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Subject: [tcpm] I-D Action: draft-ietf-tcpm-newcwv-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : Updating TCP to support Rate-Limited Traffic
	Author(s)       : Godred Fairhurst
                          Arjuna Sathiaseelan
                          Raffaello Secchi
	Filename        : draft-ietf-tcpm-newcwv-02.txt
	Pages           : 19
	Date            : 2013-07-01

Abstract:
   This document proposes an update to RFC 5681 to address issues that
   arise when TCP is used to support traffic that exhibits periods where
   the sending rate is limited by the application rather than the
   congestion window.  It updates TCP to allow a TCP sender to restart
   quickly following either an idle or rate-limited interval.  This
   method is expected to benefit applications that send rate-limited
   traffic using TCP, while also providing an appropriate response if
   congestion is experienced.

   It also evaluates the Experimental specification of TCP Congestion
   Window Validation, CWV, defined in RFC 2861, and concludes that RFC
   2861 sought to address important issues, but failed to deliver a
   widely used solution.  This document therefore recommends that the
   status of RFC 2861 is moved from Experimental to Historic, and that
   it is replaced by the current specification.

   NOTE: The standards status of this WG document is under review for
   consideration as either Experimental (EXP) or Proposed Standard (PS).
   This decision will be made later as the document is finalised.


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

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

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-newcwv-02


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


From gorry@erg.abdn.ac.uk  Tue Jul  2 00:20:14 2013
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Date: Tue, 2 Jul 2013 08:20:06 +0100
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Subject: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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We've just uploaded a new version of our draft. The congestion window
management algorithm has not changed since -00, but in -01 and -02 drafts
we introduce a more robust way to measure the pipeACK that attempts to
account for the wide variety of interactive TCP application behaviour.
This uses a sampling method to sample pipeACK over the last second.

Clearly the cwnd poorly reflects the capacity in the non-validated phase
and I think it would be wrong to use this for TCP control block sharing. 
We therefore also  introduce text that proposes that TCB sharing should
use the pipeACK in place of cwnd when a TCP sender is in the Nonvalidated
phase.  We think this value better reflects the capacity that the flow has
utilised in the network path - but would welcome more thought on this.
Thoughts on this would be most welcome.

Gorry

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the TCP Maintenance and Minor Extensions
> Working Group of the IETF.
>
> 	Title           : Updating TCP to support Rate-Limited Traffic
> 	Author(s)       : Godred Fairhurst
>                           Arjuna Sathiaseelan
>                           Raffaello Secchi
> 	Filename        : draft-ietf-tcpm-newcwv-02.txt
> 	Pages           : 19
> 	Date            : 2013-07-01
>
> Abstract:
>    This document proposes an update to RFC 5681 to address issues that
>    arise when TCP is used to support traffic that exhibits periods where
>    the sending rate is limited by the application rather than the
>    congestion window.  It updates TCP to allow a TCP sender to restart
>    quickly following either an idle or rate-limited interval.  This
>    method is expected to benefit applications that send rate-limited
>    traffic using TCP, while also providing an appropriate response if
>    congestion is experienced.
>
>    It also evaluates the Experimental specification of TCP Congestion
>    Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>    2861 sought to address important issues, but failed to deliver a
>    widely used solution.  This document therefore recommends that the
>    status of RFC 2861 is moved from Experimental to Historic, and that
>    it is replaced by the current specification.
>
>    NOTE: The standards status of this WG document is under review for
>    consideration as either Experimental (EXP) or Proposed Standard (PS).
>    This decision will be made later as the document is finalised.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>
>
> 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 ncardwell@google.com  Tue Jul  2 13:44:44 2013
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From: Neal Cardwell <ncardwell@google.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: [tcpm] packetdrill: a scriptable, portable network stack testing tool
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I'd like to announce the availability of the packetdrill network stack
testing tool.

The packetdrill scripting tool enables users to quickly write precise
tests for entire TCP/UDP/IPv4/IPv6 network stacks, from the system
call layer down to the NIC hardware. packetdrill currently works on
Linux, FreeBSD, OpenBSD, and NetBSD. It can test network stack
behavior over physical NICs on a LAN, or on a single machine using a
tun virtual network device.

The code is licensed under version 2 of the GPL, and available in a
git repository at:

  https://code.google.com/p/packetdrill/

Here's a USENIX 2013 paper about the tool:

  http://research.google.com/pubs/pub41316.html

This paper describes the design and implementation of the tool, and
our experiences using it to execute 657 test cases. The tool was
instrumental in our development of three new features for Linux
TCP=97Early Retransmit, Fast Open, and Loss Probes=97and allowed us to
find and fix 10 bugs in Linux. Our team uses packetdrill in all phases
of the development process.

Currently the source for the testing tool is in the git repository,
along with an example script for each supported OS. We will also be
posting tests from our team's Linux TCP test suite (described in the
paper), as time permits.

There is a mailing list for questions, discussions and patches:

  http://groups.google.com/group/packetdrill

Enjoy!

neal

From Donald.Smith@CenturyLink.com  Wed Jul  3 10:32:18 2013
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From: "Smith, Donald" <Donald.Smith@CenturyLink.com>
To: Neal Cardwell <ncardwell@google.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] packetdrill: a scriptable,	portable network stack testing tool
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Subject: Re: [tcpm] packetdrill: a scriptable, portable network stack testing tool
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Neal, this looks very interesting but how is it different from other simila=
r tools (scrapy etc...).



And THANKS for sharing we all need good tools ;)





(coffee !=3D sleep) & (!coffee =3D=3D sleep)
 Donald.Smith@centurylink.com<mailto:Donald.Smith@centurylink.com>
________________________________
From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of Neal Cardw=
ell [ncardwell@google.com]
Sent: Tuesday, July 02, 2013 2:44 PM
To: tcpm@ietf.org
Subject: [tcpm] packetdrill: a scriptable, portable network stack testing t=
ool

I'd like to announce the availability of the packetdrill network stack
testing tool.

The packetdrill scripting tool enables users to quickly write precise
tests for entire TCP/UDP/IPv4/IPv6 network stacks, from the system
call layer down to the NIC hardware. packetdrill currently works on
Linux, FreeBSD, OpenBSD, and NetBSD. It can test network stack
behavior over physical NICs on a LAN, or on a single machine using a
tun virtual network device.

The code is licensed under version 2 of the GPL, and available in a
git repository at:

  https://code.google.com/p/packetdrill/

Here's a USENIX 2013 paper about the tool:

  http://research.google.com/pubs/pub41316.html

This paper describes the design and implementation of the tool, and
our experiences using it to execute 657 test cases. The tool was
instrumental in our development of three new features for Linux
TCP=97Early Retransmit, Fast Open, and Loss Probes=97and allowed us to
find and fix 10 bugs in Linux. Our team uses packetdrill in all phases
of the development process.

Currently the source for the testing tool is in the git repository,
along with an example script for each supported OS. We will also be
posting tests from our team's Linux TCP test suite (described in the
paper), as time permits.

There is a mailing list for questions, discussions and patches:

  http://groups.google.com/group/packetdrill

Enjoy!

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

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<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Neal, this looks very interesting but how is it different from other sim=
ilar tools (scrapy<a></a> etc...).</p>
<p>&nbsp;</p>
<p>And THANKS for sharing we all need good tools ;)</p>
<p>&nbsp;</p>
<div>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 13px; FONT-FAMILY: Tahoma">
<div><font size=3D"2">(coffee !=3D sleep) &amp; (!coffee =3D=3D sleep)<br>
&nbsp;<a href=3D"mailto:Donald.Smith@centurylink.com">Donald.Smith@centuryl=
ink.com</a><a></a></font></div>
</div>
</div>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<div>
<hr tabindex=3D"-1">
<div id=3D"divRpF530374" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> tcpm-bounces@ietf.org [tcpm-bounces@=
ietf.org] on behalf of Neal Cardwell [ncardwell@google.com]<br>
<b>Sent:</b> Tuesday, July 02, 2013 2:44 PM<br>
<b>To:</b> tcpm@ietf.org<br>
<b>Subject:</b> [tcpm] packetdrill: a scriptable, portable network stack te=
sting tool<br>
</font><br>
</div>
<div></div>
</div>
<font size=3D"2"><span style=3D"FONT-SIZE: 10pt">
<div class=3D"PlainText">I'd like to announce the availability of the packe=
tdrill network stack<br>
testing tool.<br>
<br>
The packetdrill scripting tool enables users to quickly write precise<br>
tests for entire TCP/UDP/IPv4/IPv6 network stacks, from the system<br>
call layer down to the NIC hardware. packetdrill currently works on<br>
Linux, FreeBSD, OpenBSD, and NetBSD. It can test network stack<br>
behavior over physical NICs on a LAN, or on a single machine using a<br>
tun virtual network device.<br>
<br>
The code is licensed under version 2 of the GPL, and available in a<br>
git repository at:<br>
<br>
&nbsp; <a href=3D"https://code.google.com/p/packetdrill/" target=3D"_blank"=
>https://code.google.com/p/packetdrill/</a><br>
<br>
Here's a USENIX 2013 paper about the tool:<br>
<br>
&nbsp; <a href=3D"http://research.google.com/pubs/pub41316.html" target=3D"=
_blank">http://research.google.com/pubs/pub41316.html</a><br>
<br>
This paper describes the design and implementation of the tool, and<br>
our experiences using it to execute 657 test cases. The tool was<br>
instrumental in our development of three new features for Linux<br>
TCP=97Early Retransmit, Fast Open, and Loss Probes=97and allowed us to<br>
find and fix 10 bugs in Linux. Our team uses packetdrill in all phases<br>
of the development process.<br>
<br>
Currently the source for the testing tool is in the git repository,<br>
along with an example script for each supported OS. We will also be<br>
posting tests from our team's Linux TCP test suite (described in the<br>
paper), as time permits.<br>
<br>
There is a mailing list for questions, discussions and patches:<br>
<br>
&nbsp; <a href=3D"http://groups.google.com/group/packetdrill" target=3D"_bl=
ank">http://groups.google.com/group/packetdrill</a><br>
<br>
Enjoy!<br>
<br>
neal<br>
_______________________________________________<br>
tcpm mailing list<br>
tcpm@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div>
</span></font></div>
</div>
</body>
</html>

--_000_68EFACB32CF4464298EA2779B058889D0A3833F1PDDCWMBXEX503ct_--

From ncardwell@google.com  Wed Jul  3 10:57:36 2013
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Message-ID: <CADVnQy=YReb37zVZA6Mhecg-hxoBJkcPKJSsO0_frJpK4T2LuA@mail.gmail.com>
From: Neal Cardwell <ncardwell@google.com>
To: "Smith, Donald" <Donald.Smith@centurylink.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] packetdrill: a scriptable, portable network stack testing tool
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Hi,

I'm not familiar with scrapy, so I won't try to summarize the
differences, though I'd be interested if anyone wants to do that. We
discuss related work that we're aware of in the paper. We are
certainly not claiming that this is the "best" or only tool for
network stack testing; we find this to be one useful tool in our
toolbox, and are sharing it in the hope that others might find it
useful as well.

cheers,
neal


On Wed, Jul 3, 2013 at 1:31 PM, Smith, Donald
<Donald.Smith@centurylink.com> wrote:
> Neal, this looks very interesting but how is it different from other simi=
lar
> tools (scrapy etc...).
>
>
>
> And THANKS for sharing we all need good tools ;)
>
>
>
>
>
> (coffee !=3D sleep) & (!coffee =3D=3D sleep)
>  Donald.Smith@centurylink.com
> ________________________________
> From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of Neal
> Cardwell [ncardwell@google.com]
> Sent: Tuesday, July 02, 2013 2:44 PM
> To: tcpm@ietf.org
> Subject: [tcpm] packetdrill: a scriptable, portable network stack testing
> tool
>
> I'd like to announce the availability of the packetdrill network stack
> testing tool.
>
> The packetdrill scripting tool enables users to quickly write precise
> tests for entire TCP/UDP/IPv4/IPv6 network stacks, from the system
> call layer down to the NIC hardware. packetdrill currently works on
> Linux, FreeBSD, OpenBSD, and NetBSD. It can test network stack
> behavior over physical NICs on a LAN, or on a single machine using a
> tun virtual network device.
>
> The code is licensed under version 2 of the GPL, and available in a
> git repository at:
>
>   https://code.google.com/p/packetdrill/
>
> Here's a USENIX 2013 paper about the tool:
>
>   http://research.google.com/pubs/pub41316.html
>
> This paper describes the design and implementation of the tool, and
> our experiences using it to execute 657 test cases. The tool was
> instrumental in our development of three new features for Linux
> TCP=97Early Retransmit, Fast Open, and Loss Probes=97and allowed us to
> find and fix 10 bugs in Linux. Our team uses packetdrill in all phases
> of the development process.
>
> Currently the source for the testing tool is in the git repository,
> along with an example script for each supported OS. We will also be
> posting tests from our team's Linux TCP test suite (described in the
> paper), as time permits.
>
> There is a mailing list for questions, discussions and patches:
>
>   http://groups.google.com/group/packetdrill
>
> Enjoy!
>
> neal
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From trammell@tik.ee.ethz.ch  Thu Jul  4 05:14:30 2013
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From: Brian Trammell <trammell@tik.ee.ethz.ch>
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Date: Thu, 4 Jul 2013 14:14:17 +0200
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To: "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: [tcpm] Fwd: New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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Greetings, all,

We've posted a draft on ECN path probing and fallback, which we'd like =
to discuss at the TCPM meeting in Berlin. In a recent work [1], we found =
that though the ECN-readiness of popular webservers has significantly =
increased even in the last couple of years, there still exist paths on =
which attempts to use ECN after successful ECN negotiation will cause =
connection disruption.

This draft proposes an experimental approach to determine at runtime =
whether a path is usable, by sending segments after connection startup =
and ECN negotiation with the CE codepoint set, and disabling ECN if all =
probe segments were lost, on a per-flow or per-destination basis.

Experiments with an implementation of the approach within the Linux =
kernel are underway; we plan to be able to present initial results in =
Berlin.

Best regards,

Mirja and Brian

[1] Kuehlewind, M., Neuner, S., and Trammell, B., =93On the state of ECN =
and TCP Options on the Internet=94, in Proceedings of the the 2013 =
Passive and Active Measurement Conference, Hong Kong SAR, China, 19 =
March 2013.

Begin forwarded message:

> From: internet-drafts@ietf.org
> Subject: New Version Notification for =
draft-kuehlewind-tcpm-ecn-fallback-00.txt
> Date: 3 July 2013 17:40:32 GMT+02:00
> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, Brian =
Trammell <trammell@tik.ee.ethz.ch>
>=20
>=20
> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
> has been successfully submitted by Mirja Kuehlewind and posted to the
> IETF repository.
>=20
> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
> Revision:	 00
> Title:		 A Mechanism for ECN Path Probing and Fallback
> Creation date:	 2013-07-02
> Group:		 Individual Submission
> Number of pages: 5
> URL:             =
http://www.ietf.org/internet-drafts/draft-kuehlewind-tcpm-ecn-fallback-00.=
txt
> Status:          =
http://datatracker.ietf.org/doc/draft-kuehlewind-tcpm-ecn-fallback
> Htmlized:        =
http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-fallback-00
>=20
>=20
> Abstract:
>   Explicit Congestion Notification (ECN) is a TCP/IP extension that is
>   widely implemented but hardly used due to the perceived unusablilty
>   of ECN on many paths through the Internet caused by ECN-ignorant
>   routers and middleboxes.  This document specifies an ECN probing and
>   fall-back mechanism in case ECN has be successfully negotiated
>   between two connection endpoints, but might not be usable on the
>   path.
>=20
>=20
>=20
>=20
> The IETF Secretariat


From michawe@ifi.uio.no  Thu Jul  4 06:29:31 2013
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Subject: Re: [tcpm] Fwd: New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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Hi,

Without having read this draft yet, just from the description, I'd like =
to express that I applaud this approach. I have always thought that the =
TCP/IP suite is weak on "fallback" approaches - why do we have dismiss =
sooo many things once and for all, just because some middle-boxes or =
routers may play wrong with them? Why can't we simply *try* things, and =
fall back to a default behavior if something doesn't work?

ECN is one particular case where I was asking myself this question, so =
I'm happy to see this work appear - but I think that this method could =
probably be used to improve a lot of things at the transport layer.

Cheers,
Michael


On Jul 4, 2013, at 2:14 PM, Brian Trammell <trammell@tik.ee.ethz.ch> =
wrote:

> Greetings, all,
>=20
> We've posted a draft on ECN path probing and fallback, which we'd like =
to discuss at the TCPM meeting in Berlin. In a recent work [1], we found =
that though the ECN-readiness of popular webservers has significantly =
increased even in the last couple of years, there still exist paths on =
which attempts to use ECN after successful ECN negotiation will cause =
connection disruption.
>=20
> This draft proposes an experimental approach to determine at runtime =
whether a path is usable, by sending segments after connection startup =
and ECN negotiation with the CE codepoint set, and disabling ECN if all =
probe segments were lost, on a per-flow or per-destination basis.
>=20
> Experiments with an implementation of the approach within the Linux =
kernel are underway; we plan to be able to present initial results in =
Berlin.
>=20
> Best regards,
>=20
> Mirja and Brian
>=20
> [1] Kuehlewind, M., Neuner, S., and Trammell, B., =93On the state of =
ECN and TCP Options on the Internet=94, in Proceedings of the the 2013 =
Passive and Active Measurement Conference, Hong Kong SAR, China, 19 =
March 2013.
>=20
> Begin forwarded message:
>=20
>> From: internet-drafts@ietf.org
>> Subject: New Version Notification for =
draft-kuehlewind-tcpm-ecn-fallback-00.txt
>> Date: 3 July 2013 17:40:32 GMT+02:00
>> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, Brian =
Trammell <trammell@tik.ee.ethz.ch>
>>=20
>>=20
>> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
>> has been successfully submitted by Mirja Kuehlewind and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
>> Revision:	 00
>> Title:		 A Mechanism for ECN Path Probing and Fallback
>> Creation date:	 2013-07-02
>> Group:		 Individual Submission
>> Number of pages: 5
>> URL:             =
http://www.ietf.org/internet-drafts/draft-kuehlewind-tcpm-ecn-fallback-00.=
txt
>> Status:          =
http://datatracker.ietf.org/doc/draft-kuehlewind-tcpm-ecn-fallback
>> Htmlized:        =
http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-fallback-00
>>=20
>>=20
>> Abstract:
>>  Explicit Congestion Notification (ECN) is a TCP/IP extension that is
>>  widely implemented but hardly used due to the perceived unusablilty
>>  of ECN on many paths through the Internet caused by ECN-ignorant
>>  routers and middleboxes.  This document specifies an ECN probing and
>>  fall-back mechanism in case ECN has be successfully negotiated
>>  between two connection endpoints, but might not be usable on the
>>  path.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From michael.scharf@alcatel-lucent.com  Fri Jul  5 02:34:33 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: Agenda preparation for IETF 87 (Berlin)
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Cc: "tcpm-chairs@tools.ietf.org" <tcpm-chairs@tools.ietf.org>
Subject: [tcpm] FW: Agenda preparation for IETF 87 (Berlin)
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Hi all,

Heads up: We got a 2.5h slot in Berlin, and I have already received quite a=
 number of requests for presentations in TCPM. There is one week left to le=
t us know what you may want to talk about in Berlin.

Yet, it is possible that - unlike in many previous TCPM meetings - we will =
have this time an agenda category "if time permits".

As usual in the IETF, mailing list discussion prior to the meeting is very =
useful to deterime what topics absolutely require face-to-face discussion.

Thanks

Michael

=20

-----Original Message-----
From: Scharf, Michael (Michael) [mailto:michael.scharf@alcatel-lucent.com]=
=20
Sent: Tuesday, June 18, 2013 10:08 AM
To: tcpm (tcpm@ietf.org)
Cc: tcpm-chairs@tools.ietf.org
Subject: Agenda preparation for IETF 87 (Berlin)

Hi all,

If you plan to present at the planned TCPM meeting in Berlin, please let th=
e chairs know:

* Presentation title / draft
* Presenter
* Presentation time

The deadline for time slot requests is July 15.

Thanks

Michael, Pasi, Yoshifumi=

From internet-drafts@ietf.org  Thu Jul 11 05:41:20 2013
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-accecn-reqs-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : Problem Statement and Requirements for a More Accurate E=
CN Feedback
	Author(s)       : Mirja Kuehlewind
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-accecn-reqs-01.txt
	Pages           : 9
	Date            : 2013-07-11

Abstract:
   Explicit Congestion Notification (ECN) is an IP/TCP mechanism where
   network nodes can mark IP packets instead of dropping them to
   indicate congestion to the end-points.  An ECN-capable receiver will
   feedback this information to the sender.  ECN is specified for TCP in
   such a way that only one feedback signal can be transmitted per
   Round-Trip Time (RTT).  Recently, new TCP mechanisms like ConEx or
   DCTCP need more accurate ECN feedback information in the case where
   more than one marking is received in one RTT.  This documents
   specifies requirement for different ECN feedback scheme in the TCP
   header to provide more than one feedback signal per RTT.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-accecn-reqs-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-accecn-reqs-01


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


From internet-drafts@ietf.org  Thu Jul 11 05:51:48 2013
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-accecn-reqs-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : Problem Statement and Requirements for a More Accurate E=
CN Feedback
	Author(s)       : Mirja Kuehlewind
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-accecn-reqs-02.txt
	Pages           : 9
	Date            : 2013-07-11

Abstract:
   Explicit Congestion Notification (ECN) is an IP/TCP mechanism where
   network nodes can mark IP packets instead of dropping them to
   indicate congestion to the end-points.  An ECN-capable receiver will
   feedback this information to the sender.  ECN is specified for TCP in
   such a way that only one feedback signal can be transmitted per
   Round-Trip Time (RTT).  Recently, new TCP mechanisms like ConEx or
   DCTCP need more accurate ECN feedback information in the case where
   more than one marking is received in one RTT.  This documents
   specifies requirement for different ECN feedback scheme in the TCP
   header to provide more than one feedback signal per RTT.


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

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

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-accecn-reqs-02


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


From ycheng@google.com  Thu Jul 11 15:39:32 2013
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From: Yuchung Cheng <ycheng@google.com>
Date: Thu, 11 Jul 2013 15:39:04 -0700
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Cc: cheshire@apple.com
Subject: [tcpm] draft of tcp over udp
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http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes sense!

Another big advantage of tcp-over-udp is the ability to run user-mode
stack on any OS, allowing (much) faster software updates. For example,
many Linux clients are still running old kernels that don't support
Fast Open :|

Yuchung

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From: Christian Huitema <huitema@microsoft.com>
To: Yuchung Cheng <ycheng@google.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Cc: "cheshire@apple.com" <cheshire@apple.com>
Subject: Re: [tcpm] draft of tcp over udp
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> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes sense!

It certainly does, and I see the point of a transport protocol that works t=
hrough NAT. But the draft leaves a big question unaddressed. How is the con=
nection supposed to be set up? How does the remote party know that it shoul=
d use TCP over UDP?

The draft specifies using ICE, and it also specifies that the TCP over UDP =
endpoint is identified by the UDP port. But ICE operates by using multiple =
UDP ports, in order to traverse NAT. The endpoint identifier in ICE is typi=
cally a SIP endpoint. How is that supposed to work, exactly?


> Another big advantage of tcp-over-udp is the ability to run user-mode sta=
ck on any OS, allowing (much) faster software updates. For example, many > =
Linux clients are still running old kernels that don't support Fast Open :|

I am not sure that the "application bringing its own stack" is the best arg=
ument for TCP over UDP. Applications that need such functions already have =
the choice of incorporating their own transport, e.g. uTP. The real value i=
s to "reuse well tested TCP implementations." But then the draft should say=
 something about interop between TCP/UDP and TCP/IP...

-- Christian Huitema







From trammell@tik.ee.ethz.ch  Fri Jul 12 04:17:29 2013
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From: Brian Trammell <trammell@tik.ee.ethz.ch>
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Subject: Re: [tcpm] draft of tcp over udp
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Greetings, all,

As someone who's followed RFC 6951, due to similar frustration with =
SCTP's interaction with dodgy middleboxes, I'm (half-facetiously) =
beginning to wonder whether it makes sense to discuss TCP over UDP on a =
public mailing list, and publish the details thereof as Internet-Drafts, =
where any dodgy middlebox implementor might be able to read them, and =
correct its implementation to break TCP over UDP over IP as well, to =
maintain compatibility with its broken behavior on TCP over IP, thereby =
necessitating TCP over UDP over UDP over IP, and so on.

That said, I could see this escalation in the arms race being useful =
outside the NAT context as well; e.g. supporting multiple separate TCP =
stacks on the same machine without requiring virtualization (though I =
can't at the moment think of any not-terrible reason for doing this =
beyond congestion-control performance experiments). And aside from the =
_usefulness_ of it in any event, I appreciate the coolness of the hack. =
:)

A couple of other minor points to consider:

(1) Section 4 specifies a packet format reserving the high nybble value =
0x4 for "other transport protocols", so that 0x5-0xf (data offset in the =
reordered TCP header) can be used to mean "TCP". This does not seem to =
be compatible with RFC 6951, which sticks an SCTP header directly after =
the UDP header without reordering: this is the high nybble of the SCTP =
source port number, which has a 1/16 chance of being 0x4 all other =
things being equal.

Given that RFC 6951 and this draft have somewhat different assumptions =
(here there's an overlap between UDP and TCP port space, while 6951 =
assumes an "SCTP port" for UDP at a given network address), it's not =
really clear that this is a problem. But the mention of SCTP in section =
4 might be read as a proposal that this is an attempt to specify another =
SCTP over UDP encap incompatible with the first one, which seems a =
little odd.

(2) The draft seems to make the implicit assumption that if TCP port X =
is service foo, then UDP port X is free to be used as an overlapping =
port. This is not the case for e.g. IPFIX, which uses (by default) =
tcp/4739, udp/4739, and sctp/4739, and may actually be listening on all =
three of these.

This might be a special case, though; I've spent way too much time =
paying attention to IPFIX.

Cheers,

Brian

On 12 Jul 2013, at 5:53 , Christian Huitema <huitema@microsoft.com> =
wrote:

>> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes =
sense!
>=20
> It certainly does, and I see the point of a transport protocol that =
works through NAT. But the draft leaves a big question unaddressed. How =
is the connection supposed to be set up? How does the remote party know =
that it should use TCP over UDP?
>=20
> The draft specifies using ICE, and it also specifies that the TCP over =
UDP endpoint is identified by the UDP port. But ICE operates by using =
multiple UDP ports, in order to traverse NAT. The endpoint identifier in =
ICE is typically a SIP endpoint. How is that supposed to work, exactly?
>=20
>=20
>> Another big advantage of tcp-over-udp is the ability to run user-mode =
stack on any OS, allowing (much) faster software updates. For example, =
many > Linux clients are still running old kernels that don't support =
Fast Open :|
>=20
> I am not sure that the "application bringing its own stack" is the =
best argument for TCP over UDP. Applications that need such functions =
already have the choice of incorporating their own transport, e.g. uTP. =
The real value is to "reuse well tested TCP implementations." But then =
the draft should say something about interop between TCP/UDP and =
TCP/IP...
>=20
> -- Christian Huitema
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From Michael.Tuexen@lurchi.franken.de  Fri Jul 12 04:33:32 2013
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Subject: Re: [tcpm] draft of tcp over udp
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On Jul 12, 2013, at 1:17 PM, Brian Trammell <trammell@tik.ee.ethz.ch> =
wrote:

> Greetings, all,
>=20
> As someone who's followed RFC 6951, due to similar frustration with =
SCTP's interaction with dodgy middleboxes, I'm (half-facetiously) =
beginning to wonder whether it makes sense to discuss TCP over UDP on a =
public mailing list, and publish the details thereof as Internet-Drafts, =
where any dodgy middlebox implementor might be able to read them, and =
correct its implementation to break TCP over UDP over IP as well, to =
maintain compatibility with its broken behavior on TCP over IP, thereby =
necessitating TCP over UDP over UDP over IP, and so on.
The usecase for SCTP over UDP is to allow userland stack (like in the =
case for TCP over UDP) and
to provide NAT traversal for NATs not supporting SCTP (properly). =
However, the SCTP aware NAT
ID isn't still an RFC, so here is a difference to the TCP case.=20
>=20
> That said, I could see this escalation in the arms race being useful =
outside the NAT context as well; e.g. supporting multiple separate TCP =
stacks on the same machine without requiring virtualization (though I =
can't at the moment think of any not-terrible reason for doing this =
beyond congestion-control performance experiments). And aside from the =
_usefulness_ of it in any event, I appreciate the coolness of the hack. =
:)
>=20
> A couple of other minor points to consider:
>=20
> (1) Section 4 specifies a packet format reserving the high nybble =
value 0x4 for "other transport protocols", so that 0x5-0xf (data offset =
in the reordered TCP header) can be used to mean "TCP". This does not =
seem to be compatible with RFC 6951, which sticks an SCTP header =
directly after the UDP header without reordering: this is the high =
nybble of the SCTP source port number, which has a 1/16 chance of being =
0x4 all other things being equal.
This is correct. In the SCTP case we can't just reuse the UDP port =
numbers also as the SCTP port numbers.
This would only be possible for single homed SCTP associations. For =
multihomed associations,
each endpoint has multiple IP-addresses, but a single port number. So =
only using the UDP port
numbers would require all NAT boxes on different paths to sync the port =
number translation.
Something which will not happen. Therefore RFC 6951 has a complete UDP =
header with port numbers
the NAT boxes can change and a complete SCTP common header with SCTP =
port numbers the NAT box
won't touch. I hope this makes it clearer why RFC 6951 doesn't reuse the =
UDP port numbers.
>=20
> Given that RFC 6951 and this draft have somewhat different assumptions =
(here there's an overlap between UDP and TCP port space, while 6951 =
assumes an "SCTP port" for UDP at a given network address), it's not =
really clear that this is a problem. But the mention of SCTP in section =
4 might be read as a proposal that this is an attempt to specify another =
SCTP over UDP encap incompatible with the first one, which seems a =
little odd.
Correct. If you want to multiplex/demultiplex SCTP and ICE, you would =
have to restrict the source / destination
port number usages...
>=20
> (2) The draft seems to make the implicit assumption that if TCP port X =
is service foo, then UDP port X is free to be used as an overlapping =
port. This is not the case for e.g. IPFIX, which uses (by default) =
tcp/4739, udp/4739, and sctp/4739, and may actually be listening on all =
three of these.
Correct. The UDP and TCP port numbers are normally handled =
independently.
>=20
> This might be a special case, though; I've spent way too much time =
paying attention to IPFIX.
>=20
> Cheers,
Best regards
Michael
>=20
> Brian
>=20
> On 12 Jul 2013, at 5:53 , Christian Huitema <huitema@microsoft.com> =
wrote:
>=20
>>> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes =
sense!
>>=20
>> It certainly does, and I see the point of a transport protocol that =
works through NAT. But the draft leaves a big question unaddressed. How =
is the connection supposed to be set up? How does the remote party know =
that it should use TCP over UDP?
>>=20
>> The draft specifies using ICE, and it also specifies that the TCP =
over UDP endpoint is identified by the UDP port. But ICE operates by =
using multiple UDP ports, in order to traverse NAT. The endpoint =
identifier in ICE is typically a SIP endpoint. How is that supposed to =
work, exactly?
>>=20
>>=20
>>> Another big advantage of tcp-over-udp is the ability to run =
user-mode stack on any OS, allowing (much) faster software updates. For =
example, many > Linux clients are still running old kernels that don't =
support Fast Open :|
>>=20
>> I am not sure that the "application bringing its own stack" is the =
best argument for TCP over UDP. Applications that need such functions =
already have the choice of incorporating their own transport, e.g. uTP. =
The real value is to "reuse well tested TCP implementations." But then =
the draft should say something about interop between TCP/UDP and =
TCP/IP...
>>=20
>> -- Christian Huitema
>>=20
>>=20
>>=20
>>=20
>>=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


From michawe@ifi.uio.no  Fri Jul 12 06:41:45 2013
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Cc: "cheshire@apple.com" <cheshire@apple.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] draft of tcp over udp
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So,

This is pretty similar to:
http://www.remlab.net/op/draft-denis-udp-transport-00.txt
which also didn't address the necessary bits that Christian points out =
below. I wonder, what happened to this previous proposal, did the author =
not follow up or wasn't there enough interest?

It could be useful to learn from history here.

Cheers,
Michael


On Jul 12, 2013, at 5:53 AM, Christian Huitema <huitema@microsoft.com> =
wrote:

>> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes =
sense!
>=20
> It certainly does, and I see the point of a transport protocol that =
works through NAT. But the draft leaves a big question unaddressed. How =
is the connection supposed to be set up? How does the remote party know =
that it should use TCP over UDP?
>=20
> The draft specifies using ICE, and it also specifies that the TCP over =
UDP endpoint is identified by the UDP port. But ICE operates by using =
multiple UDP ports, in order to traverse NAT. The endpoint identifier in =
ICE is typically a SIP endpoint. How is that supposed to work, exactly?
>=20
>=20
>> Another big advantage of tcp-over-udp is the ability to run user-mode =
stack on any OS, allowing (much) faster software updates. For example, =
many > Linux clients are still running old kernels that don't support =
Fast Open :|
>=20
> I am not sure that the "application bringing its own stack" is the =
best argument for TCP over UDP. Applications that need such functions =
already have the choice of incorporating their own transport, e.g. uTP. =
The real value is to "reuse well tested TCP implementations." But then =
the draft should say something about interop between TCP/UDP and =
TCP/IP...
>=20
> -- Christian Huitema
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
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From Michael.Tuexen@lurchi.franken.de  Fri Jul 12 07:02:40 2013
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [tcpm] draft of tcp over udp
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On Jul 12, 2013, at 3:41 PM, Michael Welzl <michawe@ifi.uio.no> wrote:

> So,
>=20
> This is pretty similar to:
> http://www.remlab.net/op/draft-denis-udp-transport-00.txt
> which also didn't address the necessary bits that Christian points out =
below. I wonder, what happened to this previous proposal, did the author =
not follow up or wasn't there enough interest?
>=20
> It could be useful to learn from history here.

It won't work for SCTP with multihoming since the UDP ports (which =
possibly
gets changed on multiple paths without coordination between the NAT box) =
are also
interpreted as the SCTP ports. Same issue as with =
draft-cheshire-tcp-over-udp-00
and the reason why we have duplicated port numbers in RFC 6951.

Best regards
Michael
>=20
> Cheers,
> Michael
>=20
>=20
> On Jul 12, 2013, at 5:53 AM, Christian Huitema <huitema@microsoft.com> =
wrote:
>=20
>>> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes =
sense!
>>=20
>> It certainly does, and I see the point of a transport protocol that =
works through NAT. But the draft leaves a big question unaddressed. How =
is the connection supposed to be set up? How does the remote party know =
that it should use TCP over UDP?
>>=20
>> The draft specifies using ICE, and it also specifies that the TCP =
over UDP endpoint is identified by the UDP port. But ICE operates by =
using multiple UDP ports, in order to traverse NAT. The endpoint =
identifier in ICE is typically a SIP endpoint. How is that supposed to =
work, exactly?
>>=20
>>=20
>>> Another big advantage of tcp-over-udp is the ability to run =
user-mode stack on any OS, allowing (much) faster software updates. For =
example, many > Linux clients are still running old kernels that don't =
support Fast Open :|
>>=20
>> I am not sure that the "application bringing its own stack" is the =
best argument for TCP over UDP. Applications that need such functions =
already have the choice of incorporating their own transport, e.g. uTP. =
The real value is to "reuse well tested TCP implementations." But then =
the draft should say something about interop between TCP/UDP and =
TCP/IP...
>>=20
>> -- Christian Huitema
>>=20
>>=20
>>=20
>>=20
>>=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


From brandon.williams@akamai.com  Fri Jul 12 08:15:13 2013
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Subject: Re: [tcpm] draft of tcp over udp
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w.r.t. the draft's use of ICE, I'm uncertain as to whether it could 
break ICE for this proposal to explicitly reuse 0x4-0x7 in the first 
four bits for a purpose other than TURN channel identification. The 
draft should probably address this question. IIRC, 0x8-0xF are not used 
for any purpose in ICE or an associated protocol, so they could 
conceivably be used without risk of conflict.

--Brandon Williams

On 07/11/2013 11:53 PM, Christian Huitema wrote:
>> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes sense!
>
> It certainly does, and I see the point of a transport protocol that works through NAT. But the draft leaves a big question unaddressed. How is the connection supposed to be set up? How does the remote party know that it should use TCP over UDP?
>
> The draft specifies using ICE, and it also specifies that the TCP over UDP endpoint is identified by the UDP port. But ICE operates by using multiple UDP ports, in order to traverse NAT. The endpoint identifier in ICE is typically a SIP endpoint. How is that supposed to work, exactly?
>
>
>> Another big advantage of tcp-over-udp is the ability to run user-mode stack on any OS, allowing (much) faster software updates. For example, many > Linux clients are still running old kernels that don't support Fast Open :|
>
> I am not sure that the "application bringing its own stack" is the best argument for TCP over UDP. Applications that need such functions already have the choice of incorporating their own transport, e.g. uTP. The real value is to "reuse well tested TCP implementations." But then the draft should say something about interop between TCP/UDP and TCP/IP...
>
> -- Christian Huitema
>
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

From andrew.knutsen@bluecoat.com  Fri Jul 12 14:08:47 2013
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Subject: [tcpm] Setting ssthresh after idle
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    I have a question about idle-restarts.  All the congestion RFC's 
require shutting CWND after a 1RTO idle period, however (as far as I 
know) only 2861 addresses ssthresh for that case: "If RTO has elapsed,  
ssthresh is set to the maximum of 3/4 CWND and the current value of 
ssthresh" (section 3.1). This is implemented in the latest Linux stack, 
but not in many BSD-based stacks.

    The problem is that if ssthresh gets reduced due to sporadic packet 
loss, it is very slow to recover.  In TCP's which do not implement the 
rfc2861 reset, bursty traffic can end up in CA all the time, resulting 
in inappropriately bad performance.  Even with that reset, performance 
can be inhibited long after drops have stopped.

    An argument can also be made that after an idle period, ssthresh 
should be reset to its initial state, so a regular SS can be done. Or, 
perhaps the value could be adjusted depending on the length of the idle 
period, as is the result of saving ssthresh in the host-cache for a 
series of short-lived connections (sort of ;-). Another alternative is 
to set it to the full CWND value...

    So, given the silence of rfc5681 on this, I'm hoping to get a sense 
of current thinking from this list...

Andrew




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Subject: Re: [tcpm] Setting ssthresh after idle
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On Fri, Jul 12, 2013 at 2:08 PM, Andrew Knutsen
<andrew.knutsen@bluecoat.com> wrote:
>
>    I have a question about idle-restarts.  All the congestion RFC's require
> shutting CWND after a 1RTO idle period, however (as far as I know) only 2861
Not all. RFC2861 halves cwnd every RTO interval.

> addresses ssthresh for that case: "If RTO has elapsed,  ssthresh is set to
> the maximum of 3/4 CWND and the current value of ssthresh" (section 3.1).
> This is implemented in the latest Linux stack, but not in many BSD-based
> stacks.
>
>    The problem is that if ssthresh gets reduced due to sporadic packet loss,
> it is very slow to recover.  In TCP's which do not implement the rfc2861
> reset, bursty traffic can end up in CA all the time, resulting in
> inappropriately bad performance.  Even with that reset, performance can be
> inhibited long after drops have stopped.
after a lengthy idle cwnd is usually set to IW and it will likely slow
start because cwnd < ssthresh
if the loss is sporadic, ssthresh is probably not reduced too often

>
>    An argument can also be made that after an idle period, ssthresh should
> be reset to its initial state, so a regular SS can be done. Or, perhaps the
> value could be adjusted depending on the length of the idle period, as is
> the result of saving ssthresh in the host-cache for a series of short-lived
> connections (sort of ;-). Another alternative is to set it to the full CWND
> value...
>
>    So, given the silence of rfc5681 on this, I'm hoping to get a sense of
> current thinking from this list...
>
> Andrew
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From gorry@erg.abdn.ac.uk  Sat Jul 13 01:00:05 2013
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "McAlpine, Gary" <gary.mcalpine@bluecoat.com>
Subject: Re: [tcpm] Setting ssthresh after idle
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I agree that ssthresh needs to be increased when a TCP connection
validates a higher cwnd and then later reduces this. This is one of the
mechanisms proposed in:
http://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv

See section 4.4.2

 " If the sender exits the non-validated phase after this
   period, it MUST update the ssthresh:

         ssthresh = max(ssthresh, 3*cwnd/4). "

This adjustment of ssthresh ensures that the sender records that it
has safely sustained the present rate.  The change is beneficial to
rate-limited flows that encounter occasional congestion, and could
otherwise suffer an unwanted additional delay in recovering the
sending rate.

- We'd be interested to hear thoughts on this.

Gorry

> On Fri, Jul 12, 2013 at 2:08 PM, Andrew Knutsen
> <andrew.knutsen@bluecoat.com> wrote:
>>
>>    I have a question about idle-restarts.  All the congestion RFC's
>> require
>> shutting CWND after a 1RTO idle period, however (as far as I know) only
>> 2861
> Not all. RFC2861 halves cwnd every RTO interval.
>
>> addresses ssthresh for that case: "If RTO has elapsed,  ssthresh is set
>> to
>> the maximum of 3/4 CWND and the current value of ssthresh" (section
>> 3.1).
>> This is implemented in the latest Linux stack, but not in many BSD-based
>> stacks.
>>
>>    The problem is that if ssthresh gets reduced due to sporadic packet
>> loss,
>> it is very slow to recover.  In TCP's which do not implement the rfc2861
>> reset, bursty traffic can end up in CA all the time, resulting in
>> inappropriately bad performance.  Even with that reset, performance can
>> be
>> inhibited long after drops have stopped.
> after a lengthy idle cwnd is usually set to IW and it will likely slow
> start because cwnd < ssthresh
> if the loss is sporadic, ssthresh is probably not reduced too often
>
>>
>>    An argument can also be made that after an idle period, ssthresh
>> should
>> be reset to its initial state, so a regular SS can be done. Or, perhaps
>> the
>> value could be adjusted depending on the length of the idle period, as
>> is
>> the result of saving ssthresh in the host-cache for a series of
>> short-lived
>> connections (sort of ;-). Another alternative is to set it to the full
>> CWND
>> value...
>>
>>    So, given the silence of rfc5681 on this, I'm hoping to get a sense
>> of
>> current thinking from this list...
>>
>> Andrew
>>
>>
>>
>> _______________________________________________
>> 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 gorry@erg.abdn.ac.uk  Sun Jul 14 02:51:00 2013
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Subject: Re: [tcpm] Setting ssthresh after idle (ssthresh hostcache)
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I think I agree.This is one of the issues we are trying to resolve in the
new-cwv update. Are you arguing that we should also reset ssthresh to the
initial value after some (much) longer period of no loss?

