MsoNormal" style="margin-bottom:12.0pt">This variable
          actually counts bytes received. it's pretty confusing to call
          it BytesSent.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            No, this is tracking bytes that have been sent. </span></p>
      </div>
    </blockquote>
    [BB] I meant it counts bytes now acknowledged as received by the
    other end (which is not the same as bytes sent by this end). <br>
    DCTCP.BytesAcked would have been more meaningful. I'm not insisting
    on this - the original comment was just to improve
    comprehensibility.<br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><br>
          s/TCP SACK options [RFC2018] are ignored./<br>
           /The sender MUST ignore TCP SACK options [RFC2018] for this
          computation./<br>
          RATIONALE: <br>
            a) I think the idea is to ignore CE marks after a gap
          because, if the gap becomes considered a loss, the response to
          loss will override any need to count incoming CE marks for the
          following round trip. However, if the gap is filled before it
          is considered a loss {Note 1}, then CE marks that were ignored
          because they arrived after a gap will need to be "unignored".
          <br>
            b) We need to make clear that SACK is not ignored
          altogether, only for this computation.<br>
          <br>
          {Note 1}: I can think of two cases:<br>
          * Less than three dupacks then the gap is filled<br>
          * The gap is filled by a delayed segment so that an earlier
          retransmission was spurious<o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Ok I will update the text.<o:p></o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <pre>the sender MUST update cwnd as follows:<o:p></o:p></pre>
        <pre>      cwnd = cwnd * (1 - DCTCP.Alpha / 2)<o:p></o:p></pre>
        <p class="MsoNormal">s/MUST/SHOULD/<br>
          Rationale: Alternative interoperable response functions are
          possible. For instance an algorithm like Relentless TCP that
          simply reduces cwnd by half the size of any CE-marked segment.
          Then there is no Alpha and no g variable.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fixed<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          CURRENT:<o:p></o:p></p>
        <pre>   Just as specified in [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>], TCP should not react to congestion<o:p></o:p></pre>
        <pre>   indications more than once for every window of data.<o:p></o:p></pre>
        <p class="MsoNormal">SUGGESTED:<o:p></o:p></p>
        <pre>   Just as specified in [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>], DCTCP does not react to congestion<o:p></o:p></pre>
        <pre>   indications more than once for every window of data.<o:p></o:p></pre>
        <p class="MsoNormal">I don't know whether you intended this to
          be an upper-case SHOULD NOT. If so, I would argue against this
          being normative, because it isn't necessary for interop;
          therefore I've suggested avoiding he 'should' word.<o:p></o:p></p>
        <p class="MsoNormal"><o:p> </o:p></p>
        <pre>&lt;Praveen&gt; Fixed<o:p></o:p></pre>
        <p class="MsoNormal"><o:p> </o:p></p>
        <p class="MsoNormal"><o:p> </o:p></p>
        <p class="MsoNormal"><o:p> </o:p></p>
        <pre>   The setting of<o:p></o:p></pre>
        <pre>   the "Congestion Window Reduced" (CWR) bit is also exactly as per<o:p></o:p></pre>
        <pre>   [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2frfc3168&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=Ce9m5q0XzkbQNWEb1rqmXrqx12Uf4LDscrUAz5C78eY%3d" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>].<o:p></o:p></pre>
        <pre><o:p> </o:p></pre>
        <pre> <o:p></o:p></pre>
        <p class="MsoNormal">It is wrong to say "exactly". The sender
          certainly sets CWR when it does a congestion response. But in
          DCTCP the congestion response occurs every RTT, triggered by
          its own measurement-period logic, whereas in RFC3168 it is
          triggered by the arrival of an ECE after the ACK of any
          previous segment it sent with CWR set.<br>
          <br>
          I think it is worth saying that setting CWR is necessary for
          safety, otherwise in the case where there has been a
          configuration management error and the receiver complies with
          RFC 3168 not DCTCP, it will continue sending ECE for ever and
          starve the session.<br>
          <br>
          It might help to explicitly confirm that: <br>
          "Other aspects of TCP congestion avoidance and control such as
          the slow start phase are no different to those in RFC5681."<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fixed that CWR is needed for safety. Not adding the text
            around 5681, it is presumed.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>3.4 SYN, SYN-ACK, RST<br>
          </b><br>
          <tt><span style="font-size:10.0pt">   [RFC3168] requires that
              a compliant TCP MUST NOT set ECT on SYN or</span></tt><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;"><br>
            <tt>   SYN-ACK packets.</tt><br>
          </span><br>
          s/MUST NOT/'MUST NOT'/<br>
          Rationale: Best to make it clear this is quoting a normative
          statement in another RFC, because it is going to be
          contradicted.<br>
        </p>
      </div>
    </blockquote>
    [BB] I don't know whether you saw this one - it's not been changed
    in draft-02.<br>
    <br>
    I suggested adding quotes around 'MUST NOT'. This is good practice
    when quoting another RFC. Otherwise, to a careless reader (or
    non-native English speaker), it can look like it is also a
    requirement of this RFC.<br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"> <br>
          CURRENT<o:p></o:p></p>
        <pre>   These RFCs, however, are<o:p></o:p></pre>
        <pre>   intended for general Internet use, and do not directly apply to a<o:p></o:p></pre>
        <pre>   controlled datacenter environment.<o:p></o:p></pre>
        <pre>...<o:p></o:p></pre>
        <pre>   For DCTCP connections, the sender<o:p></o:p></pre>
        <pre>   SHOULD set ECT for SYN, SYN-ACK and RST packets.<o:p></o:p></pre>
        <p class="MsoNormal">SUGGESTED:<br>
          <br>
          <tt><span style="font-size:10.0pt">   Also, a SYN is sent
              before the ECN-capability has been</span></tt><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;"><br>
            <tt>   negotiated, so if it is CE-marked, a non-ECN TCP
              server would not</tt><br>
            <tt>   recognise the marking. RFC 3168 does not specify the
              ECN field on </tt><br>
            <tt>   a RST. The security concerns addressed by these RFCs
            </tt><br>
            <tt>   might not apply in controlled environments like data
              centres, and </tt><br>
            <tt>   it might not be necessary to cater for the presence
              of non-ECN </tt><br>
            <tt>   servers.</tt><br>
            <tt>...</tt></span><o:p></o:p></p>
        <pre>   Therefore, the sender MUST set ECT for SYN/ACK and RST packets and a <o:p></o:p></pre>
        <pre>   configuration option SHOULD be provided to set ECT for SYNs.<o:p></o:p></pre>
        <p class="MsoNormal">RATIONALE:<br>
             The draft should avoid assuming that security concerns do
          not apply in all controlled environments.<br>
          NOTE:<br>
             This might not reflect the way your implementation is
          coded, so you might want to make that clear.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Adding some text around security concerns.</span></p>
      </div>
    </blockquote>
    [BB] The security sentence you have added in draft-02 doesn't make
    sense where it is. It ought to have been added once sentence earlier
    (and it would be better to start a new para). Specifically:<br>
    CURRENT:<br>
    <tt>                be dropped with high probability.  For DCTCP
      connections, the sender    </tt><tt><br>
    </tt><tt>                SHOULD set ECT for SYN, SYN-ACK and RST
      packets.  The security    </tt><tt><br>
    </tt><tt>                concerns addressed by both these RFCs might
      not apply in controlled    </tt><tt><br>
    </tt><tt>                environments like datacenters, and it might
      not be necessary to cater    </tt><tt><br>
    </tt><tt>                to both the presence of non-ECN servers.</tt><tt><br>
    </tt>SUGGESTED:<br>
    <tt>                be dropped with high probability.  </tt><tt><tt><br>
        &lt;NEW PARA&gt;<br>
        <tt>                </tt>The security </tt><tt>concerns
        addressed by both these RFCs might not apply in controlled    </tt><tt><br>
      </tt><tt>                environments like datacenters, and it
        might not be necessary to cater    </tt><tt><br>
      </tt><tt>                <tt>for</tt> the presence of non-ECN
        servers. </tt>For DCTCP connections, the sender    </tt><tt><br>
    </tt><tt>                SHOULD set ECT for SYN, SYN-ACK and RST
      packets.  </tt><br>
    <br>
    <tt> </tt>(Note I also fixed a glitch after "cater".)<br>
    <br>
    <br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>4. Implementation Issues<br>
          </b><br>
          CURRENT:<o:p></o:p></p>
        <pre>   the implementation MUST choose a suitable<o:p></o:p></pre>
        <pre>   estimation gain.  [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-DCTCP10&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=33dRUUxyNa2tcX0xJtMl12nWwzIdHZa6j3kgAL1dxIM%3d" title="&quot;Data Center TCP (DCTCP)&quot;">DCTCP10</a>] provides a theoretical basis <o:p></o:p></pre>
        <p class="MsoNormal">SUGGEST:<o:p></o:p></p>
        <pre>   the implementation will need to choose a suitable<o:p></o:p></pre>
        <pre>   estimation gain.  [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-DCTCP10&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=33dRUUxyNa2tcX0xJtMl12nWwzIdHZa6j3kgAL1dxIM%3d" title="&quot;Data Center TCP (DCTCP)&quot;">ADCTCP10</a>] provides a theoretical basis ...<o:p></o:p></pre>
        <p class="MsoNormal" style="margin-bottom:12.0pt">RATIONALE:<br>
             a) As above, this is only one way to implement the cc, and
          others could well prove to be interoperable when tested.<br>
             b) More particularly, it would be better for an
          implementation to adapt the estimation gain to the RTT, so the
          spec ought not to imply that one value has to be chosen.<br>
             c) The later paper [ADCTCP10] provides a corrected way to
          determine the gain. From memory (I'm offline at the mo) it at
          least makes throughput inversely proportional to 1/RTT,
          whereas the approach in [DCTCP10] made it inversely
          proportional to 1/(RTT^2). I believe the Linux implementation
          uses the [ADCTCP10] approach.<o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">&lt;Praveen&gt;
          Fair enough. Updated.<o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
        <pre>   The implementation must also decide when to use DCTCP.<o:p></o:p></pre>
        <pre>...<o:p></o:p></pre>
        <pre>   likely in the same datacenter network.<o:p></o:p></pre>
        <p class="MsoNormal">The 2nd &amp; 3rd paras are a mix of
          implementation and deployment issues, but mostly deployment,
          so they might be better moved to the deployment issues
          section.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            I think they are mainly implementation issues – the fact
            that a lack of negotiation mechanism places a burden on the
            implementation to provide knobs or heuristics to use DCTCP.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          CURRENT:<o:p></o:p></p>
        <pre>   It is RECOMMENDED that an implementation deal with loss episodes in<o:p></o:p></pre>
        <pre>   the same way as conventional TCP.<o:p></o:p></pre>
        <pre>...<o:p></o:p></pre>
        <pre>   DCTCP implementation MAY also allow configuration of resetting the<o:p></o:p></pre>
        <pre>   value of DCTCP.Alpha as part of processing any loss episodes.<o:p></o:p></pre>
        <p class="MsoNormal">SUGGESTED: Move this whole para to 3.3
          Sender Behaviour section, and:<o:p></o:p></p>
        <pre>   An implementation MUST deal with loss episodes in<o:p></o:p></pre>
        <pre>   the same way as conventional TCP.<o:p></o:p></pre>
        <pre>...<o:p></o:p></pre>
        <pre>   DCTCP implementation SHOULD also reset the<o:p></o:p></pre>
        <pre>   value of DCTCP.Alpha as part of processing any loss episodes. <o:p></o:p></pre>
        <pre>   This aspect of the behaviour MAY be configurable.<o:p></o:p></pre>
        <p class="MsoNormal" style="margin-bottom:12.0pt">RATIONALE:<br>
             a) There would never be a case for not falling back to
          conventional TCP  (or do you have one?), so "RECOMMENDED" is
          too weak.<br>
             b) I have tried to interpret what you might have meant
          about resetting Alpha. I'm not sure why you think it might
          need to be configurable though?<o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">&lt;Praveen&gt;
          Yes this is a useful suggestion. Making loss handling a must
          in sender section. Reset of alpha was discussed in tcpm list,
          and Linux implementation supports it.<o:p> <br>
          </o:p></p>
      </div>
    </blockquote>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
        <pre>   ...it is also RECOMMENDED that an implementation allow<o:p></o:p></pre>
        <pre>   configuration of restarting the congestion window (cwnd) of idle<o:p></o:p></pre>
        <pre>   DCTCP connections<o:p></o:p></pre>
        <p class="MsoNormal">Where special implementation measures are
          necessary for low RTT, these should be related to the measured
          RTT, not hard-coded. There is an assumption in a number of
          places  that "data centre" implies low RTT. Of course this is
          true in a single DC, but I believe the the single admin aspect
          of a DC is what characterises DCTCP, not the low RTT aspect.
          With the modifications in [ADCTCP], DCTCP generalises to a
          network with larger RTTs as long as it is still  operated by a
          single admin. <br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            this paragraph has now been removed per Michaels’ feedback.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          CURRENT:<br>
          <tt><span style="font-size:10.0pt">   [RFC3168] forbids the
              ECN-marking of pure ACK packets, because of the</span></tt><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;"><br>
            <tt>   inability of TCP to mitigate ACK-path congestion and
              protocol-wise</tt><br>
            <tt>   preferential treatment by routers.  However, dropping
              pure ACKs -</tt><br>
            <tt>   rather than ECN marking them - has disadvantages for
              typical</tt><br>
            <tt>   datacenter traffic patterns.  Because of the
              prevalence of bursty</tt><br>
            <tt>   traffic patterns that feature transient congestion,
              dropping of ACKs</tt><br>
            <tt>   causes subsequent retransmissions. It is RECOMMENDED
              that an</tt><br>
            <tt>   implementation provide a configuration knob that will
              cause ECT to be set</tt><br>
            <tt>   on such control packes, which can be used in
              environments where such </tt> <br>
            <tt>   concerns do not apply.</tt><br>
          </span>SUGGESTED:<br>
          <tt><span style="font-size:10.0pt">   [RFC3168] forbids the
              ECN-marking of pure ACK packets, because of the</span></tt><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;"><br>
            <tt>   inability of TCP to mitigate ACK-path congestion and
              the extra </tt><br>
            <tt>   advantage to injection attackers that ECN is
              perceived to offer. </tt><br>
            <tt>   For the latter reason RFC 3168 also forbids
              ECN-marking of </tt><br>
            <tt>   retransmissions, window probes and RSTs. </tt><br>
            <br>
            <tt>   However, dropping all these control packets -</tt><br>
            <tt>   rather than ECN marking them - has considerable
              performance </tt><br>
            <tt>   disadvantages.  It is RECOMMENDED that an</tt><br>
            <tt>   implementation provide a configuration knob that will
              cause ECT to be set</tt><br>
            <tt>   on such control packes, which can be used in
              environments where such </tt> <br>
            <tt>   concerns do not apply.</tt></span><br>
          NOTE:<br>
             This might not reflect the way your implementation is
          coded, so you might want to make that clear.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Fixed</span></p>
      </div>
    </blockquote>
    [BB] A new comment...<br>
    <br>
    As pointed out in draft-bagulo-tsvwg-generalized-ecn-00, RFC3168
    only talks about injection attacks with retransmissions (not ACKs
    etc). But you imply injection attacks were the concern for other
    control packets as well.<br>
    <br>
    I'm not insisting you change this. However, unwarranted accusations
    of security vulnerabilities tend to stick, whether or not they are
    correct. So it would be better not to throw mud around onto even
    more control packets than RFC3168 did.<br>
    <br>
    Cheers<br>
    <br>
    <br>
    Bob<br>
    <br>
    <blockquote
cite="mid:BN1PR03MB008D16E15DCBE49E7C21A4AB6350@BN1PR03MB008.namprd03.prod.outlook.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><br>
          <br>
          <b>5. Deployment Issues</b><o:p></o:p></p>
        <pre>   If<o:p></o:p></pre>
        <pre>   the traffic in the datacenter is a mix of conventional TCP and DCTCP,<o:p></o:p></pre>
        <pre>   it is RECOMMENDED that DCTCP traffic be segregated from conventional<o:p></o:p></pre>
        <pre>   TCP traffic.<o:p></o:p></pre>
        <p class="MsoNormal" style="margin-bottom:12.0pt">In a data
          centre (or anywhere), is there any case where not segregating
          them would make sense? If not, this needs to be a MUST, not
          just a RECOMMENDED. I can imagine cases where there is no
          long-running conventional TCP or no long-running DCTCP, but
          that's covered by the "if" at the start of the sentence.<br>
          <br>
          You might want to refer to draft-briscoe-aqm-dualq-coupled as
          another segregation approach here too.<o:p></o:p></p>
        <pre>   Since DCTCP relies on congestion marking by the switches, DCTCP can<o:p></o:p></pre>
        <pre>   only be deployed in datacenters where the entire network<o:p></o:p></pre>
        <pre>   infrastructure supports ECN.<o:p></o:p></pre>
        <p class="MsoNormal" style="margin-bottom:12.0pt">Surely, the
          "fall-back to conventional TCP on loss" rule means you can
          delete this
constraint.                                                                       <o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">&lt;Praveen&gt;
          Reworded to imply that DCTCP will only work as expected in
          such cases.<o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
        <pre>   A<o:p></o:p></pre>
        <pre>   variant of DCTCP that can be deployed unilaterally and only requires<o:p></o:p></pre>
        <pre>   standard ECN behavior has been described in [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-ODCTCP&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=ZwfJ0LgHQ3vk%2bMD0wiia5yLrcMMSUBcGgPYXmeRu8hE%3d" title="&quot;Improving Transmission Performance with One- Sided Datacenter TCP&quot;">ODCTCP</a>][BSDCAN],<o:p></o:p></pre>
        <p class="MsoNormal"><br>
          Which part of "standard ECN behaviour" do you mean? The host
          part? The network part? The receiver part? And "can be
          deployed unilaterally" ought to be in the active voice not the
          passive, which would help resolve the ambiguity too.<br>
          <br>
          I haven't read [ODCTCP] and [BSDCAN] (I'm offline on a plane
          at the mo). Are they similar to "Instant ECN" in [Wu12] listed
          below? If not, that could be referred to as well for a
          technique that uses the regular ECN host behaviour.<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            This edit predates me and  I don’t have the full context, I
            am leaving this as is.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>6. Known Issues<br>
          </b><br>
          First para could refer to appendix A of RFC7560, which
          describes the problem precisely.<o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <pre>A method for improving the fairness of DCTCP has been proposed<o:p></o:p></pre>
        <pre>   in [<a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=https%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-ietf-tcpm-dctcp-01%23ref-ADCTCP&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=edU2TPGpLvyFGvgocDRJiYRQyjCT%2fWEOgS2MwmwzYuU%3d" title="&quot;Analysis of DCTCP: Stability, Convergence, and Fairness&quot;">ADCTCP</a>], but requires additional experimental evaluation.<o:p></o:p></pre>
        <p class="MsoNormal">s/fairness/RTT fairness/<br>
          <br>
          (In our experiments, it's much better than without.)<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Goal is to cite other work that can address potential
            issues. It is not a recommendation. s/fairness/RTT fairness/
            Fixed in next revision.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          <b>11. References<br>
          </b><br>
          I think the following are cited normatively, not just
          informatively:<br>
          RFC5681, RFC5562.<br>
          <br>
          Additional informational Reference:<br>
          <br>
          [Wu12] Wu, H., Ju, J., Lu, G., Guo, C., Xiong, Y. &amp; Zhang,
          Y., "Tuning ECN for Data Center Networks," In: Proceedings of
          the 8th International Conference on Emerging Networking
          Experiments and Technologies CoNEXT '12 pp.25-36 ACM (2012)<br>
          <br>
          <span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D">&lt;Praveen&gt;
            Citation to RFCs updated as normative. Not including the new
            reference since even though it may be relevant – it was not
            used as a reference in writing the draft.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><br>
          That's it. HTH.<br>
          <br>
          <br>
          Bob<br>
          <br>
          <o:p></o:p></p>
        <pre>-- <o:p></o:p></pre>
        <pre>________________________________________________________________<o:p></o:p></pre>
        <pre>Bob Briscoe                               <a moz-do-not-send="true" href="https://na01.safelinks.protection.outlook.com/?url=http%3a%2f%2fbobbriscoe.net%2f&amp;data=01%7c01%7cpravb%40microsoft.com%7ca572b0ba8e1b47d3e44a08d2e6d7e03f%7c72f988bf86f141af91ab2d7cd011db47%7c1&amp;sdata=iEvhWj9FCPozWAp8rnxk5VZdNF%2fv7W7aD1Cp4MxOiSs%3d">http://bobbriscoe.net/</a><o:p></o:p></pre>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
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--------------24759AAECE54BB7370BFF4EE--


From nobody Thu Aug 18 09:56:44 2016
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To: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>, Mark Nottingham <mnot@mnot.net>, "tcpm@ietf.org" <tcpm@ietf.org>
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Speaking as someone who commented on an earlier version:

I think it is good to offer advice on how TCP is used for web traffic. I 
can see there is likely useful information that can document how some 
things are done, and perhaps also recommend some changes. I support 
continuing work on this.