Or do you think a reset to 3/4*cwnd before window reduction is sufficient?

One thing that could be useful perhaps is to specify some guidance on
implementing an ssthresh hostcache. What do people think we should say?

Gorry

>
> The nasty problem is with long lived persistent high data rate WAN
> connections. A congestion event or some other event can cause ssthresh to
> get ratcheted down to a very low level and if ssthresh only gets updated
> on packet drops, it may never recover until the connection is closed and
> re-established. If an ssthresh hostcache is in use, it may never recover
> because the re-established connection will assume the ssthresh from the
> previous connection.
>
> Recovery requires a sufficiently long transfer to ramp the flight-size up
> (at a CA ramp rate) to cause another packet loss. At that point ssthresh
> will get set to the proper level for congestion avoidance at the current
> congestion level. But on a high data persistent WAN connection, it may
> never transfer a sufficiently large file to cause another packet drop. So
> the connection gets stuck in a very low performance mode. Tests with
> setting ssthresh = max(ssthresh, 3*cwnd/4) during an idle period showed
> that it recovered until the transfers were too short to push CWND
> sufficiently above ssthresh so the 3/4 CWND was higher than ssthresh.
>
> As written, it looks to me like RFC5681 was intended for the slow-start,
> congestion avoidance, fast retransmit, and fast recovery mechanisms to be
> applied to each transfer or set of transfers that occur without a
> sufficiently long idle period (i.e. 1 RTO or more). But, after an idle
> period (which could be 1 RTO or a couple days on a long lived connection),
> slow-start is intended to be used to re-acquire "ACK clock" and determine
> the available bandwidth at the current congestion level. Although not
> explicitly specified, the below statement in section 3.1 should be
> re-applied after each sufficiently long idle period. Otherwise, only the
> first transfer(s) of a long-lived connection will adjust to the current
> network conditions. The rest will only adjust to the current network
> conditions as long as ssthresh happens to be sufficiently high.
>
> "The initial value of ssthresh SHOULD be set arbitrarily high (e.g.,
>    to the size of the largest possible advertised window), but ssthresh
>    MUST be reduced in response to congestion.  Setting ssthresh as high
>    as possible allows the network conditions, rather than some arbitrary
>    host limit, to dictate the sending rate.
> "
>
> Gary
>
> -----Original Message-----
> From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]
> Sent: Saturday, July 13, 2013 1:00 AM
> To: Yuchung Cheng
> Cc: Knutsen, Andrew; tcpm@ietf.org Extensions; McAlpine, Gary
> Subject: Re: [tcpm] Setting ssthresh after idle
>
> I agree that ssthresh needs to be increased when a TCP connection
> validates a higher cwnd and then later reduces this. This is one of the
> mechanisms proposed in:
> http://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>
> See section 4.4.2
>
>  " If the sender exits the non-validated phase after this
>    period, it MUST update the ssthresh:
>
>          ssthresh = max(ssthresh, 3*cwnd/4). "
>
> This adjustment of ssthresh ensures that the sender records that it has
> safely sustained the present rate.  The change is beneficial to
> rate-limited flows that encounter occasional congestion, and could
> otherwise suffer an unwanted additional delay in recovering the sending
> rate.
>
> - We'd be interested to hear thoughts on this.
>
> Gorry
>
>> On Fri, Jul 12, 2013 at 2:08 PM, Andrew Knutsen
>> <andrew.knutsen@bluecoat.com> wrote:
>>>
>>>    I have a question about idle-restarts.  All the congestion RFC's
>>> require shutting CWND after a 1RTO idle period, however (as far as I
>>> know) only
>>> 2861
>> Not all. RFC2861 halves cwnd every RTO interval.
>>
>>> addresses ssthresh for that case: "If RTO has elapsed,  ssthresh is
>>> set to the maximum of 3/4 CWND and the current value of ssthresh"
>>> (section 3.1).
>>> This is implemented in the latest Linux stack, but not in many
>>> BSD-based stacks.
>>>
>>>    The problem is that if ssthresh gets reduced due to sporadic
>>> packet loss, it is very slow to recover.  In TCP's which do not
>>> implement the rfc2861 reset, bursty traffic can end up in CA all the
>>> time, resulting in inappropriately bad performance.  Even with that
>>> reset, performance can be inhibited long after drops have stopped.
>> after a lengthy idle cwnd is usually set to IW and it will likely slow
>> start because cwnd < ssthresh if the loss is sporadic, ssthresh is
>> probably not reduced too often
>>
>>>
>>>    An argument can also be made that after an idle period, ssthresh
>>> should be reset to its initial state, so a regular SS can be done.
>>> Or, perhaps the value could be adjusted depending on the length of
>>> the idle period, as is the result of saving ssthresh in the
>>> host-cache for a series of short-lived connections (sort of ;-).
>>> Another alternative is to set it to the full CWND value...
>>>
>>>    So, given the silence of rfc5681 on this, I'm hoping to get a
>>> sense of current thinking from this list...
>>>
>>> Andrew
>>>
>>>
>>>
>>> _______________________________________________
>>> 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 internet-drafts@ietf.org  Mon Jul 15 09:05:54 2013
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-accecn-reqs-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : Problem Statement and Requirements for a More Accurate E=
CN Feedback
	Author(s)       : Mirja Kuehlewind
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-accecn-reqs-03.txt
	Pages           : 9
	Date            : 2013-07-15

Abstract:
   Explicit Congestion Notification (ECN) is an IP/TCP mechanism where
   network nodes can mark IP packets instead of dropping them to
   indicate congestion to the end-points.  An ECN-capable receiver will
   feedback this information to the sender.  ECN is specified for TCP in
   such a way that only one feedback signal can be transmitted per
   Round-Trip Time (RTT).  Recently, new TCP mechanisms like ConEx or
   DCTCP need more accurate ECN feedback information in the case where
   more than one marking is received in one RTT.  This documents
   specifies requirement for different ECN feedback scheme in the TCP
   header to provide more than one feedback signal per RTT.


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

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

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-accecn-reqs-03


Internet-Drafts are also available by anonymous FTP at:
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From michael.scharf@alcatel-lucent.com  Mon Jul 15 10:15:38 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: draft-scharf-tcpm-reordering-00
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Dear all,

Given the recent discussions about head-of-line blocking in TCP, the follow=
ing small ID might be of interest.

Since this stuff is hopefully well-known and obvious, I don't plan to ask f=
or a time slot in Berlin - it would probably be a waste of scarce meeting t=
ime. But I encourage everybody to discuss the impact of head-of-line blocki=
ng here on the mailing list.

Michael (of course, with chair hat off)


-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] =
On Behalf Of internet-drafts@ietf.org
Sent: Monday, July 15, 2013 7:06 PM
To: i-d-announce@ietf.org
Subject: I-D Action: draft-scharf-tcpm-reordering-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.


	Title           : Quantifying Head-of-Line Blocking in TCP and SCTP
	Author(s)       : Michael Scharf
                          Sebastian Kiesel
	Filename        : draft-scharf-tcpm-reordering-00.txt
	Pages           : 10
	Date            : 2013-07-15

Abstract:
   In order to quantify the impact of head-of-line blocking on
   application latencies, this memo provides simple analytical models
   for a "back of the envelop" estimation of the delay impact for
   reliable transport over a single TCP connection, multiple TCP
   connections, multiple SCTP streams, and unordered transport.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-scharf-tcpm-reordering-00


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_______________________________________________
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Date: Mon, 15 Jul 2013 11:31:55 -0700
Cc: tcpm@ietf.org
Subject: [tcpm] I-D Action: draft-ietf-tcpm-fastopen-04.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : TCP Fast Open
	Author(s)       : Yuchung Cheng
                          Jerry Chu
                          Arvind Jain
	Filename        : draft-ietf-tcpm-fastopen-04.txt
	Pages           : 23
	Date            : 2013-07-15

Abstract:
   TCP Fast Open (TFO) allows data to be carried in the SYN and SYN-ACK
   packets and consumed by the receiving end during the initial
   connection handshake, thus saving up to one full round trip time
   (RTT) compared to the standard TCP, which requires a three-way
   handshake (3WHS) to complete before data can be exchanged. However
   TFO deviates from the standard TCP semantics in that the data in the
   SYN could be replayed to an application in some rare circumstances.
   Applications should not use TFO unless they can tolerate this issue,
   which is detailed in the Applicability section.

Terminology

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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-fastopen-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-fastopen-04


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


From ycheng@google.com  Mon Jul 15 11:44:16 2013
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 15 Jul 2013 11:43:46 -0700
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-fastopen-04.txt
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On Mon, Jul 15, 2013 at 11:31 AM,  <internet-drafts@ietf.org> wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.
>
>         Title           : TCP Fast Open
>         Author(s)       : Yuchung Cheng
>                           Jerry Chu
>                           Arvind Jain
>         Filename        : draft-ietf-tcpm-fastopen-04.txt
>         Pages           : 23
>         Date            : 2013-07-15
>
> Abstract:
>    TCP Fast Open (TFO) allows data to be carried in the SYN and SYN-ACK
>    packets and consumed by the receiving end during the initial
>    connection handshake, thus saving up to one full round trip time
>    (RTT) compared to the standard TCP, which requires a three-way
>    handshake (3WHS) to complete before data can be exchanged. However
>    TFO deviates from the standard TCP semantics in that the data in the
>    SYN could be replayed to an application in some rare circumstances.
>    Applications should not use TFO unless they can tolerate this issue,
>    which is detailed in the Applicability section.
Hi tcpm members

We have made several major changes to address comments from last
meeting and the list:
1) a new section of open areas for experimentation
2) a new appendix of proposed socket API
3) reminders of the applicability and semantic change in the abstract
4) a new subsection of browser pre-connect in the relate work
5) elaborate further on attacks in NAT in security section
6) a ton of minor formatting and word-smithing edits

Thanks for all the comments so far.

>
> Terminology
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-fastopen
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tcpm-fastopen-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-fastopen-04
>
>
> 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

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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 15 Jul 2013 14:49:13 -0700
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Subject: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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Hi

We are proposing an outrageous idea to send parity data in TCP
packet to promote sub 1-RTT loss recovery. This requires a separate
sequence space which we aren't quite sure how to do this yet without
upsetting middle-boxes. More data can be found in our new paper
"Reducing Web Latency: the Virtue of Gentle Aggression" sigcomm 2013
http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf

Comments are very welcome. Linux patches release are under way.


---------- Forwarded message ----------
From:  <internet-drafts@ietf.org>
Date: Sun, Jul 14, 2013 at 4:59 PM
Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
To: Tobias Flach <flach@usc.edu>, Nandita Dukkipati
<nanditad@google.com>, Yuchung Cheng <ycheng@google.com>, Barath
Raghavan <barath@google.com>



A new version of I-D, draft-flach-tcpm-fec-00.txt
has been successfully submitted by Tobias Flach and posted to the
IETF repository.

Filename:        draft-flach-tcpm-fec
Revision:        00
Title:           TCP Instant Recovery: Incorporating Forward Error
Correction in TCP
Creation date:   2013-07-15
Group:           Individual Submission
Number of pages: 15
URL:             http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt
Status:          http://datatracker.ietf.org/doc/draft-flach-tcpm-fec
Htmlized:        http://tools.ietf.org/html/draft-flach-tcpm-fec-00


Abstract:
   Ordinary TCP loss recovery takes at least one round-trip time and as
   such can increase application-perceived latency, especially for short
   flows such as Web transactions.  TCP Instant Recovery (TCP-IR) is an
   experimental algorithm that allows a receiving end to recover lost
   packets without retransmissions, thus potentially saving at least one
   full round-trip time compared to standard TCP.  TCP-IR achieves this
   by judiciously injecting encoded data segments within a TCP stream.
   This document describes the TCP-IR algorithm at the sending and
   receiving ends, along with the required protocol changes.




The IETF Secretariat

From touch@isi.edu  Mon Jul 15 14:59:26 2013
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You might take a look at TCP Boston, which is a much more elegant FEC 
than just parity.

But what use is parity? If a packet is corrupted, most link layers will 
drop it anyway (esp. because you don't even know if the packet header is 
correct, thus whether you're sending it to the right connection).

Joe

On 7/15/2013 2:49 PM, Yuchung Cheng wrote:
> Hi
>
> We are proposing an outrageous idea to send parity data in TCP
> packet to promote sub 1-RTT loss recovery. This requires a separate
> sequence space which we aren't quite sure how to do this yet without
> upsetting middle-boxes. More data can be found in our new paper
> "Reducing Web Latency: the Virtue of Gentle Aggression" sigcomm 2013
> http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf
>
> Comments are very welcome. Linux patches release are under way.
>
>
> ---------- Forwarded message ----------
> From:  <internet-drafts@ietf.org>
> Date: Sun, Jul 14, 2013 at 4:59 PM
> Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
> To: Tobias Flach <flach@usc.edu>, Nandita Dukkipati
> <nanditad@google.com>, Yuchung Cheng <ycheng@google.com>, Barath
> Raghavan <barath@google.com>
>
>
>
> A new version of I-D, draft-flach-tcpm-fec-00.txt
> has been successfully submitted by Tobias Flach and posted to the
> IETF repository.
>
> Filename:        draft-flach-tcpm-fec
> Revision:        00
> Title:           TCP Instant Recovery: Incorporating Forward Error
> Correction in TCP
> Creation date:   2013-07-15
> Group:           Individual Submission
> Number of pages: 15
> URL:             http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-flach-tcpm-fec
> Htmlized:        http://tools.ietf.org/html/draft-flach-tcpm-fec-00
>
>
> Abstract:
>     Ordinary TCP loss recovery takes at least one round-trip time and as
>     such can increase application-perceived latency, especially for short
>     flows such as Web transactions.  TCP Instant Recovery (TCP-IR) is an
>     experimental algorithm that allows a receiving end to recover lost
>     packets without retransmissions, thus potentially saving at least one
>     full round-trip time compared to standard TCP.  TCP-IR achieves this
>     by judiciously injecting encoded data segments within a TCP stream.
>     This document describes the TCP-IR algorithm at the sending and
>     receiving ends, along with the required protocol changes.
>
>
>
>
> The IETF Secretariat
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From touch@isi.edu  Mon Jul 15 15:09:52 2013
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Subject: Re: [tcpm] draft of tcp over udp
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A big disadvantage is that middleboxes don't know when to tear down state.

And why would this work through a NAT better than TCP?

Sure, it makes sense to run non-TCP transports over UDP (to avoid CC 
interactions, and to enable NAT traversal for unsupported transports 
such as SCTP), but doing this for TCP doesn't make sense.

If you want to run TCP out of user space, then just open a raw IP 
socket. If you're running things that provide services on ports <1024, 
you ought to have root anyway.

Others have noted more than a few problems with the port number 
assumptions in this doc:
	- not all services that are assigned TCP ports are also
	assigned corresponding UDP ports
	- in some cases, the UDP port of a service is already actively
	used
	- in some recent cases, the matching UDP port is assigned '
	only for a discovery service, which could already be running

Other issues:
	UDP allows a zero checksum (IPv4), but TCP does not

Joe

On 7/11/2013 3:39 PM, Yuchung Cheng wrote:
> http://tools.ietf.org/html/draft-cheshire-tcp-over-udp-00 makes sense!
>
> Another big advantage of tcp-over-udp is the ability to run user-mode
> stack on any OS, allowing (much) faster software updates. For example,
> many Linux clients are still running old kernels that don't support
> Fast Open :|
>
> Yuchung
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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--f46d04427270a8ec6404e1965928
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There are indeed far more elegant and sophisticated coding schemes out
there than the XOR approach we used. TCP Boston as well as other published
approaches like "Network Coded TCP" (http://arxiv.org/abs/1212.2291) are
capable to mask losses better than XOR, but XOR has the advantage that it
is easy to implement and results in a relatively low computational
overhead. In addition, we do not limit our proposal to XOR, but rather use
it to demonstrate how TCP-IR works in general, and we can easily extend the
module to support other coding schemes in the future.

Regarding the second question: Regular payloads and parity data are not
mixed in the same packet. Parity data is sent separately which enables
immediate recovery if only one packet in the encoding range is lost. As
said, more sophisticated coding schemes might do better here, but our
experiments have shown that even simple XOR can improve latency a lot -
especially for short flows.

- Tobias


On Mon, Jul 15, 2013 at 5:58 PM, Joe Touch <touch@isi.edu> wrote:

> You might take a look at TCP Boston, which is a much more elegant FEC than
> just parity.
>
> But what use is parity? If a packet is corrupted, most link layers will
> drop it anyway (esp. because you don't even know if the packet header is
> correct, thus whether you're sending it to the right connection).
>
> Joe
>
>
> On 7/15/2013 2:49 PM, Yuchung Cheng wrote:
>
>> Hi
>>
>> We are proposing an outrageous idea to send parity data in TCP
>> packet to promote sub 1-RTT loss recovery. This requires a separate
>> sequence space which we aren't quite sure how to do this yet without
>> upsetting middle-boxes. More data can be found in our new paper
>> "Reducing Web Latency: the Virtue of Gentle Aggression" sigcomm 2013
>> http://static.**googleusercontent.com/**external_content/untrusted_**
>> dlcp/research.google.com/en/**us/pubs/archive/41217.pdf<http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf>
>>
>> Comments are very welcome. Linux patches release are under way.
>>
>>
>> ---------- Forwarded message ----------
>> From:  <internet-drafts@ietf.org>
>> Date: Sun, Jul 14, 2013 at 4:59 PM
>> Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
>> To: Tobias Flach <flach@usc.edu>, Nandita Dukkipati
>> <nanditad@google.com>, Yuchung Cheng <ycheng@google.com>, Barath
>> Raghavan <barath@google.com>
>>
>>
>>
>> A new version of I-D, draft-flach-tcpm-fec-00.txt
>> has been successfully submitted by Tobias Flach and posted to the
>> IETF repository.
>>
>> Filename:        draft-flach-tcpm-fec
>> Revision:        00
>> Title:           TCP Instant Recovery: Incorporating Forward Error
>> Correction in TCP
>> Creation date:   2013-07-15
>> Group:           Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-**
>> drafts/draft-flach-tcpm-fec-**00.txt<http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt>
>> Status:          http://datatracker.ietf.org/**doc/draft-flach-tcpm-fec<http://datatracker.ietf.org/doc/draft-flach-tcpm-fec>
>> Htmlized:        http://tools.ietf.org/html/**draft-flach-tcpm-fec-00<http://tools.ietf.org/html/draft-flach-tcpm-fec-00>
>>
>>
>> Abstract:
>>     Ordinary TCP loss recovery takes at least one round-trip time and as
>>     such can increase application-perceived latency, especially for short
>>     flows such as Web transactions.  TCP Instant Recovery (TCP-IR) is an
>>     experimental algorithm that allows a receiving end to recover lost
>>     packets without retransmissions, thus potentially saving at least one
>>     full round-trip time compared to standard TCP.  TCP-IR achieves this
>>     by judiciously injecting encoded data segments within a TCP stream.
>>     This document describes the TCP-IR algorithm at the sending and
>>     receiving ends, along with the required protocol changes.
>>
>>
>>
>>
>> The IETF Secretariat
>> ______________________________**_________________
>> tcpm mailing list
>> 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<https://www.ietf.org/mailman/listinfo/tcpm>
>

--f46d04427270a8ec6404e1965928
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>There are indeed far more elegant and sophisticated c=
oding schemes out there than the XOR approach we used. TCP Boston as well a=
s other published approaches like &quot;Network Coded TCP&quot; (<a href=3D=
"http://arxiv.org/abs/1212.2291" target=3D"_blank">http://arxiv.org/abs/121=
2.2291</a>) are capable to mask losses better than XOR, but XOR has the adv=
antage that it is easy to implement and results in a relatively low computa=
tional overhead. In addition, we do not limit our proposal to XOR, but rath=
er use it to demonstrate how TCP-IR works in general, and we can easily ext=
end the module to support other coding schemes in the future.</div>
<div><br></div><div>Regarding the second question: Regular payloads and par=
ity data are not mixed in the same packet. Parity data is sent separately w=
hich enables immediate recovery if only one packet in the encoding range is=
 lost. As said, more sophisticated coding schemes might do better here, but=
 our experiments have shown that even simple XOR can improve latency a lot =
- especially for short flows.</div>
<div><br></div><div>- Tobias</div>
<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Jul 1=
5, 2013 at 5:58 PM, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch=
@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">
You might take a look at TCP Boston, which is a much more elegant FEC than =
just parity.<br>
<br>
But what use is parity? If a packet is corrupted, most link layers will dro=
p it anyway (esp. because you don&#39;t even know if the packet header is c=
orrect, thus whether you&#39;re sending it to the right connection).<span c=
lass=3D"HOEnZb"><font color=3D"#888888"><br>

<br>
Joe</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 7/15/2013 2:49 PM, Yuchung Cheng wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi<br>
<br>
We are proposing an outrageous idea to send parity data in TCP<br>
packet to promote sub 1-RTT loss recovery. This requires a separate<br>
sequence space which we aren&#39;t quite sure how to do this yet without<br=
>
upsetting middle-boxes. More data can be found in our new paper<br>
&quot;Reducing Web Latency: the Virtue of Gentle Aggression&quot; sigcomm 2=
013<br>
<a href=3D"http://static.googleusercontent.com/external_content/untrusted_d=
lcp/research.google.com/en/us/pubs/archive/41217.pdf" target=3D"_blank">htt=
p://static.<u></u>googleusercontent.com/<u></u>external_content/untrusted_<=
u></u>dlcp/research.google.com/en/<u></u>us/pubs/archive/41217.pdf</a><br>

<br>
Comments are very welcome. Linux patches release are under way.<br>
<br>
<br>
---------- Forwarded message ----------<br>
From: =A0&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">=
internet-drafts@ietf.org</a>&gt;<br>
Date: Sun, Jul 14, 2013 at 4:59 PM<br>
Subject: New Version Notification for draft-flach-tcpm-fec-00.txt<br>
To: Tobias Flach &lt;<a href=3D"mailto:flach@usc.edu" target=3D"_blank">fla=
ch@usc.edu</a>&gt;, Nandita Dukkipati<br>
&lt;<a href=3D"mailto:nanditad@google.com" target=3D"_blank">nanditad@googl=
e.com</a>&gt;, Yuchung Cheng &lt;<a href=3D"mailto:ycheng@google.com" targe=
t=3D"_blank">ycheng@google.com</a>&gt;, Barath<br>
Raghavan &lt;<a href=3D"mailto:barath@google.com" target=3D"_blank">barath@=
google.com</a>&gt;<br>
<br>
<br>
<br>
A new version of I-D, draft-flach-tcpm-fec-00.txt<br>
has been successfully submitted by Tobias Flach and posted to the<br>
IETF repository.<br>
<br>
Filename: =A0 =A0 =A0 =A0draft-flach-tcpm-fec<br>
Revision: =A0 =A0 =A0 =A000<br>
Title: =A0 =A0 =A0 =A0 =A0 TCP Instant Recovery: Incorporating Forward Erro=
r<br>
Correction in TCP<br>
Creation date: =A0 2013-07-15<br>
Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 15<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts=
/draft-flach-tcpm-fec-00.txt" target=3D"_blank">http://www.ietf.org/interne=
t-<u></u>drafts/draft-flach-tcpm-fec-<u></u>00.txt</a><br>
Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/draft=
-flach-tcpm-fec" target=3D"_blank">http://datatracker.ietf.org/<u></u>doc/d=
raft-flach-tcpm-fec</a><br>
Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-flach-=
tcpm-fec-00" target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-flac=
h-tcpm-fec-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0 Ordinary TCP loss recovery takes at least one round-trip time and a=
s<br>
=A0 =A0 such can increase application-perceived latency, especially for sho=
rt<br>
=A0 =A0 flows such as Web transactions. =A0TCP Instant Recovery (TCP-IR) is=
 an<br>
=A0 =A0 experimental algorithm that allows a receiving end to recover lost<=
br>
=A0 =A0 packets without retransmissions, thus potentially saving at least o=
ne<br>
=A0 =A0 full round-trip time compared to standard TCP. =A0TCP-IR achieves t=
his<br>
=A0 =A0 by judiciously injecting encoded data segments within a TCP stream.=
<br>
=A0 =A0 This document describes the TCP-IR algorithm at the sending and<br>
=A0 =A0 receiving ends, along with the required protocol changes.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
______________________________<u></u>_________________<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" target=3D"_blank">ht=
tps://www.ietf.org/mailman/<u></u>listinfo/tcpm</a><br>
<br>
</blockquote>
______________________________<u></u>_________________<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" target=3D"_blank">ht=
tps://www.ietf.org/mailman/<u></u>listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div>

--f46d04427270a8ec6404e1965928--

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Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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Well, FEC-style redundant transmission is fine so long as you subtract 
those XOR packets from your congestion window. I.e., either you send 10 
data packets, or 5 data and 5 XOR.

If you're just pumping data into the net in the hope that something 
makes it, we've seen that before too - Roadrunner tried it 15 years ago. 
We all know it works, but I hope we all know why it's not a good idea.

BTW, these sorts of tricks are a LOT older than 1-2 years.

Joe

On 7/15/2013 5:49 PM, Tobias Flach wrote:
> There are indeed far more elegant and sophisticated coding schemes out
> there than the XOR approach we used. TCP Boston as well as other
> published approaches like "Network Coded TCP"
> (http://arxiv.org/abs/1212.2291) are capable to mask losses better than
> XOR, but XOR has the advantage that it is easy to implement and results
> in a relatively low computational overhead. In addition, we do not limit
> our proposal to XOR, but rather use it to demonstrate how TCP-IR works
> in general, and we can easily extend the module to support other coding
> schemes in the future.
>
> Regarding the second question: Regular payloads and parity data are not
> mixed in the same packet. Parity data is sent separately which enables
> immediate recovery if only one packet in the encoding range is lost. As
> said, more sophisticated coding schemes might do better here, but our
> experiments have shown that even simple XOR can improve latency a lot -
> especially for short flows.
>
> - Tobias
>
>
> On Mon, Jul 15, 2013 at 5:58 PM, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
>
>     You might take a look at TCP Boston, which is a much more elegant
>     FEC than just parity.
>
>     But what use is parity? If a packet is corrupted, most link layers
>     will drop it anyway (esp. because you don't even know if the packet
>     header is correct, thus whether you're sending it to the right
>     connection).
>
>     Joe
>
>
>     On 7/15/2013 2:49 PM, Yuchung Cheng wrote:
>
>         Hi
>
>         We are proposing an outrageous idea to send parity data in TCP
>         packet to promote sub 1-RTT loss recovery. This requires a separate
>         sequence space which we aren't quite sure how to do this yet without
>         upsetting middle-boxes. More data can be found in our new paper
>         "Reducing Web Latency: the Virtue of Gentle Aggression" sigcomm 2013
>         http://static.__googleusercontent.com/__external_content/untrusted___dlcp/research.google.com/en/__us/pubs/archive/41217.pdf
>         <http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf>
>
>         Comments are very welcome. Linux patches release are under way.
>
>
>         ---------- Forwarded message ----------
>         From:  <internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>>
>         Date: Sun, Jul 14, 2013 at 4:59 PM
>         Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
>         To: Tobias Flach <flach@usc.edu <mailto:flach@usc.edu>>, Nandita
>         Dukkipati
>         <nanditad@google.com <mailto:nanditad@google.com>>, Yuchung
>         Cheng <ycheng@google.com <mailto:ycheng@google.com>>, Barath
>         Raghavan <barath@google.com <mailto:barath@google.com>>
>
>
>
>         A new version of I-D, draft-flach-tcpm-fec-00.txt
>         has been successfully submitted by Tobias Flach and posted to the
>         IETF repository.
>
>         Filename:        draft-flach-tcpm-fec
>         Revision:        00
>         Title:           TCP Instant Recovery: Incorporating Forward Error
>         Correction in TCP
>         Creation date:   2013-07-15
>         Group:           Individual Submission
>         Number of pages: 15
>         URL:
>         http://www.ietf.org/internet-__drafts/draft-flach-tcpm-fec-__00.txt
>         <http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt>
>         Status: http://datatracker.ietf.org/__doc/draft-flach-tcpm-fec
>         <http://datatracker.ietf.org/doc/draft-flach-tcpm-fec>
>         Htmlized: http://tools.ietf.org/html/__draft-flach-tcpm-fec-00
>         <http://tools.ietf.org/html/draft-flach-tcpm-fec-00>
>
>
>         Abstract:
>              Ordinary TCP loss recovery takes at least one round-trip
>         time and as
>              such can increase application-perceived latency, especially
>         for short
>              flows such as Web transactions.  TCP Instant Recovery
>         (TCP-IR) is an
>              experimental algorithm that allows a receiving end to
>         recover lost
>              packets without retransmissions, thus potentially saving at
>         least one
>              full round-trip time compared to standard TCP.  TCP-IR
>         achieves this
>              by judiciously injecting encoded data segments within a TCP
>         stream.
>              This document describes the TCP-IR algorithm at the sending and
>              receiving ends, along with the required protocol changes.
>
>
>
>
>         The IETF Secretariat
>         _________________________________________________
>         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 ycheng@google.com  Mon Jul 15 18:34:53 2013
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 15 Jul 2013 18:34:31 -0700
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>
Subject: Re: [tcpm] draft-scharf-tcpm-reordering-00
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On Mon, Jul 15, 2013 at 10:15 AM, Scharf, Michael (Michael)
<michael.scharf@alcatel-lucent.com> wrote:
> Dear all,
>
> Given the recent discussions about head-of-line blocking in TCP, the following small ID might be of interest.
>
> Since this stuff is hopefully well-known and obvious, I don't plan to ask for a time slot in Berlin - it would probably be a waste of scarce meeting time. But I encourage everybody to discuss the impact of head-of-line blocking here on the mailing list.
>
The issue may be well known but I am not aware of any empirical data.
There are many protocols that can address this problem:
minion, structured streams transport, SCTP, and including new QUIC
from Google. It'd be great if we can get data to justify this issue.
The metric may change for non-reliable real-time delivery such as
video conf. Deep instrumentation at the receiver queue in the stack is
a good starting point.

But overall I am glad Michael is taking the first stab.

> Michael (of course, with chair hat off)
>
>
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> Sent: Monday, July 15, 2013 7:06 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-scharf-tcpm-reordering-00.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>
>
>         Title           : Quantifying Head-of-Line Blocking in TCP and SCTP
>         Author(s)       : Michael Scharf
>                           Sebastian Kiesel
>         Filename        : draft-scharf-tcpm-reordering-00.txt
>         Pages           : 10
>         Date            : 2013-07-15
>
> Abstract:
>    In order to quantify the impact of head-of-line blocking on
>    application latencies, this memo provides simple analytical models
>    for a "back of the envelop" estimation of the delay impact for
>    reliable transport over a single TCP connection, multiple TCP
>    connections, multiple SCTP streams, and unordered transport.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-scharf-tcpm-reordering
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-scharf-tcpm-reordering-00
>
>
> 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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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Yuchung Cheng <ycheng@google.com>
Thread-Topic: [tcpm] draft-scharf-tcpm-reordering-00
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Subject: Re: [tcpm] draft-scharf-tcpm-reordering-00
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> -----Original Message-----
> From: Yuchung Cheng [mailto:ycheng@google.com]=20
> Sent: Tuesday, July 16, 2013 3:35 AM
> To: Scharf, Michael (Michael)
> Cc: tcpm (tcpm@ietf.org)
> Subject: Re: [tcpm] draft-scharf-tcpm-reordering-00
>=20
> On Mon, Jul 15, 2013 at 10:15 AM, Scharf, Michael (Michael)=20
> <michael.scharf@alcatel-lucent.com> wrote:
> > Dear all,
> >
> > Given the recent discussions about head-of-line blocking in=20
> TCP, the following small ID might be of interest.
> >
> > Since this stuff is hopefully well-known and obvious, I=20
> don't plan to ask for a time slot in Berlin - it would=20
> probably be a waste of scarce meeting time. But I encourage=20
> everybody to discuss the impact of head-of-line blocking here=20
> on the mailing list.
> >
> The issue may be well known but I am not aware of any empirical data.

As far as I know, transport protocol HOL was studied mainly for "realtime" =
signaling applications (e.g., SS7) in the past, typically using SCTP as tra=
nsport. There are a couple of empirical studies for such signaling traffic =
- some older ones are referenced in our papers.

> There are many protocols that can address this problem:
> minion, structured streams transport, SCTP, and including new=20
> QUIC from Google. It'd be great if we can get data to justify=20
> this issue.
> The metric may change for non-reliable real-time delivery=20
> such as video conf. Deep instrumentation at the receiver=20
> queue in the stack is a good starting point.

The application data arrival pattern of a real-time application such as vid=
eo conferencing (without adaptive streaming) may not be too different to ce=
rtain "realtime" signaling applications. This is why some of the old HOL st=
udies may partly applicable. That is one of the key points of our small ID.

Yet, I am not aware of published work on a deep instrumentation of the rece=
iver queue. Indeed, this would be very interesting.

Michael


> But overall I am glad Michael is taking the first stab.
>=20
> > Michael (of course, with chair hat off)
> >
> >
> > -----Original Message-----
> > From: i-d-announce-bounces@ietf.org=20
> > [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
> > internet-drafts@ietf.org
> > Sent: Monday, July 15, 2013 7:06 PM
> > To: i-d-announce@ietf.org
> > Subject: I-D Action: draft-scharf-tcpm-reordering-00.txt
> >
> >
> > A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
> >
> >
> >         Title           : Quantifying Head-of-Line Blocking=20
> in TCP and SCTP
> >         Author(s)       : Michael Scharf
> >                           Sebastian Kiesel
> >         Filename        : draft-scharf-tcpm-reordering-00.txt
> >         Pages           : 10
> >         Date            : 2013-07-15
> >
> > Abstract:
> >    In order to quantify the impact of head-of-line blocking on
> >    application latencies, this memo provides simple=20
> analytical models
> >    for a "back of the envelop" estimation of the delay impact for
> >    reliable transport over a single TCP connection, multiple TCP
> >    connections, multiple SCTP streams, and unordered transport.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-scharf-tcpm-reordering
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-scharf-tcpm-reordering-00
> >
> >
> > 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
> =

From michael.scharf@alcatel-lucent.com  Tue Jul 16 02:38:43 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Hi Gorry,

Regarding Section 4.4.1. Response to congestion in the nonvalidated phase:

>   At the end of the recovery phase, the TCP sender MUST reset the cwnd
>   using the method below:
>           cwnd =3D ((LossFlightSize - R)/2).
>
>   Where, R is the volume of data that was retransmitted during the
>   recovery phase.  This follows the method proposed for Jump Start
>   [Liu07].  The inclusion of the term R makes this adjustment more
>   conservative than standard TCP.  (This is required, since the sender
>   may have sent more segments than a Standard TCP sender would have
>   done.  The additional reduction is beneficial when the LossFlightSize
>   significantly overshoots the available path capacity incurring
>   significant loss, for instance an intense traffic burst following a
>   non-validated period.)

As already mentioned in http://www.ietf.org/mail-archive/web/tcpm/current/m=
sg08015.html, when implementing [Liu07] in Linux some years ago, I ran into=
 situations where R was larger than LossFlightSize (possibly due to multipl=
e retransmissions of the same segment). The equation cwnd =3D ((LossFlightS=
ize - R)/2) could thus result in a negative cwnd. That issue might partly d=
epend on how to exactly R is measured, but I think that this equation requi=
res further analysis and can't be mandated by a MUST.

Even if (LossFlightSize - R) is positive, I think that M. Allman's offlist =
comment about performance loss in loss of a short burst (Section 9.1) has t=
o be addressed.

For instance, I wonder why cwnd at the end of recovery phase could not be r=
eset to the value at the beginning, i.e.,

 cwnd =3D Min(cwnd/2,Max(pipeACK,LossFlightSize))=20

If that value is a "save" value for loss recovery, why is it not save to us=
e it after loss recovery?

Thanks

Michael

=20

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
> Behalf Of gorry@erg.abdn.ac.uk
> Sent: Tuesday, July 02, 2013 9:20 AM
> To: tcpm@ietf.org
> Subject: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
>=20
> We've just uploaded a new version of our draft. The=20
> congestion window management algorithm has not changed since=20
> -00, but in -01 and -02 drafts we introduce a more robust way=20
> to measure the pipeACK that attempts to account for the wide=20
> variety of interactive TCP application behaviour.
> This uses a sampling method to sample pipeACK over the last second.
>=20
> Clearly the cwnd poorly reflects the capacity in the=20
> non-validated phase and I think it would be wrong to use this=20
> for TCP control block sharing.=20
> We therefore also  introduce text that proposes that TCB=20
> sharing should use the pipeACK in place of cwnd when a TCP=20
> sender is in the Nonvalidated phase.  We think this value=20
> better reflects the capacity that the flow has utilised in=20
> the network path - but would welcome more thought on this.
> Thoughts on this would be most welcome.
>=20
> Gorry
>=20
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts=20
> > directories.
> >  This draft is a work item of the TCP Maintenance and Minor=20
> Extensions=20
> > Working Group of the IETF.
> >
> > 	Title           : Updating TCP to support Rate-Limited Traffic
> > 	Author(s)       : Godred Fairhurst
> >                           Arjuna Sathiaseelan
> >                           Raffaello Secchi
> > 	Filename        : draft-ietf-tcpm-newcwv-02.txt
> > 	Pages           : 19
> > 	Date            : 2013-07-01
> >
> > Abstract:
> >    This document proposes an update to RFC 5681 to address=20
> issues that
> >    arise when TCP is used to support traffic that exhibits=20
> periods where
> >    the sending rate is limited by the application rather than the
> >    congestion window.  It updates TCP to allow a TCP sender=20
> to restart
> >    quickly following either an idle or rate-limited interval.  This
> >    method is expected to benefit applications that send rate-limited
> >    traffic using TCP, while also providing an appropriate=20
> response if
> >    congestion is experienced.
> >
> >    It also evaluates the Experimental specification of TCP=20
> Congestion
> >    Window Validation, CWV, defined in RFC 2861, and=20
> concludes that RFC
> >    2861 sought to address important issues, but failed to deliver a
> >    widely used solution.  This document therefore=20
> recommends that the
> >    status of RFC 2861 is moved from Experimental to=20
> Historic, and that
> >    it is replaced by the current specification.
> >
> >    NOTE: The standards status of this WG document is under=20
> review for
> >    consideration as either Experimental (EXP) or Proposed=20
> Standard (PS).
> >    This decision will be made later as the document is finalised.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-newcwv-02
> >
> >
> > 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
> >
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> =

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From: "Scheffenegger, Richard" <rs@netapp.com>
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Brian Trammell <trammell@tik.ee.ethz.ch>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] Fwd: New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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Brian, Mirja,


Having read this draft (I think the rationale for not probing in the IW is =
a good one, but perhaps a little more text in the draft would do good - i.e=
. a short flow will be effectively non-reactive anyway, thus any ECN signal=
 would be superfluous), I like it.


I have one (probably academic) point though:


(I would like to see a diagram visualizing the probing process).

In Step 4, the algo sends the next 3 segments after IW with "CE". In additi=
on to the "first hop must not have experienced congestion" scenario, a midd=
lebox might also clear CE if these three segments are pure ACKs, as per 316=
8, pure ACKs should not be marked ECN-capable... also, you expect an ACK fo=
r these. Thus you need to send 3 "data segments" not simply segments, right=
?


In Step 6, you mention an interaction with ECN Nonce... Perhaps having an e=
xplicit example for an ECN-Nonce enabled session would do well (which can d=
etect the removal of the CE-flag in probing data segments by sending one EC=
T(1) data segment).


In Step 7, CWR should be sent with the segment (not necessarily data segmen=
t) ACKing any of the 1st up to the 3rd probing CE data segment. This could =
be 1, 2 or 3 segments (the text only hints on a single such segment).

Including the optional ECN Nonce probing (4th segment with ECT1; 4th ACK wi=
th NS=3D1) in the algorithm might be good...


Richard Scheffenegger



> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Brian Trammell
> Sent: Donnerstag, 04. Juli 2013 14:14
> To: tcpm@ietf.org
> Subject: [tcpm] Fwd: New Version Notification for draft-kuehlewind-tcpm-
> ecn-fallback-00.txt
>=20
> Greetings, all,
>=20
> We've posted a draft on ECN path probing and fallback, which we'd like to
> discuss at the TCPM meeting in Berlin. In a recent work [1], we found tha=
t
> though the ECN-readiness of popular webservers has significantly increase=
d
> even in the last couple of years, there still exist paths on which
> attempts to use ECN after successful ECN negotiation will cause connectio=
n
> disruption.
>=20
> This draft proposes an experimental approach to determine at runtime
> whether a path is usable, by sending segments after connection startup an=
d
> ECN negotiation with the CE codepoint set, and disabling ECN if all probe
> segments were lost, on a per-flow or per-destination basis.
>=20
> Experiments with an implementation of the approach within the Linux kerne=
l
> are underway; we plan to be able to present initial results in Berlin.
>=20
> Best regards,
>=20
> Mirja and Brian
>=20
> [1] Kuehlewind, M., Neuner, S., and Trammell, B., "On the state of ECN an=
d
> TCP Options on the Internet", in Proceedings of the the 2013 Passive and
> Active Measurement Conference, Hong Kong SAR, China, 19 March 2013.
>=20
> Begin forwarded message:
>=20
> > From: internet-drafts@ietf.org
> > Subject: New Version Notification for
> > draft-kuehlewind-tcpm-ecn-fallback-00.txt
> > Date: 3 July 2013 17:40:32 GMT+02:00
> > To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, Brian
> > Trammell <trammell@tik.ee.ethz.ch>
> >
> >
> > A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
> > has been successfully submitted by Mirja Kuehlewind and posted to the
> > IETF repository.
> >
> > Filename:	 draft-kuehlewind-tcpm-ecn-fallback
> > Revision:	 00
> > Title:		 A Mechanism for ECN Path Probing and Fallback
> > Creation date:	 2013-07-02
> > Group:		 Individual Submission
> > Number of pages: 5
> > URL:             http://www.ietf.org/internet-drafts/draft-kuehlewind-
> tcpm-ecn-fallback-00.txt
> > Status:          http://datatracker.ietf.org/doc/draft-kuehlewind-tcpm-
> ecn-fallback
> > Htmlized:        http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-
> fallback-00
> >
> >
> > Abstract:
> >   Explicit Congestion Notification (ECN) is a TCP/IP extension that is
> >   widely implemented but hardly used due to the perceived unusablilty
> >   of ECN on many paths through the Internet caused by ECN-ignorant
> >   routers and middleboxes.  This document specifies an ECN probing and
> >   fall-back mechanism in case ECN has be successfully negotiated
> >   between two connection endpoints, but might not be usable on the
> >   path.
> >
> >
> >
> >
> > The IETF Secretariat
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From michael.scharf@alcatel-lucent.com  Tue Jul 16 07:30:02 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Yuchung Cheng <ycheng@google.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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Hi,

After a quick scan through the draft, some initial questions...