Publishing a document like this in the RFC series needs to be consistent 
with IETF consensus. The ID touches on many things that are assigned to 
TCPM. (The breadth of topics may be a good reason for a document - but 
is also a concern, because many things need more clear description 
before the recommendations become clear).

I think this document should NOT be considered as BCP. It talks about 
system configuration of endpoints, and of the recommendations I see, 
some do support current PSs, some set new precedents against current PS 
RFCs and in some cases the present text appears to endorse EXP RFCs. If 
published as a BCP, that sort of advice is likely to be confusing. Such 
information can also easily be outdated, I suggest if adopted, this 
should be Informational and clearly set the scope of the recommendations.

If this work continues, it needs to be reviewed by TCPM and HTTPbis. 
This would need much more than simply a last-call in both working groups 
- but I'm sure with proper coordination progress could be made so that 
there were no surprises for either group at the time of WGLC!

Gorry


>
> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Mark Nottingham
> Sent: Wednesday, August 17, 2016 4:03 AM
> To: tcpm@ietf.org
> Cc: Patrick McManus; Daniel Stenberg
> Subject: [tcpm] TCP Tuning for HTTP - update
>
> Hi TCPM,
>
> Just a quick note; Daniel and Tim have made an update to the TCP Tuning for HTTP draft:
>   https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>
> We've had a Call for Adoption open for this draft for a while, and will likely adopt it soon. However, we'd like to get feedback from this community first -- both about the latest version of the input document, and to see if there's interest in helping out.
>
> You can give feedback on the HTTP WG mailing list <https://lists.w3.org/Archives/Public/ietf-http-wg/>, or  by responding to this e-mail (Please leave the CC line; Patrick and I will try to summarise the feedback to the WG).
>
> Cheers,
>
> --
> Mark Nottingham   https://www.mnot.net/
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
> _______________________________________________
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Mark Nottingham <mnot@mnot.net>, Patrick McManus <pmcmanus@mozilla.com>, Daniel Stenberg <daniel@haxx.se>
Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On Thu, Aug 18, 2016 at 9:54 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk>
wrote:

> Speaking as someone who commented on an earlier version:
>
> I think it is good to offer advice on how TCP is used for web traffic. I
> can see there is likely useful information that can document how some
> things are done, and perhaps also recommend some changes. I support
> continuing work on this.
>
> Publishing a document like this in the RFC series needs to be consistent
> with IETF consensus. The ID touches on many things that are assigned to
> TCPM. (The breadth of topics may be a good reason for a document - but is
> also a concern, because many things need more clear description before the
> recommendations become clear).
>
> I think this document should NOT be considered as BCP. It talks about
> system configuration of endpoints, and of the recommendations I see, some
> do support current PSs, some set new precedents against current PS RFCs and
> in some cases the present text appears to endorse EXP RFCs. If published as
> a BCP, that sort of advice is likely to be confusing. Such information can
> also easily be outdated, I suggest if adopted, this should be Informational
> and clearly set the scope of the recommendations.
>
+1 on informational

It'd be great for the document to cite more detailed technical reports, for
readers to learn the pros and cons, and the history. IMO it'd significantly
improve this document. The reason is that TCP and network will continue to
evolve, so knowing the past and details can help readers to judge whether a
specific recommendation is still useful or not, if he is willing to spend
time to explore. For example, he can use some online tools to track the
citation of these reports and see new reports.

For the document, I'd recommend two more items that my personal experience
gives remarkable performance enhancement on reducing and recover from
losses.

1) make sure your stack supports SACK
2) enable pacing.

On disabling Nagle, there are also better APIs like Linux Cork so you don't
have to send small packets even on continuous writes. One common problem I
often see is developers unconditionally disable Nagle, but their HTTP
workload may lead to lots of small packets.

my 2c


> If this work continues, it needs to be reviewed by TCPM and HTTPbis. This
> would need much more than simply a last-call in both working groups - but
> I'm sure with proper coordination progress could be made so that there were
> no surprises for either group at the time of WGLC!
>
> Gorry
>
>
>
>
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Mark Nottingham
>> Sent: Wednesday, August 17, 2016 4:03 AM
>> To: tcpm@ietf.org
>> Cc: Patrick McManus; Daniel Stenberg
>> Subject: [tcpm] TCP Tuning for HTTP - update
>>
>> Hi TCPM,
>>
>> Just a quick note; Daniel and Tim have made an update to the TCP Tuning
>> for HTTP draft:
>>   https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>>
>> We've had a Call for Adoption open for this draft for a while, and will
>> likely adopt it soon. However, we'd like to get feedback from this
>> community first -- both about the latest version of the input document, and
>> to see if there's interest in helping out.
>>
>> You can give feedback on the HTTP WG mailing list <
>> https://lists.w3.org/Archives/Public/ietf-http-wg/>, or  by responding
>> to this e-mail (Please leave the CC line; Patrick and I will try to
>> summarise the feedback to the WG).
>>
>> Cheers,
>>
>> --
>> Mark Nottingham   https://www.mnot.net/
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--94eb2c04a26c44c108053a5bffc6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Aug 18, 2016 at 9:54 AM, Gorry Fairhurst <span dir=3D"ltr">&lt;=
<a href=3D"mailto:gorry@erg.abdn.ac.uk" target=3D"_blank">gorry@erg.abdn.ac=
.uk</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Speaking as som=
eone who commented on an earlier version:<br>
<br>
I think it is good to offer advice on how TCP is used for web traffic. I ca=
n see there is likely useful information that can document how some things =
are done, and perhaps also recommend some changes. I support continuing wor=
k on this.<br>
<br>
Publishing a document like this in the RFC series needs to be consistent wi=
th IETF consensus. The ID touches on many things that are assigned to TCPM.=
 (The breadth of topics may be a good reason for a document - but is also a=
 concern, because many things need more clear description before the recomm=
endations become clear).<br>
<br>
I think this document should NOT be considered as BCP. It talks about syste=
m configuration of endpoints, and of the recommendations I see, some do sup=
port current PSs, some set new precedents against current PS RFCs and in so=
me cases the present text appears to endorse EXP RFCs. If published as a BC=
P, that sort of advice is likely to be confusing. Such information can also=
 easily be outdated, I suggest if adopted, this should be Informational and=
 clearly set the scope of the recommendations.<br></blockquote><div>+1 on i=
nformational=C2=A0</div><div><br></div><div>It&#39;d be great for the docum=
ent to cite more detailed technical reports, for readers to learn the pros =
and cons, and the history. IMO it&#39;d significantly improve this document=
. The reason is that TCP and network will continue to evolve, so knowing th=
e past and details can help readers to judge whether a specific recommendat=
ion is still useful or not, if he is willing to spend time to explore. For =
example, he can use some online tools to track the citation of these report=
s and see new reports.</div><div><br></div><div>For the document, I&#39;d r=
ecommend two more items that my personal experience gives remarkable perfor=
mance enhancement on reducing and recover from losses.</div><div><br></div>=
<div>1) make sure your stack supports SACK</div><div>2) enable pacing.=C2=
=A0</div><div><br></div><div>On disabling Nagle, there are also better APIs=
 like Linux Cork so you don&#39;t have to send small packets even on contin=
uous writes. One common problem I often see is developers unconditionally d=
isable Nagle, but their HTTP workload may lead to lots of small packets.=C2=
=A0</div><div><br></div><div>my 2c</div><div><br></div><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex"><br>
If this work continues, it needs to be reviewed by TCPM and HTTPbis. This w=
ould need much more than simply a last-call in both working groups - but I&=
#39;m sure with proper coordination progress could be made so that there we=
re no surprises for either group at the time of WGLC!<span class=3D"m_-8639=
438188464410741HOEnZb"><font color=3D"#888888"><br>
<br>
Gorry</font></span><div class=3D"m_-8639438188464410741HOEnZb"><div class=
=3D"m_-8639438188464410741h5"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
-----Original Message-----<br>
From: tcpm [mailto:<a href=3D"mailto:tcpm-bounces@ietf.org" target=3D"_blan=
k">tcpm-bounces@ietf.org</a>] On Behalf Of Mark Nottingham<br>
Sent: Wednesday, August 17, 2016 4:03 AM<br>
To: <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br=
>
Cc: Patrick McManus; Daniel Stenberg<br>
Subject: [tcpm] TCP Tuning for HTTP - update<br>
<br>
Hi TCPM,<br>
<br>
Just a quick note; Daniel and Tim have made an update to the TCP Tuning for=
 HTTP draft:<br>
=C2=A0 <a href=3D"https://tools.ietf.org/html/draft-stenberg-httpbis-tcp" r=
el=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/dr<wbr>aft-=
stenberg-httpbis-tcp</a><br>
<br>
We&#39;ve had a Call for Adoption open for this draft for a while, and will=
 likely adopt it soon. However, we&#39;d like to get feedback from this com=
munity first -- both about the latest version of the input document, and to=
 see if there&#39;s interest in helping out.<br>
<br>
You can give feedback on the HTTP WG mailing list &lt;<a href=3D"https://li=
sts.w3.org/Archives/Public/ietf-http-wg/" rel=3D"noreferrer" target=3D"_bla=
nk">https://lists.w3.org/Archives<wbr>/Public/ietf-http-wg/</a>&gt;, or=C2=
=A0 by responding to this e-mail (Please leave the CC line; Patrick and I w=
ill try to summarise the feedback to the WG).<br>
<br>
Cheers,<br>
<br>
--<br>
Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"norefe=
rrer" target=3D"_blank">https://www.mnot.net/</a><br>
<br>
<br>
<br>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
<br>
</blockquote>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div>

--94eb2c04a26c44c108053a5bffc6--


From nobody Thu Aug 18 10:55:52 2016
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To: Willy Tarreau <w@1wt.eu>
References: <5CD67877-19E3-4E79-BBF2-3E270343A378@mnot.net> <2197232f-10d7-28cb-fcc9-05bd495e3c22@isi.edu> <20160817064545.GD16017@1wt.eu> <7f7b129c-f156-d067-bef8-4a2213f461ac@isi.edu> <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu>
From: Joe Touch <touch@isi.edu>
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Cc: tcpm@ietf.org, Matthew Kerwin <matthew@kerwin.net.au>, Daniel Stenberg <daniel@haxx.se>, Mark Nottingham <mnot@mnot.net>, HTTP Working Group <ietf-http-wg@w3.org>, Patrick McManus <pmcmanus@mozilla.com>
Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi, Willy,

On the issue of ACK compression effects:


On 8/17/2016 10:38 PM, Willy Tarreau wrote:
>>     - watch out for ACK compression effects (turn it off in favor of ABC
>> > if you can)
> It does not happen that much with HTTP. Many connections on the server side
> see only one, sometimes two requests, and most responses are small (about
> 20kB on average, with favicon fitting in a single segment). Note, I'm talking
> about observations on average web sites.
The effect is more pronounced for smaller responses, for any response
using an odd number of packets. The last odd packet will be stalled
because the client needs to timeout before it will send an ACK for a
single segment (it's waiting for the second segment).

It doesn't matter how many requests you have, but the impact can be
complicated on persistent connections.

Joe

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<html>
  <head>
    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <p>Hi, Willy,</p>
    <p>On the issue of ACK compression effects:<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 8/17/2016 10:38 PM, Willy Tarreau
      wrote:<br>
    </div>
    <blockquote cite="mid:20160818053837.GC16773@1wt.eu" type="cite">
      <blockquote type="cite" style="color: #000000;">
        <pre wrap="">    - watch out for ACK compression effects (turn it off in favor of ABC
<span class="moz-txt-citetags">&gt; </span>if you can)
</pre>
      </blockquote>
      <pre wrap="">It does not happen that much with HTTP. Many connections on the server side
see only one, sometimes two requests, and most responses are small (about
20kB on average, with favicon fitting in a single segment). Note, I'm talking
about observations on average web sites.</pre>
    </blockquote>
    The effect is more pronounced for smaller responses, for any
    response using an odd number of packets. The last odd packet will be
    stalled because the client needs to timeout before it will send an
    ACK for a single segment (it's waiting for the second segment).<br>
    <br>
    It doesn't matter how many requests you have, but the impact can be
    complicated on persistent connections.<br>
    <br>
    Joe<br>
  </body>
</html>

--------------F3F89CB6AD11B845A96AEA1A--


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Date: Thu, 18 Aug 2016 21:59:48 +0200
From: Willy Tarreau <w@1wt.eu>
To: Joe Touch <touch@isi.edu>
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References: <20160817064545.GD16017@1wt.eu> <7f7b129c-f156-d067-bef8-4a2213f461ac@isi.edu> <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu> <3c93805c-05f5-2b69-235e-e1c68363bba6@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On Thu, Aug 18, 2016 at 10:50:52AM -0700, Joe Touch wrote:
> Hi, Willy,
> 
> On the issue of ACK compression effects:
> 
> 
> On 8/17/2016 10:38 PM, Willy Tarreau wrote:
> >>     - watch out for ACK compression effects (turn it off in favor of ABC
> >> > if you can)
> > It does not happen that much with HTTP. Many connections on the server side
> > see only one, sometimes two requests, and most responses are small (about
> > 20kB on average, with favicon fitting in a single segment). Note, I'm talking
> > about observations on average web sites.
> The effect is more pronounced for smaller responses, for any response
> using an odd number of packets. The last odd packet will be stalled
> because the client needs to timeout before it will send an ACK for a
> single segment (it's waiting for the second segment).

With short keep-alive responses yes but not short-lived connections
since the server sends the FIN immediately afterwards, which hints
the client not to wait for anything else and to ACK immediately.

> It doesn't matter how many requests you have, but the impact can be
> complicated on persistent connections.

Yes I agree. Please note that usually the last segment of a message
carries a push which precisely speeds up delivery to the application
and acking (as most OSes ack on push but apparently not all), so it
is not often emphasized that much. But I've met some 3G networks
where ACK compression was causing excessive retransmits from the
sender, resulting in excess retransmits of ACKs in turn, maintaining
the ACK link congested. It generally gets worse with many parallel
connections than with a single one, and in this regard HTTP/2 has
improved things a lot.

Willy


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To: Willy Tarreau <w@1wt.eu>
References: <20160817064545.GD16017@1wt.eu> <7f7b129c-f156-d067-bef8-4a2213f461ac@isi.edu> <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu> <3c93805c-05f5-2b69-235e-e1c68363bba6@isi.edu> <20160818195948.GD17944@1wt.eu>
From: Joe Touch <touch@isi.edu>
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Date: Thu, 18 Aug 2016 14:52:31 -0700
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 8/18/2016 12:59 PM, Willy Tarreau wrote:
> On Thu, Aug 18, 2016 at 10:50:52AM -0700, Joe Touch wrote:
>> Hi, Willy,
>>
>> On the issue of ACK compression effects:
>>
>>
>> On 8/17/2016 10:38 PM, Willy Tarreau wrote:
>>>>     - watch out for ACK compression effects (turn it off in favor of ABC
>>>>> if you can)
>>> It does not happen that much with HTTP. Many connections on the server side
>>> see only one, sometimes two requests, and most responses are small (about
>>> 20kB on average, with favicon fitting in a single segment). Note, I'm talking
>>> about observations on average web sites.
>> The effect is more pronounced for smaller responses, for any response
>> using an odd number of packets. The last odd packet will be stalled
>> because the client needs to timeout before it will send an ACK for a
>> single segment (it's waiting for the second segment).
> With short keep-alive responses yes but not short-lived connections
> since the server sends the FIN immediately afterwards, which hints
> the client not to wait for anything else and to ACK immediately.
If you have connection-per-transfer, but not if you have persistent
connections (the server wouldn't issue a close()).

>> It doesn't matter how many requests you have, but the impact can be
>> complicated on persistent connections.
> Yes I agree. Please note that usually the last segment of a message
> carries a push which precisely speeds up delivery to the application
> and acking (as most OSes ack on push but apparently not all), so it
> is not often emphasized that much.
PSH is optional.

The PSH bit forces data not to be aggregated by the sending TCP (i.e.,
to gather multiple send() calls) and forces the receiver to do the same,
but I don't see anywhere that PSH forces ACK compression to be
circumvented. If you have a pointer that'd be useful (I'm speaking of an
RFC). I looked for "delayed ACK" (the term used in 4.2.3.2 of RFC 1122)
and could not find any indication of a relation to PSH there, in fact
the delayed ACK discussion has no exceptions at all.

Additionally, that were true, setting the PSH bit constantly could cause
TCPs to open their slow-start windows much faster (2x per RTT, rather
than 1.5x as currently).

> But I've met some 3G networks
> where ACK compression was causing excessive retransmits from the
> sender, resulting in excess retransmits of ACKs in turn, maintaining
> the ACK link congested. It generally gets worse with many parallel
> connections than with a single one, and in this regard HTTP/2 has
> improved things a lot.

Most TCP variant assume that loss=congestion; that's often the wrong
interpretation for wireless.

(exceptions include TCP variants tuned for wireless)

Joe


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From: Willy Tarreau <w@1wt.eu>
To: Joe Touch <touch@isi.edu>
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References: <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu> <3c93805c-05f5-2b69-235e-e1c68363bba6@isi.edu> <20160818195948.GD17944@1wt.eu> <d2ca2787-6236-fadb-80e7-53e51f42382f@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On Thu, Aug 18, 2016 at 02:52:31PM -0700, Joe Touch wrote:
> The PSH bit forces data not to be aggregated by the sending TCP (i.e.,
> to gather multiple send() calls) and forces the receiver to do the same,
> but I don't see anywhere that PSH forces ACK compression to be
> circumvented.

No that's not what I was saying, just that it was often avoiding the
extra 200ms delay before sending the ACK for the odd segment number
which further amplifies the ACK compression issue.

> If you have a pointer that'd be useful (I'm speaking of an
> RFC). I looked for "delayed ACK" (the term used in 4.2.3.2 of RFC 1122)
> and could not find any indication of a relation to PSH there, in fact
> the delayed ACK discussion has no exceptions at all.

On some systems you can force to have delayed ACKs even on PSH, I do it
on linux (disable quick-ack). That's very useful to merge the response
with the ACK for the request. But these are tricky must not be abused.

> Additionally, that were true, setting the PSH bit constantly could cause
> TCPs to open their slow-start windows much faster (2x per RTT, rather
> than 1.5x as currently).

I wasn't aware.

> > But I've met some 3G networks
> > where ACK compression was causing excessive retransmits from the
> > sender, resulting in excess retransmits of ACKs in turn, maintaining
> > the ACK link congested. It generally gets worse with many parallel
> > connections than with a single one, and in this regard HTTP/2 has
> > improved things a lot.
> 
> Most TCP variant assume that loss=congestion; that's often the wrong
> interpretation for wireless.
> (exceptions include TCP variants tuned for wireless)

Yep definitely. And it's even worse with deep buffers at the intersection
of fast and slow networks like the RAN for some mobile operators because
the stacks here tend to be tuned to assume that a loss is a loss and cause
fast retransmit which further fill the RAN's buffers. Sometimes the uplink
is filled with ACKs and I even found it could be efficient to kill one every
two ACKs in such a case. But I think this asymmetry is progressively going
away in favor of more balanced links.

Willy


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To: Willy Tarreau <w@1wt.eu>
References: <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu> <3c93805c-05f5-2b69-235e-e1c68363bba6@isi.edu> <20160818195948.GD17944@1wt.eu> <d2ca2787-6236-fadb-80e7-53e51f42382f@isi.edu> <20160818220557.GA18091@1wt.eu>
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi, Willy,

Yes - this is yet another example of TCP's "twisty maze of passages, all
alike" ;-)

Joe


On 8/18/2016 3:05 PM, Willy Tarreau wrote:
> On Thu, Aug 18, 2016 at 02:52:31PM -0700, Joe Touch wrote:
>> The PSH bit forces data not to be aggregated by the sending TCP (i.e.,
>> to gather multiple send() calls) and forces the receiver to do the same,
>> but I don't see anywhere that PSH forces ACK compression to be
>> circumvented.
> No that's not what I was saying, just that it was often avoiding the
> extra 200ms delay before sending the ACK for the odd segment number
> which further amplifies the ACK compression issue.
>
>> If you have a pointer that'd be useful (I'm speaking of an
>> RFC). I looked for "delayed ACK" (the term used in 4.2.3.2 of RFC 1122)
>> and could not find any indication of a relation to PSH there, in fact
>> the delayed ACK discussion has no exceptions at all.
> On some systems you can force to have delayed ACKs even on PSH, I do it
> on linux (disable quick-ack). That's very useful to merge the response
> with the ACK for the request. But these are tricky must not be abused.
>
>> Additionally, that were true, setting the PSH bit constantly could cause
>> TCPs to open their slow-start windows much faster (2x per RTT, rather
>> than 1.5x as currently).
> I wasn't aware.
>
>>> But I've met some 3G networks
>>> where ACK compression was causing excessive retransmits from the
>>> sender, resulting in excess retransmits of ACKs in turn, maintaining
>>> the ACK link congested. It generally gets worse with many parallel
>>> connections than with a single one, and in this regard HTTP/2 has
>>> improved things a lot.
>> Most TCP variant assume that loss=congestion; that's often the wrong
>> interpretation for wireless.
>> (exceptions include TCP variants tuned for wireless)
> Yep definitely. And it's even worse with deep buffers at the intersection
> of fast and slow networks like the RAN for some mobile operators because
> the stacks here tend to be tuned to assume that a loss is a loss and cause
> fast retransmit which further fill the RAN's buffers. Sometimes the uplink
> is filled with ACKs and I even found it could be efficient to kill one every
> two ACKs in such a case. But I think this asymmetry is progressively going
> away in favor of more balanced links.
>
> Willy


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From: Yuchung Cheng <ycheng@google.com>
Date: Thu, 18 Aug 2016 17:04:03 -0700
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Subject: Re: [tcpm] Personal change in chairing
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On Wed, Aug 17, 2016 at 8:28 AM, Mirja K=C3=BChlewind <ietf@kuehlewind.net>
wrote:

> Hi all,
>
> I have to announce a personal change in the tcpm chair position:
>
> Pasi Sarolahti is stepping down due to time restrictions. Pasi, thanks a
> lot for your work and engagement as chair! Your did a great job and we ho=
pe
> to still see you as often as possible as contributor in the working group=
!
>
Thanks for your service Pasi


>
> Michael T=C3=BCxen is starting as a new tcpm chair. Micheal, great to hav=
e you
> on broad! We look forward to work with you!
>
> Mirja, as responsible AD
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Wed, Aug 17, 2016 at 8:28 AM, Mirja K=C3=BChlewind <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:ietf@kuehlewind.net" target=3D"_blank">ietf@kuehlewi=
nd.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
I have to announce a personal change in the tcpm chair position:<br>
<br>
Pasi Sarolahti is stepping down due to time restrictions. Pasi, thanks a lo=
t for your work and engagement as chair! Your did a great job and we hope t=
o still see you as often as possible as contributor in the working group!<b=
r></blockquote><div>Thanks for your service Pasi</div><div>=C2=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">
<br>
Michael T=C3=BCxen is starting as a new tcpm chair. Micheal, great to have =
you on broad! We look forward to work with you!<br>
<br>
Mirja, as responsible AD<br>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
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From nobody Thu Aug 18 20:49:38 2016
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From: Mark Nottingham <mnot@mnot.net>
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References: <5CD67877-19E3-4E79-BBF2-3E270343A378@mnot.net> <2197232f-10d7-28cb-fcc9-05bd495e3c22@isi.edu> <20160817064545.GD16017@1wt.eu> <7f7b129c-f156-d067-bef8-4a2213f461ac@isi.edu> <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu> <7a36e025-4882-4f8b-7a83-9fdcd990a971@isi.edu>
To: Joe Touch <touch@ISI.EDU>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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> On 19 Aug 2016, at 1:07 AM, Joe Touch <touch@ISI.EDU> wrote:
>=20
> I do think that this doc needs to figure out whom it is speaking to, =
what advice they actually need, etc.
>=20
> If the result is a set of recommendations that involve the word =
"sysctl", I remain skeptical it is appropriate as an RFC


I think there's broad agreement on both of these points.

I'm wondering if it makes sense to aim it primarily at HTTP implementers =
rather than administrators, with the notion that it would inform:

- Their implementation decisions
- The configuration choices they offer to administrators / users
- Their documentation (e.g., advice to their administrators when the =
implementation can't change the appropriate parts of the OS)

Would that help?=20

If so, it might make sense to organise it into sections for clients and =
servers (and intermediaries, if there's anything that isn't covered by =
the combination of the first two). Although IIRC Daniel was already =
talking about doing that.

Cheers,

--
Mark Nottingham   https://www.mnot.net/


From nobody Fri Aug 19 00:48:02 2016
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Date: Fri, 19 Aug 2016 08:47:44 +0100
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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On 18/08/2016 18:34, Yuchung Cheng wrote:
>
>
> On Thu, Aug 18, 2016 at 9:54 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk 
> <mailto:gorry@erg.abdn.ac.uk>> wrote:
>
>     Speaking as someone who commented on an earlier version:
>
>     I think it is good to offer advice on how TCP is used for web
>     traffic. I can see there is likely useful information that can
>     document how some things are done, and perhaps also recommend some
>     changes. I support continuing work on this.
>
>     Publishing a document like this in the RFC series needs to be
>     consistent with IETF consensus. The ID touches on many things that
>     are assigned to TCPM. (The breadth of topics may be a good reason
>     for a document - but is also a concern, because many things need
>     more clear description before the recommendations become clear).
>
>     I think this document should NOT be considered as BCP. It talks
>     about system configuration of endpoints, and of the
>     recommendations I see, some do support current PSs, some set new
>     precedents against current PS RFCs and in some cases the present
>     text appears to endorse EXP RFCs. If published as a BCP, that sort
>     of advice is likely to be confusing. Such information can also
>     easily be outdated, I suggest if adopted, this should be
>     Informational and clearly set the scope of the recommendations.
>
> +1 on informational
>
> It'd be great for the document to cite more detailed technical 
> reports, for readers to learn the pros and cons, and the history. IMO 
> it'd significantly improve this document. The reason is that TCP and 
> network will continue to evolve, so knowing the past and details can 
> help readers to judge whether a specific recommendation is still 
> useful or not, if he is willing to spend time to explore. For example, 
> he can use some online tools to track the citation of these reports 
> and see new reports.
>
I agree
> For the document, I'd recommend two more items that my personal 
> experience gives remarkable performance enhancement on reducing and 
> recover from losses.
>
> 1) make sure your stack supports SACK
> 2) enable pacing.
>
+1

> On disabling Nagle, there are also better APIs like Linux Cork so you 
> don't have to send small packets even on continuous writes. One common 
> problem I often see is developers unconditionally disable Nagle, but 
> their HTTP workload may lead to lots of small packets.
>
Worth metioning.
> my 2c
>
>
>     If this work continues, it needs to be reviewed by TCPM and
>     HTTPbis. This would need much more than simply a last-call in both
>     working groups - but I'm sure with proper coordination progress
>     could be made so that there were no surprises for either group at
>     the time of WGLC!
>
>     Gorry
>
>
>
>
>         -----Original Message-----
>         From: tcpm [mailto:tcpm-bounces@ietf.org
>         <mailto:tcpm-bounces@ietf.org>] On Behalf Of Mark Nottingham
>         Sent: Wednesday, August 17, 2016 4:03 AM
>         To: tcpm@ietf.org <mailto:tcpm@ietf.org>
>         Cc: Patrick McManus; Daniel Stenberg
>         Subject: [tcpm] TCP Tuning for HTTP - update
>
>         Hi TCPM,
>
>         Just a quick note; Daniel and Tim have made an update to the
>         TCP Tuning for HTTP draft:
>         https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>         <https://tools.ietf.org/html/draft-stenberg-httpbis-tcp>
>
>         We've had a Call for Adoption open for this draft for a while,
>         and will likely adopt it soon. However, we'd like to get
>         feedback from this community first -- both about the latest
>         version of the input document, and to see if there's interest
>         in helping out.
>
>         You can give feedback on the HTTP WG mailing list
>         <https://lists.w3.org/Archives/Public/ietf-http-wg/
>         <https://lists.w3.org/Archives/Public/ietf-http-wg/>>, or  by
>         responding to this e-mail (Please leave the CC line; Patrick
>         and I will try to summarise the feedback to the WG).
>
>         Cheers,
>
>         --
>         Mark Nottingham https://www.mnot.net/
>
>
>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>     <https://www.ietf.org/mailman/listinfo/tcpm>
>
>


From nobody Fri Aug 19 01:13:56 2016
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi all,=20

I agree with the recommendation on enabling pacing.
If flows are multiplexed such as in HTTP2, losses at slow start can be very=
 damaging.=20
I have proposed some text inline.=20

Cheers,

Nico

-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de Gorry Fairhurst
Envoy=E9=A0: vendredi 19 ao=FBt 2016 09:48
=C0=A0: Yuchung Cheng
Cc=A0: tcpm@ietf.org; Mark Nottingham; Patrick McManus; Daniel Stenberg
Objet=A0: Re: [tcpm] TCP Tuning for HTTP - update

On 18/08/2016 18:34, Yuchung Cheng wrote:
>
>
> On Thu, Aug 18, 2016 at 9:54 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk=20
> <mailto:gorry@erg.abdn.ac.uk>> wrote:
>
>     Speaking as someone who commented on an earlier version:
>
>     I think it is good to offer advice on how TCP is used for web
>     traffic. I can see there is likely useful information that can
>     document how some things are done, and perhaps also recommend some
>     changes. I support continuing work on this.
>
>     Publishing a document like this in the RFC series needs to be
>     consistent with IETF consensus. The ID touches on many things that
>     are assigned to TCPM. (The breadth of topics may be a good reason
>     for a document - but is also a concern, because many things need
>     more clear description before the recommendations become clear).
>
>     I think this document should NOT be considered as BCP. It talks
>     about system configuration of endpoints, and of the
>     recommendations I see, some do support current PSs, some set new
>     precedents against current PS RFCs and in some cases the present
>     text appears to endorse EXP RFCs. If published as a BCP, that sort
>     of advice is likely to be confusing. Such information can also
>     easily be outdated, I suggest if adopted, this should be
>     Informational and clearly set the scope of the recommendations.
>
> +1 on informational
>
> It'd be great for the document to cite more detailed technical=20
> reports, for readers to learn the pros and cons, and the history. IMO=20
> it'd significantly improve this document. The reason is that TCP and=20
> network will continue to evolve, so knowing the past and details can=20
> help readers to judge whether a specific recommendation is still=20
> useful or not, if he is willing to spend time to explore. For example,=20
> he can use some online tools to track the citation of these reports=20
> and see new reports.
>
I agree
> For the document, I'd recommend two more items that my personal=20
> experience gives remarkable performance enhancement on reducing and=20
> recover from losses.
>
> 1) make sure your stack supports SACK
> 2) enable pacing.
>
+1

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
[NK]
Regarding the point on enabling pacing, I would suggest text changes for th=
e section "3.2.  Initial Window".

From
"
   There has been experimentation with larger initial windows in
   combination with packet pacing, however IW10 has been reported to
   perform fairly well even in both general and high volume use cases.
"

To=20
"
  IW10 has been reported to
  perform fairly well even in both general and high volume use cases.=20
  There has been experimentation with larger initial windows in
  combination with packet pacing. Packet pacing at slow start
  reduces congestion losses at the beginning of the connection.
  Moreover, losses at the beginning of the connection result in retransmiss=
ion=20
  delay and an early slow start exit. In this case, degraded performances w=
hen flows are=20
  multiplexed such as in HTTP2 can be expected.  =20
  Packet pacing at slow start also provides clear benefits for challenged n=
etworks,=20
  such as capacity-limited or long-RTT networks.=20
"
Then, there could be text mentioning that "enabling packet pacing is recomm=
ended".
Or something similar.=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

> On disabling Nagle, there are also better APIs like Linux Cork so you=20
> don't have to send small packets even on continuous writes. One common=20
> problem I often see is developers unconditionally disable Nagle, but=20
> their HTTP workload may lead to lots of small packets.
>
Worth metioning.
> my 2c
>
>
>     If this work continues, it needs to be reviewed by TCPM and
>     HTTPbis. This would need much more than simply a last-call in both
>     working groups - but I'm sure with proper coordination progress
>     could be made so that there were no surprises for either group at
>     the time of WGLC!
>
>     Gorry
>
>
>
>
>         -----Original Message-----
>         From: tcpm [mailto:tcpm-bounces@ietf.org
>         <mailto:tcpm-bounces@ietf.org>] On Behalf Of Mark Nottingham
>         Sent: Wednesday, August 17, 2016 4:03 AM
>         To: tcpm@ietf.org <mailto:tcpm@ietf.org>
>         Cc: Patrick McManus; Daniel Stenberg
>         Subject: [tcpm] TCP Tuning for HTTP - update
>
>         Hi TCPM,
>
>         Just a quick note; Daniel and Tim have made an update to the
>         TCP Tuning for HTTP draft:
>         https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>         <https://tools.ietf.org/html/draft-stenberg-httpbis-tcp>
>
>         We've had a Call for Adoption open for this draft for a while,
>         and will likely adopt it soon. However, we'd like to get
>         feedback from this community first -- both about the latest
>         version of the input document, and to see if there's interest
>         in helping out.
>
>         You can give feedback on the HTTP WG mailing list
>         <https://lists.w3.org/Archives/Public/ietf-http-wg/
>         <https://lists.w3.org/Archives/Public/ietf-http-wg/>>, or  by
>         responding to this e-mail (Please leave the CC line; Patrick
>         and I will try to summarise the feedback to the WG).
>
>         Cheers,
>
>         --
>         Mark Nottingham https://www.mnot.net/
>
>
>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>     <https://www.ietf.org/mailman/listinfo/tcpm>
>
>

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


From nobody Fri Aug 19 05:17:57 2016
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From: SALLANTIN Renaud <renaud.sallantin@thalesaleniaspace.com>
To: 'Kuhn Nicolas' <Nicolas.Kuhn@cnes.fr>, "'gorry@erg.abdn.ac.uk'" <gorry@erg.abdn.ac.uk>, 'Yuchung Cheng' <ycheng@google.com>
Date: Fri, 19 Aug 2016 14:17:51 +0200
Thread-Topic: [tcpm] TCP Tuning for HTTP - update
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Hi all,=20

IMO, such a document could be very useful,=20
But it can also be dangerous.

For instance, the following statement seems to shut the door on other solut=
ions than IW10. Unfortunately, if IW10 improved TCP startup efficiency, it =
remains not efficient in case of large RTT networks (such as satellite syst=
ems for instance).

   There has been experimentation with larger initial windows in
   combination with packet pacing, however IW10 has been reported to
   perform fairly well even in both general and high volume use cases.=20

IMO, this doc must describe the best current practices, but also emphasize =
the remaining issues.

BR,=20
Renaud



-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de Kuhn Nicolas
Envoy=E9=A0: vendredi 19 ao=FBt 2016 10:14
=C0=A0: gorry@erg.abdn.ac.uk; Yuchung Cheng
Cc=A0: Mark Nottingham; tcpm@ietf.org; Patrick McManus; Daniel Stenberg
Objet=A0: Re: [tcpm] TCP Tuning for HTTP - update

Hi all,=20

I agree with the recommendation on enabling pacing.
If flows are multiplexed such as in HTTP2, losses at slow start can be very=
 damaging.=20
I have proposed some text inline.=20

Cheers,

Nico

-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de Gorry Fairhurst En=
voy=E9=A0: vendredi 19 ao=FBt 2016 09:48 =C0=A0: Yuchung Cheng Cc=A0: tcpm@=
ietf.org; Mark Nottingham; Patrick McManus; Daniel Stenberg Objet=A0: Re: [=
tcpm] TCP Tuning for HTTP - update

On 18/08/2016 18:34, Yuchung Cheng wrote:
>
>
> On Thu, Aug 18, 2016 at 9:54 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk=20
> <mailto:gorry@erg.abdn.ac.uk>> wrote:
>
>     Speaking as someone who commented on an earlier version:
>
>     I think it is good to offer advice on how TCP is used for web
>     traffic. I can see there is likely useful information that can
>     document how some things are done, and perhaps also recommend some
>     changes. I support continuing work on this.
>
>     Publishing a document like this in the RFC series needs to be
>     consistent with IETF consensus. The ID touches on many things that
>     are assigned to TCPM. (The breadth of topics may be a good reason
>     for a document - but is also a concern, because many things need
>     more clear description before the recommendations become clear).
>
>     I think this document should NOT be considered as BCP. It talks
>     about system configuration of endpoints, and of the
>     recommendations I see, some do support current PSs, some set new
>     precedents against current PS RFCs and in some cases the present
>     text appears to endorse EXP RFCs. If published as a BCP, that sort
>     of advice is likely to be confusing. Such information can also
>     easily be outdated, I suggest if adopted, this should be
>     Informational and clearly set the scope of the recommendations.
>
> +1 on informational
>
> It'd be great for the document to cite more detailed technical=20
> reports, for readers to learn the pros and cons, and the history. IMO=20
> it'd significantly improve this document. The reason is that TCP and=20
> network will continue to evolve, so knowing the past and details can=20
> help readers to judge whether a specific recommendation is still=20
> useful or not, if he is willing to spend time to explore. For example,=20
> he can use some online tools to track the citation of these reports=20
> and see new reports.
>
I agree
> For the document, I'd recommend two more items that my personal=20
> experience gives remarkable performance enhancement on reducing and=20
> recover from losses.
>
> 1) make sure your stack supports SACK
> 2) enable pacing.
>
+1

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
[NK]
Regarding the point on enabling pacing, I would suggest text changes for th=
e section "3.2.  Initial Window".