>   1.  Periodically in every round-trip time, a TCP-IR sender places the
>   XOR of newly transmitted segments into a single MSS-length checksum
>   packet.  The XOR is only computed for new segments not previously
>   included in checksums.

Is it correct that there are more than one XOR packets per RTT if there are=
 more than 8 or 16 outstanding packets, i.e., CWND>8 or 16?

>   3.  The encoded XOR packet uses the same sequence number as the first
>   segment it encodes.  The encoded packet carries a flag in the TCP-IR
>   option signaling that the payload is encoded.  A receiver uses the
>   flag to disambiguate an encoded packet from a regular
>   (re)transmission, since both segments carry the same sequence number.
>   The option also includes the number of bytes that the payload
>   encodes.

So, on the wire there are different bytes for the same sequence number, rig=
ht?

I recall lengthy discussions on that in the MPTCP WG.

BTW: Have you thought of using MPTCP, i.e., sending the XOR packets on a se=
parate MPTCP subflow between the same IP addresses? This could be an intere=
sting application of MPTCP. MPTCP already allows today sending packets redu=
ndantly on two subflows - adding coding to MPTCP seems doable.

>   4.  For the purpose of recovering lost segments, a receiver buffers
>   the last fifteen in-order MSS blocks that it ACKed, even if the
>   application layer has already consumed these blocks.  Because an
>   encoded packet is the XOR of at most sixteen MSS segments, the
>   receiver can recover any single lost packet by computing the XOR of
>   the encoded payload and the buffered data in the encoding range.

What about TCP flow control? Are those 15 packets reduced from the advertis=
ed RWND?

>   6.  TCP-IR does not circumvent congestion control.  If the receiver
>   [...]

As already pointed out by Joe, not circumventing congestion control implies=
 to me that XOR packets are substracted from CWND. I wonder why this is not=
 explicitly mentioned in the draft. Since packet loss is mainly due to cong=
estion in most parts of the Internet, increasing the transmission rate on a=
 congested link beyond what is allowed by congestion control is a no-go.

>   TCP-IR adds a short delay in the transmission of encoded packets to
>   the reduce the probability of losing both the original transmission
>   and the encoded packet in the same loss burst.

So, the recovery delay is actually not too different to TLP?

>   The last 15 ACKed MSS length segments remain in the buffer, even if
>   the application layer has already consumed these segments.  Segments
>   received out-of-order are already buffered by default and cannot be
>   consumed by the application layer.  Since a single TCP-IR packet
>   encodes at most 8 (interleaved XOR) or 16 (basic XOR) MSS length
>   segments, any single segment loss (up to MSS length) in the encoding
>   range can be recovered by the decoder.

With this scheme, can a malicious sender attack a receiver so that it runs =
out of buffer space? For instance, what does a receiver do if a sender send=
s only one out of 16 packets?

>   Some problems caused by middlebox interference (and their solutions
>   in TCP-IR) are not discussed in the current version of this draft:
>
>   o  Rewriting of the acknowledgement number if the acknowledged
>      segment was not observed by the middlebox.  With TCP-IR this can
>      occur after recovering a lost segment.  This issue can be
>      circumvented by retransmitting the recovered segment, even though
>      it is not needed by the other endpoint anymore.  This plugs the
>      "sequence hole" in the state of the middlebox.
>   o  Rewriting payloads of previously seen segments.
>   o  Packet coalescing and splitting.

Out of my head, these were major design constraints for MPTCP. This is why =
I wonder if transfering the XOR packets over a separate MPTCP subflow could=
 maybe help.

>   We conducted two kinds of experiments.  The first was in an emulated
>   setting using loss patterns similar to those observed in our
>   measurement of real traffic.  We used the netem module to emulate a
>   200 ms RTT and both random and correlated loss rates of 2%. TCP-IR
>   reduced the latency for short transfers in lossy environments by 28%
>   in the 90th percentile.  The benefits diminish as the minimum number

Could you comment on the transmission overhead or goodput?

At first sight, the amount of sent bytes is increased by at least 1/16 =3D =
6% (or 1/8 =3D 12% with interleaving) even if a connection faces no congest=
ion at all. Plus the per-packet overhead of the TCP option and the consumpt=
ion of scarce SYN bytes.

Of course, one could reduce the overhead by selectively activating the FEC =
scheme only if a link is congested. But this still means that the goodput o=
n a congested link is reduced.

>   of RTTs necessary to complete the transaction increases (due to the
>   message size) because the time to recover from losses no longer
>   dominates the overall transmission time.  TCP-IR is better suited for
>   small transfers common in today's Web.

Really? If a sender only has, say, 4 KB to send, i.e., about 3 packets for =
MTU 1500 byte, sending an additional coded packet results in an overhead of=
 about 25%, right?

And if an app only transfers a single byte, is the overhead then MSS / 1byt=
e =3D 150000%?

With such an overhead, is TCP-IR really well-suited for small transfers?

>   The two Options for TCP-IR used during negotiation and subsequently
>   in every packet of the connection require IANA allocate one value
>   from the TCP option Kind namespace.  Experimentation prior to the
>   allocation SHOULD follow [EXPOPT] and use experimental option kind
>   254 and two magic bytes 0xDC60, and migrate to the new option after
>   the allocation accordingly.

If experiments are performed over Internet paths, a registration at http://=
www.iana.org/assignments/tcp-parameters/tcp-parameters.xhtml#tcp-exids woul=
d be highly welcome.

Sorry if I missed something in my first reading.

Thanks

Michael

=20

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
> Behalf Of Yuchung Cheng
> Sent: Monday, July 15, 2013 11:49 PM
> To: tcpm@ietf.org Extensions
> Subject: [tcpm] Fwd: New Version Notification for=20
> draft-flach-tcpm-fec-00.txt
>=20
> Hi
>=20
> We are proposing an outrageous idea to send parity data in=20
> TCP packet to promote sub 1-RTT loss recovery. This requires=20
> a separate sequence space which we aren't quite sure how to=20
> do this yet without upsetting middle-boxes. More data can be=20
> found in our new paper "Reducing Web Latency: the Virtue of=20
> Gentle Aggression" sigcomm 2013=20
> http://static.googleusercontent.com/external_content/untrusted
> _dlcp/research.google.com/en/us/pubs/archive/41217.pdf
>=20
> Comments are very welcome. Linux patches release are under way.
>=20
> ---------- Forwarded message ----------
> From:  <internet-drafts@ietf.org>
> Date: Sun, Jul 14, 2013 at 4:59 PM
> Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
> To: Tobias Flach <flach@usc.edu>, Nandita Dukkipati=20
> <nanditad@google.com>, Yuchung Cheng <ycheng@google.com>,=20
> Barath Raghavan <barath@google.com>
>=20
>=20
>=20
> A new version of I-D, draft-flach-tcpm-fec-00.txt has been=20
> successfully submitted by Tobias Flach and posted to the IETF=20
> repository.
>=20
> Filename:        draft-flach-tcpm-fec
> Revision:        00
> Title:           TCP Instant Recovery: Incorporating Forward Error
> Correction in TCP
> Creation date:   2013-07-15
> Group:           Individual Submission
> Number of pages: 15
> URL:            =20
> http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-flach-tcpm-fec
> Htmlized:        http://tools.ietf.org/html/draft-flach-tcpm-fec-00
>=20
>=20
> Abstract:
>    Ordinary TCP loss recovery takes at least one round-trip=20
> time and as
>    such can increase application-perceived latency,=20
> especially for short
>    flows such as Web transactions.  TCP Instant Recovery=20
> (TCP-IR) is an
>    experimental algorithm that allows a receiving end to recover lost
>    packets without retransmissions, thus potentially saving=20
> at least one
>    full round-trip time compared to standard TCP.  TCP-IR=20
> achieves this
>    by judiciously injecting encoded data segments within a TCP stream.
>    This document describes the TCP-IR algorithm at the sending and
>    receiving ends, along with the required protocol changes.
>=20
>=20
>=20
>=20
> The IETF Secretariat
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> =

From gary.mcalpine@bluecoat.com  Sun Jul 14 19:28:45 2013
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From: "McAlpine, Gary" <gary.mcalpine@bluecoat.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tcpm] Setting ssthresh after idle (ssthresh hostcache)
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] Setting ssthresh after idle (ssthresh hostcache)
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I've been testing a version where we set ssthresh to the receivers advertis=
ed window initially and after each idle period so that it does a full slow =
start at the beginning of each round of transfers. The results look really =
good. After each idle period, the throughput ramps up at the slow start rat=
e until it either starts being flow controlled by the receiver's advertised=
 window or it detects a packet drop. If it detects a packet drop, then it g=
oes into congestion control mode until the next idle period. Then CWND and =
ssthresh get re-initialized for the next round of transfers. This version p=
roperly adjusted each round of transfers to the current network conditions.=
 To me, it looks like it does what RFC 5681 intended.

I also tested versions where we set ssthresh to either 3/4*cwnd or to cwnd,=
 but both of those were still very slow to recover (if at all). The version=
 that used 3/4*cwnd tended to quit recovering fairly quickly unless we kept=
 ramping up the sizes of the transfers.

I am not a fan of ssthresh hostcache. If ssthresh can get into an inappropr=
iately low state, why would you want to remember it and keep perpetuating i=
t from connection to connection? I think you want to let each new connectio=
n re-initialize cwnd and ssthresh to allow the connection to adjust to the =
current network conditions.

Thanks,
Gary

-----Original Message-----
From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]=20
Sent: Sunday, July 14, 2013 2:51 AM
To: McAlpine, Gary
Cc: gorry@erg.abdn.ac.uk; Yuchung Cheng; Knutsen, Andrew; tcpm@ietf.org Ext=
ensions
Subject: RE: [tcpm] Setting ssthresh after idle (ssthresh hostcache)

I think I agree.This is one of the issues we are trying to resolve in the n=
ew-cwv update. Are you arguing that we should also reset ssthresh to the in=
itial value after some (much) longer period of no loss?

Or do you think a reset to 3/4*cwnd before window reduction is sufficient?

One thing that could be useful perhaps is to specify some guidance on imple=
menting an ssthresh hostcache. What do people think we should say?

Gorry

>
> The nasty problem is with long lived persistent high data rate WAN=20
> connections. A congestion event or some other event can cause ssthresh=20
> to get ratcheted down to a very low level and if ssthresh only gets=20
> updated on packet drops, it may never recover until the connection is=20
> closed and re-established. If an ssthresh hostcache is in use, it may=20
> never recover because the re-established connection will assume the=20
> ssthresh from the previous connection.
>
> Recovery requires a sufficiently long transfer to ramp the flight-size=20
> up (at a CA ramp rate) to cause another packet loss. At that point=20
> ssthresh will get set to the proper level for congestion avoidance at=20
> the current congestion level. But on a high data persistent WAN=20
> connection, it may never transfer a sufficiently large file to cause=20
> another packet drop. So the connection gets stuck in a very low=20
> performance mode. Tests with setting ssthresh =3D max(ssthresh,=20
> 3*cwnd/4) during an idle period showed that it recovered until the=20
> transfers were too short to push CWND sufficiently above ssthresh so the =
3/4 CWND was higher than ssthresh.
>
> As written, it looks to me like RFC5681 was intended for the=20
> slow-start, congestion avoidance, fast retransmit, and fast recovery=20
> mechanisms to be applied to each transfer or set of transfers that=20
> occur without a sufficiently long idle period (i.e. 1 RTO or more).=20
> But, after an idle period (which could be 1 RTO or a couple days on a=20
> long lived connection), slow-start is intended to be used to=20
> re-acquire "ACK clock" and determine the available bandwidth at the=20
> current congestion level. Although not explicitly specified, the below=20
> statement in section 3.1 should be re-applied after each sufficiently=20
> long idle period. Otherwise, only the first transfer(s) of a=20
> long-lived connection will adjust to the current network conditions.=20
> The rest will only adjust to the current network conditions as long as ss=
thresh happens to be sufficiently high.
>
> "The initial value of ssthresh SHOULD be set arbitrarily high (e.g.,
>    to the size of the largest possible advertised window), but ssthresh
>    MUST be reduced in response to congestion.  Setting ssthresh as high
>    as possible allows the network conditions, rather than some arbitrary
>    host limit, to dictate the sending rate.
> "
>
> Gary
>
> -----Original Message-----
> From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]
> Sent: Saturday, July 13, 2013 1:00 AM
> To: Yuchung Cheng
> Cc: Knutsen, Andrew; tcpm@ietf.org Extensions; McAlpine, Gary
> Subject: Re: [tcpm] Setting ssthresh after idle
>
> I agree that ssthresh needs to be increased when a TCP connection=20
> validates a higher cwnd and then later reduces this. This is one of=20
> the mechanisms proposed in:
> http://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>
> See section 4.4.2
>
>  " If the sender exits the non-validated phase after this
>    period, it MUST update the ssthresh:
>
>          ssthresh =3D max(ssthresh, 3*cwnd/4). "
>
> This adjustment of ssthresh ensures that the sender records that it=20
> has safely sustained the present rate.  The change is beneficial to=20
> rate-limited flows that encounter occasional congestion, and could=20
> otherwise suffer an unwanted additional delay in recovering the=20
> sending rate.
>
> - We'd be interested to hear thoughts on this.
>
> Gorry
>
>> On Fri, Jul 12, 2013 at 2:08 PM, Andrew Knutsen=20
>> <andrew.knutsen@bluecoat.com> wrote:
>>>
>>>    I have a question about idle-restarts.  All the congestion RFC's=20
>>> require shutting CWND after a 1RTO idle period, however (as far as I
>>> know) only
>>> 2861
>> Not all. RFC2861 halves cwnd every RTO interval.
>>
>>> addresses ssthresh for that case: "If RTO has elapsed,  ssthresh is=20
>>> set to the maximum of 3/4 CWND and the current value of ssthresh"
>>> (section 3.1).
>>> This is implemented in the latest Linux stack, but not in many=20
>>> BSD-based stacks.
>>>
>>>    The problem is that if ssthresh gets reduced due to sporadic=20
>>> packet loss, it is very slow to recover.  In TCP's which do not=20
>>> implement the rfc2861 reset, bursty traffic can end up in CA all the=20
>>> time, resulting in inappropriately bad performance.  Even with that=20
>>> reset, performance can be inhibited long after drops have stopped.
>> after a lengthy idle cwnd is usually set to IW and it will likely=20
>> slow start because cwnd < ssthresh if the loss is sporadic, ssthresh=20
>> is probably not reduced too often
>>
>>>
>>>    An argument can also be made that after an idle period, ssthresh=20
>>> should be reset to its initial state, so a regular SS can be done.
>>> Or, perhaps the value could be adjusted depending on the length of=20
>>> the idle period, as is the result of saving ssthresh in the=20
>>> host-cache for a series of short-lived connections (sort of ;-).
>>> Another alternative is to set it to the full CWND value...
>>>
>>>    So, given the silence of rfc5681 on this, I'm hoping to get a=20
>>> sense of current thinking from this list...
>>>
>>> Andrew
>>>
>>>
>>>
>>> _______________________________________________
>>> 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 gary.mcalpine@bluecoat.com  Mon Jul 15 08:44:15 2013
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From: "McAlpine, Gary" <gary.mcalpine@bluecoat.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tcpm] Setting ssthresh after idle (ssthresh hostcache)
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Subject: Re: [tcpm] Setting ssthresh after idle (ssthresh hostcache)
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With regard to ssthresh hostcache, the only problem I see with it is that s=
sthresh can get set to an inappropriately low value with no mechanism to re=
cover reasonably fast. If re-initializing "ssthresh =3D rwnd" during each i=
dle period is too aggressive, then something less aggressive could be used.=
 I would like to see something time based or something like a binary search=
 based on rwnd or a combination thereof. A binary search would be simple; i=
.e. during each idle, set ssthresh =3D ssthresh(current) + (rwnd - sshtresh=
(current))/2. Or, if the idle time was tracked, then make a similar adjustm=
ent for each increment (say 1 RTO) of idle time.

Thanks,
Gary

-----Original Message-----
From: McAlpine, Gary=20
Sent: Sunday, July 14, 2013 7:29 PM
To: 'gorry@erg.abdn.ac.uk'
Cc: Yuchung Cheng; Knutsen, Andrew; tcpm@ietf.org Extensions
Subject: RE: [tcpm] Setting ssthresh after idle (ssthresh hostcache)

I've been testing a version where we set ssthresh to the receivers advertis=
ed window initially and after each idle period so that it does a full slow =
start at the beginning of each round of transfers. The results look really =
good. After each idle period, the throughput ramps up at the slow start rat=
e until it either starts being flow controlled by the receiver's advertised=
 window or it detects a packet drop. If it detects a packet drop, then it g=
oes into congestion control mode until the next idle period. Then CWND and =
ssthresh get re-initialized for the next round of transfers. This version p=
roperly adjusted each round of transfers to the current network conditions.=
 To me, it looks like it does what RFC 5681 intended.

I also tested versions where we set ssthresh to either 3/4*cwnd or to cwnd,=
 but both of those were still very slow to recover (if at all). The version=
 that used 3/4*cwnd tended to quit recovering fairly quickly unless we kept=
 ramping up the sizes of the transfers.

I am not a fan of ssthresh hostcache. If ssthresh can get into an inappropr=
iately low state, why would you want to remember it and keep perpetuating i=
t from connection to connection? I think you want to let each new connectio=
n re-initialize cwnd and ssthresh to allow the connection to adjust to the =
current network conditions.

Thanks,
Gary

-----Original Message-----
From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]
Sent: Sunday, July 14, 2013 2:51 AM
To: McAlpine, Gary
Cc: gorry@erg.abdn.ac.uk; Yuchung Cheng; Knutsen, Andrew; tcpm@ietf.org Ext=
ensions
Subject: RE: [tcpm] Setting ssthresh after idle (ssthresh hostcache)

I think I agree.This is one of the issues we are trying to resolve in the n=
ew-cwv update. Are you arguing that we should also reset ssthresh to the in=
itial value after some (much) longer period of no loss?

Or do you think a reset to 3/4*cwnd before window reduction is sufficient?

One thing that could be useful perhaps is to specify some guidance on imple=
menting an ssthresh hostcache. What do people think we should say?

Gorry

>
> The nasty problem is with long lived persistent high data rate WAN=20
> connections. A congestion event or some other event can cause ssthresh=20
> to get ratcheted down to a very low level and if ssthresh only gets=20
> updated on packet drops, it may never recover until the connection is=20
> closed and re-established. If an ssthresh hostcache is in use, it may=20
> never recover because the re-established connection will assume the=20
> ssthresh from the previous connection.
>
> Recovery requires a sufficiently long transfer to ramp the flight-size=20
> up (at a CA ramp rate) to cause another packet loss. At that point=20
> ssthresh will get set to the proper level for congestion avoidance at=20
> the current congestion level. But on a high data persistent WAN=20
> connection, it may never transfer a sufficiently large file to cause=20
> another packet drop. So the connection gets stuck in a very low=20
> performance mode. Tests with setting ssthresh =3D max(ssthresh,
> 3*cwnd/4) during an idle period showed that it recovered until the=20
> transfers were too short to push CWND sufficiently above ssthresh so the =
3/4 CWND was higher than ssthresh.
>
> As written, it looks to me like RFC5681 was intended for the=20
> slow-start, congestion avoidance, fast retransmit, and fast recovery=20
> mechanisms to be applied to each transfer or set of transfers that=20
> occur without a sufficiently long idle period (i.e. 1 RTO or more).
> But, after an idle period (which could be 1 RTO or a couple days on a=20
> long lived connection), slow-start is intended to be used to=20
> re-acquire "ACK clock" and determine the available bandwidth at the=20
> current congestion level. Although not explicitly specified, the below=20
> statement in section 3.1 should be re-applied after each sufficiently=20
> long idle period. Otherwise, only the first transfer(s) of a=20
> long-lived connection will adjust to the current network conditions.
> The rest will only adjust to the current network conditions as long as ss=
thresh happens to be sufficiently high.
>
> "The initial value of ssthresh SHOULD be set arbitrarily high (e.g.,
>    to the size of the largest possible advertised window), but ssthresh
>    MUST be reduced in response to congestion.  Setting ssthresh as high
>    as possible allows the network conditions, rather than some arbitrary
>    host limit, to dictate the sending rate.
> "
>
> Gary
>
> -----Original Message-----
> From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]
> Sent: Saturday, July 13, 2013 1:00 AM
> To: Yuchung Cheng
> Cc: Knutsen, Andrew; tcpm@ietf.org Extensions; McAlpine, Gary
> Subject: Re: [tcpm] Setting ssthresh after idle
>
> I agree that ssthresh needs to be increased when a TCP connection=20
> validates a higher cwnd and then later reduces this. This is one of=20
> the mechanisms proposed in:
> http://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>
> See section 4.4.2
>
>  " If the sender exits the non-validated phase after this
>    period, it MUST update the ssthresh:
>
>          ssthresh =3D max(ssthresh, 3*cwnd/4). "
>
> This adjustment of ssthresh ensures that the sender records that it=20
> has safely sustained the present rate.  The change is beneficial to=20
> rate-limited flows that encounter occasional congestion, and could=20
> otherwise suffer an unwanted additional delay in recovering the=20
> sending rate.
>
> - We'd be interested to hear thoughts on this.
>
> Gorry
>
>> On Fri, Jul 12, 2013 at 2:08 PM, Andrew Knutsen=20
>> <andrew.knutsen@bluecoat.com> wrote:
>>>
>>>    I have a question about idle-restarts.  All the congestion RFC's=20
>>> require shutting CWND after a 1RTO idle period, however (as far as I
>>> know) only
>>> 2861
>> Not all. RFC2861 halves cwnd every RTO interval.
>>
>>> addresses ssthresh for that case: "If RTO has elapsed,  ssthresh is=20
>>> set to the maximum of 3/4 CWND and the current value of ssthresh"
>>> (section 3.1).
>>> This is implemented in the latest Linux stack, but not in many=20
>>> BSD-based stacks.
>>>
>>>    The problem is that if ssthresh gets reduced due to sporadic=20
>>> packet loss, it is very slow to recover.  In TCP's which do not=20
>>> implement the rfc2861 reset, bursty traffic can end up in CA all the=20
>>> time, resulting in inappropriately bad performance.  Even with that=20
>>> reset, performance can be inhibited long after drops have stopped.
>> after a lengthy idle cwnd is usually set to IW and it will likely=20
>> slow start because cwnd < ssthresh if the loss is sporadic, ssthresh=20
>> is probably not reduced too often
>>
>>>
>>>    An argument can also be made that after an idle period, ssthresh=20
>>> should be reset to its initial state, so a regular SS can be done.
>>> Or, perhaps the value could be adjusted depending on the length of=20
>>> the idle period, as is the result of saving ssthresh in the=20
>>> host-cache for a series of short-lived connections (sort of ;-).
>>> Another alternative is to set it to the full CWND value...
>>>
>>>    So, given the silence of rfc5681 on this, I'm hoping to get a=20
>>> sense of current thinking from this list...
>>>
>>> Andrew
>>>
>>>
>>>
>>> _______________________________________________
>>> 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 rs@netapp.com  Tue Jul 16 09:11:54 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Thread-Topic: [tcpm] some questions related to PAWS
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Date: Tue, 16 Jul 2013 16:11:37 +0000
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Hi Yoshifumi,


> Sorry. As you mentioned, PAWS is receiver logic. What I meant was=20
> disabling PAWS check.=20
>
> Something like, "I want to have TS for RTTM, but I don't want to=20
> drop packets without TS unless the sender sends data 4GB in less=20
> than 2MSL". In this case, the node needs to measure receiving rate=20
> to check if PAWS check is required.=20
>
> It might be tricky, but I still think it's one possibility. one=20
> might prefer it rather than dropping packets without TS blindly.


After re-reading this thread, I believe the update should state how the mec=
hanism SHOULD work; if a receiver decides to unilaterally accept segments, =
that are according to this document deemed not acceptable, so be it...

Further, coming up with a mechanism to selectively negotiate for certain fu=
nctions to be used with TSopt clearly goes beyond the scope of the -bis upd=
ate. You might want to keep an eye on the TS exposure / negotiation drafts =
that Mirja, Brian and I are collectively working on (and would allow extens=
ibility for such functionality).

Best regards,


Richard Scheffenegger


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] some questions related to PAWS
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Subject: Re: [tcpm] some questions related to PAWS
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For what it'S worth, I reworded that MUST and MAY to a SHOULD and SHOULD NO=
T:

   Once TSopt has been successfully negotiated (sent and received)
   during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
   non-<RST> segment for the duration of the connection, and SHOULD be
   sent in an <RST> segment (see Section 4.2 for details).  If a non-
~  <RST> segment is received without a TSopt, a TCP SHOULD drop the
~  segment and SHOULD NOT send an <ACK> for the last in-sequence
   segment.  A TCP MUST NOT abort a TCP connection because any segment
   lacks an expected TSopt.



And added the following paragraph suggested by Joe as last point in the Sec=
urity/Middlebox section, to point out that the -bis update may uncover brok=
en middleboxes, potentially leading to stale TCP sessions.


~     *  It must be noted that [RFC1323] doesn't address the case of the
~        Timestamps option being dropped or selectively omitted after
~        being negotiated, and that the update in this document may
~        cause some broken middlebox behavior to be detected
~        (potentially unresponsive TCP sessions).


I think these two updates reflect what has been discussed on list...

(These are deltas vs. http://www.ietf.org/id/draft-ietf-tcpm-1323bis-14.txt=
 ).


Best regards,


Richard Scheffenegger

> -----Original Message-----
> From: Pasi Sarolahti [mailto:pasi.sarolahti@iki.fi]
> Sent: Dienstag, 04. Juni 2013 18:28
> To: Scheffenegger, Richard
> Cc: Yoshifumi Nishida; tcpm@ietf.org
> Subject: Re: [tcpm] some questions related to PAWS
>=20
> On Jun 4, 2013, at 3:05 PM, "Scheffenegger, Richard" <rs@netapp.com>
> wrote:
>=20
> >> It might be tricky, but I still think it's one possibility.
> >> one might prefer it rather than dropping packets without TS blindly.
> >
> > Probably. The sender MUST still provide a TSopt in every segment,
> assuming that the receiver is strictly enforcing PAWS. If a receiver
> chooses to open a broader window for misuse (as apparently two dominant
> stacks currently do), that's their business. Specifying a resiliency
> mechanism and then stipulating it is completely optional to use it, is no=
t
> the best approach for an RFC IMHO. That's an implementers choice.
>=20
> Right. But we should also remember PAWS failure should be very rare case,
> where two uncommon events coincide:
> * there is a old segment with wrapped sequence number that happens to hit
> the window
> * the segment has timestamps option removed for some reason
>=20
> > I need to find some time to summarize the discussion around this topic
> > that the WG had in the last few days, and pour it into some text for
> > the 1323bis draft. (I'm always happy if someone want's to donate
> > text..:)
>=20
> I don't think there has been much disagreement about the fact that to
> ensure reliable operation of PAWS, all segments must have timestamps. I
> tend to agree with you that standardizing an algorithm that works
> unreliably is not elegant, and majority of mailing list feedback seems to
> support this, so this seems clear.
>=20
> The trickier thing is, what language to add about the current state, but =
I
> think we should point out what we know: there are implementations that
> accept segments without timestamp after its successful negotiation (in
> fact no one so far has reported implementation that would discard segment=
s
> without timestamps), and changing to follow the "MUST discard" rule may
> bring up some communication problems as a result of faulty network
> behavior (and there was some reported evidence of this actually
> happening). I'm not suggesting this as a text to be used, but hopefully i=
t
> is an acceptable summary to everyone.
>=20
> Not to be said in the document, but just wanted to say it out: even if
> segments without timestamps were accepted, PAWS could still be able to
> detect most cases of sequence wrapping, assuming that timestamps removal
> is a rare case. If user runs into communication problems that can be
> solved by turning timestamps off entirely, the natural reaction is to do
> so. After that there is no PAWS protection at all.
>=20
> - Pasi


From andre@freebsd.org  Tue Jul 16 10:42:23 2013
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Date: Tue, 16 Jul 2013 19:42:10 +0200
From: Andre Oppermann <andre@freebsd.org>
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Subject: [tcpm] Improved SYN cookies
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This is a FYI on a recent enhancement I did to FreeBSDs SYN cookie
implementation to encode all necessary information in the ISN bits
instead of relying on the mostly absent TCP timestamp.

With it pretty much all previous shortcomings of SYN cookies are solved.

  http://svnweb.freebsd.org/base?view=revision&revision=253210
  http://svnweb.freebsd.org/base/head/sys/netinet/tcp_syncache.c?r1=253210&r2=253209&pathrev=253210

Additional comments always welcome of course.  The comments in the code
provide additional insight.

Abstract:

Improve SYN cookies by encoding the MSS, WSCALE (window scaling) and SACK
information into the ISN (initial sequence number) without the additional
use of timestamp bits and switching to the very fast and cryptographically
strong SipHash-2-4 MAC hash algorithm to protect the SYN cookie against
forgeries.

The purpose of SYN cookies is to encode all necessary session state in
the 32 bits of our initial sequence number to avoid storing any information
locally in memory.  This is especially important when under heavy spoofed
SYN attacks where we would either run out of memory or the syncache would
fill with bogus connection attempts swamping out legitimate connections.

The original SYN cookies method only stored an indexed MSS values in the
cookie.  This isn't sufficient anymore and breaks down in the presence of
WSCALE information which is only exchanged during SYN and SYN-ACK.  If we
can't keep track of it then we may severely underestimate the available
send or receive window. This is compounded with large windows whose size
information on the TCP segment header is even lower numerically.  A number
of years back SYN cookies were extended to store the additional state in
the TCP timestamp fields, if available on a connection.  While timestamps
are common among the BSD, Linux and other *nix systems Windows never enabled
them by default and thus are not present for the vast majority of clients
seen on the Internet.

The common parameters used on TCP sessions have changed quite a bit since
SYN cookies very invented some 17 years ago.  Today we have a lot more
bandwidth available making the use window scaling almost mandatory.  Also
SACK has become standard making recovering from packet loss much more
efficient.

This change moves all necessary information into the ISS removing the need
for timestamps.  Both the MSS (16 bits) and send WSCALE (4 bits) are stored
in 3 bit indexed form together with a single bit for SACK.  While this is
significantly less than the original range, it is sufficient to encode all
common values with minimal rounding.

The MSS depends on the MTU of the path and with the dominance of ethernet
the main value seen is around 1460 bytes.  Encapsulations for DSL lines
and some other overheads reduce it by a few more bytes for many connections
seen.  Rounding down to the next lower value in some cases isn't a problem
as we send only slightly more packets for the same amount of data.

The send WSCALE index is bit more tricky as rounding down under-estimates
the available send space available towards the remote host, however a small
number values dominate and are carefully selected again.

The receive WSCALE isn't encoded at all but recalculated based on the local
receive socket buffer size when a valid SYN cookie returns.  A listen socket
buffer size is unlikely to change while active.

The index values for MSS and WSCALE are selected for minimal rounding errors
based on large traffic surveys.  These values have to be periodically
validated against newer traffic surveys adjusting the arrays tcp_sc_msstab[]
and tcp_sc_wstab[] if necessary.

In addition the hash MAC to protect the SYN cookies is changed from MD5
to SipHash-2-4, a much faster and cryptographically secure algorithm.

-- 
Andre

From barath@google.com  Tue Jul 16 16:08:40 2013
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From: Barath Raghavan <barath@google.com>
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Subject: Re: [tcpm] Improved SYN cookies
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This seems like a very nice enhancement of SYN cookies.  I was reading
through the patch and had a couple of questions and comments.

In tcp_syncache.c, you say: "The MAC is computed over
(faddr||laddr||fport||lport||irs||flags||secmod)".  What is secmod?
Is it used for salting / key selection?  I found it in the code but
wasn't entirely clear on what it was supposed to represent.  Also, how
long are secmod, irs, and flags?

My comment is that you might avoid using an unvetted hash algorithm
(e.g., SipHash) even if it has a good pedigree, and when using a hash
function to build a MAC, you might prefer using a known-good
construction like HMAC.

One good alternative to using a custom hash-based MAC here is to use
AES.  Since AES is standard and the data you are computing the MAC
over is small (can be squeezed into 128 bits from what I can tell), a
nice approach to creating a fast and secure MAC is to use AES as a
secure pseudorandom function (PRF); all secure PRFs are secure MACs.
That is, given the 128-bit secret key k (that's given to SipHash in
the current design) and the message m (data to compute the MAC over),
you can compute AES_k(m) and truncate the output.  You can rotate the
secret key k periodically as in the current design.

I like your analysis that's under the headings "Vector 1" and "Vector
2", and think it's sound.

Thanks!

-Barath

> This is a FYI on a recent enhancement I did to FreeBSDs SYN cookie
> implementation to encode all necessary information in the ISN bits
> instead of relying on the mostly absent TCP timestamp.
>
> With it pretty much all previous shortcomings of SYN cookies are solved.
>
>  http://svnweb.freebsd.org/base?view=revision&revision=253210
>  http://svnweb.freebsd.org/base/head/sys/netinet/tcp_syncache.c?r1=253210&r2=253209&pathrev=253210
>
> Additional comments always welcome of course.  The comments in the code
> provide additional insight.
>
> Abstract:
>
> Improve SYN cookies by encoding the MSS, WSCALE (window scaling) and SACK
> information into the ISN (initial sequence number) without the additional
> use of timestamp bits and switching to the very fast and cryptographically
> strong SipHash-2-4 MAC hash algorithm to protect the SYN cookie against
> forgeries.
>
> The purpose of SYN cookies is to encode all necessary session state in
> the 32 bits of our initial sequence number to avoid storing any information
> locally in memory.  This is especially important when under heavy spoofed
> SYN attacks where we would either run out of memory or the syncache would
> fill with bogus connection attempts swamping out legitimate connections.
>
> The original SYN cookies method only stored an indexed MSS values in the
> cookie.  This isn't sufficient anymore and breaks down in the presence of
> WSCALE information which is only exchanged during SYN and SYN-ACK.  If we
> can't keep track of it then we may severely underestimate the available
> send or receive window. This is compounded with large windows whose size
> information on the TCP segment header is even lower numerically.  A number
> of years back SYN cookies were extended to store the additional state in
> the TCP timestamp fields, if available on a connection.  While timestamps
> are common among the BSD, Linux and other *nix systems Windows never enabled
> them by default and thus are not present for the vast majority of clients
> seen on the Internet.
>
> The common parameters used on TCP sessions have changed quite a bit since
> SYN cookies very invented some 17 years ago.  Today we have a lot more
> bandwidth available making the use window scaling almost mandatory.  Also
> SACK has become standard making recovering from packet loss much more
> efficient.
>
> This change moves all necessary information into the ISS removing the need
> for timestamps.  Both the MSS (16 bits) and send WSCALE (4 bits) are stored
> in 3 bit indexed form together with a single bit for SACK.  While this is
> significantly less than the original range, it is sufficient to encode all
> common values with minimal rounding.
>
> The MSS depends on the MTU of the path and with the dominance of ethernet
> the main value seen is around 1460 bytes.  Encapsulations for DSL lines
> and some other overheads reduce it by a few more bytes for many connections
> seen.  Rounding down to the next lower value in some cases isn't a problem
> as we send only slightly more packets for the same amount of data.
>
> The send WSCALE index is bit more tricky as rounding down under-estimates
> the available send space available towards the remote host, however a small
> number values dominate and are carefully selected again.
>
> The receive WSCALE isn't encoded at all but recalculated based on the local
> receive socket buffer size when a valid SYN cookie returns.  A listen socket
> buffer size is unlikely to change while active.
>
> The index values for MSS and WSCALE are selected for minimal rounding errors
> based on large traffic surveys.  These values have to be periodically
> validated against newer traffic surveys adjusting the arrays tcp_sc_msstab[]
> and tcp_sc_wstab[] if necessary.
>
> In addition the hash MAC to protect the SYN cookies is changed from MD5
> to SipHash-2-4, a much faster and cryptographically secure algorithm.
>
> --
> Andre
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From nishida@sfc.wide.ad.jp  Tue Jul 16 22:50:36 2013
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] some questions related to PAWS
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--001a11c34772cff0fa04e1aeabc3
Content-Type: text/plain; charset=ISO-8859-1

Hi Richard,

On Tue, Jul 16, 2013 at 9:11 AM, Scheffenegger, Richard <rs@netapp.com>wrote:

> Hi Yoshifumi,
>
> > Sorry. As you mentioned, PAWS is receiver logic. What I meant was
> > disabling PAWS check.
> >
> > Something like, "I want to have TS for RTTM, but I don't want to
> > drop packets without TS unless the sender sends data 4GB in less
> > than 2MSL". In this case, the node needs to measure receiving rate
> > to check if PAWS check is required.
> >
> > It might be tricky, but I still think it's one possibility. one
> > might prefer it rather than dropping packets without TS blindly.
>
> After re-reading this thread, I believe the update should state how the
> mechanism SHOULD work; if a receiver decides to unilaterally accept
> segments, that are according to this document deemed not acceptable, so be
> it...
>

I agree. I would like to think about PAWS, but updating text for this was
not my intention.


> Further, coming up with a mechanism to selectively negotiate for certain
> functions to be used with TSopt clearly goes beyond the scope of the -bis
> update. You might want to keep an eye on the TS exposure / negotiation
> drafts that Mirja, Brian and I are collectively working on (and would allow
> extensibility for such functionality).


Yep, I will.

--
Yoshifumi

--001a11c34772cff0fa04e1aeabc3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Richard,<br><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Tue, Jul 16, 2013 at 9:11 AM, Scheffenegger, Richard <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:rs@netapp.com" target=3D"_blank">rs@n=
etapp.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Yoshifumi,<div class=3D"im">
<br>
&gt; Sorry. As you mentioned, PAWS is receiver logic. What I meant was<br>
&gt; disabling PAWS check.<br>
&gt;<br>
&gt; Something like, &quot;I want to have TS for RTTM, but I don&#39;t want=
 to<br>
&gt; drop packets without TS unless the sender sends data 4GB in less<br>
&gt; than 2MSL&quot;. In this case, the node needs to measure receiving rat=
e<br>
&gt; to check if PAWS check is required.<br>
&gt;<br>
&gt; It might be tricky, but I still think it&#39;s one possibility. one<br=
>
&gt; might prefer it rather than dropping packets without TS blindly.<br><b=
r>
</div>After re-reading this thread, I believe the update should state how t=
he mechanism SHOULD work; if a receiver decides to unilaterally accept segm=
ents, that are according to this document deemed not acceptable, so be it..=
.<br>
</blockquote><div><br></div><div>I agree. I would like to think about PAWS,=
 but updating text for this was not my intention.</div><div>=A0</div><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">

Further, coming up with a mechanism to selectively negotiate for certain fu=
nctions to be used with TSopt clearly goes beyond the scope of the -bis upd=
ate. You might want to keep an eye on the TS exposure / negotiation drafts =
that Mirja, Brian and I are collectively working on (and would allow extens=
ibility for such functionality).=A0</blockquote>
<div><br></div><div>Yep, I will.</div><div><br></div><div>--<br></div><div>=
Yoshifumi</div></div></div></div>

--001a11c34772cff0fa04e1aeabc3--

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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: "Scheffenegger, Richard" <rs@netapp.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] some questions related to PAWS
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--14dae94ee03123ec4904e1aff551
Content-Type: text/plain; charset=ISO-8859-1

Hi Richard,

Sorry. I might be too picky, but one point..
"SHOULD NOT send an <ACK> for the last in-sequence segment." sounds to me
that I might be able to send other types of segments.
I'm guessing what you meant here is SHOULD NOT send any segments.
But, this means if TS is accidentally dropped from the segment, the draft
explicitly prohibit notifying it.
I'm a bit curious if we need to be explicit here.