From
"
   There has been experimentation with larger initial windows in
   combination with packet pacing, however IW10 has been reported to
   perform fairly well even in both general and high volume use cases.
"

To
"
  IW10 has been reported to
  perform fairly well even in both general and high volume use cases.=20
  There has been experimentation with larger initial windows in
  combination with packet pacing. Packet pacing at slow start
  reduces congestion losses at the beginning of the connection.
  Moreover, losses at the beginning of the connection result in retransmiss=
ion
  delay and an early slow start exit. In this case, degraded performances w=
hen flows are=20
  multiplexed such as in HTTP2 can be expected.  =20
  Packet pacing at slow start also provides clear benefits for challenged n=
etworks,
  such as capacity-limited or long-RTT networks.=20
"
Then, there could be text mentioning that "enabling packet pacing is recomm=
ended".
Or something similar.=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

> On disabling Nagle, there are also better APIs like Linux Cork so you=20
> don't have to send small packets even on continuous writes. One common=20
> problem I often see is developers unconditionally disable Nagle, but=20
> their HTTP workload may lead to lots of small packets.
>
Worth metioning.
> my 2c
>
>
>     If this work continues, it needs to be reviewed by TCPM and
>     HTTPbis. This would need much more than simply a last-call in both
>     working groups - but I'm sure with proper coordination progress
>     could be made so that there were no surprises for either group at
>     the time of WGLC!
>
>     Gorry
>
>
>
>
>         -----Original Message-----
>         From: tcpm [mailto:tcpm-bounces@ietf.org
>         <mailto:tcpm-bounces@ietf.org>] On Behalf Of Mark Nottingham
>         Sent: Wednesday, August 17, 2016 4:03 AM
>         To: tcpm@ietf.org <mailto:tcpm@ietf.org>
>         Cc: Patrick McManus; Daniel Stenberg
>         Subject: [tcpm] TCP Tuning for HTTP - update
>
>         Hi TCPM,
>
>         Just a quick note; Daniel and Tim have made an update to the
>         TCP Tuning for HTTP draft:
>         https://tools.ietf.org/html/draft-stenberg-httpbis-tcp
>         <https://tools.ietf.org/html/draft-stenberg-httpbis-tcp>
>
>         We've had a Call for Adoption open for this draft for a while,
>         and will likely adopt it soon. However, we'd like to get
>         feedback from this community first -- both about the latest
>         version of the input document, and to see if there's interest
>         in helping out.
>
>         You can give feedback on the HTTP WG mailing list
>         <https://lists.w3.org/Archives/Public/ietf-http-wg/
>         <https://lists.w3.org/Archives/Public/ietf-http-wg/>>, or  by
>         responding to this e-mail (Please leave the CC line; Patrick
>         and I will try to summarise the feedback to the WG).
>
>         Cheers,
>
>         --
>         Mark Nottingham https://www.mnot.net/
>
>
>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>     <https://www.ietf.org/mailman/listinfo/tcpm>
>
>

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

_______________________________________________
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References: <20160810183654.05358B80C3A@rfc-editor.org> <CAOp4FwTogyBXLYdjHrrnM3-Uz2wpSX31eZg+GJwUP5LBnqu=sQ@mail.gmail.com> <7ac89d58-fc3a-9e12-9d22-0602944f8677@isi.edu> <e8d9584f-a069-ec4f-d6f1-f8e199fdb071@isi.edu> <CAK6E8=ewOCUBu7Ko9aK78_e2rmR70G_yvUpZKLm3n0tYdLs=Yw@mail.gmail.com> <CADVnQy=-6RFns-CDLwJkueA2+d9Ov7SbSXmFgkTvrrNMAsUo6g@mail.gmail.com> <bc8861bd-b9a8-153b-7bd8-ffbda9aa85ad@isi.edu> <57B31020.60506@tik.ee.ethz.ch> <CAK6E8=d4Q8DbUT3ShwbquJH7XL47p8HqyTpCt0WadcEw4Z_qfg@mail.gmail.com>
To: Yuchung Cheng <ycheng@google.com>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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> sounds fine with me but do documents really get updated?
That depends on the working group. However the point here is, if the doc =
gets updated, then this is still listed.

Mirja


> Am 16.08.2016 um 19:42 schrieb Yuchung Cheng <ycheng@google.com>:
>=20
>=20
>=20
> On Tue, Aug 16, 2016 at 6:07 AM, Mirja K=C3=BChlewind =
<mirja.kuehlewind@tik.ee.ethz.ch> wrote:
> Hi all,
>=20
> let me try to wrap this up:
>=20
> To me this does not seem to be a case for an errata. For sure some =
clarification seems to be helpful here but that's not what erratas are =
for.
>=20
> Therefore, I would suggest to simply set this into "Held for Document =
Update" state such that it is noticed in case we decide to update the =
document at some point. Does that make sense?
> sounds fine with me but do documents really get updated?
> =20
>=20
> Thanks,
> Mirja
>=20
>=20
>=20
>=20
> On 12.08.2016 22:19, Joe Touch wrote:
>=20
>=20
> On 8/12/2016 12:17 PM, Neal Cardwell wrote:
> On Fri, Aug 12, 2016 at 9:54 AM, Yuchung Cheng <ycheng@google.com> =
wrote:
> I think an errata on recommending per-connection limit and cite the
> research finding is worthy.
> I agree that it's worthwhile to have an errata to clarify that a
> per-connection limit is best. At the very least there seems
> some risky ambiguity here.
>=20
> I disagree, as per below...
>=20
> In fact, AFAICT RFC 5961, Section 7, page 13 --
>=20
>    https://tools.ietf.org/html/rfc5961#section-7
>=20
> -- has several passages that seem to implicitly suggest that
> the rate-limiting mechanism it specifies as a SHOULD is not
> per-connection but rather global.
>=20
> There's plenty of benefit to setting a single configuration for an
> entire system vs. setting per-connection limits. That's completely
> different from whether the ACKs are tallied globally or per =
connection.
>=20
> First:
>=20
>     ... an
>     administrator who is more interested in faster cleanup of stale
>     connections (i.e., concerned about excess TCP state) may decide to
>     set a higher value thus allowing more RST's to be processed in any
>     given time period.
>=20
> If this is a single-connection rate limit, then what practical reason
> would an administrator have to allow "more RST's to be processed in
> any given time period"? I can't think of a case where a legit live
> connection would send more than one RST.
> A single connection might process many RSTs because some can be out of
> window (they'd be dropped) or not at the end of the window (dropped
> under some configurations). I.e., receiving and evaluating RSTs =
doesn't
> always result in resetting the connection.
>=20
> If this is about a single
> connection, then allowing one mid-window RST (and responding to that
> one with a single challenge ACK) should be enough,
> It would be IF the other end issued a legitimate RST. If this is a
> spoofing attack, the data can still keep coming which means that the
> window might still be moving forward by the time the next attack RST
> arrives. As a result, this sort of 'catch up' can go on until the
> connection ends. The point of the text here is to avoid wasting
> resources in such cases.
> in most cases, to
> send a challenge ACK that elicits a proper RST from the remote
> endpoint. It's not clear to me what the motivation would be for
> intentionally sending a challenge ACK in response to more than one
> mid-window RST in a single connection.
> Protecting against that case is the motivation *of the entire =
document*.
>=20
> Thus AFAICT "allowing more RST's to be processed in any given time
> period" suggests that the rate limit is across connections.
>=20
> There's no benefit to doing this across connections at all, unless you
> think that there's some reason that legitimate connections shouldn't =
be
> reset by valid RSTs (there's no benefit to that).
>=20
> Second:
>=20
>     The time limit SHOULD be tunable to help timeout brute force =
attacks
>     faster than a potential legitimate flood of RSTs.
>=20
> I may be missing something, but if this rate-limiting mechanism is
> only about a single connection, then what is a "legitimate flood of
> RSTs"?
> A sequence of RSTs that pick different values, as described in RFC =
4953.
>=20
> Or a sequence of RSTs aimed at responses to the ACK-after-RST that is
> chasing legitimate data, as described above.
>=20
>   Why would it be important for a single connection to allow a
> "legitimate flood of RSTs" destinted for itself?
> It wouldn't - that's the point of this text. You should treat a flood =
of
> RSTs for a single connection as an attack and throttle them =
accordingly.
>=20
> A single legitimate
> live connection should not send out a flood of RSTs; it should send
> out one. Once there is one legitimate RST that arrives and is
> processed, the receiver deletes the connection, and any other arriving
> RSTs don't match any existing connection, so are ignored and
> dropped, in which case is no challenge ACK to rate-limit.
> Again, the point of this document are attacks, not legitimate =
connections.
> By contrast, if this rate limit applies across connections then "a
> potential legitimate flood of RSTs" makes sense, since a group of
> legitimate live connections that all close in a burst may, in certain
> cases, generate a legitimate flood of RSTs. So tuning the limit
> upward would make sense.
>=20
> There's no case I can imagine where a set of attacks on one connection
> should be cause for concern on another necessarily.
>=20
> Yes, there's no real need to set per-connection limits (i.e., a global
> per-connection limit seems fine), but counting them as a group serves =
no
> purpose against any threat, unless you think that "if I'm attacked on
> one connection, I will be attacked on another" - but that's an =
incorrect
> inference IMO.
>=20
>=20
> Sorry if I'm missing something obvious. But it seems like there are at
> least real ambiguities there, on a point significant enough to be =
worth
> clearing up.
>=20
> I think you're missing the point of the document, but that's just my
> interpretation of the exchange.
>=20
> Joe
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>=20
>=20


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To: Mark Nottingham <mnot@mnot.net>, Joe Touch <touch@ISI.EDU>
References: <5CD67877-19E3-4E79-BBF2-3E270343A378@mnot.net> <2197232f-10d7-28cb-fcc9-05bd495e3c22@isi.edu> <20160817064545.GD16017@1wt.eu> <7f7b129c-f156-d067-bef8-4a2213f461ac@isi.edu> <20160817180802.GA16773@1wt.eu> <4ab7c5b0-3722-1346-f481-a8d76de70034@isi.edu> <20160817211317.GA16929@1wt.eu> <c928d1ca-fc89-d0b0-4e1a-8a0bd960d2bb@isi.edu> <CACweHNC1qFH5DMnZRE87bAE5sk_P+1z1Fzm-9YEu=E2DULkaYQ@mail.gmail.com> <27b58b64-48cd-39af-78b3-ef583c585fa6@isi.edu> <20160818053837.GC16773@1wt.eu> <7a36e025-4882-4f8b-7a83-9fdcd990a971@isi.edu> <BFE4FC25-CCCA-4390-AADD-BCDBFE798790@mnot.net>
From: Eliot Lear <lear@cisco.com>
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Date: Fri, 19 Aug 2016 18:52:10 +0200
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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From: Eliot Lear <lear@cisco.com>
To: Mark Nottingham <mnot@mnot.net>, Joe Touch <touch@ISI.EDU>
Cc: Willy Tarreau <w@1wt.eu>, Matthew Kerwin <matthew@kerwin.net.au>,
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Just to be the stick in the mud for a moment...


On 8/19/16 5:49 AM, Mark Nottingham wrote:
>> On 19 Aug 2016, at 1:07 AM, Joe Touch <touch@ISI.EDU> wrote:
>>
>> I do think that this doc needs to figure out whom it is speaking to, w=
hat advice they actually need, etc.
>>
>> If the result is a set of recommendations that involve the word "sysct=
l", I remain skeptical it is appropriate as an RFC
>
> I think there's broad agreement on both of these points.
>
> I'm wondering if it makes sense to aim it primarily at HTTP implementer=
s rather than administrators, with the notion that it would inform:
>
> - Their implementation decisions
> - The configuration choices they offer to administrators / users
> - Their documentation (e.g., advice to their administrators when the im=
plementation can't change the appropriate parts of the OS)
>
> Would that help?=20
>
> If so, it might make sense to organise it into sections for clients and=
 servers (and intermediaries, if there's anything that isn't covered by t=
he combination of the first two). Although IIRC Daniel was already talkin=
g about doing that.
>
>

If there are very specific settings for Linux and they are well vetted,
I see no reason not to continue to include them in an appendix, as they
are now.  Cheat sheets are nice, and there is nothing wrong with a cheat
sheet being in an RFC (and I wouoldn't mind a few being RFCs).

Eliot


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Subject: Re: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-tcp-low-rtt-00.txt
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On Thu, Aug 18, 2016 at 1:05 AM, Karen Elisabeth Egede Nielsen
<karen.nielsen@tieto.com> wrote:
> [Karen Elisabeth Egede Nielsen] I agree completely with your view and
> approach.
>
> In respect to the draft draft-bagnulo-tcpm-tcp-low-rtt-00.txt and the given
> recommendations for low RTO
> is that they do not (likely) take 2. Into account - meaning that they are
> looking for the low RTO to solve 2 for tail recovery.
> The bad thing is that they then also get the CWND reductions (as you also
> say).
>
>
> I would like for that this aspect is taken into consideration/given thoughts
> in the draft draft-bagnulo-tcpm-tcp-low-rtt-00.txt and the general 4LS
> effort.
> If we end with very low recommendations for RTO in order to handle 2. we
> risk undermining the RTO-retransmission function as well as the TLP function
> for now and forever.
>
> It might be that one for TCPs, that do not support TLP(/RACK), need to
> operate with very low RTOs to handle tail loss's "fast enough", but the
> recommendation for
> "future standard ?" TCP with TLP always enabled might be different. In this
> context lowering the RTO to handle 2. should be considered as a "temporary
> hack" ?

One feature worth discussion in the draft is the penalty of cwnd=1 on
spurious timeouts.

For DC, the RTT is so fairly low that we may consider a false reset of
cwnd to 1 is not a big deal. But consider in world of 100Gbps BDP will
still be hundreds of packets so it is going to be a non-trivial
penalty.

Therefore it is important to discuss the timeouts with the remedy
operations: how well can the connection detect spurious timeouts:
F-RTO or Eifel are the two algorithms by IETF, Linux also supports
DSACK-based undo.

However, both F-RTO and Eifel have their limitations: F-RTO requires
the sender to have new data to transmit after the first DUPACK of
timeout. In the DC world, the requests or responses are often short
enough to prevent that. Eifel relies on timestamps which is ticked at
an interval higher than the DC RTT (e.g., Linux == jiffies == 1ms
often), therefore the timestamps fail to distinguish original and
retransmit if RTO was fired before the next tick.

I plan to cover these aspects in next RACK rev.

>
> BR, Karen


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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Mark,


On 8/18/2016 8:49 PM, Mark Nottingham wrote:
>> On 19 Aug 2016, at 1:07 AM, Joe Touch <touch@ISI.EDU> wrote:
>>
>> I do think that this doc needs to figure out whom it is speaking to, what advice they actually need, etc.
>>
>> If the result is a set of recommendations that involve the word "sysctl", I remain skeptical it is appropriate as an RFC
>
> I think there's broad agreement on both of these points.
>
> I'm wondering if it makes sense to aim it primarily at HTTP implementers rather than administrators, with the notion that it would inform:
>
> - Their implementation decisions
> - The configuration choices they offer to administrators / users
> - Their documentation (e.g., advice to their administrators when the implementation can't change the appropriate parts of the OS)
>
> Would that help? 

I think so. I also think it would be very important to differentiate the
issues that are unique to HTTP (if any) vs. those that are generic to
any transaction system.

> If so, it might make sense to organise it into sections for clients and servers (and intermediaries, if there's anything that isn't covered by the combination of the first two). Although IIRC Daniel was already talking about doing that.

Clients, servers, proxies, and (even though a violation of so many parts
of the Internet) transparent proxies.

Joe


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From: Jana Iyengar <jri@google.com>
Date: Fri, 19 Aug 2016 13:05:41 -0700
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Subject: Re: [tcpm] TCP Tuning for HTTP - update
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--001a113f5b9a23cd0f053a7239b3
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Mark,

[I have not have read the entire thread in detail, but responding to one
question.]

I'd especially like to hear from other people in the TCP community; we've
> heard from Joe and Michael; do others share their opinions, or have another
> view?
>

Responding to this specific question: I think this document is useful.
There's tremendous value in having a document that discusses protocol
interactions, especially for the most important protocol interaction today
(HTTP/TCP).  Pursuing this document as Informational makes complete sense,
and is well within IETF norms.

Implementation notes/advice is actually quite a useful bit that
traditionally shows up in appendices, but arguably it really doesn't matter
where it shows up. It's valuable. I would remove specific APIs and instead
talk about mechanisms that are common enough (remove setsockopt() and
sysctls, but talk about Nagle, socket buffers, etc.) but there are a number
of commonly deployed mechanisms that it makes sense to talk about them
here. I would suggest that the document discuss important interactions from
a browser's point of view as separate from those at a web server.

I am happy to review and suggest changes to the document, but I think it's
a fine fit.

- jana

--001a113f5b9a23cd0f053a7239b3
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>Mark,</div><div><br></div><div>[I have not have read the entire thread in =
detail, but responding to one question.]</div><div><br></div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid=
 rgb(204,204,204);padding-left:1ex">
I&#39;d especially like to hear from other people in the TCP community; we&=
#39;ve heard from Joe and Michael; do others share their opinions, or have =
another view?<br></blockquote><div><br></div><div>Responding to this specif=
ic question: I think this document is useful. There&#39;s tremendous value =
in having a document that discusses protocol interactions, especially for t=
he most important protocol interaction today (HTTP/TCP).=C2=A0 Pursuing thi=
s document as Informational makes complete sense, and is well within IETF n=
orms.</div><div><br></div><div>Implementation notes/advice is actually quit=
e a useful bit that traditionally shows up in appendices, but arguably it r=
eally doesn&#39;t matter where it shows up. It&#39;s valuable. I would remo=
ve specific APIs and instead talk about mechanisms that are common enough (=
remove setsockopt() and sysctls, but talk about Nagle, socket buffers, etc.=
) but there are a number of commonly deployed mechanisms that it makes sens=
e to talk about them here. I would suggest that the document discuss import=
ant interactions from a browser&#39;s point of view as separate from those =
at a web server.<br></div><div><br></div><div>I am happy to review and sugg=
est changes to the document, but I think it&#39;s a fine fit.</div><div><br=
></div><div>- jana</div></div></div></div>

--001a113f5b9a23cd0f053a7239b3--


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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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--94eb2c18f87860de54053a9836e4
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Hi Neal,

Thanks for the info.
So, it seems to me that the linux code has the issue Kobby pointed out.
Or, am I missing something?
--
Yoshi


On Mon, Aug 15, 2016 at 6:18 PM, Neal Cardwell <ncardwell@google.com> wrote:

> On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida
> <nishida@sfc.wide.ad.jp> wrote:
> > Hello,
> > I personally think this is an interesting corner case for discussion.
> > It looks a minor one, but I'm not not very sure if we can leave it for
> each
> > implementation.
> > I also guess a question would be if the BSD's fix is the best way for the
> > issue.
>
> Yes, I agree this is an interesting case for discussion.
>
> FWIW, as a point of comparison for discussion, Linux's approach is a
> little different: in Linux, the sender does not rewind SND.NXT on
> retransmissions (RTO or Fast Recovery). Then the sender usually uses
> SND.NXT for the seq field of outgoing pure ACKs. I say "usually"
> because the Linux code has some code to deal with the case where the
> receiver has withdrawn the receive window, so that SND.NXT is now
> beyond the receive window. The code tcp_send_ack() uses to pick a seq
> for outgoing pure ACKs looks like:
>
> /* SND.NXT, if window was not shrunk.
>  * If window has been shrunk, what should we make? It is not clear at
> all.
>  * Using SND.UNA we will fail to open window, SND.NXT is out of
> window. :-(
>  * Anything in between SND.UNA...SND.UNA+SND.WND also can be already
>  * invalid. OK, let's make this for now:
>  */
> static inline __u32 tcp_acceptable_seq(const struct sock *sk)
> {
>         const struct tcp_sock *tp = tcp_sk(sk);
>
>         if (!before(tcp_wnd_end(tp), tp->snd_nxt))
>                 return tp->snd_nxt;
>         else
>                 return tcp_wnd_end(tp);
> }
>
> neal
>