--
Yoshifumi

On Tue, Jul 16, 2013 at 9:23 AM, Scheffenegger, Richard <rs@netapp.com>
wrote:
>
>
>
> For what it'S worth, I reworded that MUST and MAY to a SHOULD and SHOULD
NOT:
>
>    Once TSopt has been successfully negotiated (sent and received)
>    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>    non-<RST> segment for the duration of the connection, and SHOULD be
>    sent in an <RST> segment (see Section 4.2 for details).  If a non-
> ~  <RST> segment is received without a TSopt, a TCP SHOULD drop the
> ~  segment and SHOULD NOT send an <ACK> for the last in-sequence
>    segment.  A TCP MUST NOT abort a TCP connection because any segment
>    lacks an expected TSopt.
>
>
>
> And added the following paragraph suggested by Joe as last point in the
Security/Middlebox section, to point out that the -bis update may uncover
broken middleboxes, potentially leading to stale TCP sessions.
>
>
> ~     *  It must be noted that [RFC1323] doesn't address the case of the
> ~        Timestamps option being dropped or selectively omitted after
> ~        being negotiated, and that the update in this document may
> ~        cause some broken middlebox behavior to be detected
> ~        (potentially unresponsive TCP sessions).
>
>
> I think these two updates reflect what has been discussed on list...
>
> (These are deltas vs.
http://www.ietf.org/id/draft-ietf-tcpm-1323bis-14.txt ).
>
>
> Best regards,
>
>
> Richard Scheffenegger
>
> > -----Original Message-----
> > From: Pasi Sarolahti [mailto:pasi.sarolahti@iki.fi]
> > Sent: Dienstag, 04. Juni 2013 18:28
> > To: Scheffenegger, Richard
> > Cc: Yoshifumi Nishida; tcpm@ietf.org
> > Subject: Re: [tcpm] some questions related to PAWS
> >
> > On Jun 4, 2013, at 3:05 PM, "Scheffenegger, Richard" <rs@netapp.com>
> > wrote:
> >
> > >> It might be tricky, but I still think it's one possibility.
> > >> one might prefer it rather than dropping packets without TS blindly.
> > >
> > > Probably. The sender MUST still provide a TSopt in every segment,
> > assuming that the receiver is strictly enforcing PAWS. If a receiver
> > chooses to open a broader window for misuse (as apparently two dominant
> > stacks currently do), that's their business. Specifying a resiliency
> > mechanism and then stipulating it is completely optional to use it, is
not
> > the best approach for an RFC IMHO. That's an implementers choice.
> >
> > Right. But we should also remember PAWS failure should be very rare
case,
> > where two uncommon events coincide:
> > * there is a old segment with wrapped sequence number that happens to
hit
> > the window
> > * the segment has timestamps option removed for some reason
> >
> > > I need to find some time to summarize the discussion around this topic
> > > that the WG had in the last few days, and pour it into some text for
> > > the 1323bis draft. (I'm always happy if someone want's to donate
> > > text..:)
> >
> > I don't think there has been much disagreement about the fact that to
> > ensure reliable operation of PAWS, all segments must have timestamps. I
> > tend to agree with you that standardizing an algorithm that works
> > unreliably is not elegant, and majority of mailing list feedback seems
to
> > support this, so this seems clear.
> >
> > The trickier thing is, what language to add about the current state,
but I
> > think we should point out what we know: there are implementations that
> > accept segments without timestamp after its successful negotiation (in
> > fact no one so far has reported implementation that would discard
segments
> > without timestamps), and changing to follow the "MUST discard" rule may
> > bring up some communication problems as a result of faulty network
> > behavior (and there was some reported evidence of this actually
> > happening). I'm not suggesting this as a text to be used, but hopefully
it
> > is an acceptable summary to everyone.
> >
> > Not to be said in the document, but just wanted to say it out: even if
> > segments without timestamps were accepted, PAWS could still be able to
> > detect most cases of sequence wrapping, assuming that timestamps removal
> > is a rare case. If user runs into communication problems that can be
> > solved by turning timestamps off entirely, the natural reaction is to do
> > so. After that there is no PAWS protection at all.
> >
> > - Pasi
>

--14dae94ee03123ec4904e1aff551
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Richard,<br><br>Sorry. I might be too picky, but one po=
int..<div>&quot;SHOULD NOT send an &lt;ACK&gt; for the last in-sequence=A0s=
egment.&quot; sounds to me that I might be able to send other types of segm=
ents.</div>
<div><div>I&#39;m guessing what you meant here is SHOULD NOT send any segme=
nts.</div><div>But, this means if TS is accidentally dropped from the segme=
nt, the draft explicitly prohibit notifying it.=A0</div><div>I&#39;m a bit =
curious if we need to be explicit here.</div>
<div><br></div><div><div>--</div><div>Yoshifumi<br><br>On Tue, Jul 16, 2013=
 at 9:23 AM, Scheffenegger, Richard &lt;<a href=3D"mailto:rs@netapp.com">rs=
@netapp.com</a>&gt; wrote:<br>&gt;<br>&gt;<br>&gt;<br>&gt; For what it&#39;=
S worth, I reworded that MUST and MAY to a SHOULD and SHOULD NOT:<br>
&gt;<br>&gt; =A0 =A0Once TSopt has been successfully negotiated (sent and r=
eceived)<br>&gt; =A0 =A0during the &lt;SYN&gt;, &lt;SYN,ACK&gt; exchange, T=
Sopt MUST be sent in every<br>&gt; =A0 =A0non-&lt;RST&gt; segment for the d=
uration of the connection, and SHOULD be<br>
&gt; =A0 =A0sent in an &lt;RST&gt; segment (see Section 4.2 for details). =
=A0If a non-<br>&gt; ~ =A0&lt;RST&gt; segment is received without a TSopt, =
a TCP SHOULD drop the<br>&gt; ~ =A0segment and SHOULD NOT send an &lt;ACK&g=
t; for the last in-sequence<br>
&gt; =A0 =A0segment. =A0A TCP MUST NOT abort a TCP connection because any s=
egment<br>&gt; =A0 =A0lacks an expected TSopt.<br>&gt;<br>&gt;<br>&gt;<br>&=
gt; And added the following paragraph suggested by Joe as last point in the=
 Security/Middlebox section, to point out that the -bis update may uncover =
broken middleboxes, potentially leading to stale TCP sessions.<br>
&gt;<br>&gt;<br>&gt; ~ =A0 =A0 * =A0It must be noted that [RFC1323] doesn&#=
39;t address the case of the<br>&gt; ~ =A0 =A0 =A0 =A0Timestamps option bei=
ng dropped or selectively omitted after<br>&gt; ~ =A0 =A0 =A0 =A0being nego=
tiated, and that the update in this document may<br>
&gt; ~ =A0 =A0 =A0 =A0cause some broken middlebox behavior to be detected<b=
r>&gt; ~ =A0 =A0 =A0 =A0(potentially unresponsive TCP sessions).<br>&gt;<br=
>&gt;<br>&gt; I think these two updates reflect what has been discussed on =
list...<br>&gt;<br>
&gt; (These are deltas vs. <a href=3D"http://www.ietf.org/id/draft-ietf-tcp=
m-1323bis-14.txt">http://www.ietf.org/id/draft-ietf-tcpm-1323bis-14.txt</a>=
 ).<br>&gt;<br>&gt;<br>&gt; Best regards,<br>&gt;<br>&gt;<br>&gt; Richard S=
cheffenegger<br>
&gt;<br>&gt; &gt; -----Original Message-----<br>&gt; &gt; From: Pasi Sarola=
hti [mailto:<a href=3D"mailto:pasi.sarolahti@iki.fi">pasi.sarolahti@iki.fi<=
/a>]<br>&gt; &gt; Sent: Dienstag, 04. Juni 2013 18:28<br>&gt; &gt; To: Sche=
ffenegger, Richard<br>
&gt; &gt; Cc: Yoshifumi Nishida; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf=
.org</a><br>&gt; &gt; Subject: Re: [tcpm] some questions related to PAWS<br=
>&gt; &gt;<br>&gt; &gt; On Jun 4, 2013, at 3:05 PM, &quot;Scheffenegger, Ri=
chard&quot; &lt;<a href=3D"mailto:rs@netapp.com">rs@netapp.com</a>&gt;<br>
&gt; &gt; wrote:<br>&gt; &gt;<br>&gt; &gt; &gt;&gt; It might be tricky, but=
 I still think it&#39;s one possibility.<br>&gt; &gt; &gt;&gt; one might pr=
efer it rather than dropping packets without TS blindly.<br>&gt; &gt; &gt;<=
br>
&gt; &gt; &gt; Probably. The sender MUST still provide a TSopt in every seg=
ment,<br>&gt; &gt; assuming that the receiver is strictly enforcing PAWS. I=
f a receiver<br>&gt; &gt; chooses to open a broader window for misuse (as a=
pparently two dominant<br>
&gt; &gt; stacks currently do), that&#39;s their business. Specifying a res=
iliency<br>&gt; &gt; mechanism and then stipulating it is completely option=
al to use it, is not<br>&gt; &gt; the best approach for an RFC IMHO. That&#=
39;s an implementers choice.<br>
&gt; &gt;<br>&gt; &gt; Right. But we should also remember PAWS failure shou=
ld be very rare case,<br>&gt; &gt; where two uncommon events coincide:<br>&=
gt; &gt; * there is a old segment with wrapped sequence number that happens=
 to hit<br>
&gt; &gt; the window<br>&gt; &gt; * the segment has timestamps option remov=
ed for some reason<br>&gt; &gt;<br>&gt; &gt; &gt; I need to find some time =
to summarize the discussion around this topic<br>&gt; &gt; &gt; that the WG=
 had in the last few days, and pour it into some text for<br>
&gt; &gt; &gt; the 1323bis draft. (I&#39;m always happy if someone want&#39=
;s to donate<br>&gt; &gt; &gt; text..:)<br>&gt; &gt;<br>&gt; &gt; I don&#39=
;t think there has been much disagreement about the fact that to<br>&gt; &g=
t; ensure reliable operation of PAWS, all segments must have timestamps. I<=
br>
&gt; &gt; tend to agree with you that standardizing an algorithm that works=
<br>&gt; &gt; unreliably is not elegant, and majority of mailing list feedb=
ack seems to<br>&gt; &gt; support this, so this seems clear.<br>&gt; &gt;<b=
r>
&gt; &gt; The trickier thing is, what language to add about the current sta=
te, but I<br>&gt; &gt; think we should point out what we know: there are im=
plementations that<br>&gt; &gt; accept segments without timestamp after its=
 successful negotiation (in<br>
&gt; &gt; fact no one so far has reported implementation that would discard=
 segments<br>&gt; &gt; without timestamps), and changing to follow the &quo=
t;MUST discard&quot; rule may<br>&gt; &gt; bring up some communication prob=
lems as a result of faulty network<br>
&gt; &gt; behavior (and there was some reported evidence of this actually<b=
r>&gt; &gt; happening). I&#39;m not suggesting this as a text to be used, b=
ut hopefully it<br>&gt; &gt; is an acceptable summary to everyone.<br>&gt; =
&gt;<br>
&gt; &gt; Not to be said in the document, but just wanted to say it out: ev=
en if<br>&gt; &gt; segments without timestamps were accepted, PAWS could st=
ill be able to<br>&gt; &gt; detect most cases of sequence wrapping, assumin=
g that timestamps removal<br>
&gt; &gt; is a rare case. If user runs into communication problems that can=
 be<br>&gt; &gt; solved by turning timestamps off entirely, the natural rea=
ction is to do<br>&gt; &gt; so. After that there is no PAWS protection at a=
ll.<br>
&gt; &gt;<br>&gt; &gt; - Pasi<br>&gt;<br></div></div></div></div>

--14dae94ee03123ec4904e1aff551--

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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: Draft agenda for IETF 87 in Berlin
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Hi all,

The first draft of the TCPM agenda can be found at https://datatracker.ietf=
.org/meeting/87/agenda/tcpm/ and is also copied below.

We tried to take into account all requests, but as a result, the agenda is =
pretty full. This implies:

- The slots are *total* time. Authors, please reserve 50% of your time for =
questions and discussion.

- We got time slot requests for drafts that were not be updated since the l=
ast meeting, typically promising new results for the presentation in the me=
eting. Mailing list discussion in the next week would be *highly* welcome i=
n order to justify a time slot on our full agenda.

Please let us know if you have any comments, suggestions, or if we missed s=
omething.

Happy agenda bashing!

Michael, Pasi, Yoshifumi



***************************************************************************=
**********

TCPM Agenda
IETF 87 in Berlin, Germany
Tuesday, July 30, 2013, 09:00-11:30
***********************************

WG status
---------

09:00
TCPM status
Chairs
5min


WG items
--------

09:05
draft-ietf-tcpm-1323bis-14
Richard Scheffenegger
20min

09:25
draft-ietf-tcpm-fastopen-04
Jerry Chu / Yuchung Cheng
15mins

09:40
draft-ietf-tcpm-newcwv-02
Rafaello Secchi / Gorry Fairhurst
20min=20

10:00
draft-ietf-tcpm-rtorestart-00 (no new draft)
Anna Brunstrom
10min

10:10
draft-ietf-tcpm-accecn-reqs-02
Mirja Kuelewind
5min


Non-WG items
------------

10:15
draft-zimmermann-tcpm-tcp-rfc4614bis-02
Alexander Zimmermann
10min

10:25
draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)
Yuchung Cheng
10min

10:35
draft-flach-tcpm-fec-00
Tobias Flach
15min

10:50
draft-gont-tcpm-tcp-seq-validation-00 (no new draft)
Fernando Gont
10min

11:00
draft-kuehlewind-tcpm-ecn-fallback-00
draft-kuehlewind-tcpm-accurate-ecn-02
Mirja Kuehlewind
15min


If time permits
---------------

Wired PRR effects
Mirja Kuehlewind
5min

Dealing with sequence-number randomizing firewalls
Christoph Paasch
5min

packetdrill: Scriptable Network Stack Testing, from Sockets to Packets
Yuchung Cheng
5min

From rs@netapp.com  Wed Jul 17 06:07:02 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Thread-Topic: [tcpm] some questions related to PAWS
Thread-Index: AQHOX5nlI6SiHSreO0iaNFSHTu3955ki6zhQgAKVkoCAAEEK/IBB/htQgAFzB4D//+oGkA==
Date: Wed, 17 Jul 2013 13:06:57 +0000
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Subject: Re: [tcpm] some questions related to PAWS
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Hi Yoshifumi,

Well, that unacceptable segment SHOULD NOT trigger an ACK; the sender is fr=
ee to send a segment for other reasons though (window probe, keepalive, new=
 data).


Perhaps that sentence should better read like:

If a non-<RST> segment is received without a TSopt, a TCP SHOULD silently d=
rop the segment.


Best regards,

Richard Scheffenegger


From: Yoshifumi Nishida [mailto:nishida@sfc.wide.ad.jp]=20
Sent: Mittwoch, 17. Juli 2013 09:23
To: Scheffenegger, Richard
Cc: Pasi Sarolahti; Yoshifumi Nishida; tcpm@ietf.org
Subject: Re: [tcpm] some questions related to PAWS

Hi Richard,

Sorry. I might be too picky, but one point..
"SHOULD NOT send an <ACK> for the last in-sequence=A0segment." sounds to me=
 that I might be able to send other types of segments.
I'm guessing what you meant here is SHOULD NOT send any segments.
But, this means if TS is accidentally dropped from the segment, the draft e=
xplicitly prohibit notifying it.=A0
I'm a bit curious if we need to be explicit here.

--
Yoshifumi

On Tue, Jul 16, 2013 at 9:23 AM, Scheffenegger, Richard <rs@netapp.com> wro=
te:
>
>
>
> For what it'S worth, I reworded that MUST and MAY to a SHOULD and SHOULD =
NOT:
>
> =A0 =A0Once TSopt has been successfully negotiated (sent and received)
> =A0 =A0during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
> =A0 =A0non-<RST> segment for the duration of the connection, and SHOULD b=
e
> =A0 =A0sent in an <RST> segment (see Section 4.2 for details). =A0If a no=
n-
> ~ =A0<RST> segment is received without a TSopt, a TCP SHOULD drop the
> ~ =A0segment and SHOULD NOT send an <ACK> for the last in-sequence
> =A0 =A0segment. =A0A TCP MUST NOT abort a TCP connection because any segm=
ent
> =A0 =A0lacks an expected TSopt.
>
>
>
> And added the following paragraph suggested by Joe as last point in the S=
ecurity/Middlebox section, to point out that the -bis update may uncover br=
oken middleboxes, potentially leading to stale TCP sessions.
>
>
> ~ =A0 =A0 * =A0It must be noted that [RFC1323] doesn't address the case o=
f the
> ~ =A0 =A0 =A0 =A0Timestamps option being dropped or selectively omitted a=
fter
> ~ =A0 =A0 =A0 =A0being negotiated, and that the update in this document m=
ay
> ~ =A0 =A0 =A0 =A0cause some broken middlebox behavior to be detected
> ~ =A0 =A0 =A0 =A0(potentially unresponsive TCP sessions).
>
>
> I think these two updates reflect what has been discussed on list...
>
> (These are deltas vs. http://www.ietf.org/id/draft-ietf-tcpm-1323bis-14.t=
xt ).
>
>
> Best regards,
>
>
> Richard Scheffenegger
>
> > -----Original Message-----
> > From: Pasi Sarolahti [mailto:pasi.sarolahti@iki.fi]
> > Sent: Dienstag, 04. Juni 2013 18:28
> > To: Scheffenegger, Richard
> > Cc: Yoshifumi Nishida; tcpm@ietf.org
> > Subject: Re: [tcpm] some questions related to PAWS
> >
> > On Jun 4, 2013, at 3:05 PM, "Scheffenegger, Richard" <rs@netapp.com>
> > wrote:
> >
> > >> It might be tricky, but I still think it's one possibility.
> > >> one might prefer it rather than dropping packets without TS blindly.
> > >
> > > Probably. The sender MUST still provide a TSopt in every segment,
> > assuming that the receiver is strictly enforcing PAWS. If a receiver
> > chooses to open a broader window for misuse (as apparently two dominant
> > stacks currently do), that's their business. Specifying a resiliency
> > mechanism and then stipulating it is completely optional to use it, is =
not
> > the best approach for an RFC IMHO. That's an implementers choice.
> >
> > Right. But we should also remember PAWS failure should be very rare cas=
e,
> > where two uncommon events coincide:
> > * there is a old segment with wrapped sequence number that happens to h=
it
> > the window
> > * the segment has timestamps option removed for some reason
> >
> > > I need to find some time to summarize the discussion around this topi=
c
> > > that the WG had in the last few days, and pour it into some text for
> > > the 1323bis draft. (I'm always happy if someone want's to donate
> > > text..:)
> >
> > I don't think there has been much disagreement about the fact that to
> > ensure reliable operation of PAWS, all segments must have timestamps. I
> > tend to agree with you that standardizing an algorithm that works
> > unreliably is not elegant, and majority of mailing list feedback seems =
to
> > support this, so this seems clear.
> >
> > The trickier thing is, what language to add about the current state, bu=
t I
> > think we should point out what we know: there are implementations that
> > accept segments without timestamp after its successful negotiation (in
> > fact no one so far has reported implementation that would discard segme=
nts
> > without timestamps), and changing to follow the "MUST discard" rule may
> > bring up some communication problems as a result of faulty network
> > behavior (and there was some reported evidence of this actually
> > happening). I'm not suggesting this as a text to be used, but hopefully=
 it
> > is an acceptable summary to everyone.
> >
> > Not to be said in the document, but just wanted to say it out: even if
> > segments without timestamps were accepted, PAWS could still be able to
> > detect most cases of sequence wrapping, assuming that timestamps remova=
l
> > is a rare case. If user runs into communication problems that can be
> > solved by turning timestamps off entirely, the natural reaction is to d=
o
> > so. After that there is no PAWS protection at all.
> >
> > - Pasi
>

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From: Tobias Flach <flach@usc.edu>
To: Joe Touch <touch@isi.edu>
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--047d7bdc7c906bef8f04e1b58f97
Content-Type: text/plain; charset=ISO-8859-1

I agree that if we introduce a significant amount of redundant
transmissions, these packets need to be accounted for by the congestion
window. However, the way we designed our algorithm, the amount of
redundancy is relatively small (usually 16 data packets are encoded by 1
XOR packet), which is why in our prototype the XOR packets are transmitted
outside of the window.

And you are right that these schemes in general are pretty old, but to our
knowledge no one has actually thought about the protocol level details of
incorporating FEC into TCP before, including all the trouble middleboxes
are causing.

- Tobias

On Mon, Jul 15, 2013 at 9:32 PM, Joe Touch <touch@isi.edu> wrote:

> Well, FEC-style redundant transmission is fine so long as you subtract
> those XOR packets from your congestion window. I.e., either you send 10
> data packets, or 5 data and 5 XOR.
>
> If you're just pumping data into the net in the hope that something makes
> it, we've seen that before too - Roadrunner tried it 15 years ago. We all
> know it works, but I hope we all know why it's not a good idea.
>
> BTW, these sorts of tricks are a LOT older than 1-2 years.
>
> Joe
>
>
> On 7/15/2013 5:49 PM, Tobias Flach wrote:
>
>> There are indeed far more elegant and sophisticated coding schemes out
>> there than the XOR approach we used. TCP Boston as well as other
>> published approaches like "Network Coded TCP"
>> (http://arxiv.org/abs/1212.**2291 <http://arxiv.org/abs/1212.2291>) are
>> capable to mask losses better than
>> XOR, but XOR has the advantage that it is easy to implement and results
>> in a relatively low computational overhead. In addition, we do not limit
>> our proposal to XOR, but rather use it to demonstrate how TCP-IR works
>> in general, and we can easily extend the module to support other coding
>> schemes in the future.
>>
>> Regarding the second question: Regular payloads and parity data are not
>> mixed in the same packet. Parity data is sent separately which enables
>> immediate recovery if only one packet in the encoding range is lost. As
>> said, more sophisticated coding schemes might do better here, but our
>> experiments have shown that even simple XOR can improve latency a lot -
>> especially for short flows.
>>
>> - Tobias
>>
>>
>> On Mon, Jul 15, 2013 at 5:58 PM, Joe Touch <touch@isi.edu
>> <mailto:touch@isi.edu>> wrote:
>>
>>     You might take a look at TCP Boston, which is a much more elegant
>>     FEC than just parity.
>>
>>     But what use is parity? If a packet is corrupted, most link layers
>>     will drop it anyway (esp. because you don't even know if the packet
>>     header is correct, thus whether you're sending it to the right
>>     connection).
>>
>>     Joe
>>
>>
>>     On 7/15/2013 2:49 PM, Yuchung Cheng wrote:
>>
>>         Hi
>>
>>         We are proposing an outrageous idea to send parity data in TCP
>>         packet to promote sub 1-RTT loss recovery. This requires a
>> separate
>>         sequence space which we aren't quite sure how to do this yet
>> without
>>         upsetting middle-boxes. More data can be found in our new paper
>>         "Reducing Web Latency: the Virtue of Gentle Aggression" sigcomm
>> 2013
>>         http://static.__googleusercont**ent.com/__external_content/**
>> untrusted___dlcp/research.**google.com/en/__us/pubs/**archive/41217.pdf<http://googleusercontent.com/__external_content/untrusted___dlcp/research.google.com/en/__us/pubs/archive/41217.pdf>
>>
>>         <http://static.**googleusercontent.com/**
>> external_content/untrusted_**dlcp/research.google.com/en/**
>> us/pubs/archive/41217.pdf<http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf>
>> >
>>
>>         Comments are very welcome. Linux patches release are under way.
>>
>>
>>         ---------- Forwarded message ----------
>>         From:  <internet-drafts@ietf.org <mailto:internet-drafts@ietf.**
>> org <internet-drafts@ietf.org>>>
>>         Date: Sun, Jul 14, 2013 at 4:59 PM
>>         Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
>>         To: Tobias Flach <flach@usc.edu <mailto:flach@usc.edu>>, Nandita
>>         Dukkipati
>>         <nanditad@google.com <mailto:nanditad@google.com>>, Yuchung
>>         Cheng <ycheng@google.com <mailto:ycheng@google.com>>, Barath
>>         Raghavan <barath@google.com <mailto:barath@google.com>>
>>
>>
>>
>>         A new version of I-D, draft-flach-tcpm-fec-00.txt
>>         has been successfully submitted by Tobias Flach and posted to the
>>         IETF repository.
>>
>>         Filename:        draft-flach-tcpm-fec
>>         Revision:        00
>>         Title:           TCP Instant Recovery: Incorporating Forward Error
>>         Correction in TCP
>>         Creation date:   2013-07-15
>>         Group:           Individual Submission
>>         Number of pages: 15
>>         URL:
>>         http://www.ietf.org/internet-_**_drafts/draft-flach-tcpm-fec-_**
>> _00.txt<http://www.ietf.org/internet-__drafts/draft-flach-tcpm-fec-__00.txt>
>>         <http://www.ietf.org/internet-**drafts/draft-flach-tcpm-fec-**
>> 00.txt <http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt>>
>>         Status: http://datatracker.ietf.org/__**doc/draft-flach-tcpm-fec<http://datatracker.ietf.org/__doc/draft-flach-tcpm-fec>
>>         <http://datatracker.ietf.org/**doc/draft-flach-tcpm-fec<http://datatracker.ietf.org/doc/draft-flach-tcpm-fec>
>> >
>>         Htmlized: http://tools.ietf.org/html/__**draft-flach-tcpm-fec-00<http://tools.ietf.org/html/__draft-flach-tcpm-fec-00>
>>
>>         <http://tools.ietf.org/html/**draft-flach-tcpm-fec-00<http://tools.ietf.org/html/draft-flach-tcpm-fec-00>
>> >
>>
>>
>>         Abstract:
>>              Ordinary TCP loss recovery takes at least one round-trip
>>         time and as
>>              such can increase application-perceived latency, especially
>>         for short
>>              flows such as Web transactions.  TCP Instant Recovery
>>         (TCP-IR) is an
>>              experimental algorithm that allows a receiving end to
>>         recover lost
>>              packets without retransmissions, thus potentially saving at
>>         least one
>>              full round-trip time compared to standard TCP.  TCP-IR
>>         achieves this
>>              by judiciously injecting encoded data segments within a TCP
>>         stream.
>>              This document describes the TCP-IR algorithm at the sending
>> and
>>              receiving ends, along with the required protocol changes.
>>
>>
>>
>>
>>         The IETF Secretariat
>>         ______________________________**___________________
>>         tcpm mailing list
>>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>>         https://www.ietf.org/mailman/_**_listinfo/tcpm<https://www.ietf.org/mailman/__listinfo/tcpm>
>>         <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>
>>     <https://www.ietf.org/mailman/**listinfo/tcpm<https://www.ietf.org/mailman/listinfo/tcpm>
>> >
>>
>>
>>

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<div dir=3D"ltr">I agree that if we introduce a significant amount of redun=
dant transmissions, these packets need to be accounted for by the congestio=
n window. However, the way we designed our algorithm, the amount of redunda=
ncy is relatively small (usually 16 data packets are encoded by 1 XOR packe=
t), which is why in our prototype the XOR packets are transmitted outside o=
f the window.<div>
<br></div><div>And you are right that these schemes in general are pretty o=
ld, but to our knowledge no one has actually thought about the protocol lev=
el details of incorporating FEC into TCP before, including all the trouble =
middleboxes are causing.</div>
<div><br></div><div>- Tobias<div class=3D"gmail_extra"><br><div class=3D"gm=
ail_quote">On Mon, Jul 15, 2013 at 9:32 PM, Joe Touch <span dir=3D"ltr">&lt=
;<a href=3D"mailto:touch@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</=
span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Well, FEC-style redundant transmission is fi=
ne so long as you subtract those XOR packets from your congestion window. I=
.e., either you send 10 data packets, or 5 data and 5 XOR.<br>

<br>
If you&#39;re just pumping data into the net in the hope that something mak=
es it, we&#39;ve seen that before too - Roadrunner tried it 15 years ago. W=
e all know it works, but I hope we all know why it&#39;s not a good idea.<b=
r>

<br>
BTW, these sorts of tricks are a LOT older than 1-2 years.<br>
<br>
Joe<div class=3D"im"><br>
<br>
On 7/15/2013 5:49 PM, Tobias Flach wrote:<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"im">
There are indeed far more elegant and sophisticated coding schemes out<br>
there than the XOR approach we used. TCP Boston as well as other<br>
published approaches like &quot;Network Coded TCP&quot;<br>
(<a href=3D"http://arxiv.org/abs/1212.2291" target=3D"_blank">http://arxiv.=
org/abs/1212.<u></u>2291</a>) are capable to mask losses better than<br>
XOR, but XOR has the advantage that it is easy to implement and results<br>
in a relatively low computational overhead. In addition, we do not limit<br=
>
our proposal to XOR, but rather use it to demonstrate how TCP-IR works<br>
in general, and we can easily extend the module to support other coding<br>
schemes in the future.<br>
<br>
Regarding the second question: Regular payloads and parity data are not<br>
mixed in the same packet. Parity data is sent separately which enables<br>
immediate recovery if only one packet in the encoding range is lost. As<br>
said, more sophisticated coding schemes might do better here, but our<br>
experiments have shown that even simple XOR can improve latency a lot -<br>
especially for short flows.<br>
<br>
- Tobias<br>
<br>
<br>
On Mon, Jul 15, 2013 at 5:58 PM, Joe Touch &lt;<a href=3D"mailto:touch@isi.=
edu" target=3D"_blank">touch@isi.edu</a><br></div><div class=3D"im">
&lt;mailto:<a href=3D"mailto:touch@isi.edu" target=3D"_blank">touch@isi.edu=
</a>&gt;&gt; wrote:<br>
<br>
=A0 =A0 You might take a look at TCP Boston, which is a much more elegant<b=
r>
=A0 =A0 FEC than just parity.<br>
<br>
=A0 =A0 But what use is parity? If a packet is corrupted, most link layers<=
br>
=A0 =A0 will drop it anyway (esp. because you don&#39;t even know if the pa=
cket<br>
=A0 =A0 header is correct, thus whether you&#39;re sending it to the right<=
br>
=A0 =A0 connection).<br>
<br>
=A0 =A0 Joe<br>
<br>
<br>
=A0 =A0 On 7/15/2013 2:49 PM, Yuchung Cheng wrote:<br>
<br>
=A0 =A0 =A0 =A0 Hi<br>
<br>
=A0 =A0 =A0 =A0 We are proposing an outrageous idea to send parity data in =
TCP<br>
=A0 =A0 =A0 =A0 packet to promote sub 1-RTT loss recovery. This requires a =
separate<br>
=A0 =A0 =A0 =A0 sequence space which we aren&#39;t quite sure how to do thi=
s yet without<br>
=A0 =A0 =A0 =A0 upsetting middle-boxes. More data can be found in our new p=
aper<br>
=A0 =A0 =A0 =A0 &quot;Reducing Web Latency: the Virtue of Gentle Aggression=
&quot; sigcomm 2013<br></div>
=A0 =A0 =A0 =A0 <a href=3D"http://static." target=3D"_blank">http://static.=
</a>__<a href=3D"http://googleusercontent.com/__external_content/untrusted_=
__dlcp/research.google.com/en/__us/pubs/archive/41217.pdf" target=3D"_blank=
">googleusercont<u></u>ent.com/__external_content/<u></u>untrusted___dlcp/r=
esearch.<u></u>google.com/en/__us/pubs/<u></u>archive/41217.pdf</a><div cla=
ss=3D"im">
<br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"http://static.googleusercontent.com/external=
_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf" t=
arget=3D"_blank">http://static.<u></u>googleusercontent.com/<u></u>external=
_content/untrusted_<u></u>dlcp/research.google.com/en/<u></u>us/pubs/archiv=
e/41217.pdf</a>&gt;<br>

<br>
=A0 =A0 =A0 =A0 Comments are very welcome. Linux patches release are under =
way.<br>
<br>
<br>
=A0 =A0 =A0 =A0 ---------- Forwarded message ----------<br></div><div class=
=3D"im">
=A0 =A0 =A0 =A0 From: =A0&lt;<a href=3D"mailto:internet-drafts@ietf.org" ta=
rget=3D"_blank">internet-drafts@ietf.org</a> &lt;mailto:<a href=3D"mailto:i=
nternet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.<u></u>org<=
/a>&gt;&gt;<br>

=A0 =A0 =A0 =A0 Date: Sun, Jul 14, 2013 at 4:59 PM<br>
=A0 =A0 =A0 =A0 Subject: New Version Notification for draft-flach-tcpm-fec-=
00.txt<br></div><div class=3D"im">
=A0 =A0 =A0 =A0 To: Tobias Flach &lt;<a href=3D"mailto:flach@usc.edu" targe=
t=3D"_blank">flach@usc.edu</a> &lt;mailto:<a href=3D"mailto:flach@usc.edu" =
target=3D"_blank">flach@usc.edu</a>&gt;&gt;, Nandita<br>
=A0 =A0 =A0 =A0 Dukkipati<br></div><div class=3D"im">
=A0 =A0 =A0 =A0 &lt;<a href=3D"mailto:nanditad@google.com" target=3D"_blank=
">nanditad@google.com</a> &lt;mailto:<a href=3D"mailto:nanditad@google.com"=
 target=3D"_blank">nanditad@google.com</a>&gt;&gt;, Yuchung<br>
=A0 =A0 =A0 =A0 Cheng &lt;<a href=3D"mailto:ycheng@google.com" target=3D"_b=
lank">ycheng@google.com</a> &lt;mailto:<a href=3D"mailto:ycheng@google.com"=
 target=3D"_blank">ycheng@google.com</a>&gt;&gt;, Barath<br>
=A0 =A0 =A0 =A0 Raghavan &lt;<a href=3D"mailto:barath@google.com" target=3D=
"_blank">barath@google.com</a> &lt;mailto:<a href=3D"mailto:barath@google.c=
om" target=3D"_blank">barath@google.com</a>&gt;&gt;<br>
<br>
<br>
<br>
=A0 =A0 =A0 =A0 A new version of I-D, draft-flach-tcpm-fec-00.txt<br>
=A0 =A0 =A0 =A0 has been successfully submitted by Tobias Flach and posted =
to the<br>
=A0 =A0 =A0 =A0 IETF repository.<br>
<br>
=A0 =A0 =A0 =A0 Filename: =A0 =A0 =A0 =A0draft-flach-tcpm-fec<br>
=A0 =A0 =A0 =A0 Revision: =A0 =A0 =A0 =A000<br>
=A0 =A0 =A0 =A0 Title: =A0 =A0 =A0 =A0 =A0 TCP Instant Recovery: Incorporat=
ing Forward Error<br>
=A0 =A0 =A0 =A0 Correction in TCP<br>
=A0 =A0 =A0 =A0 Creation date: =A0 2013-07-15<br>
=A0 =A0 =A0 =A0 Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
=A0 =A0 =A0 =A0 Number of pages: 15<br>
=A0 =A0 =A0 =A0 URL:<br></div>
=A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-__drafts/draft-flac=
h-tcpm-fec-__00.txt" target=3D"_blank">http://www.ietf.org/internet-_<u></u=
>_drafts/draft-flach-tcpm-fec-_<u></u>_00.txt</a><br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"http://www.ietf.org/internet-drafts/draft-fl=
ach-tcpm-fec-00.txt" target=3D"_blank">http://www.ietf.org/internet-<u></u>=
drafts/draft-flach-tcpm-fec-<u></u>00.txt</a>&gt;<br>
=A0 =A0 =A0 =A0 Status: <a href=3D"http://datatracker.ietf.org/__doc/draft-=
flach-tcpm-fec" target=3D"_blank">http://datatracker.ietf.org/__<u></u>doc/=
draft-flach-tcpm-fec</a><br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"http://datatracker.ietf.org/doc/draft-flach-=
tcpm-fec" target=3D"_blank">http://datatracker.ietf.org/<u></u>doc/draft-fl=
ach-tcpm-fec</a>&gt;<br>
=A0 =A0 =A0 =A0 Htmlized: <a href=3D"http://tools.ietf.org/html/__draft-fla=
ch-tcpm-fec-00" target=3D"_blank">http://tools.ietf.org/html/__<u></u>draft=
-flach-tcpm-fec-00</a><div class=3D"im"><br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"http://tools.ietf.org/html/draft-flach-tcpm-=
fec-00" target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-flach-tcp=
m-fec-00</a>&gt;<br>
<br>
<br>
=A0 =A0 =A0 =A0 Abstract:<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0Ordinary TCP loss recovery takes at least one ro=
und-trip<br>
=A0 =A0 =A0 =A0 time and as<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0such can increase application-perceived latency,=
 especially<br>
=A0 =A0 =A0 =A0 for short<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0flows such as Web transactions. =A0TCP Instant R=
ecovery<br>
=A0 =A0 =A0 =A0 (TCP-IR) is an<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0experimental algorithm that allows a receiving e=
nd to<br>
=A0 =A0 =A0 =A0 recover lost<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0packets without retransmissions, thus potentiall=
y saving at<br>
=A0 =A0 =A0 =A0 least one<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0full round-trip time compared to standard TCP. =
=A0TCP-IR<br>
=A0 =A0 =A0 =A0 achieves this<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0by judiciously injecting encoded data segments w=
ithin a TCP<br>
=A0 =A0 =A0 =A0 stream.<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0This document describes the TCP-IR algorithm at =
the sending and<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0receiving ends, along with the required protocol=
 changes.<br>
<br>
<br>
<br>
<br>
=A0 =A0 =A0 =A0 The IETF Secretariat<br></div>
=A0 =A0 =A0 =A0 ______________________________<u></u>___________________<br=
>
=A0 =A0 =A0 =A0 tcpm mailing list<br>
=A0 =A0 =A0 =A0 <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@iet=
f.org</a> &lt;mailto:<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcp=
m@ietf.org</a>&gt;<br>
=A0 =A0 =A0 =A0 <a href=3D"https://www.ietf.org/mailman/__listinfo/tcpm" ta=
rget=3D"_blank">https://www.ietf.org/mailman/_<u></u>_listinfo/tcpm</a><br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" =
target=3D"_blank">https://www.ietf.org/mailman/<u></u>listinfo/tcpm</a>&gt;=
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=A0 =A0 ______________________________<u></u>___________________<br>
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=A0 =A0 <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a=
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rg</a>&gt;<br>
=A0 =A0 <a href=3D"https://www.ietf.org/mailman/__listinfo/tcpm" target=3D"=
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<br>
<br>
</blockquote>
</blockquote></div><br></div></div></div>

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Thanks for your comments.
Responses inline:


On Tue, Jul 16, 2013 at 10:29 AM, Scharf, Michael (Michael) <
michael.scharf@alcatel-lucent.com> wrote:

> Hi,
>
> After a quick scan through the draft, some initial questions...
>
> >   1.  Periodically in every round-trip time, a TCP-IR sender places the
> >   XOR of newly transmitted segments into a single MSS-length checksum
> >   packet.  The XOR is only computed for new segments not previously
> >   included in checksums.
>
> Is it correct that there are more than one XOR packets per RTT if there
> are more than 8 or 16 outstanding packets, i.e., CWND>8 or 16?
>

That is correct. If there are more than 8 (or 16) unencoded packets in the
queue when the timer fires, multiple XOR packets are generated.
Additionally, if new packets are transmitted after the timer fired, the
timer is armed again (set to RTT/4), so for a sender transmitting data
continuously the timer would fire up to four times per RTT.


>
> >   3.  The encoded XOR packet uses the same sequence number as the first
> >   segment it encodes.  The encoded packet carries a flag in the TCP-IR
> >   option signaling that the payload is encoded.  A receiver uses the
> >   flag to disambiguate an encoded packet from a regular
> >   (re)transmission, since both segments carry the same sequence number.
> >   The option also includes the number of bytes that the payload
> >   encodes.
>
> So, on the wire there are different bytes for the same sequence number,
> right?
>
>
Yes.


> I recall lengthy discussions on that in the MPTCP WG.
>
> BTW: Have you thought of using MPTCP, i.e., sending the XOR packets on a
> separate MPTCP subflow between the same IP addresses? This could be an
> interesting application of MPTCP. MPTCP already allows today sending
> packets redundantly on two subflows - adding coding to MPTCP seems doable.
>

We thought about MPTCP in our initial design phase but our take was that
MPTCP has too much complexity for our purpose. The question we tried to
answer was: How would we augment TCP in the simplest possible way to inject
FEC packets within the data stream. Once MPTCP implementations are more
mature and widely deployed we certainly have to revisit this question.


>
> >   4.  For the purpose of recovering lost segments, a receiver buffers
> >   the last fifteen in-order MSS blocks that it ACKed, even if the
> >   application layer has already consumed these blocks.  Because an
> >   encoded packet is the XOR of at most sixteen MSS segments, the
> >   receiver can recover any single lost packet by computing the XOR of
> >   the encoded payload and the buffered data in the encoding range.
>
> What about TCP flow control? Are those 15 packets reduced from the
> advertised RWND?
>

Yes. In the draft and prototype we do not mention this yet, but will
include it in future revisions.


>
> >   6.  TCP-IR does not circumvent congestion control.  If the receiver
> >   [...]
>
> As already pointed out by Joe, not circumventing congestion control
> implies to me that XOR packets are substracted from CWND. I wonder why this
> is not explicitly mentioned in the draft. Since packet loss is mainly due
> to congestion in most parts of the Internet, increasing the transmission
> rate on a congested link beyond what is allowed by congestion control is a
> no-go.
>

By not circumventing CC, we mean reducing cwnd in the event that a loss is
transparently recovered at the receiver using FEC segments.
The next version of the draft will include a clarification on whether FEC
segments should be within or outside of cwnd. Both are viable options.


>
> >   TCP-IR adds a short delay in the transmission of encoded packets to
> >   the reduce the probability of losing both the original transmission
> >   and the encoded packet in the same loss burst.
>
> So, the recovery delay is actually not too different to TLP?
>

The delay added for FEC segments is only a 1/4 RTT. While TLP waits for 2
RTTs before sending a new or retransmitted segment. Besides that, you are
right that there is some similarity between the two schemes. TLP could be
considered an additional encoding type, which only "encodes" the last
packet in flight.


>
> >   The last 15 ACKed MSS length segments remain in the buffer, even if
> >   the application layer has already consumed these segments.  Segments
> >   received out-of-order are already buffered by default and cannot be
> >   consumed by the application layer.  Since a single TCP-IR packet
> >   encodes at most 8 (interleaved XOR) or 16 (basic XOR) MSS length
> >   segments, any single segment loss (up to MSS length) in the encoding
> >   range can be recovered by the decoder.
>
> With this scheme, can a malicious sender attack a receiver so that it runs
> out of buffer space? For instance, what does a receiver do if a sender
> sends only one out of 16 packets?
>

I might not follow this correctly, but XOR packets are always transmitted
after the original data segments, and right now they are discarded if there
is not enough data buffered data at the receiver to recover segments. So if
a sender sends 1 out of 16 packets, and then the XOR for these 16 packets,
the receiver will not recover any data, and respond with an ACK having the
R_FAIL flag set. Does that answer your question?


>
> >   Some problems caused by middlebox interference (and their solutions
> >   in TCP-IR) are not discussed in the current version of this draft:
> >
> >   o  Rewriting of the acknowledgement number if the acknowledged
> >      segment was not observed by the middlebox.  With TCP-IR this can
> >      occur after recovering a lost segment.  This issue can be
> >      circumvented by retransmitting the recovered segment, even though
> >      it is not needed by the other endpoint anymore.  This plugs the
> >      "sequence hole" in the state of the middlebox.
> >   o  Rewriting payloads of previously seen segments.
> >   o  Packet coalescing and splitting.
>
> Out of my head, these were major design constraints for MPTCP. This is why
> I wonder if transfering the XOR packets over a separate MPTCP subflow could
> maybe help.
>

I am sure it will work if FEC segments are transferred over an MPTCP
subflow. The downside is this is not immediately deployable. Therefore, we
are interested in investigating extensions to TCP to accommodate simple XOR
segments.