--94eb2c18f87860de54053a9836e4
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">Hi Neal,<div><br></div><div>Thanks for the info.</div><div=
>So, it seems to me that the linux code has the issue Kobby pointed out.</d=
iv><div>Or, am I missing something?</div><div>--<br></div><div>Yoshi</div><=
div><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, A=
ug 15, 2016 at 6:18 PM, Neal Cardwell <span dir=3D"ltr">&lt;<a href=3D"mail=
to:ncardwell@google.com" target=3D"_blank">ncardwell@google.com</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><span>On Mon, Aug 15, 2016 at =
6:39 PM, Yoshifumi Nishida<br>
&lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">nishida@sfc=
.wide.ad.jp</a>&gt; wrote:<br>
&gt; Hello,<br>
&gt; I personally think this is an interesting corner case for discussion.<=
br>
&gt; It looks a minor one, but I&#39;m not not very sure if we can leave it=
 for each<br>
&gt; implementation.<br>
&gt; I also guess a question would be if the BSD&#39;s fix is the best way =
for the<br>
&gt; issue.<br>
<br>
</span>Yes, I agree this is an interesting case for discussion.<br>
<br>
FWIW, as a point of comparison for discussion, Linux&#39;s approach is a<br=
>
little different: in Linux, the sender does not rewind SND.NXT on<br>
retransmissions (RTO or Fast Recovery). Then the sender usually uses<br>
SND.NXT for the seq field of outgoing pure ACKs. I say &quot;usually&quot;<=
br>
because the Linux code has some code to deal with the case where the<br>
receiver has withdrawn the receive window, so that SND.NXT is now<br>
beyond the receive window. The code tcp_send_ack() uses to pick a seq<br>
for outgoing pure ACKs looks like:<br>
<br>
/* SND.NXT, if window was not shrunk.<br>
=C2=A0* If window has been shrunk, what should we make? It is not clear at<=
br>
all.<br>
=C2=A0* Using SND.UNA we will fail to open window, SND.NXT is out of<br>
window. :-(<br>
=C2=A0* Anything in between SND.UNA...SND.UNA+SND.WND also can be already<b=
r>
=C2=A0* invalid. OK, let&#39;s make this for now:<br>
=C2=A0*/<br>
static inline __u32 tcp_acceptable_seq(const struct sock *sk)<br>
{<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 const struct tcp_sock *tp =3D tcp_sk(sk);<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 if (!before(tcp_wnd_end(tp), tp-&gt;snd_nxt))<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 return tp-&gt;snd_n=
xt;<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 else<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 return tcp_wnd_end(=
tp);<br>
}<br>
<span><font color=3D"#888888"><br>
neal<br>
</font></span></blockquote></div><br></div></div></div>

--94eb2c18f87860de54053a9836e4--


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References: <MWHPR11MB1374A50BC599B093EA09668984070@MWHPR11MB1374.namprd11.prod.outlook.com> <7070553C-65D1-46EE-95F4-DAE82E1F5A5E@weston.borman.com> <CY4PR11MB187848FCCEF4DB140F85913E841A0@CY4PR11MB1878.namprd11.prod.outlook.com> <2D524A8D-A5CA-45A6-B94D-FA1DA0CEE609@weston.borman.com> <CY4PR11MB1878E7E911194A1032E32428841E0@CY4PR11MB1878.namprd11.prod.outlook.com> <CAO249yeTNRFuFcWn5ga854g24GM_7DAjeKOSw3d7h=HQQYKX1Q@mail.gmail.com> <CADVnQymeyst9De4F8Zqc6wGfLEdzmGsypSC-ZKZ7bXT6PO=J4g@mail.gmail.com> <CAO249yeeEo3B9H3SyGqA8aiWOWjoHYji=JXEh1GHOHSsf+RszA@mail.gmail.com>
From: Neal Cardwell <ncardwell@google.com>
Date: Sun, 21 Aug 2016 15:44:54 -0400
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To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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I am pretty sure Linux does not have the issue Kobby pointed out in this thread.

At a high level Linux should be OK because it follows the principle
David Borman laid out in his August 16 email: "ACK-only packets should
be sent with the largest in-window sequence number that has ever been
sent."

Linux obeys that principle by using tp->snd_nxt to store the largest
sequence number that has ever been sent, and having
tcp_acceptable_seq() use tp->snd_nxt but clamp the outgoing sequence
number to make sure it is in-window. To be able to do this, in Linux,
the sender does not rewind tp->snd_nxt on retransmissions.

neal

On Sun, Aug 21, 2016 at 1:25 PM, Yoshifumi Nishida
<nishida@sfc.wide.ad.jp> wrote:
> Hi Neal,
>
> Thanks for the info.
> So, it seems to me that the linux code has the issue Kobby pointed out.
> Or, am I missing something?
> --
> Yoshi
>
>
> On Mon, Aug 15, 2016 at 6:18 PM, Neal Cardwell <ncardwell@google.com> wrote:
>>
>> On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida
>> <nishida@sfc.wide.ad.jp> wrote:
>> > Hello,
>> > I personally think this is an interesting corner case for discussion.
>> > It looks a minor one, but I'm not not very sure if we can leave it for
>> > each
>> > implementation.
>> > I also guess a question would be if the BSD's fix is the best way for
>> > the
>> > issue.
>>
>> Yes, I agree this is an interesting case for discussion.
>>
>> FWIW, as a point of comparison for discussion, Linux's approach is a
>> little different: in Linux, the sender does not rewind SND.NXT on
>> retransmissions (RTO or Fast Recovery). Then the sender usually uses
>> SND.NXT for the seq field of outgoing pure ACKs. I say "usually"
>> because the Linux code has some code to deal with the case where the
>> receiver has withdrawn the receive window, so that SND.NXT is now
>> beyond the receive window. The code tcp_send_ack() uses to pick a seq
>> for outgoing pure ACKs looks like:
>>
>> /* SND.NXT, if window was not shrunk.
>>  * If window has been shrunk, what should we make? It is not clear at
>> all.
>>  * Using SND.UNA we will fail to open window, SND.NXT is out of
>> window. :-(
>>  * Anything in between SND.UNA...SND.UNA+SND.WND also can be already
>>  * invalid. OK, let's make this for now:
>>  */
>> static inline __u32 tcp_acceptable_seq(const struct sock *sk)
>> {
>>         const struct tcp_sock *tp = tcp_sk(sk);
>>
>>         if (!before(tcp_wnd_end(tp), tp->snd_nxt))
>>                 return tp->snd_nxt;
>>         else
>>                 return tcp_wnd_end(tp);
>> }
>>
>> neal
>
>


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Subject: Re: [tcpm] Personal change in chairing
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Thanks, Mirja, and thanks to everyone for collaboration! I think Michael =
is an excellent choice for a new TCPM co-chair.

Hope to see many of you in future some time!

- Pasi


> On 17 Aug 2016, at 18:28, Mirja K=C3=BChlewind <ietf@kuehlewind.net> =
wrote:
>=20
> Hi all,
>=20
> I have to announce a personal change in the tcpm chair position:
>=20
> Pasi Sarolahti is stepping down due to time restrictions. Pasi, thanks =
a lot for your work and engagement as chair! Your did a great job and we =
hope to still see you as often as possible as contributor in the working =
group!
>=20
> Michael T=C3=BCxen is starting as a new tcpm chair. Micheal, great to =
have you on broad! We look forward to work with you!
>=20
> Mirja, as responsible AD


From nobody Mon Aug 22 23:33:14 2016
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Subject: Re: [tcpm] Personal change in chairing
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> On 22 Aug 2016, at 10:14, Pasi Sarolahti <pasi.sarolahti@iki.fi> =
wrote:
>=20
> Thanks, Mirja, and thanks to everyone for collaboration! I think =
Michael is an excellent choice for a new TCPM co-chair.
>=20
> Hope to see many of you in future some time!
Hi Pasi,

thank you very much for serving as WG chair! I hope to see you soon.

Best regards
Michael
>=20
> - Pasi
>=20
>=20
>> On 17 Aug 2016, at 18:28, Mirja K=C3=BChlewind <ietf@kuehlewind.net> =
wrote:
>>=20
>> Hi all,
>>=20
>> I have to announce a personal change in the tcpm chair position:
>>=20
>> Pasi Sarolahti is stepping down due to time restrictions. Pasi, =
thanks a lot for your work and engagement as chair! Your did a great job =
and we hope to still see you as often as possible as contributor in the =
working group!
>>=20
>> Michael T=C3=BCxen is starting as a new tcpm chair. Micheal, great to =
have you on broad! We look forward to work with you!
>>=20
>> Mirja, as responsible AD
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Tue, 23 Aug 2016 00:06:48 -0700
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To: Neal Cardwell <ncardwell@google.com>
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Cc: Kobby Carmona <kobby.Carmona@qlogic.com>, "tcpm@ietf.org" <tcpm@ietf.org>, David Borman <dab@weston.borman.com>
Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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--001a113d1268ed3477053ab7ced2
Content-Type: text/plain; charset=UTF-8

Hi Neal,

Oh. I see. Thanks for the explanation.
I'd like to think about it a bit more.
Thanks,
--
Yoshi

On Sun, Aug 21, 2016 at 12:44 PM, Neal Cardwell <ncardwell@google.com>
wrote:

> I am pretty sure Linux does not have the issue Kobby pointed out in this
> thread.
>
> At a high level Linux should be OK because it follows the principle
> David Borman laid out in his August 16 email: "ACK-only packets should
> be sent with the largest in-window sequence number that has ever been
> sent."
>
> Linux obeys that principle by using tp->snd_nxt to store the largest
> sequence number that has ever been sent, and having
> tcp_acceptable_seq() use tp->snd_nxt but clamp the outgoing sequence
> number to make sure it is in-window. To be able to do this, in Linux,
> the sender does not rewind tp->snd_nxt on retransmissions.
>
> neal
>
> On Sun, Aug 21, 2016 at 1:25 PM, Yoshifumi Nishida
> <nishida@sfc.wide.ad.jp> wrote:
> > Hi Neal,
> >
> > Thanks for the info.
> > So, it seems to me that the linux code has the issue Kobby pointed out.
> > Or, am I missing something?
> > --
> > Yoshi
> >
> >
> > On Mon, Aug 15, 2016 at 6:18 PM, Neal Cardwell <ncardwell@google.com>
> wrote:
> >>
> >> On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida
> >> <nishida@sfc.wide.ad.jp> wrote:
> >> > Hello,
> >> > I personally think this is an interesting corner case for discussion.
> >> > It looks a minor one, but I'm not not very sure if we can leave it for
> >> > each
> >> > implementation.
> >> > I also guess a question would be if the BSD's fix is the best way for
> >> > the
> >> > issue.
> >>
> >> Yes, I agree this is an interesting case for discussion.
> >>
> >> FWIW, as a point of comparison for discussion, Linux's approach is a
> >> little different: in Linux, the sender does not rewind SND.NXT on
> >> retransmissions (RTO or Fast Recovery). Then the sender usually uses
> >> SND.NXT for the seq field of outgoing pure ACKs. I say "usually"
> >> because the Linux code has some code to deal with the case where the
> >> receiver has withdrawn the receive window, so that SND.NXT is now
> >> beyond the receive window. The code tcp_send_ack() uses to pick a seq
> >> for outgoing pure ACKs looks like:
> >>
> >> /* SND.NXT, if window was not shrunk.
> >>  * If window has been shrunk, what should we make? It is not clear at
> >> all.
> >>  * Using SND.UNA we will fail to open window, SND.NXT is out of
> >> window. :-(
> >>  * Anything in between SND.UNA...SND.UNA+SND.WND also can be already
> >>  * invalid. OK, let's make this for now:
> >>  */
> >> static inline __u32 tcp_acceptable_seq(const struct sock *sk)
> >> {
> >>         const struct tcp_sock *tp = tcp_sk(sk);
> >>
> >>         if (!before(tcp_wnd_end(tp), tp->snd_nxt))
> >>                 return tp->snd_nxt;
> >>         else
> >>                 return tcp_wnd_end(tp);
> >> }
> >>
> >> neal
> >
> >
>

--001a113d1268ed3477053ab7ced2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Neal,<div><br></div><div>Oh. I see. Thanks for the expl=
anation.</div><div>I&#39;d like to think about it a bit more.</div><div>Tha=
nks,</div><div>--<br></div><div>Yoshi<br><div class=3D"gmail_extra"><br><di=
v class=3D"gmail_quote">On Sun, Aug 21, 2016 at 12:44 PM, Neal Cardwell <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:ncardwell@google.com" target=3D"_blank=
">ncardwell@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">I am pretty sure Linux does not have the issue Kobby pointed out in thi=
s thread.<br>
<br>
At a high level Linux should be OK because it follows the principle<br>
David Borman laid out in his August 16 email: &quot;ACK-only packets should=
<br>
be sent with the largest in-window sequence number that has ever been<br>
sent.&quot;<br>
<br>
Linux obeys that principle by using tp-&gt;snd_nxt to store the largest<br>
sequence number that has ever been sent, and having<br>
tcp_acceptable_seq() use tp-&gt;snd_nxt but clamp the outgoing sequence<br>
number to make sure it is in-window. To be able to do this, in Linux,<br>
the sender does not rewind tp-&gt;snd_nxt on retransmissions.<br>
<br>
neal<br>
<br>
On Sun, Aug 21, 2016 at 1:25 PM, Yoshifumi Nishida<br>
<div><div>&lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">n=
ishida@sfc.wide.ad.jp</a>&gt; wrote:<br>
&gt; Hi Neal,<br>
&gt;<br>
&gt; Thanks for the info.<br>
&gt; So, it seems to me that the linux code has the issue Kobby pointed out=
.<br>
&gt; Or, am I missing something?<br>
&gt; --<br>
&gt; Yoshi<br>
&gt;<br>
&gt;<br>
&gt; On Mon, Aug 15, 2016 at 6:18 PM, Neal Cardwell &lt;<a href=3D"mailto:n=
cardwell@google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<=
br>
&gt;&gt;<br>
&gt;&gt; On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida<br>
&gt;&gt; &lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">ni=
shida@sfc.wide.ad.jp</a>&gt; wrote:<br>
&gt;&gt; &gt; Hello,<br>
&gt;&gt; &gt; I personally think this is an interesting corner case for dis=
cussion.<br>
&gt;&gt; &gt; It looks a minor one, but I&#39;m not not very sure if we can=
 leave it for<br>
&gt;&gt; &gt; each<br>
&gt;&gt; &gt; implementation.<br>
&gt;&gt; &gt; I also guess a question would be if the BSD&#39;s fix is the =
best way for<br>
&gt;&gt; &gt; the<br>
&gt;&gt; &gt; issue.<br>
&gt;&gt;<br>
&gt;&gt; Yes, I agree this is an interesting case for discussion.<br>
&gt;&gt;<br>
&gt;&gt; FWIW, as a point of comparison for discussion, Linux&#39;s approac=
h is a<br>
&gt;&gt; little different: in Linux, the sender does not rewind SND.NXT on<=
br>
&gt;&gt; retransmissions (RTO or Fast Recovery). Then the sender usually us=
es<br>
&gt;&gt; SND.NXT for the seq field of outgoing pure ACKs. I say &quot;usual=
ly&quot;<br>
&gt;&gt; because the Linux code has some code to deal with the case where t=
he<br>
&gt;&gt; receiver has withdrawn the receive window, so that SND.NXT is now<=
br>
&gt;&gt; beyond the receive window. The code tcp_send_ack() uses to pick a =
seq<br>
&gt;&gt; for outgoing pure ACKs looks like:<br>
&gt;&gt;<br>
&gt;&gt; /* SND.NXT, if window was not shrunk.<br>
&gt;&gt;=C2=A0 * If window has been shrunk, what should we make? It is not =
clear at<br>
&gt;&gt; all.<br>
&gt;&gt;=C2=A0 * Using SND.UNA we will fail to open window, SND.NXT is out =
of<br>
&gt;&gt; window. :-(<br>
&gt;&gt;=C2=A0 * Anything in between SND.UNA...SND.UNA+SND.WND also can be =
already<br>
&gt;&gt;=C2=A0 * invalid. OK, let&#39;s make this for now:<br>
&gt;&gt;=C2=A0 */<br>
&gt;&gt; static inline __u32 tcp_acceptable_seq(const struct sock *sk)<br>
&gt;&gt; {<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0const struct tcp_sock *tp =3D tcp=
_sk(sk);<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0if (!before(tcp_wnd_end(tp), tp-&=
gt;snd_nxt))<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0retur=
n tp-&gt;snd_nxt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0else<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0retur=
n tcp_wnd_end(tp);<br>
&gt;&gt; }<br>
&gt;&gt;<br>
&gt;&gt; neal<br>
&gt;<br>
&gt;<br>
</div></div></blockquote></div><br></div></div></div>

--001a113d1268ed3477053ab7ced2--


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Subject: [tcpm] Fwd: [Technical Errata Reported] RFC0793 (4785)
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FYI.

> Anfang der weitergeleiteten Nachricht:
> 
> Von: RFC Errata System <rfc-editor@rfc-editor.org>
> Betreff: [Technical Errata Reported] RFC0793 (4785)
> Datum: 20. August 2016 um 15:29:55 MESZ
> An: iesg@ietf.org
> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
> 
> The following errata report has been submitted for RFC0793,
> "Transmission Control Protocol".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=793&eid=4785
> 
> --------------------------------------
> Type: Technical
> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
> 
> Section: 3.9  p. 72
> 
> Original Text
> -------------
> 
> 
> If the ACK is a duplicate
> (SEG.ACK < SND.UNA), it can be ignored.
> 
> Corrected Text
> --------------
> If the ACK is a duplicate
> (SEG.ACK =< SND.UNA), it can be ignored except when equality is met, 
> then window should be updated. This can happen when there are 
> segments in flight but the receiver shrinks its RCV.BUF to drop all 
> of them and send an ACK carrying zero window update. Upon its 
> arrival at the sending TCP, condition SND.UNA = SEG.ACK is met and 
> we must update SND.WND <- 0. Then sender starts persist timer for 
> sending zero-window probes [Ref. RFC 1122 Section 4.2.2.17, page 92]
> 
> 
> 
> 
> 
> Notes
> -----
> The condition is corrected as Duplicate ACK in 
> RFC 1122, Section 4.2.2.20 (g) p. 94. Accordingly old text must be 
> modified and new text should also be present to support the corrected 
> condition in RFC 793. 
> 
> This is one case where duplicate acknowledgment cannot be ignored. i.e. 
> when SEG.ACK == SND.UNA and advertised window in the incoming ACK is 0
> in which case sender needs to :
> 1. update window
> 2. start persist timer
> 3. send zero window probe segments. 
> 
> 
> Note:
> Such ACKs should not be called as duplicates as it fails condition (e) 
> of definition of "DUPLICATE ACKNOWLEDGMENT" in Ref 5681 Section 2, pg 4
> 
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary. 
> 
> --------------------------------------
> RFC0793 (no draft string recorded)
> --------------------------------------
> Title               : Transmission Control Protocol
> Publication Date    : September 1981
> Author(s)           : J. Postel
> Category            : INTERNET STANDARD
> Source              : Legacy
> Area                : Legacy
> Stream              : IETF
> Verifying Party     : IESG
> 


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References: <20160820132955.49560B80EE3@rfc-editor.org> <471798EE-293E-44E6-A7A3-EAB2ACF6CAEC@kuehlewind.net>
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Subject: Re: [tcpm] Fwd: [Technical Errata Reported] RFC0793 (4785)
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As noted, RFC1122 already has a correction on this.  The fix from 1122 
is in the 793bis draft.


Original 793 text:

           If the ACK is a duplicate
           (SEG.ACK < SND.UNA), it can be ignored.  If the ACK acks
           something not yet sent (SEG.ACK > SND.NXT) then send an ACK,
           drop the segment, and return.

           If SND.UNA < SEG.ACK =< SND.NXT, the send window should be
           updated.  If (SND.WL1 < SEG.SEQ or (SND.WL1 = SEG.SEQ and
           SND.WL2 =< SEG.ACK)), set SND.WND <- SEG.WND, set
           SND.WL1 <- SEG.SEQ, and set SND.WL2 <- SEG.ACK.