>
> >   We conducted two kinds of experiments.  The first was in an emulated
> >   setting using loss patterns similar to those observed in our
> >   measurement of real traffic.  We used the netem module to emulate a
> >   200 ms RTT and both random and correlated loss rates of 2%. TCP-IR
> >   reduced the latency for short transfers in lossy environments by 28%
> >   in the 90th percentile.  The benefits diminish as the minimum number
>
> Could you comment on the transmission overhead or goodput?
>
> At first sight, the amount of sent bytes is increased by at least 1/16 =
> 6% (or 1/8 = 12% with interleaving) even if a connection faces no
> congestion at all. Plus the per-packet overhead of the TCP option and the
> consumption of scarce SYN bytes.
>
> Of course, one could reduce the overhead by selectively activating the FEC
> scheme only if a link is congested. But this still means that the goodput
> on a congested link is reduced.
>

That is correct. For longer flows, or flows which usually don't see
congestion, FEC is only causing performance degradation. We don't envision
that every flow uses this, but rather have it as a tuning knob for
scenarios where a reduction in throughput is acceptable if that reduces
overall latency (by reducing the probability of an RTO).


>
> >   of RTTs necessary to complete the transaction increases (due to the
> >   message size) because the time to recover from losses no longer
> >   dominates the overall transmission time.  TCP-IR is better suited for
> >   small transfers common in today's Web.
>
> Really? If a sender only has, say, 4 KB to send, i.e., about 3 packets for
> MTU 1500 byte, sending an additional coded packet results in an overhead of
> about 25%, right?


Yes. The key is that a loss of the third packet can take a very long time
to recover from. With TLP we need to wait at least 2 RTTs thus completing
the transmission after 3 RTTs. With TCP-IR we are done after 1.25 RTTs. Now
it is up to the user whether avoiding a 25% overhead or reducing latency by
58% in the event of a tail loss is more important.


>


> And if an app only transfers a single byte, is the overhead then MSS /
> 1byte = 150000%?
>

We have not specified this yet, but if there is less than MSS data to
encode, the encoded packet does not need to use padding to fill up unused
space. So in this particular case the encoding range would be 1, and the
encoded packet would carry a 1-byte payload. Overhead can therefore be 100%
in the worst case.


>
> With such an overhead, is TCP-IR really well-suited for small transfers?
>
> >   The two Options for TCP-IR used during negotiation and subsequently
> >   in every packet of the connection require IANA allocate one value
> >   from the TCP option Kind namespace.  Experimentation prior to the
> >   allocation SHOULD follow [EXPOPT] and use experimental option kind
> >   254 and two magic bytes 0xDC60, and migrate to the new option after
> >   the allocation accordingly.
>
> If experiments are performed over Internet paths, a registration at
> http://www.iana.org/assignments/tcp-parameters/tcp-parameters.xhtml#tcp-exidswould be highly welcome.
>

Thanks for the pointer!

- Tobias


>
> Sorry if I missed something in my first reading.
>
> Thanks
>
> Michael
>
>
>
> > -----Original Message-----
> > From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
> > Behalf Of Yuchung Cheng
> > Sent: Monday, July 15, 2013 11:49 PM
> > To: tcpm@ietf.org Extensions
> > Subject: [tcpm] Fwd: New Version Notification for
> > draft-flach-tcpm-fec-00.txt
> >
> > Hi
> >
> > We are proposing an outrageous idea to send parity data in
> > TCP packet to promote sub 1-RTT loss recovery. This requires
> > a separate sequence space which we aren't quite sure how to
> > do this yet without upsetting middle-boxes. More data can be
> > found in our new paper "Reducing Web Latency: the Virtue of
> > Gentle Aggression" sigcomm 2013
> > http://static.googleusercontent.com/external_content/untrusted
> > _dlcp/research.google.com/en/us/pubs/archive/41217.pdf
> >
> > Comments are very welcome. Linux patches release are under way.
> >
> > ---------- Forwarded message ----------
> > From:  <internet-drafts@ietf.org>
> > Date: Sun, Jul 14, 2013 at 4:59 PM
> > Subject: New Version Notification for draft-flach-tcpm-fec-00.txt
> > To: Tobias Flach <flach@usc.edu>, Nandita Dukkipati
> > <nanditad@google.com>, Yuchung Cheng <ycheng@google.com>,
> > Barath Raghavan <barath@google.com>
> >
> >
> >
> > A new version of I-D, draft-flach-tcpm-fec-00.txt has been
> > successfully submitted by Tobias Flach and posted to the IETF
> > repository.
> >
> > Filename:        draft-flach-tcpm-fec
> > Revision:        00
> > Title:           TCP Instant Recovery: Incorporating Forward Error
> > Correction in TCP
> > Creation date:   2013-07-15
> > Group:           Individual Submission
> > Number of pages: 15
> > URL:
> > http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt
> > Status:          http://datatracker.ietf.org/doc/draft-flach-tcpm-fec
> > Htmlized:        http://tools.ietf.org/html/draft-flach-tcpm-fec-00
> >
> >
> > Abstract:
> >    Ordinary TCP loss recovery takes at least one round-trip
> > time and as
> >    such can increase application-perceived latency,
> > especially for short
> >    flows such as Web transactions.  TCP Instant Recovery
> > (TCP-IR) is an
> >    experimental algorithm that allows a receiving end to recover lost
> >    packets without retransmissions, thus potentially saving
> > at least one
> >    full round-trip time compared to standard TCP.  TCP-IR
> > achieves this
> >    by judiciously injecting encoded data segments within a TCP stream.
> >    This document describes the TCP-IR algorithm at the sending and
> >    receiving ends, along with the required protocol changes.
> >
> >
> >
> >
> > The IETF Secretariat
> > _______________________________________________
> > 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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Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks for your comments.<div>Responses inline:<br><div cl=
ass=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, Jul 16, 2013=
 at 10:29 AM, Scharf, Michael (Michael) <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:michael.scharf@alcatel-lucent.com" target=3D"_blank">michael.scharf@al=
catel-lucent.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Hi,<br>
<br>
After a quick scan through the draft, some initial questions...<br>
<br>
&gt; =A0 1. =A0Periodically in every round-trip time, a TCP-IR sender place=
s the<br>
&gt; =A0 XOR of newly transmitted segments into a single MSS-length checksu=
m<br>
&gt; =A0 packet. =A0The XOR is only computed for new segments not previousl=
y<br>
&gt; =A0 included in checksums.<br>
<br>
Is it correct that there are more than one XOR packets per RTT if there are=
 more than 8 or 16 outstanding packets, i.e., CWND&gt;8 or 16?<br></blockqu=
ote><div><br></div><div>That is correct. If there are more than 8 (or 16) u=
nencoded packets in the queue when the timer fires, multiple XOR packets ar=
e generated. Additionally, if new packets are transmitted after the timer f=
ired, the timer is armed again (set to RTT/4), so for a sender transmitting=
 data continuously the timer would fire up to four times per RTT.</div>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 3. =A0The encoded XOR packet uses the same sequence number as the =
first<br>
&gt; =A0 segment it encodes. =A0The encoded packet carries a flag in the TC=
P-IR<br>
&gt; =A0 option signaling that the payload is encoded. =A0A receiver uses t=
he<br>
&gt; =A0 flag to disambiguate an encoded packet from a regular<br>
&gt; =A0 (re)transmission, since both segments carry the same sequence numb=
er.<br>
&gt; =A0 The option also includes the number of bytes that the payload<br>
&gt; =A0 encodes.<br>
<br>
So, on the wire there are different bytes for the same sequence number, rig=
ht?<br>
<br></blockquote><div><br></div><div>Yes.</div><div>=A0</div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px=
;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1e=
x">


I recall lengthy discussions on that in the MPTCP WG.<br>
<br>
BTW: Have you thought of using MPTCP, i.e., sending the XOR packets on a se=
parate MPTCP subflow between the same IP addresses? This could be an intere=
sting application of MPTCP. MPTCP already allows today sending packets redu=
ndantly on two subflows - adding coding to MPTCP seems doable.<br>

</blockquote><div><br></div><div>We thought about MPTCP in our initial desi=
gn phase but our take was that MPTCP has too much complexity for our purpos=
e. The question we tried to answer was: How would we augment TCP in the sim=
plest possible way to inject FEC packets within the data stream. Once MPTCP=
 implementations are more mature and widely deployed we certainly have to r=
evisit this question.</div>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 4. =A0For the purpose of recovering lost segments, a receiver buff=
ers<br>
&gt; =A0 the last fifteen in-order MSS blocks that it ACKed, even if the<br=
>
&gt; =A0 application layer has already consumed these blocks. =A0Because an=
<br>
&gt; =A0 encoded packet is the XOR of at most sixteen MSS segments, the<br>
&gt; =A0 receiver can recover any single lost packet by computing the XOR o=
f<br>
&gt; =A0 the encoded payload and the buffered data in the encoding range.<b=
r>
<br>
What about TCP flow control? Are those 15 packets reduced from the advertis=
ed RWND?<br></blockquote><div><br></div><div>Yes. In the draft and prototyp=
e we do not mention this yet, but will include it in future revisions.</div=
>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 6. =A0TCP-IR does not circumvent congestion control. =A0If the rec=
eiver<br>
&gt; =A0 [...]<br>
<br>
As already pointed out by Joe, not circumventing congestion control implies=
 to me that XOR packets are substracted from CWND. I wonder why this is not=
 explicitly mentioned in the draft. Since packet loss is mainly due to cong=
estion in most parts of the Internet, increasing the transmission rate on a=
 congested link beyond what is allowed by congestion control is a no-go.<br=
>

</blockquote><div><br></div><div><span style=3D"font-family:arial,sans-seri=
f;font-size:13px">By not circumventing CC, we mean reducing cwnd in the eve=
nt that a loss is transparently recovered at the receiver using FEC segment=
s.=A0</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px">The next v=
ersion of the draft will include a clarification on whether FEC segments sh=
ould be within or outside of cwnd. Both are viable options.</span><br></div=
>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 TCP-IR adds a short delay in the transmission of encoded packets t=
o<br>
&gt; =A0 the reduce the probability of losing both the original transmissio=
n<br>
&gt; =A0 and the encoded packet in the same loss burst.<br>
<br>
So, the recovery delay is actually not too different to TLP?<br></blockquot=
e><div><br></div><div><span style=3D"font-family:arial,sans-serif;font-size=
:13px">The delay added for FEC segments is only a 1/4 RTT. While TLP waits =
for 2 RTTs before sending a new or retransmitted segment. Besides that, you=
 are right that there is some similarity between the two schemes. TLP could=
 be considered an additional encoding type, which only &quot;encodes&quot; =
the last packet in flight.</span><br>

</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex">
<br>
&gt; =A0 The last 15 ACKed MSS length segments remain in the buffer, even i=
f<br>
&gt; =A0 the application layer has already consumed these segments. =A0Segm=
ents<br>
&gt; =A0 received out-of-order are already buffered by default and cannot b=
e<br>
&gt; =A0 consumed by the application layer. =A0Since a single TCP-IR packet=
<br>
&gt; =A0 encodes at most 8 (interleaved XOR) or 16 (basic XOR) MSS length<b=
r>
&gt; =A0 segments, any single segment loss (up to MSS length) in the encodi=
ng<br>
&gt; =A0 range can be recovered by the decoder.<br>
<br>
With this scheme, can a malicious sender attack a receiver so that it runs =
out of buffer space? For instance, what does a receiver do if a sender send=
s only one out of 16 packets?<br></blockquote><div><br></div><div>I might n=
ot follow this correctly, but XOR packets are always transmitted after the =
original data segments, and right now they are discarded if there is not en=
ough data buffered data at the receiver to recover segments. So if a sender=
 sends 1 out of 16 packets, and then the XOR for these 16 packets, the rece=
iver will not recover any data, and respond with an ACK having the R_FAIL f=
lag set. Does that answer your question?</div>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 Some problems caused by middlebox interference (and their solution=
s<br>
&gt; =A0 in TCP-IR) are not discussed in the current version of this draft:=
<br>
&gt;<br>
&gt; =A0 o =A0Rewriting of the acknowledgement number if the acknowledged<b=
r>
&gt; =A0 =A0 =A0segment was not observed by the middlebox. =A0With TCP-IR t=
his can<br>
&gt; =A0 =A0 =A0occur after recovering a lost segment. =A0This issue can be=
<br>
&gt; =A0 =A0 =A0circumvented by retransmitting the recovered segment, even =
though<br>
&gt; =A0 =A0 =A0it is not needed by the other endpoint anymore. =A0This plu=
gs the<br>
&gt; =A0 =A0 =A0&quot;sequence hole&quot; in the state of the middlebox.<br=
>
&gt; =A0 o =A0Rewriting payloads of previously seen segments.<br>
&gt; =A0 o =A0Packet coalescing and splitting.<br>
<br>
Out of my head, these were major design constraints for MPTCP. This is why =
I wonder if transfering the XOR packets over a separate MPTCP subflow could=
 maybe help.<br></blockquote><div><br></div><div><span style=3D"font-family=
:arial,sans-serif;font-size:13px">I am sure it will work if FEC segments ar=
e transferred over an MPTCP subflow. The downside is this is not immediatel=
y deployable. Therefore, we are interested in investigating extensions to T=
CP to accommodate simple XOR segments.</span></div>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 We conducted two kinds of experiments. =A0The first was in an emul=
ated<br>
&gt; =A0 setting using loss patterns similar to those observed in our<br>
&gt; =A0 measurement of real traffic. =A0We used the netem module to emulat=
e a<br>
&gt; =A0 200 ms RTT and both random and correlated loss rates of 2%. TCP-IR=
<br>
&gt; =A0 reduced the latency for short transfers in lossy environments by 2=
8%<br>
&gt; =A0 in the 90th percentile. =A0The benefits diminish as the minimum nu=
mber<br>
<br>
Could you comment on the transmission overhead or goodput?<br>
<br>
At first sight, the amount of sent bytes is increased by at least 1/16 =3D =
6% (or 1/8 =3D 12% with interleaving) even if a connection faces no congest=
ion at all. Plus the per-packet overhead of the TCP option and the consumpt=
ion of scarce SYN bytes.<br>


<br>
Of course, one could reduce the overhead by selectively activating the FEC =
scheme only if a link is congested. But this still means that the goodput o=
n a congested link is reduced.<br></blockquote><div><br></div><div>That is =
correct. For longer flows, or flows which usually don&#39;t see congestion,=
 FEC is only causing performance degradation. We don&#39;t envision that ev=
ery flow uses this, but rather have it as a tuning knob for scenarios where=
 a reduction in throughput is acceptable if that reduces overall latency (b=
y reducing the probability of an RTO).</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; =A0 of RTTs necessary to complete the transaction increases (due to th=
e<br>
&gt; =A0 message size) because the time to recover from losses no longer<br=
>
&gt; =A0 dominates the overall transmission time. =A0TCP-IR is better suite=
d for<br>
&gt; =A0 small transfers common in today&#39;s Web.<br>
<br>
Really? If a sender only has, say, 4 KB to send, i.e., about 3 packets for =
MTU 1500 byte, sending an additional coded packet results in an overhead of=
 about 25%, right?</blockquote><div><br></div><div>Yes. The key is that a l=
oss of the third packet can take a very long time to recover from. With TLP=
 we need to wait at least 2 RTTs thus completing the transmission after 3 R=
TTs. With TCP-IR we are done after 1.25 RTTs. Now it is up to the user whet=
her avoiding a 25% overhead or reducing latency by 58% in the event of a ta=
il loss is more important.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">=A0</blockquote><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<br>
And if an app only transfers a single byte, is the overhead then MSS / 1byt=
e =3D 150000%?<br></blockquote><div><br></div><div>We have not specified th=
is yet, but if there is less than MSS data to encode, the encoded packet do=
es not need to use padding to fill up unused space. So in this particular c=
ase the encoding range would be 1, and the encoded packet would carry a 1-b=
yte payload. Overhead can therefore be 100% in the worst case.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
With such an overhead, is TCP-IR really well-suited for small transfers?<br=
>
<br>
&gt; =A0 The two Options for TCP-IR used during negotiation and subsequentl=
y<br>
&gt; =A0 in every packet of the connection require IANA allocate one value<=
br>
&gt; =A0 from the TCP option Kind namespace. =A0Experimentation prior to th=
e<br>
&gt; =A0 allocation SHOULD follow [EXPOPT] and use experimental option kind=
<br>
&gt; =A0 254 and two magic bytes 0xDC60, and migrate to the new option afte=
r<br>
&gt; =A0 the allocation accordingly.<br>
<br>
If experiments are performed over Internet paths, a registration at <a href=
=3D"http://www.iana.org/assignments/tcp-parameters/tcp-parameters.xhtml#tcp=
-exids" target=3D"_blank">http://www.iana.org/assignments/tcp-parameters/tc=
p-parameters.xhtml#tcp-exids</a> would be highly welcome.<br>
</blockquote><div><br></div><div>Thanks for the pointer!</div><div><br></di=
v><div>- Tobias</div><div>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">


<br>
Sorry if I missed something in my first reading.<br>
<br>
Thanks<br>
<span><font color=3D"#888888"><br>
Michael<br>
</font></span><div><br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:tcpm-bounces@ietf.org" target=3D"_blank">tcpm-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:tcpm-bounces@ietf.org" targe=
t=3D"_blank">tcpm-bounces@ietf.org</a>] On<br>
</div><div>&gt; Behalf Of Yuchung Cheng<br>
&gt; Sent: Monday, July 15, 2013 11:49 PM<br>
&gt; To: <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</=
a> Extensions<br>
&gt; Subject: [tcpm] Fwd: New Version Notification for<br>
&gt; draft-flach-tcpm-fec-00.txt<br>
&gt;<br>
</div><div><div>&gt; Hi<br>
&gt;<br>
&gt; We are proposing an outrageous idea to send parity data in<br>
&gt; TCP packet to promote sub 1-RTT loss recovery. This requires<br>
&gt; a separate sequence space which we aren&#39;t quite sure how to<br>
&gt; do this yet without upsetting middle-boxes. More data can be<br>
&gt; found in our new paper &quot;Reducing Web Latency: the Virtue of<br>
&gt; Gentle Aggression&quot; sigcomm 2013<br>
&gt; <a href=3D"http://static.googleusercontent.com/external_content/untrus=
ted" target=3D"_blank">http://static.googleusercontent.com/external_content=
/untrusted</a><br>
&gt; _dlcp/<a href=3D"http://research.google.com/en/us/pubs/archive/41217.p=
df" target=3D"_blank">research.google.com/en/us/pubs/archive/41217.pdf</a><=
br>
&gt;<br>
&gt; Comments are very welcome. Linux patches release are under way.<br>
&gt;<br>
&gt; ---------- Forwarded message ----------<br>
&gt; From: =A0&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_bl=
ank">internet-drafts@ietf.org</a>&gt;<br>
&gt; Date: Sun, Jul 14, 2013 at 4:59 PM<br>
&gt; Subject: New Version Notification for draft-flach-tcpm-fec-00.txt<br>
&gt; To: Tobias Flach &lt;<a href=3D"mailto:flach@usc.edu" target=3D"_blank=
">flach@usc.edu</a>&gt;, Nandita Dukkipati<br>
&gt; &lt;<a href=3D"mailto:nanditad@google.com" target=3D"_blank">nanditad@=
google.com</a>&gt;, Yuchung Cheng &lt;<a href=3D"mailto:ycheng@google.com" =
target=3D"_blank">ycheng@google.com</a>&gt;,<br>
&gt; Barath Raghavan &lt;<a href=3D"mailto:barath@google.com" target=3D"_bl=
ank">barath@google.com</a>&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; A new version of I-D, draft-flach-tcpm-fec-00.txt has been<br>
&gt; successfully submitted by Tobias Flach and posted to the IETF<br>
&gt; repository.<br>
&gt;<br>
&gt; Filename: =A0 =A0 =A0 =A0draft-flach-tcpm-fec<br>
&gt; Revision: =A0 =A0 =A0 =A000<br>
&gt; Title: =A0 =A0 =A0 =A0 =A0 TCP Instant Recovery: Incorporating Forward=
 Error<br>
&gt; Correction in TCP<br>
&gt; Creation date: =A0 2013-07-15<br>
&gt; Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
&gt; Number of pages: 15<br>
&gt; URL:<br>
&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00=
.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-flach-tcp=
m-fec-00.txt</a><br>
&gt; Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/=
draft-flach-tcpm-fec" target=3D"_blank">http://datatracker.ietf.org/doc/dra=
ft-flach-tcpm-fec</a><br>
&gt; Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-f=
lach-tcpm-fec-00" target=3D"_blank">http://tools.ietf.org/html/draft-flach-=
tcpm-fec-00</a><br>
&gt;<br>
&gt;<br>
&gt; Abstract:<br>
&gt; =A0 =A0Ordinary TCP loss recovery takes at least one round-trip<br>
&gt; time and as<br>
&gt; =A0 =A0such can increase application-perceived latency,<br>
&gt; especially for short<br>
&gt; =A0 =A0flows such as Web transactions. =A0TCP Instant Recovery<br>
&gt; (TCP-IR) is an<br>
&gt; =A0 =A0experimental algorithm that allows a receiving end to recover l=
ost<br>
&gt; =A0 =A0packets without retransmissions, thus potentially saving<br>
&gt; at least one<br>
&gt; =A0 =A0full round-trip time compared to standard TCP. =A0TCP-IR<br>
&gt; achieves this<br>
&gt; =A0 =A0by judiciously injecting encoded data segments within a TCP str=
eam.<br>
&gt; =A0 =A0This document describes the TCP-IR algorithm at the sending and=
<br>
&gt; =A0 =A0receiving ends, along with the required protocol changes.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The IETF Secretariat<br>
&gt; _______________________________________________<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" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt;<br>
_______________________________________________<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" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div></div>

--14dae9cc955c7cc69404e1b64e8b--

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Cc: "tcpm-chairs@tools.ietf.org" <tcpm-chairs@tools.ietf.org>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: [tcpm] Tail loss probe talk [was Re: Draft agenda for IETF 87 in Berlin]
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On Wed, Jul 17, 2013 at 2:01 AM, Scharf, Michael (Michael)
<michael.scharf@alcatel-lucent.com> wrote:
> Hi all,
>
> The first draft of the TCPM agenda can be found at https://datatracker.ie=
tf.org/meeting/87/agenda/tcpm/ and is also copied below.
>
> We tried to take into account all requests, but as a result, the agenda i=
s pretty full. This implies:
>
> - The slots are *total* time. Authors, please reserve 50% of your time fo=
r questions and discussion.
>
> - We got time slot requests for drafts that were not be updated since the=
 last meeting, typically promising new results for the presentation in the =
meeting. Mailing list discussion in the next week would be *highly* welcome=
 in order to justify a time slot on our full agenda.
Sure we should do that for TLP

>
> Please let us know if you have any comments, suggestions, or if we missed=
 something.
>
> Happy agenda bashing!
>
> Michael, Pasi, Yoshifumi
>
>
>
> *************************************************************************=
************
>
> TCPM Agenda
> IETF 87 in Berlin, Germany
> Tuesday, July 30, 2013, 09:00-11:30
> ***********************************
>
> WG status
> ---------
>
> 09:00
> TCPM status
> Chairs
> 5min
>
>
> WG items
> --------
>
> 09:05
> draft-ietf-tcpm-1323bis-14
> Richard Scheffenegger
> 20min
>
> 09:25
> draft-ietf-tcpm-fastopen-04
> Jerry Chu / Yuchung Cheng
> 15mins
>
> 09:40
> draft-ietf-tcpm-newcwv-02
> Rafaello Secchi / Gorry Fairhurst
> 20min
>
> 10:00
> draft-ietf-tcpm-rtorestart-00 (no new draft)
> Anna Brunstrom
> 10min
>
> 10:10
> draft-ietf-tcpm-accecn-reqs-02
> Mirja Kuelewind
> 5min
>
>
> Non-WG items
> ------------
>
> 10:15
> draft-zimmermann-tcpm-tcp-rfc4614bis-02
> Alexander Zimmermann
> 10min
>
> 10:25
> draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)
> Yuchung Cheng
> 10min
TCP Loss probe (TLP) is not updated the draft because we weren't sure
(yet) what to update, and the new data shows similar improvement that
we've presented
http://static.googleusercontent.com/external_content/untrusted_dlcp/researc=
h.google.com/en//pubs/archive/41217.pdf

Based on the recent WG and private discussions and the SIGCOMM
reviews, people are not sure the performance improvement justifies
this new feature, as the loss recovery is feature rich and complicated
with RFC3517, RFC6937, RFC5682, RFC5827, etc.

To simplify, I'd like to present TLP as a simple conceptual change to
RTO recovery: On first timeout, retransmit the last packet instead of
the first unacked packet, and don't change but reduce first timeout to
2 SRTT, as a "quick loss probe".

As an alternative we've tried to only crank down the RTO but the
result is not good, because reducing cwnd to 1 is catastrophic, and
TCP today has to live sudden delay spikes due to buffer bloat and
mobile delays. It's better to be very conservative before declaring
the network is busted and use LW=3D1.

We are hopeful to get more WG interests for TLP :)

>
> 10:35
> draft-flach-tcpm-fec-00
> Tobias Flach
> 15min
>
> 10:50
> draft-gont-tcpm-tcp-seq-validation-00 (no new draft)
> Fernando Gont
> 10min
>
> 11:00
> draft-kuehlewind-tcpm-ecn-fallback-00
> draft-kuehlewind-tcpm-accurate-ecn-02
> Mirja Kuehlewind
> 15min
>
>
> If time permits
> ---------------
>
> Wired PRR effects
> Mirja Kuehlewind
> 5min
>
> Dealing with sequence-number randomizing firewalls
> Christoph Paasch
> 5min
>
> packetdrill: Scriptable Network Stack Testing, from Sockets to Packets
> Yuchung Cheng
> 5min
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From andre@freebsd.org  Wed Jul 17 08:59:08 2013
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Subject: Re: [tcpm] Improved SYN cookies
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On 17.07.2013 01:08, Barath Raghavan wrote:
> This seems like a very nice enhancement of SYN cookies.  I was reading
> through the patch and had a couple of questions and comments.
>
> In tcp_syncache.c, you say: "The MAC is computed over
> (faddr||laddr||fport||lport||irs||flags||secmod)".  What is secmod?
> Is it used for salting / key selection?  I found it in the code but
> wasn't entirely clear on what it was supposed to represent.  Also, how
> long are secmod, irs, and flags?

Secmod is the memory address of the hash bucket of the SYN cache.  We
typically have 512 hash bucket heads to which the entry chains attach.
The size of secmod depends on the pointer size of the architecture and
is either 32 or 64 bits.  IRS is 32 bits and flags 8 bits.

> My comment is that you might avoid using an unvetted hash algorithm
> (e.g., SipHash) even if it has a good pedigree, and when using a hash
> function to build a MAC, you might prefer using a known-good
> construction like HMAC.

The primary motivation for SipHash is its speed.  We do not gain anything
by using a HMAC-SHA256 which is about a thousand times slower.  We have
to make sure that we do not replace the SYN cache memory exhaustion with
a high computational overhead CPU exhaustion attack.  For example SipHash-
2-4 is three times faster than the (non-HMAC) MD5 hash we previously used.

SipHash is adequate for the purpose here as the key cracking window is
only 30 seconds.  The cryptographers I asked have expressed their confidence
in using SipHash for this purpose.  SipHash itself combines a number of
cryptographic approaches used by a couple of SHA3 finalists.  See the SipHash
paper for more details.

Using a HMAC construction is mostly necessary to prevent length extension
attacks with hash algorithms that are susceptible to it (Merkle–Damgård
construction based ones like MD5 and SHA1+2).  In our case of SYN cookies
the message length is fixed and can't be extended.  A MAC thus should be
sufficient.  SipHash already is a keyed hash MAC that doesn't need a HMAC
construction for this application.

> One good alternative to using a custom hash-based MAC here is to use
> AES.  Since AES is standard and the data you are computing the MAC
> over is small (can be squeezed into 128 bits from what I can tell), a
> nice approach to creating a fast and secure MAC is to use AES as a
> secure pseudorandom function (PRF); all secure PRFs are secure MACs.
> That is, given the 128-bit secret key k (that's given to SipHash in
> the current design) and the message m (data to compute the MAC over),
> you can compute AES_k(m) and truncate the output.  You can rotate the
> secret key k periodically as in the current design.

I wary of self-invented crypto approaches as they violate rule #1.  I'm
not enough of a crypto-math guy to comment on the strength or weakness
of your proposed approach.  I just want to point out that AES is quite
a complex function with a large state.  It most likely is way slower
than even the previous simple MD5 we had.  Also AES, especially in hash
constructions, is very susceptible to side channel timing attacks.  Using
a constant-time AES implementation yet makes it slower again.  Hardware
supported AES is not available on all platforms and even then it typically
is optimized for bulk throughput and not for one-off single block encryption
with comparatively high setup times.

For SYN cookies we have to optimize for speed with reasonable hash strength.
The 30 second cookie lifetime is a big constraint for any attacker.

I believe, given all current knowledge and design goals, SipHash-2-4 is the
optimal choice for SYN cookies.

> I like your analysis that's under the headings "Vector 1" and "Vector
> 2", and think it's sound.

Thanks for the review and feedback. :)

-- 
Andre


From touch@isi.edu  Wed Jul 17 09:33:39 2013
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On 7/17/2013 7:03 AM, Tobias Flach wrote:
> I agree that if we introduce a significant amount of redundant
> transmissions, these packets need to be accounted for by the congestion
> window. However, the way we designed our algorithm, the amount of
> redundancy is relatively small (usually 16 data packets are encoded by 1
> XOR packet), which is why in our prototype the XOR packets are
> transmitted outside of the window.

There's no rule in TCP congestion control that says "oh, it's OK to send 
one more packet if that's a small increase vs. what CWND says".

I appreciate that this extension considers that extra packet "not new 
information", but to the rest of the network it's one extra packet.

> And you are right that these schemes in general are pretty old, but to
> our knowledge no one has actually thought about the protocol level
> details of incorporating FEC into TCP before,

That's incorrect.

> including all the trouble middleboxes are causing.

I haven't seen anything in TCP Boston or other similar approaches that a 
middlebox would interfere with.

Joe


>
> - Tobias
>
> On Mon, Jul 15, 2013 at 9:32 PM, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
>
>     Well, FEC-style redundant transmission is fine so long as you
>     subtract those XOR packets from your congestion window. I.e., either
>     you send 10 data packets, or 5 data and 5 XOR.
>
>     If you're just pumping data into the net in the hope that something
>     makes it, we've seen that before too - Roadrunner tried it 15 years
>     ago. We all know it works, but I hope we all know why it's not a
>     good idea.
>
>     BTW, these sorts of tricks are a LOT older than 1-2 years.
>
>     Joe
>
>
>     On 7/15/2013 5:49 PM, Tobias Flach wrote:
>
>         There are indeed far more elegant and sophisticated coding
>         schemes out
>         there than the XOR approach we used. TCP Boston as well as other
>         published approaches like "Network Coded TCP"
>         (http://arxiv.org/abs/1212.__2291
>         <http://arxiv.org/abs/1212.2291>) are capable to mask losses
>         better than
>         XOR, but XOR has the advantage that it is easy to implement and
>         results
>         in a relatively low computational overhead. In addition, we do
>         not limit
>         our proposal to XOR, but rather use it to demonstrate how TCP-IR
>         works
>         in general, and we can easily extend the module to support other
>         coding
>         schemes in the future.
>
>         Regarding the second question: Regular payloads and parity data
>         are not
>         mixed in the same packet. Parity data is sent separately which
>         enables
>         immediate recovery if only one packet in the encoding range is
>         lost. As
>         said, more sophisticated coding schemes might do better here,
>         but our
>         experiments have shown that even simple XOR can improve latency
>         a lot -
>         especially for short flows.
>
>         - Tobias
>
>
>         On Mon, Jul 15, 2013 at 5:58 PM, Joe Touch <touch@isi.edu
>         <mailto:touch@isi.edu>
>         <mailto:touch@isi.edu <mailto:touch@isi.edu>>> wrote:
>
>              You might take a look at TCP Boston, which is a much more
>         elegant
>              FEC than just parity.
>
>              But what use is parity? If a packet is corrupted, most link
>         layers
>              will drop it anyway (esp. because you don't even know if
>         the packet
>              header is correct, thus whether you're sending it to the right
>              connection).
>
>              Joe
>
>
>              On 7/15/2013 2:49 PM, Yuchung Cheng wrote:
>
>                  Hi
>
>                  We are proposing an outrageous idea to send parity data
>         in TCP
>                  packet to promote sub 1-RTT loss recovery. This
>         requires a separate
>                  sequence space which we aren't quite sure how to do
>         this yet without
>                  upsetting middle-boxes. More data can be found in our
>         new paper
>                  "Reducing Web Latency: the Virtue of Gentle Aggression"
>         sigcomm 2013
>         http://static.__googleusercont__ent.com/__external_content/__untrusted___dlcp/research.__google.com/en/__us/pubs/__archive/41217.pdf
>         <http://googleusercontent.com/__external_content/untrusted___dlcp/research.google.com/en/__us/pubs/archive/41217.pdf>
>
>
>         <http://static.__googleusercontent.com/__external_content/untrusted___dlcp/research.google.com/en/__us/pubs/archive/41217.pdf
>         <http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf>>
>
>                  Comments are very welcome. Linux patches release are
>         under way.
>
>
>                  ---------- Forwarded message ----------
>                  From:  <internet-drafts@ietf.org
>         <mailto:internet-drafts@ietf.org>
>         <mailto:internet-drafts@ietf.__org
>         <mailto:internet-drafts@ietf.org>>>
>                  Date: Sun, Jul 14, 2013 at 4:59 PM
>                  Subject: New Version Notification for
>         draft-flach-tcpm-fec-00.txt
>                  To: Tobias Flach <flach@usc.edu <mailto:flach@usc.edu>
>         <mailto:flach@usc.edu <mailto:flach@usc.edu>>>, Nandita
>                  Dukkipati
>                  <nanditad@google.com <mailto:nanditad@google.com>
>         <mailto:nanditad@google.com <mailto:nanditad@google.com>>>, Yuchung
>                  Cheng <ycheng@google.com <mailto:ycheng@google.com>
>         <mailto:ycheng@google.com <mailto:ycheng@google.com>>>, Barath
>                  Raghavan <barath@google.com <mailto:barath@google.com>
>         <mailto:barath@google.com <mailto:barath@google.com>>>
>
>
>
>                  A new version of I-D, draft-flach-tcpm-fec-00.txt
>                  has been successfully submitted by Tobias Flach and
>         posted to the
>                  IETF repository.
>
>                  Filename:        draft-flach-tcpm-fec
>                  Revision:        00
>                  Title:           TCP Instant Recovery: Incorporating
>         Forward Error
>                  Correction in TCP
>                  Creation date:   2013-07-15
>                  Group:           Individual Submission
>                  Number of pages: 15
>                  URL:
>         http://www.ietf.org/internet-____drafts/draft-flach-tcpm-fec-____00.txt
>         <http://www.ietf.org/internet-__drafts/draft-flach-tcpm-fec-__00.txt>
>
>         <http://www.ietf.org/internet-__drafts/draft-flach-tcpm-fec-__00.txt
>         <http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt>>
>                  Status:
>         http://datatracker.ietf.org/____doc/draft-flach-tcpm-fec
>         <http://datatracker.ietf.org/__doc/draft-flach-tcpm-fec>
>                  <http://datatracker.ietf.org/__doc/draft-flach-tcpm-fec
>         <http://datatracker.ietf.org/doc/draft-flach-tcpm-fec>>
>                  Htmlized:
>         http://tools.ietf.org/html/____draft-flach-tcpm-fec-00
>         <http://tools.ietf.org/html/__draft-flach-tcpm-fec-00>
>
>                  <http://tools.ietf.org/html/__draft-flach-tcpm-fec-00
>         <http://tools.ietf.org/html/draft-flach-tcpm-fec-00>>
>
>
>                  Abstract:
>                       Ordinary TCP loss recovery takes at least one
>         round-trip
>                  time and as
>                       such can increase application-perceived latency,
>         especially
>                  for short
>                       flows such as Web transactions.  TCP Instant Recovery
>                  (TCP-IR) is an
>                       experimental algorithm that allows a receiving end to
>                  recover lost
>                       packets without retransmissions, thus potentially
>         saving at
>                  least one
>                       full round-trip time compared to standard TCP.  TCP-IR
>                  achieves this
>                       by judiciously injecting encoded data segments
>         within a TCP
>                  stream.
>                       This document describes the TCP-IR algorithm at
>         the sending and
>                       receiving ends, along with the required protocol
>         changes.
>
>
>
>
>                  The IETF Secretariat
>                  ___________________________________________________
>                  tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org> <mailto:tcpm@ietf.org
>         <mailto:tcpm@ietf.org>>
>         https://www.ietf.org/mailman/____listinfo/tcpm
>         <https://www.ietf.org/mailman/__listinfo/tcpm>
>                  <https://www.ietf.org/mailman/__listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>>
>
>              ___________________________________________________
>              tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org> <mailto:tcpm@ietf.org
>         <mailto:tcpm@ietf.org>>
>         https://www.ietf.org/mailman/____listinfo/tcpm
>         <https://www.ietf.org/mailman/__listinfo/tcpm>
>              <https://www.ietf.org/mailman/__listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>>
>
>
>

From touch@isi.edu  Wed Jul 17 09:37:42 2013
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Date: Wed, 17 Jul 2013 09:36:55 -0700
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Improved SYN cookies
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Have you addressed the impact on:
	- ISN reuse and overlap
	- other protocols that use ISN (TCP-AO comes to mind)

Joe

On 7/16/2013 10:42 AM, Andre Oppermann wrote:
> This is a FYI on a recent enhancement I did to FreeBSDs SYN cookie
> implementation to encode all necessary information in the ISN bits
> instead of relying on the mostly absent TCP timestamp.
>
> With it pretty much all previous shortcomings of SYN cookies are solved.
>
>   http://svnweb.freebsd.org/base?view=revision&revision=253210
>   http://svnweb.freebsd.org/base/head/sys/netinet/tcp_syncache.c?r1=253210&r2=253209&pathrev=253210
>
> Additional comments always welcome of course.  The comments in the code
> provide additional insight.
>
> Abstract:
>
> Improve SYN cookies by encoding the MSS, WSCALE (window scaling) and SACK
> information into the ISN (initial sequence number) without the additional
> use of timestamp bits and switching to the very fast and cryptographically
> strong SipHash-2-4 MAC hash algorithm to protect the SYN cookie against
> forgeries.
>
> The purpose of SYN cookies is to encode all necessary session state in
> the 32 bits of our initial sequence number to avoid storing any information
> locally in memory.  This is especially important when under heavy spoofed
> SYN attacks where we would either run out of memory or the syncache would
> fill with bogus connection attempts swamping out legitimate connections.
>
> The original SYN cookies method only stored an indexed MSS values in the
> cookie.  This isn't sufficient anymore and breaks down in the presence of
> WSCALE information which is only exchanged during SYN and SYN-ACK.  If we
> can't keep track of it then we may severely underestimate the available
> send or receive window. This is compounded with large windows whose size
> information on the TCP segment header is even lower numerically.  A number
> of years back SYN cookies were extended to store the additional state in
> the TCP timestamp fields, if available on a connection.  While timestamps
> are common among the BSD, Linux and other *nix systems Windows never
> enabled
> them by default and thus are not present for the vast majority of clients
> seen on the Internet.
>
> The common parameters used on TCP sessions have changed quite a bit since
> SYN cookies very invented some 17 years ago.  Today we have a lot more
> bandwidth available making the use window scaling almost mandatory.  Also
> SACK has become standard making recovering from packet loss much more
> efficient.
>
> This change moves all necessary information into the ISS removing the need
> for timestamps.  Both the MSS (16 bits) and send WSCALE (4 bits) are stored
> in 3 bit indexed form together with a single bit for SACK.  While this is
> significantly less than the original range, it is sufficient to encode all
> common values with minimal rounding.
>
> The MSS depends on the MTU of the path and with the dominance of ethernet
> the main value seen is around 1460 bytes.  Encapsulations for DSL lines
> and some other overheads reduce it by a few more bytes for many connections
> seen.  Rounding down to the next lower value in some cases isn't a problem
> as we send only slightly more packets for the same amount of data.
>
> The send WSCALE index is bit more tricky as rounding down under-estimates
> the available send space available towards the remote host, however a small
> number values dominate and are carefully selected again.
>
> The receive WSCALE isn't encoded at all but recalculated based on the local
> receive socket buffer size when a valid SYN cookie returns.  A listen
> socket
> buffer size is unlikely to change while active.
>
> The index values for MSS and WSCALE are selected for minimal rounding
> errors
> based on large traffic surveys.  These values have to be periodically
> validated against newer traffic surveys adjusting the arrays
> tcp_sc_msstab[]
> and tcp_sc_wstab[] if necessary.
>
> In addition the hash MAC to protect the SYN cookies is changed from MD5
> to SipHash-2-4, a much faster and cryptographically secure algorithm.
>

From nishida@sfc.wide.ad.jp  Wed Jul 17 09:54:43 2013
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] some questions related to PAWS
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--001a11c3c96e75b20604e1b7f2b1
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Hi Richard,

On Wed, Jul 17, 2013 at 6:06 AM, Scheffenegger, Richard <rs@netapp.com>wrote:

>
> Hi Yoshifumi,
>
> Well, that unacceptable segment SHOULD NOT trigger an ACK; the sender is
> free to send a segment for other reasons though (window probe, keepalive,
> new data).
>
>
> Perhaps that sentence should better read like:
>
> If a non-<RST> segment is received without a TSopt, a TCP SHOULD silently
> drop the segment.
>

Yes, I agree. I prefer this one.
--
Yoshifumi

--001a11c3c96e75b20604e1b7f2b1
Content-Type: text/html; charset=ISO-8859-1

<div dir="ltr">Hi Richard,<br><div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 17, 2013 at 6:06 AM, Scheffenegger, Richard <span dir="ltr">&lt;<a href="mailto:rs@netapp.com" target="_blank">rs@netapp.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
Hi Yoshifumi,<br>
<br>
Well, that unacceptable segment SHOULD NOT trigger an ACK; the sender is free to send a segment for other reasons though (window probe, keepalive, new data).<br>
<br>
<br>
Perhaps that sentence should better read like:<br>
<br>
If a non-&lt;RST&gt; segment is received without a TSopt, a TCP SHOULD silently drop the segment.<br></blockquote><div><br></div><div>Yes, I agree. I prefer this one. <br>--<br></div><div>Yoshifumi <br></div><br></div></div>
</div></div>

--001a11c3c96e75b20604e1b7f2b1--

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Date: Wed, 17 Jul 2013 10:54:14 -0700
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From: Nandita Dukkipati <nanditad@google.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Subject: Re: [tcpm] Tail loss probe talk [was Re: Draft agenda for IETF 87 in Berlin]
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--001a11c2e76a1be21b04e1b8c824
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> > 10:25
> > draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)
> > Yuchung Cheng
> > 10min
> TCP Loss probe (TLP) is not updated the draft because we weren't sure
> (yet) what to update, and the new data shows similar improvement that
> we've presented
>
> http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en//pubs/archive/41217.pdf
>
> Based on the recent WG and private discussions and the SIGCOMM
> reviews, people are not sure the performance improvement justifies
> this new feature, as the loss recovery is feature rich and complicated
> with RFC3517, RFC6937, RFC5682, RFC5827, etc.
>
> To simplify, I'd like to present TLP as a simple conceptual change to
> RTO recovery: On first timeout, retransmit the last packet instead of
> the first unacked packet, and don't change but reduce first timeout to
> 2 SRTT, as a "quick loss probe".
>
> As an alternative we've tried to only crank down the RTO but the
> result is not good, because reducing cwnd to 1 is catastrophic, and
> TCP today has to live sudden delay spikes due to buffer bloat and
> mobile delays. It's better to be very conservative before declaring
> the network is busted and use LW=1.
>
> We are hopeful to get more WG interests for TLP :)
>

That's right, it would be great to have a simplified TLP adopted as WG
item. FWIW, if anyone would like to give the TLP algorithm a spin in their
tests, here are the two commits:

Basic TLP algorithm (Section 2 of
http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01)
Commit:
http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=6ba8a3b19e764b6a65e4030ab0999be50c291e6c

Detecting recovered losses in TLP (Section 3 of I-D)
Commit:
http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=9b717a8d245075ffb8e95a2dfb4ee97ce4747457


Nandita

--001a11c2e76a1be21b04e1b8c824
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">

&gt; 10:25<br>
&gt; draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)<br>
&gt; Yuchung Cheng<br>
&gt; 10min<br>
TCP Loss probe (TLP) is not updated the draft because we weren&#39;t sure<b=
r>
(yet) what to update, and the new data shows similar improvement that<br>
we&#39;ve presented<br>
<a href=3D"http://static.googleusercontent.com/external_content/untrusted_d=
lcp/research.google.com/en//pubs/archive/41217.pdf" target=3D"_blank">http:=
//static.googleusercontent.com/external_content/untrusted_dlcp/research.goo=
gle.com/en//pubs/archive/41217.pdf</a><br>

<br>
Based on the recent WG and private discussions and the SIGCOMM<br>
reviews, people are not sure the performance improvement justifies<br>
this new feature, as the loss recovery is feature rich and complicated<br>
with RFC3517, RFC6937, RFC5682, RFC5827, etc.<br>
<br>
To simplify, I&#39;d like to present TLP as a simple conceptual change to<b=
r>
RTO recovery: On first timeout, retransmit the last packet instead of<br>
the first unacked packet, and don&#39;t change but reduce first timeout to<=
br>
2 SRTT, as a &quot;quick loss probe&quot;.<br>
<br>
As an alternative we&#39;ve tried to only crank down the RTO but the<br>
result is not good, because reducing cwnd to 1 is catastrophic, and<br>
TCP today has to live sudden delay spikes due to buffer bloat and<br>
mobile delays. It&#39;s better to be very conservative before declaring<br>
the network is busted and use LW=3D1.<br>
<br>
We are hopeful to get more WG interests for TLP :)<br></blockquote><div><br=
></div><div>That&#39;s right, it would be great to have a simplified TLP ad=
opted as WG item. FWIW, if anyone would like to give the TLP algorithm a sp=
in in their tests, here are the two commits:</div>
<div><br></div><div>Basic TLP algorithm (Section 2 of=A0<a href=3D"http://t=
ools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01">http://tools.iet=
f.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01</a>)</div><div>Commit:=A0=
<a href=3D"http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/c=
ommit/?id=3D6ba8a3b19e764b6a65e4030ab0999be50c291e6c">http://git.kernel.org=
/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=3D6ba8a3b19e764b6a65e4=
030ab0999be50c291e6c</a><br>
</div><div><br></div><div>Detecting recovered losses in TLP (Section 3 of I=
-D)</div><div>Commit:=A0<a href=3D"http://git.kernel.org/cgit/linux/kernel/=
git/torvalds/linux.git/commit/?id=3D9b717a8d245075ffb8e95a2dfb4ee97ce474745=
7">http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?i=
d=3D9b717a8d245075ffb8e95a2dfb4ee97ce4747457</a>=A0<br>
</div><div><br></div><div>Nandita</div></div></div></div>

--001a11c2e76a1be21b04e1b8c824--

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From: Yuchung Cheng <ycheng@google.com>
Date: Wed, 17 Jul 2013 11:13:24 -0700
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Cc: "tcpm-chairs@tools.ietf.org" <tcpm-chairs@tools.ietf.org>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] Draft agenda for IETF 87 in Berlin
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On Wed, Jul 17, 2013 at 2:01 AM, Scharf, Michael (Michael)
<michael.scharf@alcatel-lucent.com> wrote:
> Hi all,
>
> The first draft of the TCPM agenda can be found at https://datatracker.ie=
tf.org/meeting/87/agenda/tcpm/ and is also copied below.
>
> We tried to take into account all requests, but as a result, the agenda i=
s pretty full. This implies:
>
> - The slots are *total* time. Authors, please reserve 50% of your time fo=
r questions and discussion.
>
> - We got time slot requests for drafts that were not be updated since the=
 last meeting, typically promising new results for the presentation in the =
meeting. Mailing list discussion in the next week would be *highly* welcome=
 in order to justify a time slot on our full agenda.
>
> Please let us know if you have any comments, suggestions, or if we missed=
 something.
>
> Happy agenda bashing!
>
> Michael, Pasi, Yoshifumi
>
>
>
> *************************************************************************=
************
>
> TCPM Agenda
> IETF 87 in Berlin, Germany
> Tuesday, July 30, 2013, 09:00-11:30
> ***********************************
>
> WG status
> ---------
>
> 09:00
> TCPM status
> Chairs
> 5min
>
>
> WG items
> --------
>
> 09:05
> draft-ietf-tcpm-1323bis-14
> Richard Scheffenegger
> 20min
>
> 09:25
> draft-ietf-tcpm-fastopen-04
> Jerry Chu / Yuchung Cheng
> 15mins
>
> 09:40
> draft-ietf-tcpm-newcwv-02
> Rafaello Secchi / Gorry Fairhurst
> 20min
>
> 10:00
> draft-ietf-tcpm-rtorestart-00 (no new draft)
> Anna Brunstrom
> 10min
>
> 10:10
> draft-ietf-tcpm-accecn-reqs-02
> Mirja Kuelewind
> 5min
>
>
> Non-WG items
> ------------
>
> 10:15
> draft-zimmermann-tcpm-tcp-rfc4614bis-02
> Alexander Zimmermann
> 10min
>
> 10:25
> draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)
> Yuchung Cheng
> 10min
>
> 10:35
> draft-flach-tcpm-fec-00
> Tobias Flach
> 15min
>
> 10:50
> draft-gont-tcpm-tcp-seq-validation-00 (no new draft)
> Fernando Gont
> 10min
>
> 11:00
> draft-kuehlewind-tcpm-ecn-fallback-00
> draft-kuehlewind-tcpm-accurate-ecn-02
> Mirja Kuehlewind
> 15min
>
>
> If time permits
> ---------------
>
> Wired PRR effects
> Mirja Kuehlewind
> 5min
>
> Dealing with sequence-number randomizing firewalls
> Christoph Paasch
> 5min
>
> packetdrill: Scriptable Network Stack Testing, from Sockets to Packets
> Yuchung Cheng
> 5min
I will present packetdrill in iccrg instead. Please drop my talk slot. Than=
ks.


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

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From: Yuchung Cheng <ycheng@google.com>
Date: Wed, 17 Jul 2013 17:05:47 -0700
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To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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On Wed, Jul 17, 2013 at 9:32 AM, Joe Touch <touch@isi.edu> wrote:
>
>
> On 7/17/2013 7:03 AM, Tobias Flach wrote:
>>
>> I agree that if we introduce a significant amount of redundant
>> transmissions, these packets need to be accounted for by the congestion
>> window. However, the way we designed our algorithm, the amount of
>> redundancy is relatively small (usually 16 data packets are encoded by 1
>> XOR packet), which is why in our prototype the XOR packets are
>> transmitted outside of the window.
>
>
> There's no rule in TCP congestion control that says "oh, it's OK to send one
> more packet if that's a small increase vs. what CWND says".
>
> I appreciate that this extension considers that extra packet "not new
> information", but to the rest of the network it's one extra packet.
Good point. We'll reconsider that.

>
>
>> And you are right that these schemes in general are pretty old, but to
>> our knowledge no one has actually thought about the protocol level
>> details of incorporating FEC into TCP before,
>
>
> That's incorrect.
I think what Tobias meant was there were no real implementation of FEC
in TCP itself.
There are many designs of similar sort (e.g. boston), but we are not
aware of any publicly accessible real stack

>
>
>> including all the trouble middleboxes are causing.
>
>
> I haven't seen anything in TCP Boston or other similar approaches that a
> middlebox would interfere with.
This calls for more information to compare other works in the mtg (and
in the draft).

>
> Joe
>
>
>>
>> - Tobias
>>
>> On Mon, Jul 15, 2013 at 9:32 PM, Joe Touch <touch@isi.edu
>> <mailto:touch@isi.edu>> wrote:
>>
>>     Well, FEC-style redundant transmission is fine so long as you
>>     subtract those XOR packets from your congestion window. I.e., either
>>     you send 10 data packets, or 5 data and 5 XOR.
>>
>>     If you're just pumping data into the net in the hope that something
>>     makes it, we've seen that before too - Roadrunner tried it 15 years
>>     ago. We all know it works, but I hope we all know why it's not a
>>     good idea.
>>
>>     BTW, these sorts of tricks are a LOT older than 1-2 years.
>>
>>     Joe
>>
>>
>>     On 7/15/2013 5:49 PM, Tobias Flach wrote:
>>
>>         There are indeed far more elegant and sophisticated coding
>>         schemes out
>>         there than the XOR approach we used. TCP Boston as well as other
>>         published approaches like "Network Coded TCP"
>>         (http://arxiv.org/abs/1212.__2291
>>         <http://arxiv.org/abs/1212.2291>) are capable to mask losses
>>
>>         better than
>>         XOR, but XOR has the advantage that it is easy to implement and
>>         results
>>         in a relatively low computational overhead. In addition, we do
>>         not limit
>>         our proposal to XOR, but rather use it to demonstrate how TCP-IR
>>         works
>>         in general, and we can easily extend the module to support other
>>         coding
>>         schemes in the future.
>>
>>         Regarding the second question: Regular payloads and parity data
>>         are not
>>         mixed in the same packet. Parity data is sent separately which
>>         enables
>>         immediate recovery if only one packet in the encoding range is
>>         lost. As
>>         said, more sophisticated coding schemes might do better here,
>>         but our
>>         experiments have shown that even simple XOR can improve latency
>>         a lot -
>>         especially for short flows.
>>
>>         - Tobias
>>
>>
>>         On Mon, Jul 15, 2013 at 5:58 PM, Joe Touch <touch@isi.edu
>>         <mailto:touch@isi.edu>
>>         <mailto:touch@isi.edu <mailto:touch@isi.edu>>> wrote:
>>
>>              You might take a look at TCP Boston, which is a much more
>>         elegant
>>              FEC than just parity.
>>
>>              But what use is parity? If a packet is corrupted, most link
>>         layers
>>              will drop it anyway (esp. because you don't even know if
>>         the packet
>>              header is correct, thus whether you're sending it to the
>> right
>>              connection).
>>
>>              Joe
>>
>>
>>              On 7/15/2013 2:49 PM, Yuchung Cheng wrote:
>>
>>                  Hi
>>
>>                  We are proposing an outrageous idea to send parity data
>>         in TCP
>>                  packet to promote sub 1-RTT loss recovery. This
>>         requires a separate
>>                  sequence space which we aren't quite sure how to do
>>         this yet without
>>                  upsetting middle-boxes. More data can be found in our
>>         new paper
>>                  "Reducing Web Latency: the Virtue of Gentle Aggression"
>>         sigcomm 2013
>>
>> http://static.__googleusercont__ent.com/__external_content/__untrusted___dlcp/research.__google.com/en/__us/pubs/__archive/41217.pdf
>>
>> <http://googleusercontent.com/__external_content/untrusted___dlcp/research.google.com/en/__us/pubs/archive/41217.pdf>
>>
>>
>>
>>
>> <http://static.__googleusercontent.com/__external_content/untrusted___dlcp/research.google.com/en/__us/pubs/archive/41217.pdf
>>
>> <http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en/us/pubs/archive/41217.pdf>>
>>
>>                  Comments are very welcome. Linux patches release are
>>         under way.
>>
>>
>>                  ---------- Forwarded message ----------
>>                  From:  <internet-drafts@ietf.org
>>         <mailto:internet-drafts@ietf.org>
>>         <mailto:internet-drafts@ietf.__org
>>         <mailto:internet-drafts@ietf.org>>>
>>                  Date: Sun, Jul 14, 2013 at 4:59 PM
>>                  Subject: New Version Notification for
>>         draft-flach-tcpm-fec-00.txt
>>                  To: Tobias Flach <flach@usc.edu <mailto:flach@usc.edu>
>>         <mailto:flach@usc.edu <mailto:flach@usc.edu>>>, Nandita
>>                  Dukkipati
>>                  <nanditad@google.com <mailto:nanditad@google.com>
>>         <mailto:nanditad@google.com <mailto:nanditad@google.com>>>,
>> Yuchung
>>                  Cheng <ycheng@google.com <mailto:ycheng@google.com>
>>         <mailto:ycheng@google.com <mailto:ycheng@google.com>>>, Barath
>>                  Raghavan <barath@google.com <mailto:barath@google.com>
>>         <mailto:barath@google.com <mailto:barath@google.com>>>
>>
>>
>>
>>
>>                  A new version of I-D, draft-flach-tcpm-fec-00.txt
>>                  has been successfully submitted by Tobias Flach and
>>         posted to the
>>                  IETF repository.
>>
>>                  Filename:        draft-flach-tcpm-fec
>>                  Revision:        00
>>                  Title:           TCP Instant Recovery: Incorporating
>>         Forward Error
>>                  Correction in TCP
>>                  Creation date:   2013-07-15
>>                  Group:           Individual Submission
>>                  Number of pages: 15
>>                  URL:
>>
>> http://www.ietf.org/internet-____drafts/draft-flach-tcpm-fec-____00.txt
>>
>> <http://www.ietf.org/internet-__drafts/draft-flach-tcpm-fec-__00.txt>
>>
>>
>>
>> <http://www.ietf.org/internet-__drafts/draft-flach-tcpm-fec-__00.txt
>>         <http://www.ietf.org/internet-drafts/draft-flach-tcpm-fec-00.txt>>
>>                  Status:
>>         http://datatracker.ietf.org/____doc/draft-flach-tcpm-fec
>>         <http://datatracker.ietf.org/__doc/draft-flach-tcpm-fec>
>>
>>                  <http://datatracker.ietf.org/__doc/draft-flach-tcpm-fec
>>         <http://datatracker.ietf.org/doc/draft-flach-tcpm-fec>>
>>                  Htmlized:
>>         http://tools.ietf.org/html/____draft-flach-tcpm-fec-00
>>         <http://tools.ietf.org/html/__draft-flach-tcpm-fec-00>
>>
>>
>>                  <http://tools.ietf.org/html/__draft-flach-tcpm-fec-00
>>         <http://tools.ietf.org/html/draft-flach-tcpm-fec-00>>
>>
>>
>>                  Abstract:
>>                       Ordinary TCP loss recovery takes at least one
>>         round-trip
>>                  time and as
>>                       such can increase application-perceived latency,
>>         especially
>>                  for short
>>                       flows such as Web transactions.  TCP Instant
>> Recovery
>>                  (TCP-IR) is an
>>                       experimental algorithm that allows a receiving end
>> to
>>                  recover lost
>>                       packets without retransmissions, thus potentially
>>         saving at
>>                  least one
>>                       full round-trip time compared to standard TCP.
>> TCP-IR
>>                  achieves this
>>                       by judiciously injecting encoded data segments
>>         within a TCP
>>                  stream.
>>                       This document describes the TCP-IR algorithm at
>>         the sending and
>>                       receiving ends, along with the required protocol
>>         changes.
>>
>>
>>
>>
>>                  The IETF Secretariat
>>                  ___________________________________________________
>>                  tcpm mailing list
>>         tcpm@ietf.org <mailto:tcpm@ietf.org> <mailto:tcpm@ietf.org
>>         <mailto:tcpm@ietf.org>>
>>         https://www.ietf.org/mailman/____listinfo/tcpm
>>         <https://www.ietf.org/mailman/__listinfo/tcpm>
>>                  <https://www.ietf.org/mailman/__listinfo/tcpm
>>         <https://www.ietf.org/mailman/listinfo/tcpm>>
>>
>>              ___________________________________________________
>>              tcpm mailing list
>>         tcpm@ietf.org <mailto:tcpm@ietf.org> <mailto:tcpm@ietf.org
>>         <mailto:tcpm@ietf.org>>
>>         https://www.ietf.org/mailman/____listinfo/tcpm
>>         <https://www.ietf.org/mailman/__listinfo/tcpm>
>>              <https://www.ietf.org/mailman/__listinfo/tcpm
>>         <https://www.ietf.org/mailman/listinfo/tcpm>>
>>
>>
>>
>

From gorry@erg.abdn.ac.uk  Wed Jul 17 23:38:46 2013
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Date: Thu, 18 Jul 2013 07:38:40 +0100
From: gorry@erg.abdn.ac.uk
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Cc: raffaello@erg.abdn.ac.uk, "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Hi Michael,

I feel I should start by making sure we are on the same page with respect
to "R".
In CWV, we consider apps that do not always fill cwnd to probe for
capacity, hence there can be cases where a TCP flow can change its
behaviour. If we allow a cwnd that corresponds to much more than the
average sending rate (which we do), then I suggest we need to
significantly reduce the cwnd when there is a significant "mistake" in
sending more than the current path capacity. This is the role of "R".

Originally we thought "R" should be computed from SACK Information. That
is what we had simulated before, and I think follows part of Mark Allman's
JS work - but we now think this can be troublesome.   Instead, we now use
a method that counts retransmitted segments during the recovery phase for
the first window of data. We'd be very happy to talk about how we
currently do this, especially if you have experience (Raffaello can say
more, in cc).

See in-line for the second half of the answer.

Gorry


> Hi Gorry,
>
> Regarding Section 4.4.1. Response to congestion in the nonvalidated phase:
>
>>   At the end of the recovery phase, the TCP sender MUST reset the cwnd
>>   using the method below:
>>           cwnd = ((LossFlightSize - R)/2).
>>
>>   Where, R is the volume of data that was retransmitted during the
>>   recovery phase.  This follows the method proposed for Jump Start
>>   [Liu07].  The inclusion of the term R makes this adjustment more
>>   conservative than standard TCP.  (This is required, since the sender
>>   may have sent more segments than a Standard TCP sender would have
>>   done.  The additional reduction is beneficial when the LossFlightSize
>>   significantly overshoots the available path capacity incurring
>>   significant loss, for instance an intense traffic burst following a
>>   non-validated period.)
>
> As already mentioned in
> http://www.ietf.org/mail-archive/web/tcpm/current/msg08015.html, when
> implementing [Liu07] in Linux some years ago, I ran into situations where
> R was larger than LossFlightSize (possibly due to multiple retransmissions
> of the same segment). The equation cwnd = ((LossFlightSize - R)/2) could
> thus result in a negative cwnd. That issue might partly depend on how to
> exactly R is measured, but I think that this equation requires further
> analysis and can't be mandated by a MUST.
>
> Even if (LossFlightSize - R) is positive, I think that M. Allman's offlist
> comment about performance loss in loss of a short burst (Section 9.1) has
> to be addressed.
>
> For instance, I wonder why cwnd at the end of recovery phase could not be
> reset to the value at the beginning, i.e.,
>
>  cwnd = Min(cwnd/2,Max(pipeACK,LossFlightSize))
>
> If that value is a "safe" value for loss recovery, why is it not save to
> use it after loss recovery?
>
I think this formula can't be used AFTER recovery.

This is why we chose to use this during recovery:

- At the start of loss recovery, the sender needs to react at least the
same as normal TCP, by halving cwnd. But: cwnd/2 is halving something that
was not validated, and did not provide any indication of the actual data
sent - in the NVP, the cwnd is only an upper limit.

The second term above reflects recently used capacity;

-  pipeACK is intended as a measure of capacity used in the recent past,
but it can be quite different to (D-R) in cases when the traffic pattern
changes inducing congestion. It helps though in not under-estimating
capacity.

- LossFlightSize (D) reflects the pattern that induced the congestion -
this is already in-flight during loss recovery.

The value of cwnd at the end of the loss recovery should not be dependent
on an initial cwnd that was not itself validated.

> Thanks
>
> Michael
>
>
>
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
>> Behalf Of gorry@erg.abdn.ac.uk
>> Sent: Tuesday, July 02, 2013 9:20 AM
>> To: tcpm@ietf.org
>> Subject: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
>>
>> We've just uploaded a new version of our draft. The
>> congestion window management algorithm has not changed since
>> -00, but in -01 and -02 drafts we introduce a more robust way
>> to measure the pipeACK that attempts to account for the wide
>> variety of interactive TCP application behaviour.
>> This uses a sampling method to sample pipeACK over the last second.
>>
>> Clearly the cwnd poorly reflects the capacity in the
>> non-validated phase and I think it would be wrong to use this
>> for TCP control block sharing.
>> We therefore also  introduce text that proposes that TCB
>> sharing should use the pipeACK in place of cwnd when a TCP
>> sender is in the Nonvalidated phase.  We think this value
>> better reflects the capacity that the flow has utilised in
>> the network path - but would welcome more thought on this.
>> Thoughts on this would be most welcome.
>>
>> Gorry
>>
>> >
>> > A New Internet-Draft is available from the on-line Internet-Drafts
>> > directories.
>> >  This draft is a work item of the TCP Maintenance and Minor
>> Extensions
>> > Working Group of the IETF.
>> >
>> > 	Title           : Updating TCP to support Rate-Limited Traffic
>> > 	Author(s)       : Godred Fairhurst
>> >                           Arjuna Sathiaseelan
>> >                           Raffaello Secchi
>> > 	Filename        : draft-ietf-tcpm-newcwv-02.txt
>> > 	Pages           : 19
>> > 	Date            : 2013-07-01
>> >
>> > Abstract:
>> >    This document proposes an update to RFC 5681 to address
>> issues that
>> >    arise when TCP is used to support traffic that exhibits
>> periods where
>> >    the sending rate is limited by the application rather than the
>> >    congestion window.  It updates TCP to allow a TCP sender
>> to restart
>> >    quickly following either an idle or rate-limited interval.  This
>> >    method is expected to benefit applications that send rate-limited
>> >    traffic using TCP, while also providing an appropriate
>> response if
>> >    congestion is experienced.
>> >
>> >    It also evaluates the Experimental specification of TCP
>> Congestion
>> >    Window Validation, CWV, defined in RFC 2861, and
>> concludes that RFC
>> >    2861 sought to address important issues, but failed to deliver a
>> >    widely used solution.  This document therefore
>> recommends that the
>> >    status of RFC 2861 is moved from Experimental to
>> Historic, and that
>> >    it is replaced by the current specification.
>> >
>> >    NOTE: The standards status of this WG document is under
>> review for
>> >    consideration as either Experimental (EXP) or Proposed
>> Standard (PS).
>> >    This decision will be made later as the document is finalised.
>> >
>> >
>> > The IETF datatracker status page for this draft is:
>> > https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>> >
>> > There's also a htmlized version available at:
>> > http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>> >
>> > A diff from the previous version is available at:
>> > http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>> >
>> >
>> > 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
>> >
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>



From michael.scharf@alcatel-lucent.com  Thu Jul 18 07:12:57 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Hi Gorry,=20

> -----Original Message-----
> From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]=20
> Sent: Thursday, July 18, 2013 8:39 AM
> To: Scharf, Michael (Michael)
> Cc: gorry@erg.abdn.ac.uk; raffaello@erg.abdn.ac.uk; tcpm@ietf.org
> Subject: RE: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
>=20
> Hi Michael,
>=20
> I feel I should start by making sure we are on the same page=20
> with respect to "R".
> In CWV, we consider apps that do not always fill cwnd to=20
> probe for capacity, hence there can be cases where a TCP flow=20
> can change its behaviour. If we allow a cwnd that corresponds=20
> to much more than the average sending rate (which we do),=20
> then I suggest we need to significantly reduce the cwnd when=20
> there is a significant "mistake" in sending more than the=20
> current path capacity. This is the role of "R".
>
> Originally we thought "R" should be computed from SACK=20
> Information. That is what we had simulated before, and I=20
> think follows part of Mark Allman's
> JS work - but we now think this can be troublesome.  =20
> Instead, we now use
> a method that counts retransmitted segments during the=20
> recovery phase for the first window of data. We'd be very=20
> happy to talk about how we currently do this, especially if=20
> you have experience (Raffaello can say more, in cc).

My main concern is that a substraction of two counters can be negative or 0=
, and this is not addressed well, as far as I can tell.

Unless it is clear by definition of "R" that "R" is *always* larger than Lo=
ssFlightSize (more precisely, that the resulting cwnd is at least the RW=3D=
1 MSS), imho one has to deal with the case that the term cwnd =3D ((LossFli=
ghtSize - R)/2) can return a negative result or 0.

I don't fully understand what "retransmitted segments during the recovery p=
hase for the first window of data" implies exactly, i.e., whether the cwnd =
could be negative or not.


> See in-line for the second half of the answer.
>=20
> Gorry
>=20
>=20
> > Hi Gorry,
> >
> > Regarding Section 4.4.1. Response to congestion in the=20
> nonvalidated phase:
> >
> >>   At the end of the recovery phase, the TCP sender MUST=20
> reset the cwnd
> >>   using the method below:
> >>           cwnd =3D ((LossFlightSize - R)/2).
> >>
> >>   Where, R is the volume of data that was retransmitted during the
> >>   recovery phase.  This follows the method proposed for Jump Start
> >>   [Liu07].  The inclusion of the term R makes this adjustment more
> >>   conservative than standard TCP.  (This is required,=20
> since the sender
> >>   may have sent more segments than a Standard TCP sender would have
> >>   done.  The additional reduction is beneficial when the=20
> LossFlightSize
> >>   significantly overshoots the available path capacity incurring
> >>   significant loss, for instance an intense traffic burst=20
> following a
> >>   non-validated period.)
> >
> > As already mentioned in
> >=20
> http://www.ietf.org/mail-archive/web/tcpm/current/msg08015.html, when=20
> > implementing [Liu07] in Linux some years ago, I ran into situations=20
> > where R was larger than LossFlightSize (possibly due to multiple=20
> > retransmissions of the same segment). The equation cwnd =3D=20
> > ((LossFlightSize - R)/2) could thus result in a negative cwnd. That=20
> > issue might partly depend on how to exactly R is measured,=20
> but I think=20
> > that this equation requires further analysis and can't be=20
> mandated by a MUST.
> >
> > Even if (LossFlightSize - R) is positive, I think that M. Allman's=20
> > offlist comment about performance loss in loss of a short burst=20
> > (Section 9.1) has to be addressed.
> >
> > For instance, I wonder why cwnd at the end of recovery=20
> phase could not=20
> > be reset to the value at the beginning, i.e.,
> >
> >  cwnd =3D Min(cwnd/2,Max(pipeACK,LossFlightSize))
> >
> > If that value is a "safe" value for loss recovery, why is=20
> it not save=20
> > to use it after loss recovery?
> >
> I think this formula can't be used AFTER recovery.
>=20
> This is why we chose to use this during recovery:
>=20
> - At the start of loss recovery, the sender needs to react at=20
> least the same as normal TCP, by halving cwnd. But: cwnd/2 is=20
> halving something that was not validated, and did not provide=20
> any indication of the actual data sent - in the NVP, the cwnd=20
> is only an upper limit.
>=20
> The second term above reflects recently used capacity;
>=20
> -  pipeACK is intended as a measure of capacity used in the=20
> recent past, but it can be quite different to (D-R) in cases=20
> when the traffic pattern changes inducing congestion. It=20
> helps though in not under-estimating capacity.
>
> - LossFlightSize (D) reflects the pattern that induced the=20
> congestion - this is already in-flight during loss recovery.
>=20
> The value of cwnd at the end of the loss recovery should not=20
> be dependent on an initial cwnd that was not itself validated.

OK, this seems to make sense. But, still, why to calculate cwnd only based =
on LossFlightSize and not on pipeACK?

As far as I understand Mark's comment in Section 9.1, his concern is that p=
ipeACK might be a more appropriate estimation than FlightSize in some cases=
. Maybe I miss something basic here, but this is not about "R" - it is abou=
t pipeACK vs. Flightsize.
=20
For instance, something like this (I have not fully thought of it - maybe I=
 am too naive here?):

  cwnd =3D Max(RW, Max(pipeACK-R, LossFlightSize-R)/2 )

Michael




> > Thanks
> >
> > Michael
> >
> >
> >
> >> -----Original Message-----
> >> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org]=20
> On Behalf=20
> >> Of gorry@erg.abdn.ac.uk
> >> Sent: Tuesday, July 02, 2013 9:20 AM
> >> To: tcpm@ietf.org
> >> Subject: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
> >>
> >> We've just uploaded a new version of our draft. The=20
> congestion window=20
> >> management algorithm has not changed since -00, but in -01 and -02=20
> >> drafts we introduce a more robust way to measure the pipeACK that=20
> >> attempts to account for the wide variety of interactive TCP=20
> >> application behaviour.
> >> This uses a sampling method to sample pipeACK over the last second.
> >>
> >> Clearly the cwnd poorly reflects the capacity in the non-validated=20
> >> phase and I think it would be wrong to use this for TCP=20
> control block=20
> >> sharing.
> >> We therefore also  introduce text that proposes that TCB sharing=20
> >> should use the pipeACK in place of cwnd when a TCP sender=20
> is in the=20
> >> Nonvalidated phase.  We think this value better reflects=20
> the capacity=20
> >> that the flow has utilised in the network path - but would welcome=20
> >> more thought on this.
> >> Thoughts on this would be most welcome.
> >>
> >> Gorry
> >>
> >> >
> >> > A New Internet-Draft is available from the on-line=20
> Internet-Drafts=20
> >> > directories.
> >> >  This draft is a work item of the TCP Maintenance and Minor
> >> Extensions
> >> > Working Group of the IETF.
> >> >
> >> > 	Title           : Updating TCP to support Rate-Limited Traffic
> >> > 	Author(s)       : Godred Fairhurst
> >> >                           Arjuna Sathiaseelan
> >> >                           Raffaello Secchi
> >> > 	Filename        : draft-ietf-tcpm-newcwv-02.txt
> >> > 	Pages           : 19
> >> > 	Date            : 2013-07-01
> >> >
> >> > Abstract:
> >> >    This document proposes an update to RFC 5681 to address
> >> issues that
> >> >    arise when TCP is used to support traffic that exhibits
> >> periods where
> >> >    the sending rate is limited by the application rather than the
> >> >    congestion window.  It updates TCP to allow a TCP sender
> >> to restart
> >> >    quickly following either an idle or rate-limited=20
> interval.  This
> >> >    method is expected to benefit applications that send=20
> rate-limited
> >> >    traffic using TCP, while also providing an appropriate
> >> response if
> >> >    congestion is experienced.
> >> >
> >> >    It also evaluates the Experimental specification of TCP
> >> Congestion
> >> >    Window Validation, CWV, defined in RFC 2861, and
> >> concludes that RFC
> >> >    2861 sought to address important issues, but failed=20
> to deliver a
> >> >    widely used solution.  This document therefore
> >> recommends that the
> >> >    status of RFC 2861 is moved from Experimental to
> >> Historic, and that
> >> >    it is replaced by the current specification.
> >> >
> >> >    NOTE: The standards status of this WG document is under
> >> review for
> >> >    consideration as either Experimental (EXP) or Proposed
> >> Standard (PS).
> >> >    This decision will be made later as the document is finalised.
> >> >
> >> >
> >> > The IETF datatracker status page for this draft is:
> >> > https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
> >> >
> >> > There's also a htmlized version available at:
> >> > http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
> >> >
> >> > A diff from the previous version is available at:
> >> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-newcwv-02
> >> >
> >> >
> >> > 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
> >> >
> >>
> >>
> >> _______________________________________________
> >> tcpm mailing list
> >> tcpm@ietf.org
> >> https://www.ietf.org/mailman/listinfo/tcpm
> >>
>=20
>=20
> =

From rs@netapp.com  Thu Jul 18 13:15:10 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Nandita Dukkipati <nanditad@google.com>
Thread-Topic: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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Hi Nandita,

> Thanks for your comments, Richard. do you mind if I cc the list?

Not really, if you think they are valuable :)


>> Why use a ECN-like handshake for the recovery signal? This=20
>> limits your reactions to one bit per RTT, but a RTT could=20
>> cover many FEC "episodes" (consecutive segments all covered=20
>> by a single FEC).
>=20
> The analogy to ECN's mechanism is only in the context of=20
> sender reducing congestion window just once per RTT, for=20
> segments recovered silently at the receiver.

I understand. My context was in the area of adaptive FEC episodes... Perhap=
s an example helps:

Say the window is 12(+3) segments [as I believe Joe has a very valid point =
with the FEC segments need to be accounted for against CWND], and the FEC e=
pisode is 4 segments. With a single handshake-like mechanism, the sender wi=
ll not know if each of the three in-flight FEC episodes had to be recovered=
, or only the first... Of course, the CC reaction will be identical (just l=
ike having experienced one, two or three losses in the same window - all le=
ading to a single CC reaction). But for an adaptive FEC episode, you don't =
have a good signal to decrease, or increase the FEC episode length. I.e. if=
 only one of the three FEC episodes had to be recovered, a three-time large=
r FEC episode would be just as good (or a slight increase in the current on=
e would have done), reducing the relative overhead from, say, 4+1 to 6+1 or=
 8+1; in the opposite case, where each episode had to be recovered, a decre=
ase in the episode length would be advisable to maintain the FEC properties=
...

In that scenario, your current handshake mechanism seems to fail to deliver=
 a good signal to differentiate between these cases.=20

Of course, if the FEC episode length (do you have a name for that) is fixed=
 (socket option) and not dynamic (even though the draft seems to allow for =
that), you wouldn't need a more fine-grained feedback...




Richard Scheffenegger


From: Nandita Dukkipati [mailto:nanditad@google.com]=20
Sent: Mittwoch, 17. Juli 2013 21:13
To: Scheffenegger, Richard
Cc: Tobias Flach
Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-=
00.txt

Thanks for your comments, Richard. do you mind if I cc the list?

> On Tue, Jul 16, 2013 at 6:40 AM, Scheffenegger, Richard <rs@netapp.com> w=
rote:
> Hi Tobias, Nandita,
=A0
> Two observations:
>=A0
> Why use a ECN-like handshake for the recovery signal? This limits your re=
actions to one bit per RTT, but a RTT could cover many FEC "episodes" (cons=
ecutive segments all covered by a single FEC).

The analogy to ECN's mechanism is only in the context of sender reducing co=
ngestion window just once per RTT, for segments recovered silently at the r=
eceiver.
=A0
=A0
> I'm wondering, if a unilateral feedback like we propose in AccECN wouldn'=
t yield more informantion, ie. to let the sender reduce the FEC episodes=A0=
 before an actual R_FAIL occurs (covering less than 16 segments per FEC). Y=
ou'd need a counter of some width for this, but as it would count up by one=
 only once per FEC episode (1/1 to 1/16 segments), I would think a small nu=
mber of bits (4-6) would be sufficient.

What you are proposing makes sense to me. While our focus for the first cut=
 was to come up with the simplest design that achieves most of the performa=
nce benefits, extensions such as what you propose should certainly be on th=
e table. I think the key is to weigh in the complexity versus the benefit..=
.

=A0
> Second, you state that FEC has to be disabled when MSS changes.. But you =
don't explicitly state, how to do this (apparently, the FEC option would st=
ill need to be sent, but no FEC packets anymore). But is this really necess=
ary on shrinking MSS?=A0 I can see, that supporting increases in MSS would =
be difficult to support (but MSS shouldn't be growing after 3WHS), but for =
a shrinking MSS the effect would be the same as sending a couple non-MSS si=
zed packets, that are FEC-covered. AS you cann't assume that all (but the l=
ast) segment is MSS sized, and padding the payload with zero up to MSS size=
 is mentioned for smaller segments, why not have this on shrinking MSS too =
- keep the padding, but expand this to FEC segments as well. Of course, thi=
s is easy with XOR encoding, other encoding schemes might not work so well.

Nice! I like this idea of handling the shrinking MSS - should include in th=
e next draft version.
=A0
=A0
> Finally, I have mentioned this earlier, we (NetApp) might want to have ou=
r RAID-DP scheme in there as well. It does have a few drawbacks operational=
ly and functionally (larger FEC episode - longer latency until recovery; fu=
nctionally - only prime-1 FEC episodes can natively be supported (formally,=
 there can be "unsent" "padding" segments, to arrive at FEC episodes of len=
gth prime-1 though).=20
> =A0
> In that light, it might be interesting to dynamically switch between enco=
dings (non-interleaved xor, interleaved xor, double-xor), just as the sende=
r is free to choose the FEC episode length dynamically.

The way it stands now is the encoding scheme negotiation is done in 3WHS, a=
nd this is stuck with for the rest of the connection. But if the encoding t=
ype is carried in the option on every segment, I think it's feasible to swi=
tch between encodings during the connection course.
=A0
=A0
> BTW: are overlapping FECs (up to the limit of 16 segments) allowed? Ie, s=
end one FEC covering 1-4 as the 5th packet; 1-8 as the 10th packet, 1-16 as=
 the 20th packet? Just asking because I have recently read a paper showing =
such overlapped FEC schemes to subsequently cover more and more data.

Yes, the current design allows this. Although we haven't actually tried thi=
s in our implementation.

Nandita
=A0
=A0


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Tobias Flach <flach@usc.edu>, "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
Thread-Topic: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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Hi Tobias,

>>   1.  Periodically in every round-trip time, a TCP-IR sender places the
>>   XOR of newly transmitted segments into a single MSS-length checksum
>>   packet.  The XOR is only computed for new segments not previously
>>   included in checksums.

> Is it correct that there are more than one XOR packets per
> RTT if there are more than 8 or 16 outstanding packets,
> i.e., CWND>8 or 16?
>
> That is correct. If there are more than 8 (or 16) unencoded
> packets in the queue when the timer fires, multiple XOR packets
> are generated. Additionally, if new packets are transmitted after
> the timer fired, the timer is armed again (set to RTT/4), so for
> a sender transmitting data continuously the timer would fire up
> to four times per RTT.


So, you are ONLY sending with the RTT/4 timer? Wouldn't it be more useful t=
o send the FEC segments immediately following the segments that are covered=
 by it?

1 2 3 4 Xa 5 6 7 [pause] Xb

Vs=20

1 2 3 4 5 6 7 [pause] Xa Xb

(FEC episode of 4; as the client only captures up to 16 segment - i assume =
of the first episode that cann't be recovered? - for larger windows, where =
a whole bunch of XOR segment will get sent out at the RTT/4 timer, most of =
them will be pretty useless, even if they could theoretically could have re=
covered some loss... Or am I getting something wrong here?

Best regards,


Richard Scheffenegger



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From: Tobias Flach <flach@usc.edu>
To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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--089e0102fa08642ac204e1cfde07
Content-Type: text/plain; charset=ISO-8859-1

Right now FEC packets are only transmitted when the timer fires but your
proposal sounds interesting. As long as there is more outstanding data
which would be transmitted right after an FEC packet, it might be okay to
transmit the FEC packet immediately. And we would still delay if there is
no more data to send, e.g.

1 2 3 4 Xa 5 6 7 8 [pause] Xb

The only disadvantage I see is that having FEC packets in the middle of the
burst, is that the burst size increases, which might be okay.

Regarding your second question, let's use the example above. If we set the
maximum encoding range to 4, the receiver only needs to keep the last 3
in-order packets buffered.

Examples (* denotes loss of the packet):
* 2 3 4 Xa --> 1 can be recovered, because 2 - 4 remained in the OOO queue
1 2 3 * Xa --> 4 can be recovered, because 1 - 3 remained in the in-order
queue (because they are the last 3 in-order packets)
1 2 3 * 5 6 7 * Xa Xb --> 4 can be recovered by Xa, because packets 1 - 3
are the last three 3 in-order packets buffered, 8 can be recovered by Xb
because 5 - 7 are in the OOO queue.
1 2 * * 5 6 7 * Xa Xb --> 3 and 4 cannot be recovered by Xa, but 8 can
still be recovered and goes into the OOO queue.

Any FEC packet which would require earlier segments for a recovery, is
actually only encoding packets which have already been received.
Does this answer your question?