Text in 793bis:

           If the ACK is a duplicate (SEG.ACK =<
           SND.UNA), it can be ignored.  If the ACK acks
           something not yet sent (SEG.ACK > SND.NXT) then send
           an ACK, drop the segment, and return.

           If SND.UNA =< SEG.ACK =< SND.NXT, the send window
           should be updated.  If (SND.WL1 < SEG.SEQ or (SND.WL1
           = SEG.SEQ and SND.WL2 =< SEG.ACK)), set SND.WND <-
           SEG.WND, set SND.WL1 <- SEG.SEQ, and set SND.WL2 <-
           SEG.ACK.





On 8/24/2016 7:21 PM, Mirja Kuehlewind (IETF) wrote:
> FYI.
>
>> Anfang der weitergeleiteten Nachricht:
>>
>> Von: RFC Errata System <rfc-editor@rfc-editor.org>
>> Betreff: [Technical Errata Reported] RFC0793 (4785)
>> Datum: 20. August 2016 um 15:29:55 MESZ
>> An: iesg@ietf.org
>> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
>>
>> The following errata report has been submitted for RFC0793,
>> "Transmission Control Protocol".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=793&eid=4785
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
>>
>> Section: 3.9  p. 72
>>
>> Original Text
>> -------------
>>
>>
>> If the ACK is a duplicate
>> (SEG.ACK < SND.UNA), it can be ignored.
>>
>> Corrected Text
>> --------------
>> If the ACK is a duplicate
>> (SEG.ACK =< SND.UNA), it can be ignored except when equality is met,
>> then window should be updated. This can happen when there are
>> segments in flight but the receiver shrinks its RCV.BUF to drop all
>> of them and send an ACK carrying zero window update. Upon its
>> arrival at the sending TCP, condition SND.UNA = SEG.ACK is met and
>> we must update SND.WND <- 0. Then sender starts persist timer for
>> sending zero-window probes [Ref. RFC 1122 Section 4.2.2.17, page 92]
>>
>>
>>
>>
>>
>> Notes
>> -----
>> The condition is corrected as Duplicate ACK in
>> RFC 1122, Section 4.2.2.20 (g) p. 94. Accordingly old text must be
>> modified and new text should also be present to support the corrected
>> condition in RFC 793.
>>
>> This is one case where duplicate acknowledgment cannot be ignored. i.e.
>> when SEG.ACK == SND.UNA and advertised window in the incoming ACK is 0
>> in which case sender needs to :
>> 1. update window
>> 2. start persist timer
>> 3. send zero window probe segments.
>>
>>
>> Note:
>> Such ACKs should not be called as duplicates as it fails condition (e)
>> of definition of "DUPLICATE ACKNOWLEDGMENT" in Ref 5681 Section 2, pg 4
>>
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC0793 (no draft string recorded)
>> --------------------------------------
>> Title               : Transmission Control Protocol
>> Publication Date    : September 1981
>> Author(s)           : J. Postel
>> Category            : INTERNET STANDARD
>> Source              : Legacy
>> Area                : Legacy
>> Stream              : IETF
>> Verifying Party     : IESG
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From Cheng.Cui@netapp.com  Wed Aug 24 14:06:30 2016
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From: "Cui, Cheng" <Cheng.Cui@netapp.com>
To: "david.borman@windriver.com" <david.borman@windriver.com>, "dab@weston.borman.com" <dab@weston.borman.com>, "david.borman@quantum.com" <david.borman@quantum.com>
Thread-Topic: question about if a recent Linux patch is compliant with RFC7323 on window scaling
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Date: Wed, 24 Aug 2016 21:06:09 +0000
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Subject: [tcpm] question about if a recent Linux patch is compliant with RFC7323 on window scaling
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Hello David,

I hope this email could reach you well, because I found related
discussions about this topic on window scaling and the case of window
shrinking (or retraction or loss of precision). And I try to make this
question simple.

There is a recent Linux patch at receiver side to round-up advertised
window due to precision loss of window scaling. It reaches my attention
because the same problem could also happen between a pair of Linux and
FreeBSD nodes, and I am not aware of any similar patch in FreeBSD yet.

The Linux patch is this:
http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=
=3D6
07bfbf2d55dd1cfe5368b41c2a81a8c9ccf4723

And I quote some description of the Linux patch below:
> If the sender uses up the entire window before it is shrunk, this can
>have
> chaotic effects on the connection. When sending ACKs,
>tcp_acceptable_seq()
> will notice that the window has been shrunk since tcp_wnd_end() is before
> tp->snd_nxt, which makes it choose tcp_wnd_end() as sequence number.
> This will fail the receivers checks in tcp_sequence() however since it
> is before it's tp->rcv_wup, making it respond with a dupack.


If the Linux sender=B9s behavior is right ("ACK-only packets should be sent
with the largest in-window sequence number that has ever been sent." ref:
https://www.ietf.org/mail-archive/web/tcpm/current/msg10512.html), it
actually chooses "tp->snd_una+tp->snd_wnd" (tcp_wnd_end()) instead of
tp->snd_nxt, as it thought tp->snd_nxt is out of window, in case of
precision loss which made the receiver=B9s advertise-window smaller.
However, the RFC7323 explicitly specifies the receiver should right shift
the exact scale factor, without mentioning any further round-up:

https://tools.ietf.org/html/rfc7323#page-9
>    o  The window field (SEG.WND) of every outgoing segment, with the
>       exception of <SYN> segments, MUST be right-shifted by
>       Rcv.Wind.Shift bits:
>=20
>                     SEG.WND =3D RCV.WND >> Rcv.Wind.Shift

And also if I am understanding correctly, RFC7323 page 10 (2.4. Addressing
Window Retraction) specifies it is the sender=B9s responsibility to handle
the sequence number out of window:
https://tools.ietf.org/html/rfc7323#page-10
>    4)  On first retransmission, or if the sequence number is out of
>        window by less than 2^Rcv.Wind.Shift, then do normal
>        retransmission(s) without regard to the receiver window as long
>        as the original segment was in window when it was sent.
>=20
>    5)  Subsequent retransmissions MAY only be sent if they are within
>        the window announced by the most recent <ACK>.

In your discussion of the link below, if I am understanding correctly, you
were also in favor of "Yes. It is ok to have more receive buffer space
available than you advertise in the window, but not ok to advertise a
larger window than you are willing to accept." So I think you were not in
favor of rounding-up the advertise-window.

https://www.ietf.org/mail-archive/web/tsvwg/current/msg06475.html


So my question is: Is the Linux patch (on receiver side to round-up the
advertise-window) RFC compliant? Or is it just the right thing (vendor
specific) and other systems like FreeBSD have to follow up, so that their
talks can have no problem?

Thanks and apologize in advance if I did not do enough research,
--Cheng Cui




From sanjeev.ranot.81@gmail.com  Thu Aug 25 01:29:33 2016
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From: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Date: Thu, 25 Aug 2016 13:59:30 +0530
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Subject: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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--94eb2c11aff463a473053ae132f7
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Hi,

please can the team explain why we don't use 15 as a shift count instead
of 14 as mentioned in RFC 7323. Is it because of machines not supporting
unsigned storage or is it more than sufficient guard for denying old or
unwanted segment retransmissions from the previous window.

My understanding:
We know that 2^32 sequence counter (sequence counter is the the sequence
field in TCP header) has valid sequence numbers from 0 to 4294967295. To
determine if a data segment is "old" or "new" it sets the window size to
floor((2^32-1 )/2) +1 since sender's and receiver's window can be out of
phase by at-most 1 window.
1st part has valid range of sequence numbers from = 0 to 2147483647 and
since its contained inside size of of 2^32 therefore we get another valid
range of sequence numbers from 2147483648 to 4294967295. So if the sender
has sent all its packets within the window from 0 to 2147483647 and receiver
has accepted all the packets and sent cumulative acks as well for all of
these then it advances RCV.NXT to start of next maximum available range
of sequence numbers.
Upon receiving all cumulative acks the sender also advance to next set of
available sequence numbers. Otherwise, if considering the scenario when
spurious retransmission occurs or another scenario where all cumulative acks
are lost and retransmission timeout or another scenario when cumulative ack
carrying widow update information is lost, then in such cases the sender
will retransmit segment with sequence number that will fail receiver's
segment acceptability test.

If there is a reason behind this then please cite in response. Also please
tell if such change is valid then do we need to report it as correction or
as a
change for RFC 7323.

Thanks & Regards,
Sanjeev Ranot

--94eb2c11aff463a473053ae132f7
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<div dir=3D"ltr">Hi,<br><br>please can the team explain why we don&#39;t us=
e 15 as a shift count instead<br>of 14 as mentioned in RFC 7323. Is it beca=
use of machines not supporting<br>unsigned storage or is it more than suffi=
cient guard for denying old or<br>unwanted segment retransmissions from the=
 previous window.<br><br>My understanding:<br>We know that 2^32 sequence co=
unter (sequence counter is the the sequence<br>field in TCP header) has val=
id sequence numbers from 0 to 4294967295. To<br>determine if a data segment=
 is &quot;old&quot; or &quot;new&quot; it sets the window size to<br>floor(=
(2^32-1 )/2) +1 since sender&#39;s and receiver&#39;s window can be out of<=
br>phase by at-most 1 window.<br>1st part has valid range of sequence numbe=
rs from =3D 0 to 2147483647 and<br>since its contained inside size of of 2^=
32 therefore we get another valid<br>range of sequence numbers from 2147483=
648 to 4294967295. So if the sender<br>has sent all its packets within the =
window from 0 to 2147483647 and receiver<br>has accepted all the packets an=
d sent cumulative acks as well for all of<br>these then it advances RCV.NXT=
 to start of next maximum available range<br>of sequence numbers.<br>Upon r=
eceiving all cumulative acks the sender also advance to next set of <br>ava=
ilable sequence numbers. Otherwise, if considering the scenario when <br>sp=
urious retransmission occurs or another scenario where all cumulative acks<=
br>are lost and retransmission timeout or another scenario when cumulative =
ack<br>carrying widow update information is lost, then in such cases the se=
nder <br>will retransmit segment with sequence number that will fail receiv=
er&#39;s <br>segment acceptability test.<br><br>If there is a reason behind=
 this then please cite in response. Also please <br>tell if such change is =
valid then do we need to report it as correction or as a <br>change for RFC=
 7323.<br><br>Thanks &amp; Regards,<br>Sanjeev Ranot<br></div>

--94eb2c11aff463a473053ae132f7--


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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Thu, 25 Aug 2016 12:01:04 -0700
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hello Sanjeev,

I wrote a draft about it before.
I am still working on it as I need to do some studies a bit more, but I
think it's probably ok to use 15 as a shift count.
A question would be how to migrate it even if using 15 is possible.
Or, we can publish this draft as reference information as we've seen this
question from time to time.

https://tools.ietf.org/html/draft-nishida-tcpm-maxwin-00

Thanks,
--
Yoshi

On Thu, Aug 25, 2016 at 1:29 AM, Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
wrote:

> Hi,
>
> please can the team explain why we don't use 15 as a shift count instead
> of 14 as mentioned in RFC 7323. Is it because of machines not supporting
> unsigned storage or is it more than sufficient guard for denying old or
> unwanted segment retransmissions from the previous window.
>
> My understanding:
> We know that 2^32 sequence counter (sequence counter is the the sequence
> field in TCP header) has valid sequence numbers from 0 to 4294967295. To
> determine if a data segment is "old" or "new" it sets the window size to
> floor((2^32-1 )/2) +1 since sender's and receiver's window can be out of
> phase by at-most 1 window.
> 1st part has valid range of sequence numbers from = 0 to 2147483647 and
> since its contained inside size of of 2^32 therefore we get another valid
> range of sequence numbers from 2147483648 to 4294967295. So if the sender
> has sent all its packets within the window from 0 to 2147483647 and
> receiver
> has accepted all the packets and sent cumulative acks as well for all of
> these then it advances RCV.NXT to start of next maximum available range
> of sequence numbers.
> Upon receiving all cumulative acks the sender also advance to next set of
> available sequence numbers. Otherwise, if considering the scenario when
> spurious retransmission occurs or another scenario where all cumulative
> acks
> are lost and retransmission timeout or another scenario when cumulative ack
> carrying widow update information is lost, then in such cases the sender
> will retransmit segment with sequence number that will fail receiver's
> segment acceptability test.
>
> If there is a reason behind this then please cite in response. Also please
> tell if such change is valid then do we need to report it as correction or
> as a
> change for RFC 7323.
>
> Thanks & Regards,
> Sanjeev Ranot
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>

--94eb2c0568b00841f3053aea0559
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<div dir=3D"ltr">Hello Sanjeev,<div><br></div><div>I wrote a draft about it=
 before.=C2=A0</div><div>I am still working on it as I need to do some stud=
ies a bit more, but I think it&#39;s probably ok to use 15 as a shift count=
.</div><div>A question would be how to migrate it even if using 15 is possi=
ble.=C2=A0</div><div>Or, we can publish this draft as reference information=
 as we&#39;ve seen this question from time to time.</div><div><br></div><di=
v><a href=3D"https://tools.ietf.org/html/draft-nishida-tcpm-maxwin-00">http=
s://tools.ietf.org/html/draft-nishida-tcpm-maxwin-00</a></div><div><br></di=
v><div>Thanks,</div><div class=3D"gmail_extra">--</div><div class=3D"gmail_=
extra">Yoshi</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Thu, Aug 25, 2016 at 1:29 AM, Sanjeev Ranot <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:sanjeev.ranot.81@gmail.com" target=3D"_blank">sanjeev.ranot.81=
@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex"><div dir=3D"ltr">Hi,<br><br>please can the team explain why we d=
on&#39;t use 15 as a shift count instead<br>of 14 as mentioned in RFC 7323.=
 Is it because of machines not supporting<br>unsigned storage or is it more=
 than sufficient guard for denying old or<br>unwanted segment retransmissio=
ns from the previous window.<br><br>My understanding:<br>We know that 2^32 =
sequence counter (sequence counter is the the sequence<br>field in TCP head=
er) has valid sequence numbers from 0 to 4294967295. To<br>determine if a d=
ata segment is &quot;old&quot; or &quot;new&quot; it sets the window size t=
o<br>floor((2^32-1 )/2) +1 since sender&#39;s and receiver&#39;s window can=
 be out of<br>phase by at-most 1 window.<br>1st part has valid range of seq=
uence numbers from =3D 0 to 2147483647 and<br>since its contained inside si=
ze of of 2^32 therefore we get another valid<br>range of sequence numbers f=
rom 2147483648 to 4294967295. So if the sender<br>has sent all its packets =
within the window from 0 to 2147483647 and receiver<br>has accepted all the=
 packets and sent cumulative acks as well for all of<br>these then it advan=
ces RCV.NXT to start of next maximum available range<br>of sequence numbers=
.<br>Upon receiving all cumulative acks the sender also advance to next set=
 of <br>available sequence numbers. Otherwise, if considering the scenario =
when <br>spurious retransmission occurs or another scenario where all cumul=
ative acks<br>are lost and retransmission timeout or another scenario when =
cumulative ack<br>carrying widow update information is lost, then in such c=
ases the sender <br>will retransmit segment with sequence number that will =
fail receiver&#39;s <br>segment acceptability test.<br><br>If there is a re=
ason behind this then please cite in response. Also please <br>tell if such=
 change is valid then do we need to report it as correction or as a <br>cha=
nge for RFC 7323.<br><br>Thanks &amp; Regards,<br>Sanjeev Ranot<br></div>
<br>______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a><br>
<br></blockquote></div><br></div></div>

--94eb2c0568b00841f3053aea0559--


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From: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Date: Sun, 28 Aug 2016 14:25:09 +0530
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hi Yoshi,

thanks for replying. I have done some work in these couple of days on
how exchange and interoperability of shift count 15 or greater will work.

Basically my idea is for the new TCP stack (which will support shf.cnt >=
15)
to exchange shf.cnt even in Data Transfer Phase and not only in connection
establishment. I have presented this exchange work in the document enclosed.

Please have a look at the document and let me know for any info or
clarification
required. I hope it helps in your related studies and work.

Please feel free to write and discuss.

Thanks & Regards,
Sanjeev Ranot


On 26 August 2016 at 00:31, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
wrote:

> Hello Sanjeev,
>
> I wrote a draft about it before.
> I am still working on it as I need to do some studies a bit more, but I
> think it's probably ok to use 15 as a shift count.
> A question would be how to migrate it even if using 15 is possible.
> Or, we can publish this draft as reference information as we've seen this
> question from time to time.
>
> https://tools.ietf.org/html/draft-nishida-tcpm-maxwin-00
>
> Thanks,
> --
> Yoshi
>
> On Thu, Aug 25, 2016 at 1:29 AM, Sanjeev Ranot <sanjeev.ranot.81@gmail.com
> > wrote:
>
>> Hi,
>>
>> please can the team explain why we don't use 15 as a shift count instead
>> of 14 as mentioned in RFC 7323. Is it because of machines not supporting
>> unsigned storage or is it more than sufficient guard for denying old or
>> unwanted segment retransmissions from the previous window.
>>
>> My understanding:
>> We know that 2^32 sequence counter (sequence counter is the the sequence
>> field in TCP header) has valid sequence numbers from 0 to 4294967295. To
>> determine if a data segment is "old" or "new" it sets the window size to
>> floor((2^32-1 )/2) +1 since sender's and receiver's window can be out of
>> phase by at-most 1 window.
>> 1st part has valid range of sequence numbers from = 0 to 2147483647 and
>> since its contained inside size of of 2^32 therefore we get another valid
>> range of sequence numbers from 2147483648 to 4294967295. So if the sender
>> has sent all its packets within the window from 0 to 2147483647 and
>> receiver
>> has accepted all the packets and sent cumulative acks as well for all of
>> these then it advances RCV.NXT to start of next maximum available range
>> of sequence numbers.
>> Upon receiving all cumulative acks the sender also advance to next set of
>> available sequence numbers. Otherwise, if considering the scenario when
>> spurious retransmission occurs or another scenario where all cumulative
>> acks
>> are lost and retransmission timeout or another scenario when cumulative
>> ack
>> carrying widow update information is lost, then in such cases the sender
>> will retransmit segment with sequence number that will fail receiver's
>> segment acceptability test.
>>
>> If there is a reason behind this then please cite in response. Also
>> please
>> tell if such change is valid then do we need to report it as correction
>> or as a
>> change for RFC 7323.
>>
>> Thanks & Regards,
>> Sanjeev Ranot
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>>
>