- Tobias



On Thu, Jul 18, 2013 at 4:24 PM, Scheffenegger, Richard <rs@netapp.com>wrote:

> Hi Tobias,
>
> >>   1.  Periodically in every round-trip time, a TCP-IR sender places the
> >>   XOR of newly transmitted segments into a single MSS-length checksum
> >>   packet.  The XOR is only computed for new segments not previously
> >>   included in checksums.
>
> > Is it correct that there are more than one XOR packets per
> > RTT if there are more than 8 or 16 outstanding packets,
> > i.e., CWND>8 or 16?
> >
> > That is correct. If there are more than 8 (or 16) unencoded
> > packets in the queue when the timer fires, multiple XOR packets
> > are generated. Additionally, if new packets are transmitted after
> > the timer fired, the timer is armed again (set to RTT/4), so for
> > a sender transmitting data continuously the timer would fire up
> > to four times per RTT.
>
>
> So, you are ONLY sending with the RTT/4 timer? Wouldn't it be more useful
> to send the FEC segments immediately following the segments that are
> covered by it?
>
> 1 2 3 4 Xa 5 6 7 [pause] Xb
>
> Vs
>
> 1 2 3 4 5 6 7 [pause] Xa Xb
>
> (FEC episode of 4; as the client only captures up to 16 segment - i assume
> of the first episode that cann't be recovered? - for larger windows, where
> a whole bunch of XOR segment will get sent out at the RTT/4 timer, most of
> them will be pretty useless, even if they could theoretically could have
> recovered some loss... Or am I getting something wrong here?
>
> Best regards,
>
>
> Richard Scheffenegger
>
>
>

--089e0102fa08642ac204e1cfde07
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Right now FEC packets are only transmitted when the timer =
fires but your proposal sounds interesting. As long as there is more outsta=
nding data which would be transmitted right after an FEC packet, it might b=
e okay to transmit the FEC packet immediately. And we would still delay if =
there is no more data to send, e.g.<div>
<br></div><div>1 2 3 4 Xa 5 6 7 8 [pause] Xb<br><div><br></div><div>The onl=
y disadvantage I see is that having FEC packets in the middle of the burst,=
 is that the burst size increases, which might be okay.</div><div><br></div=
>
<div>Regarding your second question, let&#39;s use the example above. If we=
 set the maximum encoding range to 4, the receiver only needs to keep the l=
ast 3 in-order packets buffered.</div><div><br></div><div>Examples (* denot=
es loss of the packet):</div>
<div>* 2 3 4 Xa --&gt; 1 can be recovered, because 2 - 4 remained in the OO=
O queue</div><div>1 2 3 * Xa --&gt; 4 can be recovered, because 1 - 3 remai=
ned in the in-order queue (because they are the last 3 in-order packets)</d=
iv>
<div>1 2 3 * 5 6 7 * Xa Xb --&gt; 4 can be recovered by Xa, because packets=
 1 - 3 are the last three 3 in-order packets buffered, 8 can be recovered b=
y Xb because 5 - 7 are in the OOO queue.</div><div>1 2 * * 5 6 7 * Xa Xb --=
&gt; 3 and 4 cannot be recovered by Xa, but 8 can still be recovered and go=
es into the OOO queue.</div>
<div><br></div><div>Any FEC packet which would require earlier segments for=
 a recovery, is actually only encoding packets which have already been rece=
ived.</div><div>Does this answer your question?</div><div><br></div><div>
- Tobias</div><div><br><div class=3D"gmail_extra"><br><br><div class=3D"gma=
il_quote">On Thu, Jul 18, 2013 at 4:24 PM, Scheffenegger, Richard <span dir=
=3D"ltr">&lt;<a href=3D"mailto:rs@netapp.com" target=3D"_blank">rs@netapp.c=
om</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Hi Tobias,<br>
<div class=3D"im"><br>
&gt;&gt; =A0 1. =A0Periodically in every round-trip time, a TCP-IR sender p=
laces the<br>
&gt;&gt; =A0 XOR of newly transmitted segments into a single MSS-length che=
cksum<br>
&gt;&gt; =A0 packet. =A0The XOR is only computed for new segments not previ=
ously<br>
&gt;&gt; =A0 included in checksums.<br>
<br>
&gt; Is it correct that there are more than one XOR packets per<br>
&gt; RTT if there are more than 8 or 16 outstanding packets,<br>
&gt; i.e., CWND&gt;8 or 16?<br>
&gt;<br>
&gt; That is correct. If there are more than 8 (or 16) unencoded<br>
&gt; packets in the queue when the timer fires, multiple XOR packets<br>
&gt; are generated. Additionally, if new packets are transmitted after<br>
&gt; the timer fired, the timer is armed again (set to RTT/4), so for<br>
&gt; a sender transmitting data continuously the timer would fire up<br>
&gt; to four times per RTT.<br>
<br>
<br>
</div>So, you are ONLY sending with the RTT/4 timer? Wouldn&#39;t it be mor=
e useful to send the FEC segments immediately following the segments that a=
re covered by it?<br>
<br>
1 2 3 4 Xa 5 6 7 [pause] Xb<br>
<br>
Vs<br>
<br>
1 2 3 4 5 6 7 [pause] Xa Xb<br>
<br>
(FEC episode of 4; as the client only captures up to 16 segment - i assume =
of the first episode that cann&#39;t be recovered? - for larger windows, wh=
ere a whole bunch of XOR segment will get sent out at the RTT/4 timer, most=
 of them will be pretty useless, even if they could theoretically could hav=
e recovered some loss... Or am I getting something wrong here?<br>

<br>
Best regards,<br>
<br>
<br>
Richard Scheffenegger<br>
<br>
<br>
</blockquote></div><br></div></div></div></div>

--089e0102fa08642ac204e1cfde07--

From rs@netapp.com  Fri Jul 19 01:57:12 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Tobias Flach <flach@usc.edu>
Thread-Topic: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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Hi Tobias,

> The only disadvantage I see is that having FEC packets=20
> in the middle of the burst, is that the burst size=20
> increases, which might be okay.

True, with the relative probabilities which packets get dropped in a burst,=
 you want to unmangle them; with TSO, you probably want to send the FEC seg=
ment at the beginning of the next TSO burst (or actually in between, since =
you need individual TCP options for each segment anyway).=20

And again, FEC segments need to account against CWND. I consider a (useful)=
 6.25% (16:1) to 50% (2:1) increase in traffic volume not insignificant tha=
t TCP CC should not apply to these segments.=20


But again, the receivers FEC queue will need to hold on to the full window =
of segments. I made myself not clear enough in the examples, if the OOO que=
ue is also only 4 deep (not 16):

> 1 2 3 * 5 6 7 * Xa Xb --> 4 can be recovered by Xa,=20
> because packets 1 - 3 are the last three 3 in-order=20
> packets buffered, 8 can be recovered by Xb because=20
> 5 - 7 are in the OOO queue.

Only 4 can be recovered; not all (most) stacks are segment oriented - this =
appears to be a silent assumption in your draft that should be made explici=
t (or have I missed that part?)...


Best regards,

Richard Scheffenegger

From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of Tob=
ias Flach
Sent: Donnerstag, 18. Juli 2013 23:27
To: Scheffenegger, Richard
Cc: tcpm@ietf.org Extensions
Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-=
00.txt

Right now FEC packets are only transmitted when the timer fires but your pr=
oposal sounds interesting. As long as there is more outstanding data which =
would be transmitted right after an FEC packet, it might be okay to transmi=
t the FEC packet immediately. And we would still delay if there is no more =
data to send, e.g.

1 2 3 4 Xa 5 6 7 8 [pause] Xb

The only disadvantage I see is that having FEC packets in the middle of the=
 burst, is that the burst size increases, which might be okay.

Regarding your second question, let's use the example above. If we set the =
maximum encoding range to 4, the receiver only needs to keep the last 3 in-=
order packets buffered.

Examples (* denotes loss of the packet):
* 2 3 4 Xa --> 1 can be recovered, because 2 - 4 remained in the OOO queue
1 2 3 * Xa --> 4 can be recovered, because 1 - 3 remained in the in-order q=
ueue (because they are the last 3 in-order packets)
1 2 3 * 5 6 7 * Xa Xb --> 4 can be recovered by Xa, because packets 1 - 3 a=
re the last three 3 in-order packets buffered, 8 can be recovered by Xb bec=
ause 5 - 7 are in the OOO queue.
1 2 * * 5 6 7 * Xa Xb --> 3 and 4 cannot be recovered by Xa, but 8 can stil=
l be recovered and goes into the OOO queue.

Any FEC packet which would require earlier segments for a recovery, is actu=
ally only encoding packets which have already been received.
Does this answer your question?

- Tobias


On Thu, Jul 18, 2013 at 4:24 PM, Scheffenegger, Richard <rs@netapp.com> wro=
te:
Hi Tobias,

>> =A0 1. =A0Periodically in every round-trip time, a TCP-IR sender places =
the
>> =A0 XOR of newly transmitted segments into a single MSS-length checksum
>> =A0 packet. =A0The XOR is only computed for new segments not previously
>> =A0 included in checksums.

> Is it correct that there are more than one XOR packets per
> RTT if there are more than 8 or 16 outstanding packets,
> i.e., CWND>8 or 16?
>
> That is correct. If there are more than 8 (or 16) unencoded
> packets in the queue when the timer fires, multiple XOR packets
> are generated. Additionally, if new packets are transmitted after
> the timer fired, the timer is armed again (set to RTT/4), so for
> a sender transmitting data continuously the timer would fire up
> to four times per RTT.

So, you are ONLY sending with the RTT/4 timer? Wouldn't it be more useful t=
o send the FEC segments immediately following the segments that are covered=
 by it?

1 2 3 4 Xa 5 6 7 [pause] Xb

Vs

1 2 3 4 5 6 7 [pause] Xa Xb

(FEC episode of 4; as the client only captures up to 16 segment - i assume =
of the first episode that cann't be recovered? - for larger windows, where =
a whole bunch of XOR segment will get sent out at the RTT/4 timer, most of =
them will be pretty useless, even if they could theoretically could have re=
covered some loss... Or am I getting something wrong here?

Best regards,


Richard Scheffenegger



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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fec-00.txt
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On Fri, Jul 19, 2013 at 1:57 AM, Scheffenegger, Richard <rs@netapp.com> wro=
te:
> Hi Tobias,
>
>> The only disadvantage I see is that having FEC packets
>> in the middle of the burst, is that the burst size
>> increases, which might be okay.
>
> True, with the relative probabilities which packets get dropped in a burs=
t, you want to unmangle them; with TSO, you probably want to send the FEC s=
egment at the beginning of the next TSO burst (or actually in between, sinc=
e you need individual TCP options for each segment anyway).
>
> And again, FEC segments need to account against CWND. I consider a (usefu=
l) 6.25% (16:1) to 50% (2:1) increase in traffic volume not insignificant t=
hat TCP CC should not apply to these segments.
>
>
> But again, the receivers FEC queue will need to hold on to the full windo=
w of segments. I made myself not clear enough in the examples, if the OOO q=
ueue is also only 4 deep (not 16):
why is OOO queue max length limited to 4? FECs are accounted in the
receiver buffer autotuning, i.e., the sender won't send us FEC packet
unless the receiver told him it can take that.

>
>> 1 2 3 * 5 6 7 * Xa Xb --> 4 can be recovered by Xa,
>> because packets 1 - 3 are the last three 3 in-order
>> packets buffered, 8 can be recovered by Xb because
>> 5 - 7 are in the OOO queue.
>
> Only 4 can be recovered; not all (most) stacks are segment oriented - thi=
s appears to be a silent assumption in your draft that should be made expli=
cit (or have I missed that part?)...

I am not sure what's a segmented oriented stack. are you referring to
segment or byte-based congestion control?

>
>
> Best regards,
>
> Richard Scheffenegger
>
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of T=
obias Flach
> Sent: Donnerstag, 18. Juli 2013 23:27
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org Extensions
> Subject: Re: [tcpm] Fwd: New Version Notification for draft-flach-tcpm-fe=
c-00.txt
>
> Right now FEC packets are only transmitted when the timer fires but your =
proposal sounds interesting. As long as there is more outstanding data whic=
h would be transmitted right after an FEC packet, it might be okay to trans=
mit the FEC packet immediately. And we would still delay if there is no mor=
e data to send, e.g.
>
> 1 2 3 4 Xa 5 6 7 8 [pause] Xb
>
> The only disadvantage I see is that having FEC packets in the middle of t=
he burst, is that the burst size increases, which might be okay.
>
> Regarding your second question, let's use the example above. If we set th=
e maximum encoding range to 4, the receiver only needs to keep the last 3 i=
n-order packets buffered.
>
> Examples (* denotes loss of the packet):
> * 2 3 4 Xa --> 1 can be recovered, because 2 - 4 remained in the OOO queu=
e
> 1 2 3 * Xa --> 4 can be recovered, because 1 - 3 remained in the in-order=
 queue (because they are the last 3 in-order packets)
> 1 2 3 * 5 6 7 * Xa Xb --> 4 can be recovered by Xa, because packets 1 - 3=
 are the last three 3 in-order packets buffered, 8 can be recovered by Xb b=
ecause 5 - 7 are in the OOO queue.
> 1 2 * * 5 6 7 * Xa Xb --> 3 and 4 cannot be recovered by Xa, but 8 can st=
ill be recovered and goes into the OOO queue.
>
> Any FEC packet which would require earlier segments for a recovery, is ac=
tually only encoding packets which have already been received.
> Does this answer your question?
>
> - Tobias
>
>
> On Thu, Jul 18, 2013 at 4:24 PM, Scheffenegger, Richard <rs@netapp.com> w=
rote:
> Hi Tobias,
>
>>>   1.  Periodically in every round-trip time, a TCP-IR sender places the
>>>   XOR of newly transmitted segments into a single MSS-length checksum
>>>   packet.  The XOR is only computed for new segments not previously
>>>   included in checksums.
>
>> Is it correct that there are more than one XOR packets per
>> RTT if there are more than 8 or 16 outstanding packets,
>> i.e., CWND>8 or 16?
>>
>> That is correct. If there are more than 8 (or 16) unencoded
>> packets in the queue when the timer fires, multiple XOR packets
>> are generated. Additionally, if new packets are transmitted after
>> the timer fired, the timer is armed again (set to RTT/4), so for
>> a sender transmitting data continuously the timer would fire up
>> to four times per RTT.
>
> So, you are ONLY sending with the RTT/4 timer? Wouldn't it be more useful=
 to send the FEC segments immediately following the segments that are cover=
ed by it?
>
> 1 2 3 4 Xa 5 6 7 [pause] Xb
>
> Vs
>
> 1 2 3 4 5 6 7 [pause] Xa Xb
>
> (FEC episode of 4; as the client only captures up to 16 segment - i assum=
e of the first episode that cann't be recovered? - for larger windows, wher=
e a whole bunch of XOR segment will get sent out at the RTT/4 timer, most o=
f them will be pretty useless, even if they could theoretically could have =
recovered some loss... Or am I getting something wrong here?
>
> Best regards,
>
>
> Richard Scheffenegger
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From fernando@gont.com.ar  Sun Jul 21 09:47:40 2013
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From: Fernando Gont <fernando@gont.com.ar>
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Subject: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Folks,

Found this by chance -- probably a datapoint that advice is needed in
this area (that is, draft-gont-tcpm-tcp-seq-validation).

P.S.: Will present results from real-world testing at the next tcpm meeting.

Cheers,
Fernando




-------- Original Message --------
From: Matt Miller <matt@matthewjmiller.net>
Date: Wed, 17 Jul 2013 07:08:26 -0400
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Message-ID:
<CAFc6gu_q1X10EzsHrvnmYuQ0ZnKz9uNXbfJJe-guva6J-QKAow@mail.gmail.com>
Subject: TCP Loopback Connections with the Same Src/Dest Port
To: FreeBSD Net <freebsd-net@freebsd.org>

Our system is based on FreeBSD 8.1.  In some tests, we were having
issues caused by connections of this form (more details below):

 TCP4      0      0      0/   0/   0    127.0.0.1.665   127.0.0.1.665
 FIN_WAIT_1
 TCP4      0      0      0/   0/   0    127.0.0.1.637   127.0.0.1.637
 FIN_WAIT_1
 TCP4      0      0      0/   0/   0    127.0.0.1.648   127.0.0.1.648
 FIN_WAIT_1

Some questions we had:

- Has anyone else ever seen these same src/dest address/port TCP
connections created?  Does anyone know of a legitimate reason why they
should be allowed?

- If there are no known use cases for this type of connection, does
anyone have more context/insight on the design here: should this type
of inpcb creation be prevented in the kernel or is it the
application's responsibility to ensure it never creates this type of
socket?

For those interested, more details of the issue seen follow.  The
connection seems to get stuck in swi_net sending and receiving pure
FIN/ACKs to itself:

#12 0xffffffff804372ce in ip_output (m=0xffffff0003ccf300,
opt=<optimized out>, ro=0xffffff8020c2b6a0, flags=0, imo=0x0,
inp=0xffffff0019933968) at ../../../../sys/netinet/ip_output.c
#13 0xffffffff804423dc in tcp_output (tp=0xffffff0019de2370) at
../../../../sys/netinet/tcp_output.c
#14 0xffffffff8043ef5d in tcp_do_segment (m=0xffffff0019af1200,
th=0x100200, so=0xffffff011ac59570, tp=0xffffff0019de2370,
drop_hdrlen=52, tlen=0, iptos=0 '\000', ti_locked=3) at
../../../../sys/netinet/tcp_input.c
#15 0xffffffff80440311 in tcp_input (m=0xffffff0019af1200,
off0=<optimized out>) at ../../../../sys/netinet/tcp_input.c
#16 0xffffffff8043530b in ip_input (m=0xffffff0019af1200) at
../../../../sys/netinet/ip_input.c
#17 0xffffffff8040889f in netisr_process_workstream_proto
(proto=<optimized out>, nwsp=<optimized out>) at
../../../../sys/net/netisr.c
#18 swi_net (arg=0xffffffff80f59800) at ../../../../sys/net/netisr.c

swi_net() just continues in this loop, ad nauseam:

 759         while ((bits = nwsp->nws_pendingbits) != 0) {
 760                 while ((prot = ffs(bits)) != 0) {
 761                         prot--;
 762                         bits &= ~(1 << prot);
 763                         (void)netisr_process_workstream_proto(nwsp,
prot);
 764                 }
 765         }

The tcp_output() being triggered in tcp_do_segment() in the case is
the one show on line 2303 below:

2212         /*
2213          * In ESTABLISHED state: drop duplicate ACKs; ACK out of range
2214          * ACKs.  If the ack is in the range
2215          *      tp->snd_una < th->th_ack <= tp->snd_max
2216          * then advance tp->snd_una to th->th_ack and drop
2217          * data from the retransmission queue.  If this ACK reflects
2218          * more up to date window information we update our
window information.
2219          */
2220         case TCPS_ESTABLISHED:
2221         case TCPS_FIN_WAIT_1:
2222         case TCPS_FIN_WAIT_2:
2223         case TCPS_CLOSE_WAIT:
2224         case TCPS_CLOSING:
2225         case TCPS_LAST_ACK:
2226                 if (SEQ_GT(th->th_ack, tp->snd_max)) {
2227                         TCPSTAT_INC(tcps_rcvacktoomuch);
2228                         goto dropafterack;
2229                 }
...
2234                 if (SEQ_LEQ(th->th_ack, tp->snd_una)) {
...
2248                         if (tlen == 0 && tiwin == tp->snd_wnd) {
2249                                 TCPSTAT_INC(tcps_rcvdupack);
...
2277                                 if (!tcp_timer_active(tp, TT_REXMT) ||
2278                                     th->th_ack != tp->snd_una)
2279                                         tp->t_dupacks = 0;
2280                                 else if (++tp->t_dupacks >
tcprexmtthresh ||
2281                                     ((V_tcp_do_newreno ||
2282                                       (tp->t_flags &
TF_SACK_PERMIT)) &&
2283                                      IN_FASTRECOVERY(tp))) {
2284                                         if ((tp->t_flags &
TF_SACK_PERMIT) &&
2285                                             IN_FASTRECOVERY(tp)) {
2286                                                 int awnd;
2287
2288                                                 /*
2289                                                  * Compute the
amount of data in flight first.
2290                                                  * We can inject
new data into the pipe iff
2291                                                  * we have less
than 1/2 the original window's
2292                                                  * worth of data in
flight.
2293                                                  */
2294                                                 awnd =
(tp->snd_nxt - tp->snd_fack) +
2295
tp->sackhint.sack_bytes_rexmit;
2296                                                 if (awnd <
tp->snd_ssthresh) {
2297
tp->snd_cwnd += tp->t_maxseg;
2298                                                         if
(tp->snd_cwnd > tp->snd_ssthresh)
2299
tp->snd_cwnd = tp->snd_ssthresh;
2300                                                 }
2301                                         } else
2302                                                 tp->snd_cwnd +=
tp->t_maxseg;
2303                                         (void) tcp_output(tp);
2304                                         goto drop;

I've noticed that we don't yet have this patch in our code:

http://svnweb.freebsd.org/base?view=revision&revision=239672

Which seems like it could be relevant here to the general case of both
ends of the connection entering FIN_WAIT_1 at the same time and
sending FIN/ACKs repeatedly (though our connections are a bizarre case
of this where both ends of the connection are actually the same
connection).

Thanks,

Matt
_______________________________________________
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-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@si6networks.com
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1






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Subject: Re: [tcpm] draft-ietf-tcpm-1323bis-14.txt
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Lots of mails have been exchanged on this since the WGLC, but here is an =
attempt to shortly summarize the main points:

* Mark Allman made a thorough set of comments during the WGLC. My =
understanding is that these comments are now addressed. Some paragraphs =
have changed quite a bit as a result, so it would be good if people can =
check if the current text looks good.

* After the WGLC there was an update on the text regarding handling of =
segments without timestamp option after they have been successfully =
negotiated (MUST drop in ver-14), and a follow-up discussion about =
possible interactions with middleboxes. In a recent mail Richard =
proposes changing this to SHOULD. This change is not yet visible in the =
I-D repository.

If I'm forgetting something important, please remind me (and the list).

I think in the meeting we should go through the main recent changes, and =
check if people are ok with the current wordings. Additional comments on =
the list are much appreciated.

- Pasi


On Jul 16, 2013, at 2:03 PM, "Scheffenegger, Richard" <rs@netapp.com> =
wrote:

> Hi,
>=20
> I would like to take this opportunity to stir up the recent =
discussions (from April/May timeframe) again.
>=20
> There have been some comments past the posting of this version on the =
list, which I wasn't able to fully qualify how they would impact this =
latest text.
>=20
> I would like to encourage the participants in these discussions to =
review the latest version, and point out if they cannot agree with any =
of the wording.
>=20
>=20
>=20
> There will probably be a slot at the upcoming IETF to try to find out =
what the consensus is on this document, too...
>=20
> Best regards,
>=20
> Richard Scheffenegger
>=20
>=20
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>> Sent: Donnerstag, 23. Mai 2013 16:10
>> To: Bob Braden; Scheffenegger, Richard; Robert T. Braden; David =
Borman;
>> Van Jacobson
>> Subject: New Version Notification for draft-ietf-tcpm-1323bis-14.txt
>>=20
>>=20
>> A new version of I-D, draft-ietf-tcpm-1323bis-14.txt has been =
successfully
>> submitted by David Borman and posted to the IETF repository.
>>=20
>> Filename:	 draft-ietf-tcpm-1323bis
>> Revision:	 14
>> Title:		 TCP Extensions for High Performance
>> Creation date:	 2013-05-23
>> Group:		 tcpm
>> Number of pages: 46
>> URL:             http://www.ietf.org/internet-drafts/draft-ietf-tcpm-
>> 1323bis-14.txt
>> Status:          =
http://datatracker.ietf.org/doc/draft-ietf-tcpm-1323bis
>> Htmlized:        =
http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-14
>> Diff:            =
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-1323bis-
>> 14
>>=20
>> Abstract:
>>   This document specifies a set of TCP extensions to improve
>>   performance over paths with a large bandwidth * delay product and =
to
>>   provide reliable operation over very high-speed paths.  It defines
>>   TCP options for scaled windows and timestamps.  The timestamps are
>>   used for two distinct mechanisms, RTTM (Round Trip Time =
Measurement)
>>   and PAWS (Protection Against Wrapped Sequences).
>>=20
>>   This document updates and obsoletes RFC 1323.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From touch@isi.edu  Sun Jul 21 17:16:24 2013
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Date: Sun, 21 Jul 2013 17:15:44 -0700
From: Joe Touch <touch@isi.edu>
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To: "Scheffenegger, Richard" <rs@netapp.com>
References: <CAO249yfdHnz0vd2rBQXpQ4RJqo2R09r0X7_-b03fkMC_csV=Yw@mail.gmail.com> <012C3117EDDB3C4781FD802A8C27DD4F24BC11A2@SACEXCMBX02-PRD.hq.netapp.com> <CAO249ycGrS3HpqJvy1DUQV=_sqpmVHEk529SW+Z0cLEhxOARiQ@mail.gmail.com> <16682_1370353034_51ADED8A_16682_97_1_012C3117EDDB3C4781FD802A8C27DD4F24BC8FA7@SACEXCMBX02-PRD.hq.netapp.com> <DE23153F-9697-48FC-BFA1-9032739ACBA0@iki.fi> <012C3117EDDB3C4781FD802A8C27DD4F25C057ED@SACEXCMBX02-PRD.hq.netapp.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] some questions related to PAWS
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If PAWS is a "best effort" assurance, then I accept your conclusion.

If PAWS is something we think provides a guarantee (and I do), then it 
is a mistake, and the previous MUST and MUST NOTs should be retained.

I've so far seen discussion talking about the practical side of this 
issue from the standpoint of non-compliant components (notably 
middleboxes). That may be true but remains irrelevant to the impact on 
the connection.

If you keep the SHOULD and SHOULD NOT below, you're effectively killing 
PAWS capability. You need to at least admit that in the text.

Joe

On 7/16/2013 9:23 AM, Scheffenegger, Richard wrote:
>
>
> For what it'S worth, I reworded that MUST and MAY to a SHOULD and SHOULD NOT:
>
>     Once TSopt has been successfully negotiated (sent and received)
>     during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>     non-<RST> segment for the duration of the connection, and SHOULD be
>     sent in an <RST> segment (see Section 4.2 for details).  If a non-
> ~  <RST> segment is received without a TSopt, a TCP SHOULD drop the
> ~  segment and SHOULD NOT send an <ACK> for the last in-sequence
>     segment.  A TCP MUST NOT abort a TCP connection because any segment
>     lacks an expected TSopt.
>
>
>
> And added the following paragraph suggested by Joe as last point in the Security/Middlebox section, to point out that the -bis update may uncover broken middleboxes, potentially leading to stale TCP sessions.
>
>
> ~     *  It must be noted that [RFC1323] doesn't address the case of the
> ~        Timestamps option being dropped or selectively omitted after
> ~        being negotiated, and that the update in this document may
> ~        cause some broken middlebox behavior to be detected
> ~        (potentially unresponsive TCP sessions).
>
>
> I think these two updates reflect what has been discussed on list...
>
> (These are deltas vs. http://www.ietf.org/id/draft-ietf-tcpm-1323bis-14.txt ).
>
>
> Best regards,
>
>
> Richard Scheffenegger
>
>> -----Original Message-----
>> From: Pasi Sarolahti [mailto:pasi.sarolahti@iki.fi]
>> Sent: Dienstag, 04. Juni 2013 18:28
>> To: Scheffenegger, Richard
>> Cc: Yoshifumi Nishida; tcpm@ietf.org
>> Subject: Re: [tcpm] some questions related to PAWS
>>
>> On Jun 4, 2013, at 3:05 PM, "Scheffenegger, Richard" <rs@netapp.com>
>> wrote:
>>
>>>> It might be tricky, but I still think it's one possibility.
>>>> one might prefer it rather than dropping packets without TS blindly.
>>>
>>> Probably. The sender MUST still provide a TSopt in every segment,
>> assuming that the receiver is strictly enforcing PAWS. If a receiver
>> chooses to open a broader window for misuse (as apparently two dominant
>> stacks currently do), that's their business. Specifying a resiliency
>> mechanism and then stipulating it is completely optional to use it, is not
>> the best approach for an RFC IMHO. That's an implementers choice.
>>
>> Right. But we should also remember PAWS failure should be very rare case,
>> where two uncommon events coincide:
>> * there is a old segment with wrapped sequence number that happens to hit
>> the window
>> * the segment has timestamps option removed for some reason
>>
>>> I need to find some time to summarize the discussion around this topic
>>> that the WG had in the last few days, and pour it into some text for
>>> the 1323bis draft. (I'm always happy if someone want's to donate
>>> text..:)
>>
>> I don't think there has been much disagreement about the fact that to
>> ensure reliable operation of PAWS, all segments must have timestamps. I
>> tend to agree with you that standardizing an algorithm that works
>> unreliably is not elegant, and majority of mailing list feedback seems to
>> support this, so this seems clear.
>>
>> The trickier thing is, what language to add about the current state, but I
>> think we should point out what we know: there are implementations that
>> accept segments without timestamp after its successful negotiation (in
>> fact no one so far has reported implementation that would discard segments
>> without timestamps), and changing to follow the "MUST discard" rule may
>> bring up some communication problems as a result of faulty network
>> behavior (and there was some reported evidence of this actually
>> happening). I'm not suggesting this as a text to be used, but hopefully it
>> is an acceptable summary to everyone.
>>
>> Not to be said in the document, but just wanted to say it out: even if
>> segments without timestamps were accepted, PAWS could still be able to
>> detect most cases of sequence wrapping, assuming that timestamps removal
>> is a rare case. If user runs into communication problems that can be
>> solved by turning timestamps off entirely, the natural reaction is to do
>> so. After that there is no PAWS protection at all.
>>
>> - Pasi
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From touch@isi.edu  Sun Jul 21 22:03:22 2013
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Date: Sun, 21 Jul 2013 22:02:31 -0700
From: Joe Touch <touch@isi.edu>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Same src/dst for the same IP makes no sense; there need to be two ends 
to a connection, and each end is supposed to be uniquely determined by 
the socket (as defined in 793, not Un*x).

IMO, it ought to be rejected by the API, just as would be one that was 
otherwise incompletely or incorrectly specified (picking a source 
address not on a local interface, picking a port range you don't have 
privilege to access, etc.).

However, loopback is a subnet (127.0.0.0/8), not just a single address. 
It ought to be feasible and correct to open a connection to yourself on 
the same port on different loopback addresses.

Joe

On 7/21/2013 9:47 AM, Fernando Gont wrote:
> Folks,
>
> Found this by chance -- probably a datapoint that advice is needed in
> this area (that is, draft-gont-tcpm-tcp-seq-validation).
>
> P.S.: Will present results from real-world testing at the next tcpm meeting.
>
> Cheers,
> Fernando
>
>
>
>
> -------- Original Message --------
> From: Matt Miller <matt@matthewjmiller.net>
> Date: Wed, 17 Jul 2013 07:08:26 -0400
> X-Google-Sender-Auth: ba5SYKKiksigElenhewyH0EffCs
> Message-ID:
> <CAFc6gu_q1X10EzsHrvnmYuQ0ZnKz9uNXbfJJe-guva6J-QKAow@mail.gmail.com>
> Subject: TCP Loopback Connections with the Same Src/Dest Port
> To: FreeBSD Net <freebsd-net@freebsd.org>
>
> Our system is based on FreeBSD 8.1.  In some tests, we were having
> issues caused by connections of this form (more details below):
>
>   TCP4      0      0      0/   0/   0    127.0.0.1.665   127.0.0.1.665
>   FIN_WAIT_1
>   TCP4      0      0      0/   0/   0    127.0.0.1.637   127.0.0.1.637
>   FIN_WAIT_1
>   TCP4      0      0      0/   0/   0    127.0.0.1.648   127.0.0.1.648
>   FIN_WAIT_1
>
> Some questions we had:
>
> - Has anyone else ever seen these same src/dest address/port TCP
> connections created?  Does anyone know of a legitimate reason why they
> should be allowed?
>
> - If there are no known use cases for this type of connection, does
> anyone have more context/insight on the design here: should this type
> of inpcb creation be prevented in the kernel or is it the
> application's responsibility to ensure it never creates this type of
> socket?
>
> For those interested, more details of the issue seen follow.  The
> connection seems to get stuck in swi_net sending and receiving pure
> FIN/ACKs to itself:
>
> #12 0xffffffff804372ce in ip_output (m=0xffffff0003ccf300,
> opt=<optimized out>, ro=0xffffff8020c2b6a0, flags=0, imo=0x0,
> inp=0xffffff0019933968) at ../../../../sys/netinet/ip_output.c
> #13 0xffffffff804423dc in tcp_output (tp=0xffffff0019de2370) at
> ../../../../sys/netinet/tcp_output.c
> #14 0xffffffff8043ef5d in tcp_do_segment (m=0xffffff0019af1200,
> th=0x100200, so=0xffffff011ac59570, tp=0xffffff0019de2370,
> drop_hdrlen=52, tlen=0, iptos=0 '\000', ti_locked=3) at
> ../../../../sys/netinet/tcp_input.c
> #15 0xffffffff80440311 in tcp_input (m=0xffffff0019af1200,
> off0=<optimized out>) at ../../../../sys/netinet/tcp_input.c
> #16 0xffffffff8043530b in ip_input (m=0xffffff0019af1200) at
> ../../../../sys/netinet/ip_input.c
> #17 0xffffffff8040889f in netisr_process_workstream_proto
> (proto=<optimized out>, nwsp=<optimized out>) at
> ../../../../sys/net/netisr.c
> #18 swi_net (arg=0xffffffff80f59800) at ../../../../sys/net/netisr.c
>
> swi_net() just continues in this loop, ad nauseam:
>
>   759         while ((bits = nwsp->nws_pendingbits) != 0) {
>   760                 while ((prot = ffs(bits)) != 0) {
>   761                         prot--;
>   762                         bits &= ~(1 << prot);
>   763                         (void)netisr_process_workstream_proto(nwsp,
> prot);
>   764                 }
>   765         }
>
> The tcp_output() being triggered in tcp_do_segment() in the case is
> the one show on line 2303 below:
>
> 2212         /*
> 2213          * In ESTABLISHED state: drop duplicate ACKs; ACK out of range
> 2214          * ACKs.  If the ack is in the range
> 2215          *      tp->snd_una < th->th_ack <= tp->snd_max
> 2216          * then advance tp->snd_una to th->th_ack and drop
> 2217          * data from the retransmission queue.  If this ACK reflects
> 2218          * more up to date window information we update our
> window information.
> 2219          */
> 2220         case TCPS_ESTABLISHED:
> 2221         case TCPS_FIN_WAIT_1:
> 2222         case TCPS_FIN_WAIT_2:
> 2223         case TCPS_CLOSE_WAIT:
> 2224         case TCPS_CLOSING:
> 2225         case TCPS_LAST_ACK:
> 2226                 if (SEQ_GT(th->th_ack, tp->snd_max)) {
> 2227                         TCPSTAT_INC(tcps_rcvacktoomuch);
> 2228                         goto dropafterack;
> 2229                 }
> ...
> 2234                 if (SEQ_LEQ(th->th_ack, tp->snd_una)) {
> ...
> 2248                         if (tlen == 0 && tiwin == tp->snd_wnd) {
> 2249                                 TCPSTAT_INC(tcps_rcvdupack);
> ...
> 2277                                 if (!tcp_timer_active(tp, TT_REXMT) ||
> 2278                                     th->th_ack != tp->snd_una)
> 2279                                         tp->t_dupacks = 0;
> 2280                                 else if (++tp->t_dupacks >
> tcprexmtthresh ||
> 2281                                     ((V_tcp_do_newreno ||
> 2282                                       (tp->t_flags &
> TF_SACK_PERMIT)) &&
> 2283                                      IN_FASTRECOVERY(tp))) {
> 2284                                         if ((tp->t_flags &
> TF_SACK_PERMIT) &&
> 2285                                             IN_FASTRECOVERY(tp)) {
> 2286                                                 int awnd;
> 2287
> 2288                                                 /*
> 2289                                                  * Compute the
> amount of data in flight first.
> 2290                                                  * We can inject
> new data into the pipe iff
> 2291                                                  * we have less
> than 1/2 the original window's
> 2292                                                  * worth of data in
> flight.
> 2293                                                  */
> 2294                                                 awnd =
> (tp->snd_nxt - tp->snd_fack) +
> 2295
> tp->sackhint.sack_bytes_rexmit;
> 2296                                                 if (awnd <
> tp->snd_ssthresh) {
> 2297
> tp->snd_cwnd += tp->t_maxseg;
> 2298                                                         if
> (tp->snd_cwnd > tp->snd_ssthresh)
> 2299
> tp->snd_cwnd = tp->snd_ssthresh;
> 2300                                                 }
> 2301                                         } else
> 2302                                                 tp->snd_cwnd +=
> tp->t_maxseg;
> 2303                                         (void) tcp_output(tp);
> 2304                                         goto drop;
>
> I've noticed that we don't yet have this patch in our code:
>
> http://svnweb.freebsd.org/base?view=revision&revision=239672
>
> Which seems like it could be relevant here to the general case of both
> ends of the connection entering FIN_WAIT_1 at the same time and
> sending FIN/ACKs repeatedly (though our connections are a bizarre case
> of this where both ends of the connection are actually the same
> connection).
>
> Thanks,
>
> Matt
> _______________________________________________
> freebsd-net@freebsd.org mailing list
> http://lists.freebsd.org/mailman/listinfo/freebsd-net
> To unsubscribe, send any mail to "freebsd-net-unsubscribe@freebsd.org"
>
>

From fernando@gont.com.ar  Sun Jul 21 22:13:55 2013
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Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Hi, Joe,

On 07/22/2013 07:02 AM, Joe Touch wrote:
> Same src/dst for the same IP makes no sense; there need to be two ends
> to a connection, and each end is supposed to be uniquely determined by
> the socket (as defined in 793, not Un*x).

It could be employed for IPC, without the need for two state machines.

There are still two ends -- it's just that they are the same.



> IMO, it ought to be rejected by the API, just as would be one that was
> otherwise incompletely or incorrectly specified (picking a source
> address not on a local interface, picking a port range you don't have
> privilege to access, etc.).

Side question: There's still the issue of what to do if, say, you're
listening on IP.port:* and received a packet IP.port:IP.port (i.e., you
cannot rely on the API, because you may receive the packet on the wire
-- e.g., if it was forged).

Thanks!

Cheers,
-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@si6networks.com
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1




From jakob.heitz@ericsson.com  Sun Jul 21 22:25:01 2013
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From: Jakob Heitz <jakob.heitz@ericsson.com>
To: Fernando Gont <fernando@gont.com.ar>, Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Suppose you could have 2 ends with the same protocol identifier.

Then you could have 3. No end knows who it's talking to anymore.

Reductio ad absurdum.



--

Jakob Heitz.

________________________________
From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of Fernando G=
ont [fernando@gont.com.ar]
Sent: Sunday, 21 July 2013 10:13 PM
To: Joe Touch
Cc: tcpm@ietf.org; Fernando Gont
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Po=
rt

Hi, Joe,

On 07/22/2013 07:02 AM, Joe Touch wrote:
> Same src/dst for the same IP makes no sense; there need to be two ends
> to a connection, and each end is supposed to be uniquely determined by
> the socket (as defined in 793, not Un*x).

It could be employed for IPC, without the need for two state machines.

There are still two ends -- it's just that they are the same.

--_000_2F3EBB88EC3A454AAB08915FBF0B8C7E02DBFAC7eusaamb109erics_
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<p>Suppose you could have 2 ends with the same protocol identifier.</p>
<p>Then you could have 3. No end knows who it's talking to anymore.</p>
<p>Reductio ad absurdum.</p>
<p>&nbsp;</p>
<p>--</p>
<div>
<div style=3D"FONT-SIZE: 13px; FONT-FAMILY: Tahoma">
<p>Jakob Heitz.</p>
</div>
</div>
<div>
<hr tabindex=3D"-1">
<div id=3D"x_divRplyFwdMsg"><font color=3D"#000000" size=3D"2" face=3D"Taho=
ma"><b>From:</b> tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of=
 Fernando Gont [fernando@gont.com.ar]<br>
<b>Sent:</b> Sunday, 21 July 2013 10:13 PM<br>
<b>To:</b> Joe Touch<br>
<b>Cc:</b> tcpm@ietf.org; Fernando Gont<br>
<b>Subject:</b> Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/=
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<div class=3D"PlainText">Hi, Joe,<br>
<br>
On 07/22/2013 07:02 AM, Joe Touch wrote:<br>
&gt; Same src/dst for the same IP makes no sense; there need to be two ends=
<br>
&gt; to a connection, and each end is supposed to be uniquely determined by=
<br>
&gt; the socket (as defined in 793, not Un*x).<br>
<br>
It could be employed for IPC, without the need for two state machines.<br>
<br>
There are still two ends -- it's just that they are the same.</div>
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</body>
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From: Jakob Heitz <jakob.heitz@ericsson.com>
To: Fernando Gont <fernando@gont.com.ar>, Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Note, anycast is different.

Multiple anycast endpoints have the same address, but they

coordinate amongst themselves to appear as a single endpoint.



--

Jakob Heitz.

________________________________
From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of Jakob Heit=
z [jakob.heitz@ericsson.com]
Sent: Sunday, 21 July 2013 10:24 PM
To: Fernando Gont; Joe Touch
Cc: tcpm@ietf.org
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Po=
rt


Suppose you could have 2 ends with the same protocol identifier.

Then you could have 3. No end knows who it's talking to anymore.

Reductio ad absurdum.



--

Jakob Heitz.

________________________________
From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of Fernando G=
ont [fernando@gont.com.ar]
Sent: Sunday, 21 July 2013 10:13 PM
To: Joe Touch
Cc: tcpm@ietf.org; Fernando Gont
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Po=
rt

Hi, Joe,

On 07/22/2013 07:02 AM, Joe Touch wrote:
> Same src/dst for the same IP makes no sense; there need to be two ends
> to a connection, and each end is supposed to be uniquely determined by
> the socket (as defined in 793, not Un*x).

It could be employed for IPC, without the need for two state machines.

There are still two ends -- it's just that they are the same.