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<div dir=3D"ltr"><div><div><div>Hi Yoshi,<br><br></div>thanks for replying.=
 I have done some work in these couple of days on <br></div>how exchange an=
d interoperability of shift count 15 or greater will work. </div><div></div=
><div><br></div><div>Basically my idea is for the new TCP stack (which will=
 support shf.cnt &gt;=3D 15)<br></div><div>to exchange shf.cnt even in Data=
 Transfer Phase and not only in connection<br></div><div>establishment. I h=
ave presented this exchange work in the document enclosed.<br><br></div><di=
v>Please have a look at the document and let me know for any info or clarif=
ication<br></div><div>required. I hope it helps in your related studies and=
 work. <br><br></div><div>Please feel free to write and discuss.<br><br></d=
iv><div>Thanks &amp; Regards,<br></div><div>Sanjeev Ranot<br></div><div><di=
v><br></div></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_=
quote">On 26 August 2016 at 00:31, Yoshifumi Nishida <span dir=3D"ltr">&lt;=
<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">nishida@sfc.wid=
e.ad.jp</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D=
"ltr">Hello Sanjeev,<div><br></div><div>I wrote a draft about it before.=C2=
=A0</div><div>I am still working on it as I need to do some studies a bit m=
ore, but I think it&#39;s probably ok to use 15 as a shift count.</div><div=
>A question would be how to migrate it even if using 15 is possible.=C2=A0<=
/div><div>Or, we can publish this draft as reference information as we&#39;=
ve seen this question from time to time.</div><div><br></div><div><a href=
=3D"https://tools.ietf.org/html/draft-nishida-tcpm-maxwin-00" target=3D"_bl=
ank">https://tools.ietf.org/html/<wbr>draft-nishida-tcpm-maxwin-00</a></div=
><div><br></div><div>Thanks,</div><div class=3D"gmail_extra">--</div><div c=
lass=3D"gmail_extra">Yoshi</div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote"><div><div class=3D"h5">On Thu, Aug 25, 2016 at 1:29 AM, Sa=
njeev Ranot <span dir=3D"ltr">&lt;<a href=3D"mailto:sanjeev.ranot.81@gmail.=
com" target=3D"_blank">sanjeev.ranot.81@gmail.com</a>&gt;</span> wrote:<br>=
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><div cla=
ss=3D"h5"><div dir=3D"ltr">Hi,<br><br>please can the team explain why we do=
n&#39;t use 15 as a shift count instead<br>of 14 as mentioned in RFC 7323. =
Is it because of machines not supporting<br>unsigned storage or is it more =
than sufficient guard for denying old or<br>unwanted segment retransmission=
s from the previous window.<br><br>My understanding:<br>We know that 2^32 s=
equence counter (sequence counter is the the sequence<br>field in TCP heade=
r) has valid sequence numbers from 0 to 4294967295. To<br>determine if a da=
ta segment is &quot;old&quot; or &quot;new&quot; it sets the window size to=
<br>floor((2^32-1 )/2) +1 since sender&#39;s and receiver&#39;s window can =
be out of<br>phase by at-most 1 window.<br>1st part has valid range of sequ=
ence numbers from =3D 0 to 2147483647 and<br>since its contained inside siz=
e of of 2^32 therefore we get another valid<br>range of sequence numbers fr=
om 2147483648 to 4294967295. So if the sender<br>has sent all its packets w=
ithin the window from 0 to 2147483647 and receiver<br>has accepted all the =
packets and sent cumulative acks as well for all of<br>these then it advanc=
es RCV.NXT to start of next maximum available range<br>of sequence numbers.=
<br>Upon receiving all cumulative acks the sender also advance to next set =
of <br>available sequence numbers. Otherwise, if considering the scenario w=
hen <br>spurious retransmission occurs or another scenario where all cumula=
tive acks<br>are lost and retransmission timeout or another scenario when c=
umulative ack<br>carrying widow update information is lost, then in such ca=
ses the sender <br>will retransmit segment with sequence number that will f=
ail receiver&#39;s <br>segment acceptability test.<br><br>If there is a rea=
son behind this then please cite in response. Also please <br>tell if such =
change is valid then do we need to report it as correction or as a <br>chan=
ge for RFC 7323.<br><br>Thanks &amp; Regards,<br>Sanjeev Ranot<br></div>
<br></div></div>______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
<br></blockquote></div><br></div></div>
</blockquote></div><br></div>

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--001a114d9492ab9370053b1de7cd--


From nobody Mon Aug 29 00:32:22 2016
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Mon, 29 Aug 2016 00:32:12 -0700
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hi Sanjeev,

On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot
<sanjeev.ranot.81@gmail.com> wrote:
> Hi Yoshi,
>
> thanks for replying. I have done some work in these couple of days on
> how exchange and interoperability of shift count 15 or greater will work.
>
> Basically my idea is for the new TCP stack (which will support shf.cnt >=
> 15)
> to exchange shf.cnt even in Data Transfer Phase and not only in connection
> establishment. I have presented this exchange work in the document enclosed.
>
> Please have a look at the document and let me know for any info or
> clarification

Thanks for the doc.
I personally think you might want to clarify why we want to change
shift count in data transfer phase a bit more.
This is because even if we use shift count=15, I think we can specify
rather small window size by using a small number for the timestamp
value. So, if the stack supports shift count=15, I think it can just
use it from the beginning of the data transfer in most cases.

If your focus is to have a feature to update shift count during data
transfer phase in general, I think it can be a separate proposal from
increasing max shift count. Because it looks a more broad topic to me.

Thanks,
--
Yoshi


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Veaceslav ROMAN <Veaceslav.Roman@orange.md>, Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] [e2e] TCP HyStart patch deployment
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Piers O'Hanlon <p.ohanlon@gmail.com>, Sangtae Ha <sangtae.ha@gmail.com>, Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Eric Dumazet <edumazet@google.com>, "end2end-interest@postel.org" <end2end-interest@postel.org>
Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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References: <81564C0D7D4D2A4B9A86C8C7404A13DA32145DE4@ESESSMB205.ericsson.se> <1FFD9A11-ADF2-4BA6-A9D3-D8997E1A13E5@gmail.com> <81564C0D7D4D2A4B9A86C8C7404A13DA34B2E666@ESESSMB205.ericsson.se> <7DBBB686E19D2049ADAACD210B474BB10166E64B65@XCHSRV01.main.orange.md> <CADVnQy=6eRAd_HGw7gcbKXdo+vHSKQ+PuyvoqpB+iNBeDjCo+A@mail.gmail.com> <7DBBB686E19D2049ADAACD210B474BB10166E65133@XCHSRV01.main.orange.md> <CADVnQymN6KYDR7jPvrv5oJsqv0tDNqxWETdHgubW=GmrPLjc0Q@mail.gmail.com> <7DBBB686E19D2049ADAACD210B474BB10166E676EF@XCHSRV01.main.orange.md> <CADVnQymtsM2cb9BkQOzrtNaHfJR7KL6BFXv753hxEnqyW5denQ@mail.gmail.com> <1441314842.8932.222.camel@edumazet-glaptop2.roam.corp.google.com> <CADVnQykn6WgCvgnUnWOVR2CkQ9r3_Bro_Vm6V_BPAo4fEUa4Gg@mail.gmail.com> <CADVnQynMTrG+C=hpdP7nz=mj6BCzN4DiE67NKhiaen7kOrYqbQ@mail.gmail.com> <1441385297.17208.6.camel@edumazet-glaptop2.roam.corp.google.com> <7DBBB686E19D2049ADAACD210B474BB10166E856CA@XCHSRV01.main.orange.md> <81564C0D7D4D2A4B9A86C8C7404A13DA34BD84BC@ESESSMB205.ericsson.se> <7DBBB686E19D2049ADAACD210B474BB10166E859FB@XCHSRV01.main.orange.md> <81564C0D7D4D2A4B9A86C8C7404A13DA34BD86EC@ESESSMB205.ericsson.se> <7DBBB686E19D2049ADAACD210B474BB10166E85C4E@XCHSRV01.main.orange.md> <81564C0D7D4D2A4B9A86C8C7404A13DA34BD880B@ESESSMB205.ericsson.se> <CANn89iJ1MHtR6RsSQs0WL9gohMQpk87eZGGG7wysxgH-CumV_Q@mail.gmail.com> <7DBBB686E19D2049ADAACD210B474BB10166E8BDBA@XCHSRV01.main.orange.md> <CADVnQynu=OA9eOb0vBx8gwFMZTkzMC6GsFHuhfiaF+mqB4OCdA@mail.gmail.com> <7DBBB686E19D2049ADAACD210B474BB10167E64F3E@XCHSRV01.main.orange.md> <81564C0D7D4D2A4B9A86C8C7404A13DA34EB08A1@ESESSMB205.ericsson.se> <7DBBB686E19D2049ADAACD210B474BB10167E8DC6E@XCHSRV01.main.orange.md> <DB4PR07MB348A247A23C586B7215B529C2E00@DB4PR07MB348.eurprd07.prod.outlook.com>
From: Neal Cardwell <ncardwell@google.com>
Date: Tue, 30 Aug 2016 10:00:24 -0400
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To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Piers O'Hanlon <p.ohanlon@gmail.com>, Sangtae Ha <sangtae.ha@gmail.com>, Eric Dumazet <edumazet@google.com>, "end2end-interest@postel.org" <end2end-interest@postel.org>
Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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Hi,

As one data point, at Google the google.com and YouTube TCP stacks use
Eric Dumazet's fq pacing qdisc:
  https://www.ietf.org/proceedings/88/slides/slides-88-tcpm-9.pdf
  https://lwn.net/Articles/564978/
Eric discovered that the Hystart Ack Train mechanism does not work
well with pacing, so for TCP at Google we have long ago disabled the
Hystart Ack Train mechanism, and just use the delay mechanism.

It is also our experience that the Hystart delay mechanism often
triggers spuriously for cellular and wifi links, and is difficult to
tune in a way that works well generally. It may be useful for someone
in the community to do a broad investigation to see if there are
HYSTART_DELAY_MIN,  HYSTART_DELAY_MAX, and threshold ratio values that
work better than the current values across a variety of important
last-mile technologies (cellular, cable, DSL, wifi,...).

neal

On Tue, Aug 30, 2016 at 3:02 AM, Ingemar Johansson S
<ingemar.s.johansson@ericsson.com> wrote:
> Hi
>
> An old thread resumed from our side as we see issues with the HyStart alg=
orithm in Cubic when we evaluate some new LTE and 5G access concepts.
> One problem is the packet train algorithm : I understand that it does not=
 work well with packet pacing and is therefore disabled in QUIC. Is the pac=
ket train algorithm used in general, I guess not in the cases when packet p=
acing is enabled or ?
>
> The other problem is the delay algorithm, my impression based on limited =
simulation experiments is that the HYSTART_DELAY_MIN needs to be set to ~16=
ms to avoid that HyStart exits prematurely. The question is if this causes =
problem with other access types (DSL ?)
>
> Comments welcome
> /Ingemar
>
>
>> -----Original Message-----
>> From: Veaceslav ROMAN [mailto:Veaceslav.Roman@orange.md]
>> Sent: den 3 november 2015 20:06
>> To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>; Neal
>> Cardwell <ncardwell@google.com>
>> Cc: Eric Dumazet <edumazet@google.com>; Eric Dumazet
>> <eric.dumazet@gmail.com>; tcpm@ietf.org; Piers O'Hanlon
>> <p.ohanlon@gmail.com>; Sangtae Ha <sangtae.ha@gmail.com>; end2end-
>> interest@postel.org
>> Subject: RE: [tcpm] [e2e] TCP HyStart patch deployment
>>
>> You are right, it wasn't modified and stayed at 16. Thank you for sugges=
tion.
>> Currently we are trying to make an experimental version were this would =
be
>> possible to make these parameters and few other things externally settab=
le.
>>
>> But, honestly, the further we dig in the code and traces the more the
>> confusion and more questions arises.
>>       Hystart rely on correct identification of round borders. For this
>> hystart_reset sets the border at snd_nxt.
>> But, if several ACK arrive at high speed when the current round border i=
s
>> crossed and the sender has not time to transmit two packets immediately
>> after ACK, then the snd_nxt does not progress and hystart_reset sets the
>> round of the border to a pretty random value and then in the next round =
the
>> hystart measure pieces of two rounds.
>>       Also, the code is calling hystart_update on each ACK before the
>> hystart_reset, as a result at the border of rounds the head of line (HOL=
) ACK
>> delay of round N+1 is accounted in the round N and, because, the HOL
>> usually has less delay, it is frequently become delay_min but not in rou=
nd
>> N+1 and, as a result early exit in round N+1 after counting delays of pa=
ckets 2
>> through 9.
>>       This should also affect train detect as, when crossing the round
>> border, instead of subtracting the time of the first and the last ACK of=
 the
>> train N there will be subtracted times of first ACK of train N and first=
 ACK of
>> the train N+1 which will be approximately equal to RTT and definitely bi=
gger
>> than RTT/2 as per expected algorithm. This shall typically lead to alway=
s exit in
>> train 2. This will affect high speed networks with RTT less than
>> hystart_ack_delta (2ms).
>>              Then call of hystart_reset is subject of check of several c=
onditions like
>> may_rise_cwnd or is_cwnd_limited, while it doesn't look that
>> hystart_update is, so that it may become totally out of sync with round
>> border.
>>
>>       We are still doing efforts to try to find parameters which would m=
ake
>> hystart better, but analysis of many traces make much impression of
>> randomness of exit and not sure whether parameters may help.
>>
>> Veaceslav Roman
>>
>> -----Original Message-----
>> From: Ingemar Johansson S [mailto:ingemar.s.johansson@ericsson.com]
>> Sent: Tuesday, 03 November 2015 15:19
>> To: Veaceslav ROMAN; Neal Cardwell
>> Cc: Eric Dumazet; Eric Dumazet; tcpm@ietf.org; Piers O'Hanlon; Sangtae H=
a;
>> end2end-interest@postel.org; Ingemar Johansson S
>> Subject: RE: [tcpm] [e2e] TCP HyStart patch deployment
>>
>> Hi
>>
>> Thanks for doing the experiments.
>> It may be possible that HyStart may need more extensive modifications.
>> Need to ask the dumb question first though. Is HYSTART_DELAY_MAX
>> modified in your experiment or is it still 16ms ?
>> I guess HYSTART_DELAY_MAX may need to be set to
>>     HYSTART_DELAY_MAX =3D MAX(16ms, HYSTART_DELAY_MIN*2) Or
>> something similar, one can always argue around the factor 2 above.
>>
>> /Ingemar
>>
>> > -----Original Message-----
>> > From: Veaceslav ROMAN [mailto:Veaceslav.Roman@orange.md]
>> > Sent: den 17 oktober 2015 22:48
>> > To: Neal Cardwell
>> > Cc: Eric Dumazet; Ingemar Johansson S; Eric Dumazet; tcpm@ietf.org;
>> > Piers O'Hanlon; Sangtae Ha; end2end-interest@postel.org
>> > Subject: RE: [tcpm] [e2e] TCP HyStart patch deployment
>> >
>> > Hi,
>> > Recompiled kernel to HYSTART_DELAY_MIN  20 ms:
>> >
>> >   Download 10MB, High Bandwidths, HYSTART_DELAY_MIN  20 ms:
>> >            Hystart average download time: 1.18 s
>> >            NoHystart average download time: 1.00  s
>> >
>> > No significant improvement against the case with HYSTART_DELAY_MIN  10
>> > ms.
>> >
>> > Sill Hystart early exit is visible.
>> >
>> > Other tests with HYSTART_DELAY_MIN  20 ms and files of 3MB.
>> >
>> >   Download 3MB, High Bandwidths, HYSTART_DELAY_MIN  20 ms:
>> >            Hystart average download time: 0.47 s
>> >            NoHystart average download time: 0.39  s
>> >
>> > compare with 4 ms
>> >
>> >   Download 3MB, High Bandwidths, HYSTART_DELAY_MIN  4 ms:
>> >            Hystart average download time: 0.65 s
>> >            NoHystart average download time: 0.37  s
>> >
>> > Few more comment: High Bandwidth is typically above 30-40 Mbps.
>> > For Low Bandwidth, when radio do not permit above 40 Mbps Hystart
>> > early exit does not have a significant impact. I do explain it with
>> > the fact that the minimum exit CWND for Hysstart is 29 which, for RTT
>> > of ~15ms (minimum possible in LTE and visible in majority of traces)
>> > leads to ~25 Mbps, so Hystart exit, at least, at 25 Mbps and then it
>> > takes not too long to cubic to reach 30-40 Mbps.
>> >
>> > Veaceslav Roman
>> >
>> > -----Original Message-----
>> > From: Neal Cardwell [mailto:ncardwell@google.com]
>> > Sent: Tuesday, 06 October 2015 04:06
>> > To: Veaceslav ROMAN
>> > Cc: Eric Dumazet; Ingemar Johansson S; Eric Dumazet; tcpm@ietf.org;
>> > Piers O'Hanlon; Sangtae Ha; end2end-interest@postel.org
>> > Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
>> >
>> > On Mon, Oct 5, 2015 at 6:07 PM, Veaceslav ROMAN
>> > <Veaceslav.Roman@orange.md> wrote:
>> > > We've managed to recompile the kernel 4.1 with tcp_cubic
>> > HYSTART_DELAY_MIN  (10U<<3) , 10ms.
>> > > Well, we had time only for one test cycle, but results are very prom=
ising:
>> > >
>> > > Download 10MB, High Bandwidths, HYSTART_DELAY_MIN  10 ms:
>> > >            Hystart average download time: 1.16 s
>> > >            NoHystart average download time: 0.93  s
>> > ...
>> > >  Download 10MB, High Bandwidths, HYSTART_DELAY_MIN  4 ms:
>> > >            Hystart average download time: 1.42 s
>> > >            NoHystart average download time: 0.89  s
>> >
>> > Thanks! That is very useful and interesting data. Would you be able to
>> > try a few other values, like 15ms and 20ms?
>> >
>> > neal


From nobody Tue Aug 30 08:33:43 2016
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From: "Cui, Cheng" <Cheng.Cui@netapp.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: question about if a recent Linux patch is compliant with RFC7323 on window scaling
Thread-Index: AQHR/ktWq33ah5ClWE2yMFWt4EpCvqBhx/sA
Date: Tue, 30 Aug 2016 14:16:47 +0000
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Subject: Re: [tcpm] question about if a recent Linux patch is compliant with RFC7323 on window scaling
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Six days have passed and no response yet. Can anyone in this board make a c=
omment?

Thanks,
--Cheng Cui
NetApp Scale Out Networking


On 8/24/16, 5:06 PM, "Cui, Cheng" <Cheng.Cui@netapp.com> wrote:

    Hello David,
   =20
    I hope this email could reach you well, because I found related
    discussions about this topic on window scaling and the case of window
    shrinking (or retraction or loss of precision). And I try to make this
    question simple.
   =20
    There is a recent Linux patch at receiver side to round-up advertised
    window due to precision loss of window scaling. It reaches my attention
    because the same problem could also happen between a pair of Linux and
    FreeBSD nodes, and I am not aware of any similar patch in FreeBSD yet.
   =20
    The Linux patch is this:
    http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?=
id=3D6
    07bfbf2d55dd1cfe5368b41c2a81a8c9ccf4723
   =20
    And I quote some description of the Linux patch below:
    > If the sender uses up the entire window before it is shrunk, this can
    >have
    > chaotic effects on the connection. When sending ACKs,
    >tcp_acceptable_seq()
    > will notice that the window has been shrunk since tcp_wnd_end() is be=
fore
    > tp->snd_nxt, which makes it choose tcp_wnd_end() as sequence number.
    > This will fail the receivers checks in tcp_sequence() however since i=
t
    > is before it's tp->rcv_wup, making it respond with a dupack.
   =20
   =20
    If the Linux sender?s behavior is right ("ACK-only packets should be se=
nt
    with the largest in-window sequence number that has ever been sent." re=
f:
    https://www.ietf.org/mail-archive/web/tcpm/current/msg10512.html), it
    actually chooses "tp->snd_una+tp->snd_wnd" (tcp_wnd_end()) instead of
    tp->snd_nxt, as it thought tp->snd_nxt is out of window, in case of
    precision loss which made the receiver?s advertise-window smaller.
    However, the RFC7323 explicitly specifies the receiver should right shi=
ft
    the exact scale factor, without mentioning any further round-up:
   =20
    https://tools.ietf.org/html/rfc7323#page-9
    >    o  The window field (SEG.WND) of every outgoing segment, with the
    >       exception of <SYN> segments, MUST be right-shifted by
    >       Rcv.Wind.Shift bits:
    >=20
    >                     SEG.WND =3D RCV.WND >> Rcv.Wind.Shift
   =20
    And also if I am understanding correctly, RFC7323 page 10 (2.4. Address=
ing
    Window Retraction) specifies it is the sender?s responsibility to handl=
e
    the sequence number out of window:
    https://tools.ietf.org/html/rfc7323#page-10
    >    4)  On first retransmission, or if the sequence number is out of
    >        window by less than 2^Rcv.Wind.Shift, then do normal
    >        retransmission(s) without regard to the receiver window as lon=
g
    >        as the original segment was in window when it was sent.
    >=20
    >    5)  Subsequent retransmissions MAY only be sent if they are within
    >        the window announced by the most recent <ACK>.
   =20
    In your discussion of the link below, if I am understanding correctly, =
you
    were also in favor of "Yes. It is ok to have more receive buffer space
    available than you advertise in the window, but not ok to advertise a
    larger window than you are willing to accept." So I think you were not =
in
    favor of rounding-up the advertise-window.
   =20
    https://www.ietf.org/mail-archive/web/tsvwg/current/msg06475.html
   =20
   =20
    So my question is: Is the Linux patch (on receiver side to round-up the
    advertise-window) RFC compliant? Or is it just the right thing (vendor
    specific) and other systems like FreeBSD have to follow up, so that the=
ir
    talks can have no problem?
   =20
    Thanks and apologize in advance if I did not do enough research,
    --Cheng Cui
   =20
   =20
   =20
   =20
   =20
   =20



From nobody Tue Aug 30 13:35:35 2016
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Changed milestone "Submit document on a time-based fast loss detection
algorithm for TCP to the IESG for publication as an Experimental RFC",
set state to active from review, accepting new milestone.

URL: https://datatracker.ietf.org/wg/tcpm/charter/


From nobody Wed Aug 31 02:23:08 2016
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A new meeting session request has just been submitted by Michael Scharf, a Chair of the tcpm working group.


---------------------------------------------------------
Working Group Name: TCP Maintenance and Minor Extensions
Area Name: Transport Area
Session Requester: Michael Scharf

Number of Sessions: 1
Length of Session(s):  2.5 Hours
Number of Attendees: 80
Conflicts to Avoid: 
 First Priority: tsvwg tsvarea taps mptcp tcpinc httpbis iccrg 
 Second Priority: rmcat rtcweb aqm dclcrg maprg
 Third Priority: ippm teas


Special Requests:
  Further conflicts: quic (if approved), plus (if approved), other BoFs in TSV
---------------------------------------------------------


From nobody Wed Aug 31 02:41:27 2016
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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] WG acceptance of draft-cheng-tcpm-rack-01
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Hi all,

I am not aware of further comments as a response to my email below. The con=
sensus is therefore to adopt draft-cheng-tcpm-rack-01 as new TCPM WG item.=
=20

@authors: Please resubmit as draft-ietf-tcpm-...

I'd suggest that the authors make a proposal whether to include TLP. We can=
 come back to this question in one of the upcoming TCPM meetings, preferabl=
y based on a text proposal.=20

Thanks

Michael=20


-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael (Nok=
ia - DE)
Sent: Tuesday, July 19, 2016 8:37 AM
To: tcpm@ietf.org
Subject: [tcpm] WG acceptance of draft-cheng-tcpm-rack-01

Hi all,

Yesterday we had uniform consensus in the room to adopt draft-cheng-tcpm-ra=
ck-01 as a new TCPM working group item. As a remark, the support was the st=
rongest I have recently seen in TCPM.=20

Therefore I'd like to confirm on the list that it is working group consensu=
s to add a new milestone to the TCPM charter

  Submit document on a time-based fast loss detection algorithm for TCP (RA=
CK) to the IESG for publication as an Experimental RFC

and to use draft-cheng-tcpm-rack-01 as a starting point.

Two additional notes:

- It has been mentioned that further experimentation is needed e.g. regardi=
ng the parameters and therefore the suggested status is Experimental

- It has been proposed to also document TLP in this document as it is timer=
-based mechanism as well

If you have any feedback or suggestions, please let us know in the next two=
 weeks.

Thanks

Michael

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


From nobody Wed Aug 31 08:01:44 2016
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From: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Date: Wed, 31 Aug 2016 09:32:05 +0530
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To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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--94eb2c12474e132df3053b56294f
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Hi Yoshi,

Yes, using shf.cnt of 15 in connection establishment phase is sufficient
enough, but  while thinking from migration and interoperability this
suggestion of changing shf.cnt during Data Transfer Phase came to mind.

Present behavior is that if window limit is reached then we cannot
transfer no matter how big cwnd is. This limits the rate of growth of
congestion window if not growth completely. Now with high speed interfaces
it will not always be the case that we increase size of MTU (as for jumbo
frames) to get the bandwidth with support of bundling (etherChannels).

Generally when shf.cnt is chosen it defines the size of SND.WND
within range from 0 to max SND.WND. SND.WND proves to be a
constraint in situation discussed below.

Note: With PAWS we need to make sure that it is sufficiently large
enough to provide ample monotonically non-decreasing numbers so that
they can be differentiated or make its use in such a way so that
its always sufficient to distinguish at-least between 2 successive
windows. RFC 7323 defines such a way by putting same TSval for all
segments belonging to the same window.

Before citing situation for the need of dynamic exchange of shf.cnt we
need to discuss why sequence number of greater than 32 bit is not needed
because of PAWS and PAWS timestamp Interface transfer speed and window
size are related

With PAWS, each wrap of timestamp must be >= 2*MSL. This MSL is based
on transfer rate. Assuming 1 timestamp tick is 1 ms, hence if we
transfer 2^30 bytes of data in each window i.e. with shf.cnt = 14 then
maximum transfer rate which we need to get this big a window is of
1 Tera bytes (TbE) with PAWS in place

1000 ms (1s) = 1000 Gb (1TbE)
1ms  = 1 Gb ~ 2^30

Therefore with PAWS timestamp tick of 1ms allows 1 Gb of window size that
we can transmit safely without data corruption due to duplicate data or
ACK segments. This minimum timestamp click needed for PAWS safety is
provided with a base consideration of sequence number of 32 bits in size.
It takes 49.6 days for a 32 bit sequence number to wrap again when
timestamp tick is 1ms.  If in future we were to increase sequence numbers
to 64 bits, then this timestamp tick can be reduced or we MAY never need
PAWS again thereafter if 64 bit sequence number is used and we can never
increase bandwidth speed no matter how much we bundle cables.

The situation discussed assume that we are never going to increase the
sequence number size and we are good with PAWS to provide us safety from
duplicate data and ACK segment or of old packets from previous incarnations
of the same TCP connection.

Another example with shf.cnt = 15 :
With shf.cnt = 15
1000ms  = 2000 Gb (2TbE)
1ms        = 2000000000 bits = 2 Gb ~ 2^31
Therefore with a speed of 2Tbps and 1 ms timestamp tick for PAWS we can
get a window size of 2 Gb.

*Migration Notes:*
This means that when we introduce a shf.cnt of 15, then we MAY either let
PAWS timestamp tick to remain same as used for 1 Gb of window size with
shf.cnt of 14 if its sufficient enough or we may think to readjust if we
want to gain more speed by lowering the previous PAWS timestamp.

Similary with a speed of 4Tbps and 1 ms timestamp tick for PAWS we can
get a window size of 4 Gb.

*Situation:*
TCP connection is going on for days using same shf.cnt which was
exchanged on day 1 when this TCP connection was established. Suppose
due to incredible high transfer rate of the inter-continent link
cwnd == SND.WND was already reached at end of day 1 and now the
transfer rate is limited by SND.WND (decided by shf.cnt) when only
very small fraction of link speed is utilized. Now suppose a very
huge surplus of memory is made available by hot-swappable memory
card inserted on both ends while TCP connection is running its
Data Transfer Phase. Shf.cnt was already decided in beginning of
this TCP connection and it was decided on the basis of amount on
total memory available at that time to be allocated to its RCV.BUF.
It required restarting this ongoing connection to have a new shf.cnt
that will now consider this new additional memory surplus (i.e.
increase RCV.BUF) and start all over again from start or further
from the last index received is area of improvement.

*My proposal is that (this is separate from the on going work on shf.cnt =
15) :*
If TCP stack have support of exchanging shf.cnt in data transfer phase
when large amount of memory is available dynamically then we can achieve
significant high transfer rates without ever having to restart the
connection.

This feature comes in heading "Dynamically adjustable TCP window" and
shf.cnt_interoperability.pdf mentions one of the possible mechanisms
for implementing this introducing 7 new state variables and 1 TCP flag.

Also, just to reiterate regarding migration purposes please look into
PAWS and discuss with stake holders if they are ok with existing
timestamp for new shf.cnt of 15.


Thanks & Regards,
Sanjeev Ranot



On 29 August 2016 at 13:02, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
wrote:

> Hi Sanjeev,
>
> On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot
> <sanjeev.ranot.81@gmail.com> wrote:
> > Hi Yoshi,
> >
> > thanks for replying. I have done some work in these couple of days on
> > how exchange and interoperability of shift count 15 or greater will work.
> >
> > Basically my idea is for the new TCP stack (which will support shf.cnt >=
> > 15)
> > to exchange shf.cnt even in Data Transfer Phase and not only in
> connection
> > establishment. I have presented this exchange work in the document
> enclosed.
> >
> > Please have a look at the document and let me know for any info or
> > clarification
>
> Thanks for the doc.
> I personally think you might want to clarify why we want to change
> shift count in data transfer phase a bit more.
> This is because even if we use shift count=15, I think we can specify
> rather small window size by using a small number for the timestamp
> value. So, if the stack supports shift count=15, I think it can just
> use it from the beginning of the data transfer in most cases.
>
> If your focus is to have a feature to update shift count during data
> transfer phase in general, I think it can be a separate proposal from
> increasing max shift count. Because it looks a more broad topic to me.
>
> Thanks,
> --
> Yoshi
>

--94eb2c12474e132df3053b56294f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hi Yoshi,<br><br>Yes, using shf.cnt of 15 in connecti=
on establishment phase is sufficient <br>enough, but=C2=A0 while thinking f=
rom migration and interoperability this <br>suggestion of changing shf.cnt =
during Data Transfer Phase came to mind.<br><br>Present behavior is that if=
 window limit is reached then we cannot <br>transfer no matter how big cwnd=
 is. This limits the rate of growth of <br>congestion window if not growth =
completely. Now with high speed interfaces <br>it will not always be the ca=
se that we increase size of MTU (as for jumbo <br>frames) to get the bandwi=
dth with support of bundling (etherChannels).<br><br>Generally when shf.cnt=
 is chosen it defines the size of SND.WND<br>within range from 0 to max SND=
.WND. SND.WND proves to be a <br>constraint in situation discussed below.<b=
r><br>Note: With PAWS we need to make sure that it is sufficiently large <b=
r>enough to provide ample monotonically non-decreasing numbers so that <br>=
they can be differentiated or make its use in such a way so that <br>its al=
ways sufficient to distinguish at-least between 2 successive <br>windows. R=
FC 7323 defines such a way by putting same TSval for all <br>segments belon=
ging to the same window.<br><br>Before citing situation for the need of dyn=
amic exchange of shf.cnt we <br>need to discuss why sequence number of grea=
ter than 32 bit is not needed <br>because of PAWS and PAWS timestamp Interf=
ace transfer speed and window <br>size are related<br><br>With PAWS, each w=
rap of timestamp must be &gt;=3D 2*MSL. This MSL is based <br>on transfer r=
ate. Assuming 1 timestamp tick is 1 ms, hence if we <br>transfer 2^30 bytes=
 of data in each window i.e. with shf.cnt =3D 14 then <br>maximum transfer =
rate which we need to get this big a window is of <br>1 Tera bytes (TbE) wi=
th PAWS in place<br><br>1000 ms (1s) =3D 1000 Gb (1TbE)<br>1ms=C2=A0 =3D 1 =
Gb ~ 2^30 <br><br>Therefore with PAWS timestamp tick of 1ms allows 1 Gb of =
window size that <br>we can transmit safely without data corruption due to =
duplicate data or<br>ACK segments. This minimum timestamp click needed for =
PAWS safety is <br>provided with a base consideration of sequence number of=
 32 bits in size. <br>It takes 49.6 days for a 32 bit sequence number to wr=
ap again when <br>timestamp tick is 1ms.=C2=A0 If in future we were to incr=
ease sequence numbers <br>to 64 bits, then this timestamp tick can be reduc=
ed or we MAY never need <br>PAWS again thereafter if 64 bit sequence number=
 is used and we can never <br>increase bandwidth speed no matter how much w=
e bundle cables. <br><br>The situation discussed assume that we are never g=
oing to increase the <br>sequence number size and we are good with PAWS to =
provide us safety from <br>duplicate data and ACK segment or of old packets=
 from previous incarnations <br>of the same TCP connection.<br><br>Another =
example with shf.cnt =3D 15 :<br>With shf.cnt =3D 15 <br>1000ms=C2=A0 =3D 2=
000 Gb (2TbE)<br>1ms=C2=A0=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0 =3D 2000000000 bi=
ts =3D 2 Gb ~ 2^31<br>Therefore with a speed of 2Tbps and 1 ms timestamp ti=
ck for PAWS we can <br>get a window size of 2 Gb. <br><br><u>Migration Note=
s:</u><br>This means that when we introduce a shf.cnt of 15, then we MAY ei=
ther let <br>PAWS timestamp tick to remain same as used for 1 Gb of window =
size with <br>shf.cnt of 14 if its sufficient enough or we may think to rea=
djust if we <br>want to gain more speed by lowering the previous PAWS times=
tamp.<br><br>Similary with a speed of 4Tbps and 1 ms timestamp tick for PAW=
S we can <br>get a window size of 4 Gb. <br><br><u>Situation:</u> <br>TCP c=
onnection is going on for days using same shf.cnt which was <br>exchanged o=
n day 1 when this TCP connection was established. Suppose <br>due to incred=
ible high transfer rate of the inter-continent link <br>cwnd =3D=3D SND.WND=
 was already reached at end of day 1 and now the <br>transfer rate is limit=
ed by SND.WND (decided by shf.cnt) when only <br>very small fraction of lin=
k speed is utilized. Now suppose a very <br>huge surplus of memory is made =
available by hot-swappable memory <br>card inserted on both ends while TCP =
connection is running its <br>Data Transfer Phase. Shf.cnt was already deci=
ded in beginning of <br>this TCP connection and it was decided on the basis=
 of amount on <br>total memory available at that time to be allocated to it=
s RCV.BUF. <br>It required restarting this ongoing connection to have a new=
 shf.cnt <br>that will now consider this new additional memory surplus (i.e=
. <br>increase RCV.BUF) and start all over again from start or further <br>=
from the last index received is area of improvement. <br><br><u>My proposal=
 is that (this is separate from the on going work on shf.cnt =3D 15) :</u><=
br>If TCP stack have support of exchanging shf.cnt in data transfer phase <=
br>when large amount of memory is available dynamically then we can achieve=
 <br>significant high transfer rates without ever having to restart the con=
nection.<br><br>This feature comes in heading &quot;Dynamically adjustable =
TCP window&quot; and <br>shf.cnt_interoperability.pdf mentions one of the p=
ossible mechanisms <br>for implementing this introducing 7 new state variab=
les and 1 TCP flag.<br><br></div><div>Also, just to reiterate regarding mig=
ration purposes please look into <br>PAWS and discuss with stake holders if=
 they are ok with existing<br></div><div>timestamp for new shf.cnt of 15. <=
br></div><div><br><br></div>Thanks &amp; Regards,<br><div>Sanjeev Ranot<br>=
</div><div><br><br></div></div><div class=3D"gmail_extra"><br><div class=3D=
"gmail_quote">On 29 August 2016 at 13:02, Yoshifumi Nishida <span dir=3D"lt=
r">&lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">nishida@=
sfc.wide.ad.jp</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi S=
anjeev,<br>
<span class=3D""><br>
On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot<br>
&lt;<a href=3D"mailto:sanjeev.ranot.81@gmail.com">sanjeev.ranot.81@gmail.co=
m</a>&gt; wrote:<br>
&gt; Hi Yoshi,<br>
&gt;<br>
&gt; thanks for replying. I have done some work in these couple of days on<=
br>
&gt; how exchange and interoperability of shift count 15 or greater will wo=
rk.<br>
&gt;<br>
&gt; Basically my idea is for the new TCP stack (which will support shf.cnt=
 &gt;=3D<br>
&gt; 15)<br>
&gt; to exchange shf.cnt even in Data Transfer Phase and not only in connec=
tion<br>
&gt; establishment. I have presented this exchange work in the document enc=
losed.<br>
&gt;<br>
&gt; Please have a look at the document and let me know for any info or<br>
&gt; clarification<br>
<br>
</span>Thanks for the doc.<br>
I personally think you might want to clarify why we want to change<br>
shift count in data transfer phase a bit more.<br>
This is because even if we use shift count=3D15, I think we can specify<br>
rather small window size by using a small number for the timestamp<br>
value. So, if the stack supports shift count=3D15, I think it can just<br>
use it from the beginning of the data transfer in most cases.<br>
<br>
If your focus is to have a feature to update shift count during data<br>
transfer phase in general, I think it can be a separate proposal from<br>
increasing max shift count. Because it looks a more broad topic to me.<br>
<br>
Thanks,<br>
--<br>
Yoshi<br>
</blockquote></div><br></div>

--94eb2c12474e132df3053b56294f--


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From: Yuchung Cheng <ycheng@google.com>
Date: Wed, 31 Aug 2016 08:24:52 -0700
Message-ID: <CAK6E8=fSTzi5ZyA-B4n7xkuUksEJxmFYuDF-fQtSVyBEBmFK2Q@mail.gmail.com>
To: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/rBheMc5HKFPl7xRxrVkdrZkc8QA>
Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Matthew Mathis <mattmathis@google.com>, "draft-cheng-tcpm-rack@ietf.org" <draft-cheng-tcpm-rack@ietf.org>
Subject: Re: [tcpm] WG acceptance of draft-cheng-tcpm-rack-01
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On Wed, Aug 31, 2016 at 2:41 AM, Scharf, Michael (Nokia - DE)
<michael.scharf@nokia.com> wrote:
>
> Hi all,
>
> I am not aware of further comments as a response to my email below. The consensus is therefore to adopt draft-cheng-tcpm-rack-01 as new TCPM WG item.
>
> @authors: Please resubmit as draft-ietf-tcpm-...
>
> I'd suggest that the authors make a proposal whether to include TLP. We can come back to this question in one of the upcoming TCPM meetings, preferably based on a text proposal.
Thanks Michael. I've discussed with the authors of TLP Nandita and
Matt. Our consensus was to first fuse TLP (as an optimization) and
RACK together in one draft to get WG feedback. We'd resubmit a draft
before the next meeting deadline.

Personally I am continue to work on RACK implementation. One problem
is the current algorithm requires scanning the scoreboard for every
ACK. This can take too much cycles on fast connection with lots of
losses and sacks. Another test I am running is to use RACK + TLP alone
without any other recovery heuristics in Linux, and make sure we don't
see regressions in boundary cases. I think both are important to make
sure for the "working code" principle for the new proposal .


>
> Thanks
>
> Michael
>
>
> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael (Nokia - DE)
> Sent: Tuesday, July 19, 2016 8:37 AM
> To: tcpm@ietf.org
> Subject: [tcpm] WG acceptance of draft-cheng-tcpm-rack-01
>
> Hi all,
>
> Yesterday we had uniform consensus in the room to adopt draft-cheng-tcpm-rack-01 as a new TCPM working group item. As a remark, the support was the strongest I have recently seen in TCPM.
>
> Therefore I'd like to confirm on the list that it is working group consensus to add a new milestone to the TCPM charter
>
>   Submit document on a time-based fast loss detection algorithm for TCP (RACK) to the IESG for publication as an Experimental RFC
>
> and to use draft-cheng-tcpm-rack-01 as a starting point.
>
> Two additional notes:
>
> - It has been mentioned that further experimentation is needed e.g. regarding the parameters and therefore the suggested status is Experimental
>
> - It has been proposed to also document TLP in this document as it is timer-based mechanism as well
>
> If you have any feedback or suggestions, please let us know in the next two weeks.
>
> Thanks
>
> Michael
>
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