--_000_2F3EBB88EC3A454AAB08915FBF0B8C7E02DBFB07eusaamb109erics_
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<p>Note, anycast is different.</p>
<p>Multiple anycast endpoints have the same address, but they</p>
<p>coordinate amongst themselves to appear as a single endpoint.</p>
<div>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 13px; FONT-FAMILY: Tahoma">
<p>--</p>
<p>Jakob Heitz.</p>
</div>
</div>
<hr tabindex=3D"-1">
<div id=3D"divRpF4971" style=3D"DIRECTION: ltr"><font color=3D"#000000" siz=
e=3D"2" face=3D"Tahoma"><b>From:</b> tcpm-bounces@ietf.org [tcpm-bounces@ie=
tf.org] on behalf of Jakob Heitz [jakob.heitz@ericsson.com]<br>
<b>Sent:</b> Sunday, 21 July 2013 10:24 PM<br>
<b>To:</b> Fernando Gont; Joe Touch<br>
<b>Cc:</b> tcpm@ietf.org<br>
<b>Subject:</b> Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/=
Dest Port<br>
</font><br>
</div>
<div></div>
<div>
<div style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Lucida Console; COLOR: #993366;=
 DIRECTION: ltr">
<p>Suppose you could have 2 ends with the same protocol identifier.</p>
<p>Then you could have 3. No end knows who it's talking to anymore.</p>
<p>Reductio ad absurdum.</p>
<p>&nbsp;</p>
<p>--</p>
<div>
<div style=3D"FONT-SIZE: 13px; FONT-FAMILY: Tahoma">
<p>Jakob Heitz.</p>
</div>
</div>
<div>
<hr tabindex=3D"-1">
<div id=3D"x_divRplyFwdMsg"><font color=3D"#000000" size=3D"2" face=3D"Taho=
ma"><b>From:</b> tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] on behalf of=
 Fernando Gont [fernando@gont.com.ar]<br>
<b>Sent:</b> Sunday, 21 July 2013 10:13 PM<br>
<b>To:</b> Joe Touch<br>
<b>Cc:</b> tcpm@ietf.org; Fernando Gont<br>
<b>Subject:</b> Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/=
Dest Port<br>
</font><br>
</div>
<div></div>
</div>
<font size=3D"2"><span style=3D"FONT-SIZE: 10pt">
<div class=3D"PlainText">Hi, Joe,<br>
<br>
On 07/22/2013 07:02 AM, Joe Touch wrote:<br>
&gt; Same src/dst for the same IP makes no sense; there need to be two ends=
<br>
&gt; to a connection, and each end is supposed to be uniquely determined by=
<br>
&gt; the socket (as defined in 793, not Un*x).<br>
<br>
It could be employed for IPC, without the need for two state machines.<br>
<br>
There are still two ends -- it's just that they are the same.</div>
</span></font></div>
</div>
</div>
</body>
</html>

--_000_2F3EBB88EC3A454AAB08915FBF0B8C7E02DBFB07eusaamb109erics_--

From fgont@si6networks.com  Sun Jul 21 23:43:40 2013
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On 07/22/2013 07:24 AM, Jakob Heitz wrote:
> Suppose you could have 2 ends with the same protocol identifier.
> 
> Then you could have 3. No end knows who it's talking to anymore.

I don't follow... could you please elaborate?

When you try to create a third endpoint, the address would be in use.

But with only two endpoints, it's fine: since you have a local and a
remote endpoint. -- the fact that both are the same is simply a cornercase.

Cheers,
-- 
Fernando Gont
SI6 Networks
e-mail: fgont@si6networks.com
PGP Fingerprint: 6666 31C6 D484 63B2 8FB1 E3C4 AE25 0D55 1D4E 7492





From dab@weston.borman.com  Mon Jul 22 07:17:56 2013
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It's a socket connected to itself, and thus only makes sense for a local =
IP address, be it the loopback address or any other address on the =
machine.  You can write to the socket, and then read the data back out.  =
If your TCP properly handles this case of a self-connected socket, then =
it handles a bunch of corner cases.  If you have to disallow them =
because it sends your system into an infinite loop of packets, then you =
have a latent bug just waiting to be tickled between a pair of sockets.  =
It's better to fix the problem than disallow the symptom.  In all these =
cases that I've seen, the problem is that there is valid ACK information =
in the packet, and the incoming TCP processing is dropping the packet =
without processing the ACK information.  Crossing SYNs, crossing FINs, =
crossing probes, they all can cause this problem if you don't process =
the ACK information.  The self-connected socket just guarantees that =
you'll hit both the crossing SYN and crossing FIN cases.

Testing that a self-connected socket works should be part of every TCP =
regression test.

			-David Borman

On Jul 22, 2013, at 12:02 AM, Joe Touch <touch@isi.edu> wrote:

> Same src/dst for the same IP makes no sense; there need to be two ends =
to a connection, and each end is supposed to be uniquely determined by =
the socket (as defined in 793, not Un*x).
>=20
> IMO, it ought to be rejected by the API, just as would be one that was =
otherwise incompletely or incorrectly specified (picking a source =
address not on a local interface, picking a port range you don't have =
privilege to access, etc.).
>=20
> However, loopback is a subnet (127.0.0.0/8), not just a single =
address. It ought to be feasible and correct to open a connection to =
yourself on the same port on different loopback addresses.
>=20
> Joe
>=20
> On 7/21/2013 9:47 AM, Fernando Gont wrote:
>> Folks,
>>=20
>> Found this by chance -- probably a datapoint that advice is needed in
>> this area (that is, draft-gont-tcpm-tcp-seq-validation).
>>=20
>> P.S.: Will present results from real-world testing at the next tcpm =
meeting.
>>=20
>> Cheers,
>> Fernando
>>=20
>>=20
>>=20
>>=20
>> -------- Original Message --------
>> From: Matt Miller <matt@matthewjmiller.net>
>> Date: Wed, 17 Jul 2013 07:08:26 -0400
>> X-Google-Sender-Auth: ba5SYKKiksigElenhewyH0EffCs
>> Message-ID:
>> <CAFc6gu_q1X10EzsHrvnmYuQ0ZnKz9uNXbfJJe-guva6J-QKAow@mail.gmail.com>
>> Subject: TCP Loopback Connections with the Same Src/Dest Port
>> To: FreeBSD Net <freebsd-net@freebsd.org>
>>=20
>> Our system is based on FreeBSD 8.1.  In some tests, we were having
>> issues caused by connections of this form (more details below):
>>=20
>> TCP4      0      0      0/   0/   0    127.0.0.1.665   127.0.0.1.665
>> FIN_WAIT_1
>> TCP4      0      0      0/   0/   0    127.0.0.1.637   127.0.0.1.637
>> FIN_WAIT_1
>> TCP4      0      0      0/   0/   0    127.0.0.1.648   127.0.0.1.648
>> FIN_WAIT_1
>>=20
>> Some questions we had:
>>=20
>> - Has anyone else ever seen these same src/dest address/port TCP
>> connections created?  Does anyone know of a legitimate reason why =
they
>> should be allowed?
>>=20
>> - If there are no known use cases for this type of connection, does
>> anyone have more context/insight on the design here: should this type
>> of inpcb creation be prevented in the kernel or is it the
>> application's responsibility to ensure it never creates this type of
>> socket?
>>=20
>> For those interested, more details of the issue seen follow.  The
>> connection seems to get stuck in swi_net sending and receiving pure
>> FIN/ACKs to itself:
>>=20
>> #12 0xffffffff804372ce in ip_output (m=3D0xffffff0003ccf300,
>> opt=3D<optimized out>, ro=3D0xffffff8020c2b6a0, flags=3D0, imo=3D0x0,
>> inp=3D0xffffff0019933968) at ../../../../sys/netinet/ip_output.c
>> #13 0xffffffff804423dc in tcp_output (tp=3D0xffffff0019de2370) at
>> ../../../../sys/netinet/tcp_output.c
>> #14 0xffffffff8043ef5d in tcp_do_segment (m=3D0xffffff0019af1200,
>> th=3D0x100200, so=3D0xffffff011ac59570, tp=3D0xffffff0019de2370,
>> drop_hdrlen=3D52, tlen=3D0, iptos=3D0 '\000', ti_locked=3D3) at
>> ../../../../sys/netinet/tcp_input.c
>> #15 0xffffffff80440311 in tcp_input (m=3D0xffffff0019af1200,
>> off0=3D<optimized out>) at ../../../../sys/netinet/tcp_input.c
>> #16 0xffffffff8043530b in ip_input (m=3D0xffffff0019af1200) at
>> ../../../../sys/netinet/ip_input.c
>> #17 0xffffffff8040889f in netisr_process_workstream_proto
>> (proto=3D<optimized out>, nwsp=3D<optimized out>) at
>> ../../../../sys/net/netisr.c
>> #18 swi_net (arg=3D0xffffffff80f59800) at =
../../../../sys/net/netisr.c
>>=20
>> swi_net() just continues in this loop, ad nauseam:
>>=20
>> 759         while ((bits =3D nwsp->nws_pendingbits) !=3D 0) {
>> 760                 while ((prot =3D ffs(bits)) !=3D 0) {
>> 761                         prot--;
>> 762                         bits &=3D ~(1 << prot);
>> 763                         =
(void)netisr_process_workstream_proto(nwsp,
>> prot);
>> 764                 }
>> 765         }
>>=20
>> The tcp_output() being triggered in tcp_do_segment() in the case is
>> the one show on line 2303 below:
>>=20
>> 2212         /*
>> 2213          * In ESTABLISHED state: drop duplicate ACKs; ACK out of =
range
>> 2214          * ACKs.  If the ack is in the range
>> 2215          *      tp->snd_una < th->th_ack <=3D tp->snd_max
>> 2216          * then advance tp->snd_una to th->th_ack and drop
>> 2217          * data from the retransmission queue.  If this ACK =
reflects
>> 2218          * more up to date window information we update our
>> window information.
>> 2219          */
>> 2220         case TCPS_ESTABLISHED:
>> 2221         case TCPS_FIN_WAIT_1:
>> 2222         case TCPS_FIN_WAIT_2:
>> 2223         case TCPS_CLOSE_WAIT:
>> 2224         case TCPS_CLOSING:
>> 2225         case TCPS_LAST_ACK:
>> 2226                 if (SEQ_GT(th->th_ack, tp->snd_max)) {
>> 2227                         TCPSTAT_INC(tcps_rcvacktoomuch);
>> 2228                         goto dropafterack;
>> 2229                 }
>> ...
>> 2234                 if (SEQ_LEQ(th->th_ack, tp->snd_una)) {
>> ...
>> 2248                         if (tlen =3D=3D 0 && tiwin =3D=3D =
tp->snd_wnd) {
>> 2249                                 TCPSTAT_INC(tcps_rcvdupack);
>> ...
>> 2277                                 if (!tcp_timer_active(tp, =
TT_REXMT) ||
>> 2278                                     th->th_ack !=3D tp->snd_una)
>> 2279                                         tp->t_dupacks =3D 0;
>> 2280                                 else if (++tp->t_dupacks >
>> tcprexmtthresh ||
>> 2281                                     ((V_tcp_do_newreno ||
>> 2282                                       (tp->t_flags &
>> TF_SACK_PERMIT)) &&
>> 2283                                      IN_FASTRECOVERY(tp))) {
>> 2284                                         if ((tp->t_flags &
>> TF_SACK_PERMIT) &&
>> 2285                                             IN_FASTRECOVERY(tp)) =
{
>> 2286                                                 int awnd;
>> 2287
>> 2288                                                 /*
>> 2289                                                  * Compute the
>> amount of data in flight first.
>> 2290                                                  * We can inject
>> new data into the pipe iff
>> 2291                                                  * we have less
>> than 1/2 the original window's
>> 2292                                                  * worth of data =
in
>> flight.
>> 2293                                                  */
>> 2294                                                 awnd =3D
>> (tp->snd_nxt - tp->snd_fack) +
>> 2295
>> tp->sackhint.sack_bytes_rexmit;
>> 2296                                                 if (awnd <
>> tp->snd_ssthresh) {
>> 2297
>> tp->snd_cwnd +=3D tp->t_maxseg;
>> 2298                                                         if
>> (tp->snd_cwnd > tp->snd_ssthresh)
>> 2299
>> tp->snd_cwnd =3D tp->snd_ssthresh;
>> 2300                                                 }
>> 2301                                         } else
>> 2302                                                 tp->snd_cwnd +=3D
>> tp->t_maxseg;
>> 2303                                         (void) tcp_output(tp);
>> 2304                                         goto drop;
>>=20
>> I've noticed that we don't yet have this patch in our code:
>>=20
>> http://svnweb.freebsd.org/base?view=3Drevision&revision=3D239672
>>=20
>> Which seems like it could be relevant here to the general case of =
both
>> ends of the connection entering FIN_WAIT_1 at the same time and
>> sending FIN/ACKs repeatedly (though our connections are a bizarre =
case
>> of this where both ends of the connection are actually the same
>> connection).
>>=20
>> Thanks,
>>=20
>> Matt
>> _______________________________________________
>> freebsd-net@freebsd.org mailing list
>> http://lists.freebsd.org/mailman/listinfo/freebsd-net
>> To unsubscribe, send any mail to =
"freebsd-net-unsubscribe@freebsd.org"
>>=20
>>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From trammell@tik.ee.ethz.ch  Mon Jul 22 08:03:26 2013
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To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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hi Richard,

Thanks for the comments! Inline...

On 16 Jul 2013, at 14:55 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:

> Brian, Mirja,
>=20
>=20
> Having read this draft (I think the rationale for not probing in the =
IW is a good one, but perhaps a little more text in the draft would do =
good - i.e. a short flow will be effectively non-reactive anyway, thus =
any ECN signal would be superfluous), I like it.
>=20
>=20
> I have one (probably academic) point though:
>=20
>=20
> (I would like to see a diagram visualizing the probing process).

Good suggestion; we'll add one in the next rev.

> In Step 4, the algo sends the next 3 segments after IW with "CE". In =
addition to the "first hop must not have experienced congestion" =
scenario, a middlebox might also clear CE if these three segments are =
pure ACKs, as per 3168, pure ACKs should not be marked ECN-capable... =
also, you expect an ACK for these. Thus you need to send 3 "data =
segments" not simply segments, right?

Yep, this is what we meant, but we need to make this more clear.

> In Step 6, you mention an interaction with ECN Nonce... Perhaps having =
an explicit example for an ECN-Nonce enabled session would do well =
(which can detect the removal of the CE-flag in probing data segments by =
sending one ECT(1) data segment).
>=20
> In Step 7, CWR should be sent with the segment (not necessarily data =
segment) ACKing any of the 1st up to the 3rd probing CE data segment. =
This could be 1, 2 or 3 segments (the text only hints on a single such =
segment).

hm, that wasn't the intention, but on review that could be clearer, as =
well. We'll make it so.

> Including the optional ECN Nonce probing (4th segment with ECT1; 4th =
ACK with NS=3D1) in the algorithm might be good...

Good suggestion; we were originally thinking of something a little more =
elaborate, and I do think we need to think about this a bit more, =
though, since an actual CE mark would obliterate the nonce. I know =
that's unlikely _now_, but one of the goals of the work is making it =
less so. :)

Cheers,

Brian


>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf =
Of
>> Brian Trammell
>> Sent: Donnerstag, 04. Juli 2013 14:14
>> To: tcpm@ietf.org
>> Subject: [tcpm] Fwd: New Version Notification for =
draft-kuehlewind-tcpm-
>> ecn-fallback-00.txt
>>=20
>> Greetings, all,
>>=20
>> We've posted a draft on ECN path probing and fallback, which we'd =
like to
>> discuss at the TCPM meeting in Berlin. In a recent work [1], we found =
that
>> though the ECN-readiness of popular webservers has significantly =
increased
>> even in the last couple of years, there still exist paths on which
>> attempts to use ECN after successful ECN negotiation will cause =
connection
>> disruption.
>>=20
>> This draft proposes an experimental approach to determine at runtime
>> whether a path is usable, by sending segments after connection =
startup and
>> ECN negotiation with the CE codepoint set, and disabling ECN if all =
probe
>> segments were lost, on a per-flow or per-destination basis.
>>=20
>> Experiments with an implementation of the approach within the Linux =
kernel
>> are underway; we plan to be able to present initial results in =
Berlin.
>>=20
>> Best regards,
>>=20
>> Mirja and Brian
>>=20
>> [1] Kuehlewind, M., Neuner, S., and Trammell, B., "On the state of =
ECN and
>> TCP Options on the Internet", in Proceedings of the the 2013 Passive =
and
>> Active Measurement Conference, Hong Kong SAR, China, 19 March 2013.
>>=20
>> Begin forwarded message:
>>=20
>>> From: internet-drafts@ietf.org
>>> Subject: New Version Notification for
>>> draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>> Date: 3 July 2013 17:40:32 GMT+02:00
>>> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, Brian
>>> Trammell <trammell@tik.ee.ethz.ch>
>>>=20
>>>=20
>>> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>> has been successfully submitted by Mirja Kuehlewind and posted to =
the
>>> IETF repository.
>>>=20
>>> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
>>> Revision:	 00
>>> Title:		 A Mechanism for ECN Path Probing and Fallback
>>> Creation date:	 2013-07-02
>>> Group:		 Individual Submission
>>> Number of pages: 5
>>> URL:             =
http://www.ietf.org/internet-drafts/draft-kuehlewind-
>> tcpm-ecn-fallback-00.txt
>>> Status:          =
http://datatracker.ietf.org/doc/draft-kuehlewind-tcpm-
>> ecn-fallback
>>> Htmlized:        =
http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-
>> fallback-00
>>>=20
>>>=20
>>> Abstract:
>>>  Explicit Congestion Notification (ECN) is a TCP/IP extension that =
is
>>>  widely implemented but hardly used due to the perceived unusablilty
>>>  of ECN on many paths through the Internet caused by ECN-ignorant
>>>  routers and middleboxes.  This document specifies an ECN probing =
and
>>>  fall-back mechanism in case ECN has be successfully negotiated
>>>  between two connection endpoints, but might not be usable on the
>>>  path.
>>>=20
>>>=20
>>>=20
>>>=20
>>> The IETF Secretariat
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


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Date: Mon, 22 Jul 2013 08:35:50 -0700
From: Joe Touch <touch@isi.edu>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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On 7/22/2013 7:17 AM, David Borman wrote:
> Testing that a self-connected socket works should be part of every TCP regression test.

 From RFC793:

     To allow for many processes within a single Host to use TCP
     communication facilities simultaneously, the TCP provides a set of
     addresses or ports within each host.  Concatenated with the network
     and host addresses from the internet communication layer, this forms
     a socket.  A pair of sockets uniquely identifies each connection.

This text says "a pair". I interpreted that as prohibiting use of a 
single socket for both ends, but I suppose you could allow it to happen.

The benefit of that situation is it would test simultaneous open and 
close, but note that some docs have written those off lately when we 
think that "can't" happen.

You'd have to prove that every protocol could handle simultaneous cases 
and state.

The key question is "why would this ever happen, or should it ever 
reasonably happen"? It's just as easily handled by having the Un*x 
socket "lie" about having TCP and just copy the buffer from send to 
receive - and what, really, is the point of that?

Joe

>
> 			-David Borman
>
> On Jul 22, 2013, at 12:02 AM, Joe Touch <touch@isi.edu> wrote:
>
>> Same src/dst for the same IP makes no sense; there need to be two ends to a connection, and each end is supposed to be uniquely determined by the socket (as defined in 793, not Un*x).
>>
>> IMO, it ought to be rejected by the API, just as would be one that was otherwise incompletely or incorrectly specified (picking a source address not on a local interface, picking a port range you don't have privilege to access, etc.).
>>
>> However, loopback is a subnet (127.0.0.0/8), not just a single address. It ought to be feasible and correct to open a connection to yourself on the same port on different loopback addresses.
>>
>> Joe
>>
>> On 7/21/2013 9:47 AM, Fernando Gont wrote:
>>> Folks,
>>>
>>> Found this by chance -- probably a datapoint that advice is needed in
>>> this area (that is, draft-gont-tcpm-tcp-seq-validation).
>>>
>>> P.S.: Will present results from real-world testing at the next tcpm meeting.
>>>
>>> Cheers,
>>> Fernando
>>>
>>>
>>>
>>>
>>> -------- Original Message --------
>>> From: Matt Miller <matt@matthewjmiller.net>
>>> Date: Wed, 17 Jul 2013 07:08:26 -0400
>>> X-Google-Sender-Auth: ba5SYKKiksigElenhewyH0EffCs
>>> Message-ID:
>>> <CAFc6gu_q1X10EzsHrvnmYuQ0ZnKz9uNXbfJJe-guva6J-QKAow@mail.gmail.com>
>>> Subject: TCP Loopback Connections with the Same Src/Dest Port
>>> To: FreeBSD Net <freebsd-net@freebsd.org>
>>>
>>> Our system is based on FreeBSD 8.1.  In some tests, we were having
>>> issues caused by connections of this form (more details below):
>>>
>>> TCP4      0      0      0/   0/   0    127.0.0.1.665   127.0.0.1.665
>>> FIN_WAIT_1
>>> TCP4      0      0      0/   0/   0    127.0.0.1.637   127.0.0.1.637
>>> FIN_WAIT_1
>>> TCP4      0      0      0/   0/   0    127.0.0.1.648   127.0.0.1.648
>>> FIN_WAIT_1
>>>
>>> Some questions we had:
>>>
>>> - Has anyone else ever seen these same src/dest address/port TCP
>>> connections created?  Does anyone know of a legitimate reason why they
>>> should be allowed?
>>>
>>> - If there are no known use cases for this type of connection, does
>>> anyone have more context/insight on the design here: should this type
>>> of inpcb creation be prevented in the kernel or is it the
>>> application's responsibility to ensure it never creates this type of
>>> socket?
>>>
>>> For those interested, more details of the issue seen follow.  The
>>> connection seems to get stuck in swi_net sending and receiving pure
>>> FIN/ACKs to itself:
>>>
>>> #12 0xffffffff804372ce in ip_output (m=0xffffff0003ccf300,
>>> opt=<optimized out>, ro=0xffffff8020c2b6a0, flags=0, imo=0x0,
>>> inp=0xffffff0019933968) at ../../../../sys/netinet/ip_output.c
>>> #13 0xffffffff804423dc in tcp_output (tp=0xffffff0019de2370) at
>>> ../../../../sys/netinet/tcp_output.c
>>> #14 0xffffffff8043ef5d in tcp_do_segment (m=0xffffff0019af1200,
>>> th=0x100200, so=0xffffff011ac59570, tp=0xffffff0019de2370,
>>> drop_hdrlen=52, tlen=0, iptos=0 '\000', ti_locked=3) at
>>> ../../../../sys/netinet/tcp_input.c
>>> #15 0xffffffff80440311 in tcp_input (m=0xffffff0019af1200,
>>> off0=<optimized out>) at ../../../../sys/netinet/tcp_input.c
>>> #16 0xffffffff8043530b in ip_input (m=0xffffff0019af1200) at
>>> ../../../../sys/netinet/ip_input.c
>>> #17 0xffffffff8040889f in netisr_process_workstream_proto
>>> (proto=<optimized out>, nwsp=<optimized out>) at
>>> ../../../../sys/net/netisr.c
>>> #18 swi_net (arg=0xffffffff80f59800) at ../../../../sys/net/netisr.c
>>>
>>> swi_net() just continues in this loop, ad nauseam:
>>>
>>> 759         while ((bits = nwsp->nws_pendingbits) != 0) {
>>> 760                 while ((prot = ffs(bits)) != 0) {
>>> 761                         prot--;
>>> 762                         bits &= ~(1 << prot);
>>> 763                         (void)netisr_process_workstream_proto(nwsp,
>>> prot);
>>> 764                 }
>>> 765         }
>>>
>>> The tcp_output() being triggered in tcp_do_segment() in the case is
>>> the one show on line 2303 below:
>>>
>>> 2212         /*
>>> 2213          * In ESTABLISHED state: drop duplicate ACKs; ACK out of range
>>> 2214          * ACKs.  If the ack is in the range
>>> 2215          *      tp->snd_una < th->th_ack <= tp->snd_max
>>> 2216          * then advance tp->snd_una to th->th_ack and drop
>>> 2217          * data from the retransmission queue.  If this ACK reflects
>>> 2218          * more up to date window information we update our
>>> window information.
>>> 2219          */
>>> 2220         case TCPS_ESTABLISHED:
>>> 2221         case TCPS_FIN_WAIT_1:
>>> 2222         case TCPS_FIN_WAIT_2:
>>> 2223         case TCPS_CLOSE_WAIT:
>>> 2224         case TCPS_CLOSING:
>>> 2225         case TCPS_LAST_ACK:
>>> 2226                 if (SEQ_GT(th->th_ack, tp->snd_max)) {
>>> 2227                         TCPSTAT_INC(tcps_rcvacktoomuch);
>>> 2228                         goto dropafterack;
>>> 2229                 }
>>> ...
>>> 2234                 if (SEQ_LEQ(th->th_ack, tp->snd_una)) {
>>> ...
>>> 2248                         if (tlen == 0 && tiwin == tp->snd_wnd) {
>>> 2249                                 TCPSTAT_INC(tcps_rcvdupack);
>>> ...
>>> 2277                                 if (!tcp_timer_active(tp, TT_REXMT) ||
>>> 2278                                     th->th_ack != tp->snd_una)
>>> 2279                                         tp->t_dupacks = 0;
>>> 2280                                 else if (++tp->t_dupacks >
>>> tcprexmtthresh ||
>>> 2281                                     ((V_tcp_do_newreno ||
>>> 2282                                       (tp->t_flags &
>>> TF_SACK_PERMIT)) &&
>>> 2283                                      IN_FASTRECOVERY(tp))) {
>>> 2284                                         if ((tp->t_flags &
>>> TF_SACK_PERMIT) &&
>>> 2285                                             IN_FASTRECOVERY(tp)) {
>>> 2286                                                 int awnd;
>>> 2287
>>> 2288                                                 /*
>>> 2289                                                  * Compute the
>>> amount of data in flight first.
>>> 2290                                                  * We can inject
>>> new data into the pipe iff
>>> 2291                                                  * we have less
>>> than 1/2 the original window's
>>> 2292                                                  * worth of data in
>>> flight.
>>> 2293                                                  */
>>> 2294                                                 awnd =
>>> (tp->snd_nxt - tp->snd_fack) +
>>> 2295
>>> tp->sackhint.sack_bytes_rexmit;
>>> 2296                                                 if (awnd <
>>> tp->snd_ssthresh) {
>>> 2297
>>> tp->snd_cwnd += tp->t_maxseg;
>>> 2298                                                         if
>>> (tp->snd_cwnd > tp->snd_ssthresh)
>>> 2299
>>> tp->snd_cwnd = tp->snd_ssthresh;
>>> 2300                                                 }
>>> 2301                                         } else
>>> 2302                                                 tp->snd_cwnd +=
>>> tp->t_maxseg;
>>> 2303                                         (void) tcp_output(tp);
>>> 2304                                         goto drop;
>>>
>>> I've noticed that we don't yet have this patch in our code:
>>>
>>> http://svnweb.freebsd.org/base?view=revision&revision=239672
>>>
>>> Which seems like it could be relevant here to the general case of both
>>> ends of the connection entering FIN_WAIT_1 at the same time and
>>> sending FIN/ACKs repeatedly (though our connections are a bizarre case
>>> of this where both ends of the connection are actually the same
>>> connection).
>>>
>>> Thanks,
>>>
>>> Matt
>>> _______________________________________________
>>> freebsd-net@freebsd.org mailing list
>>> http://lists.freebsd.org/mailman/listinfo/freebsd-net
>>> To unsubscribe, send any mail to "freebsd-net-unsubscribe@freebsd.org"
>>>
>>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>

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Hi Brian,

> > Including the optional ECN Nonce probing (4th segment with ECT1; 4th AC=
K
> with NS=3D1) in the algorithm might be good...
>=20
> Good suggestion; we were originally thinking of something a little more
> elaborate, and I do think we need to think about this a bit more, though,
> since an actual CE mark would obliterate the nonce. I know that's unlikel=
y
> _now_, but one of the goals of the work is making it less so. :)


Indeed... You would want to send the ECT(1) first, and the artificial CEs a=
fterwards; if there is an ACK,ECE for the segment with ECT(1), this would a=
lso prove the path is ECN-capable, right?

But I'd like to listen to your elaborate scheme :)


Richard Scheffenegger



> -----Original Message-----
> From: Brian Trammell [mailto:trammell@tik.ee.ethz.ch]
> Sent: Montag, 22. Juli 2013 17:03
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org; Mirja Kuehlewind
> Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-
> ecn-fallback-00.txt
>=20
> hi Richard,
>=20
> Thanks for the comments! Inline...
>=20
> On 16 Jul 2013, at 14:55 , "Scheffenegger, Richard" <rs@netapp.com> wrote=
:
>=20
> > Brian, Mirja,
> >
> >
> > Having read this draft (I think the rationale for not probing in the IW
> is a good one, but perhaps a little more text in the draft would do good =
-
> i.e. a short flow will be effectively non-reactive anyway, thus any ECN
> signal would be superfluous), I like it.
> >
> >
> > I have one (probably academic) point though:
> >
> >
> > (I would like to see a diagram visualizing the probing process).
>=20
> Good suggestion; we'll add one in the next rev.
>=20
> > In Step 4, the algo sends the next 3 segments after IW with "CE". In
> addition to the "first hop must not have experienced congestion" scenario=
,
> a middlebox might also clear CE if these three segments are pure ACKs, as
> per 3168, pure ACKs should not be marked ECN-capable... also, you expect
> an ACK for these. Thus you need to send 3 "data segments" not simply
> segments, right?
>=20
> Yep, this is what we meant, but we need to make this more clear.
>=20
> > In Step 6, you mention an interaction with ECN Nonce... Perhaps having
> an explicit example for an ECN-Nonce enabled session would do well (which
> can detect the removal of the CE-flag in probing data segments by sending
> one ECT(1) data segment).
> >
> > In Step 7, CWR should be sent with the segment (not necessarily data
> segment) ACKing any of the 1st up to the 3rd probing CE data segment. Thi=
s
> could be 1, 2 or 3 segments (the text only hints on a single such
> segment).
>=20
> hm, that wasn't the intention, but on review that could be clearer, as
> well. We'll make it so.
>=20
> > Including the optional ECN Nonce probing (4th segment with ECT1; 4th AC=
K
> with NS=3D1) in the algorithm might be good...
>=20
> Good suggestion; we were originally thinking of something a little more
> elaborate, and I do think we need to think about this a bit more, though,
> since an actual CE mark would obliterate the nonce. I know that's unlikel=
y
> _now_, but one of the goals of the work is making it less so. :)
>=20
> Cheers,
>=20
> Brian
>=20
>=20
> >> -----Original Message-----
> >> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf
> >> Of Brian Trammell
> >> Sent: Donnerstag, 04. Juli 2013 14:14
> >> To: tcpm@ietf.org
> >> Subject: [tcpm] Fwd: New Version Notification for
> >> draft-kuehlewind-tcpm- ecn-fallback-00.txt
> >>
> >> Greetings, all,
> >>
> >> We've posted a draft on ECN path probing and fallback, which we'd
> >> like to discuss at the TCPM meeting in Berlin. In a recent work [1],
> >> we found that though the ECN-readiness of popular webservers has
> >> significantly increased even in the last couple of years, there still
> >> exist paths on which attempts to use ECN after successful ECN
> >> negotiation will cause connection disruption.
> >>
> >> This draft proposes an experimental approach to determine at runtime
> >> whether a path is usable, by sending segments after connection
> >> startup and ECN negotiation with the CE codepoint set, and disabling
> >> ECN if all probe segments were lost, on a per-flow or per-destination
> basis.
> >>
> >> Experiments with an implementation of the approach within the Linux
> >> kernel are underway; we plan to be able to present initial results in
> Berlin.
> >>
> >> Best regards,
> >>
> >> Mirja and Brian
> >>
> >> [1] Kuehlewind, M., Neuner, S., and Trammell, B., "On the state of
> >> ECN and TCP Options on the Internet", in Proceedings of the the 2013
> >> Passive and Active Measurement Conference, Hong Kong SAR, China, 19
> March 2013.
> >>
> >> Begin forwarded message:
> >>
> >>> From: internet-drafts@ietf.org
> >>> Subject: New Version Notification for
> >>> draft-kuehlewind-tcpm-ecn-fallback-00.txt
> >>> Date: 3 July 2013 17:40:32 GMT+02:00
> >>> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, Brian
> >>> Trammell <trammell@tik.ee.ethz.ch>
> >>>
> >>>
> >>> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
> >>> has been successfully submitted by Mirja Kuehlewind and posted to
> >>> the IETF repository.
> >>>
> >>> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
> >>> Revision:	 00
> >>> Title:		 A Mechanism for ECN Path Probing and Fallback
> >>> Creation date:	 2013-07-02
> >>> Group:		 Individual Submission
> >>> Number of pages: 5
> >>> URL:             http://www.ietf.org/internet-drafts/draft-kuehlewind=
-
> >> tcpm-ecn-fallback-00.txt
> >>> Status:          http://datatracker.ietf.org/doc/draft-kuehlewind-
> tcpm-
> >> ecn-fallback
> >>> Htmlized:        http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn=
-
> >> fallback-00
> >>>
> >>>
> >>> Abstract:
> >>>  Explicit Congestion Notification (ECN) is a TCP/IP extension that
> >>> is  widely implemented but hardly used due to the perceived
> >>> unusablilty  of ECN on many paths through the Internet caused by
> >>> ECN-ignorant  routers and middleboxes.  This document specifies an
> >>> ECN probing and  fall-back mechanism in case ECN has be successfully
> >>> negotiated  between two connection endpoints, but might not be
> >>> usable on the  path.
> >>>
> >>>
> >>>
> >>>
> >>> The IETF Secretariat
> >>
> >> _______________________________________________
> >> 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, 22 Jul 2013 10:18:31 -0700
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, HTTP Working Group <ietf-http-wg@w3.org>
Subject: Re: [tcpm] Berlin and Hamburg Agendas
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+tcpm

On Mon, Jul 22, 2013 at 5:47 AM, Mark Nottingham <mnot@mnot.net> wrote:
> I've uploaded agendas for Berlin:
>   http://trac.tools.ietf.org/wg/httpbis/trac/browser/wg_materials/ietf87/=
agenda.txt
>
> ... and Hamburg:
>   https://github.com/http2/wg_materials/blob/master/interim-13-08/agenda.=
md
>
> As always, suggestions are welcome.
>
> Note that in Berlin, we have a session on Wednesday, where we'll talk abo=
ut our usual topics, and another on Friday, where we'll have a joint meetin=
g with the Transport area.
>
> I've purposefully kept the Hamburg agenda loose, because what we discuss =
there is going to depend a lot upon how far implementation has gotten. I wi=
ll be updating the agenda with a more targeted list of issues for us to att=
ack (as we did in San Francisco).
>
> Regards,
>
> --
> Mark Nottingham   http://www.mnot.net/
>
>
>
>

From michael.scharf@alcatel-lucent.com  Mon Jul 22 10:34:20 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: HTTP/2 and Transport Joint Meeting
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Hi all,

To make this slightly more explicit: In Berlin, there will be a HTTP/2 and =
Transport Joint Meeting. It is currently scheduled for Friday morning.

As you can see in the agenda for the Berlin meeting (referenced below), thi=
s discussion is highly related to TCPM topics. Folks, please consider atten=
ding this session!

Michael


-----Original Message-----
From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of Yuc=
hung Cheng
Sent: Monday, July 22, 2013 7:19 PM
To: Mark Nottingham
Cc: tcpm@ietf.org Extensions; HTTP Working Group
Subject: Re: [tcpm] Berlin and Hamburg Agendas

+tcpm

On Mon, Jul 22, 2013 at 5:47 AM, Mark Nottingham <mnot@mnot.net> wrote:
> I've uploaded agendas for Berlin:
>  =20
> http://trac.tools.ietf.org/wg/httpbis/trac/browser/wg_materials/ietf87
> /agenda.txt
>
> ... and Hamburg:
>  =20
> https://github.com/http2/wg_materials/blob/master/interim-13-08/agenda
> .md
>
> As always, suggestions are welcome.
>
> Note that in Berlin, we have a session on Wednesday, where we'll talk abo=
ut our usual topics, and another on Friday, where we'll have a joint meetin=
g with the Transport area.
>
> I've purposefully kept the Hamburg agenda loose, because what we discuss =
there is going to depend a lot upon how far implementation has gotten. I wi=
ll be updating the agenda with a more targeted list of issues for us to att=
ack (as we did in San Francisco).
>
> Regards,
>
> --
> Mark Nottingham   http://www.mnot.net/
>
>
>
>
_______________________________________________
tcpm mailing list
tcpm@ietf.org
https://www.ietf.org/mailman/listinfo/tcpm

From dab@weston.borman.com  Mon Jul 22 10:49:36 2013
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From: David Borman <dab@weston.borman.com>
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Date: Mon, 22 Jul 2013 12:49:25 -0500
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To: Joe Touch <touch@ISI.EDU>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] TCP Loopback Connections with the Same Src/Dest Port
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On Jul 22, 2013, at 10:35 AM, Joe Touch <touch@ISI.EDU> wrote:

>=20
>=20
> On 7/22/2013 7:17 AM, David Borman wrote:
>> Testing that a self-connected socket works should be part of every =
TCP regression test.
>=20
> =46rom RFC793:
>=20
>    To allow for many processes within a single Host to use TCP
>    communication facilities simultaneously, the TCP provides a set of
>    addresses or ports within each host.  Concatenated with the network
>    and host addresses from the internet communication layer, this =
forms
>    a socket.  A pair of sockets uniquely identifies each connection.
>=20
> This text says "a pair". I interpreted that as prohibiting use of a =
single socket for both ends, but I suppose you could allow it to happen.

Yeah..., I don't make that interpretation.  Logically it's still a pair, =
it just happens to be that the same socket is being used for both sides =
of the pair.  Self connected TCP sockets have been around and working =
for a long time (they worked in CSRG BSD, I know because I fixed 'em), =
but people make changes to code and don't test for self-connected =
sockets, so they can get broken. And when things break for self =
connected sockets, the same situation can happen with a pair of sockets. =
 You can test all the same cases with a pair of sockets, it just takes =
more work to get the timing just right.

>=20
> The benefit of that situation is it would test simultaneous open and =
close, but note that some docs have written those off lately when we =
think that "can't" happen.

And that's when the bugs creep in. :-)

>=20
> You'd have to prove that every protocol could handle simultaneous =
cases and state.

For every protocol used by sockets?  No, this is only about TCP sockets, =
not sockets in general.  Different protocols have different semantics, =
it doesn't matter what other protocols will or will not allow.

>=20
> The key question is "why would this ever happen, or should it ever =
reasonably happen"? It's just as easily handled by having the Un*x =
socket "lie" about having TCP and just copy the buffer from send to =
receive - and what, really, is the point of that?

I disagree, that question is a red herring.  It doesn't matter whether =
or not you have an application use for it, it ought to work, and as a =
test tool it's an easy way to verify that crossing SYNs and crossing =
FINs are being handled properly.

			-David Borman

P.S. A use case?  Besides testing, the only semi-useful case that pops =
into my mind would be to pass data across an exec() call in lieu of =
using a tmp file, but you can also do that with a pair of sockets and a =
little more work.

>=20
> Joe
>=20
>>=20
>> 			-David Borman
>>=20
>> On Jul 22, 2013, at 12:02 AM, Joe Touch <touch@isi.edu> wrote:
>>=20
>>> Same src/dst for the same IP makes no sense; there need to be two =
ends to a connection, and each end is supposed to be uniquely determined =
by the socket (as defined in 793, not Un*x).
>>>=20
>>> IMO, it ought to be rejected by the API, just as would be one that =
was otherwise incompletely or incorrectly specified (picking a source =
address not on a local interface, picking a port range you don't have =
privilege to access, etc.).
>>>=20
>>> However, loopback is a subnet (127.0.0.0/8), not just a single =
address. It ought to be feasible and correct to open a connection to =
yourself on the same port on different loopback addresses.
>>>=20
>>> Joe
>>>=20
>>> On 7/21/2013 9:47 AM, Fernando Gont wrote:
>>>> Folks,
>>>>=20
>>>> Found this by chance -- probably a datapoint that advice is needed =
in
>>>> this area (that is, draft-gont-tcpm-tcp-seq-validation).
>>>>=20
>>>> P.S.: Will present results from real-world testing at the next tcpm =
meeting.
>>>>=20
>>>> Cheers,
>>>> Fernando
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> -------- Original Message --------
>>>> From: Matt Miller <matt@matthewjmiller.net>
>>>> Date: Wed, 17 Jul 2013 07:08:26 -0400
>>>> X-Google-Sender-Auth: ba5SYKKiksigElenhewyH0EffCs
>>>> Message-ID:
>>>> =
<CAFc6gu_q1X10EzsHrvnmYuQ0ZnKz9uNXbfJJe-guva6J-QKAow@mail.gmail.com>
>>>> Subject: TCP Loopback Connections with the Same Src/Dest Port
>>>> To: FreeBSD Net <freebsd-net@freebsd.org>
>>>>=20
>>>> Our system is based on FreeBSD 8.1.  In some tests, we were having
>>>> issues caused by connections of this form (more details below):
>>>>=20
>>>> TCP4      0      0      0/   0/   0    127.0.0.1.665   =
127.0.0.1.665
>>>> FIN_WAIT_1
>>>> TCP4      0      0      0/   0/   0    127.0.0.1.637   =
127.0.0.1.637
>>>> FIN_WAIT_1
>>>> TCP4      0      0      0/   0/   0    127.0.0.1.648   =
127.0.0.1.648
>>>> FIN_WAIT_1
>>>>=20
>>>> Some questions we had:
>>>>=20
>>>> - Has anyone else ever seen these same src/dest address/port TCP
>>>> connections created?  Does anyone know of a legitimate reason why =
they
>>>> should be allowed?
>>>>=20
>>>> - If there are no known use cases for this type of connection, does
>>>> anyone have more context/insight on the design here: should this =
type
>>>> of inpcb creation be prevented in the kernel or is it the
>>>> application's responsibility to ensure it never creates this type =
of
>>>> socket?
>>>>=20
>>>> For those interested, more details of the issue seen follow.  The
>>>> connection seems to get stuck in swi_net sending and receiving pure
>>>> FIN/ACKs to itself:
>>>>=20
>>>> #12 0xffffffff804372ce in ip_output (m=3D0xffffff0003ccf300,
>>>> opt=3D<optimized out>, ro=3D0xffffff8020c2b6a0, flags=3D0, imo=3D0x0,=

>>>> inp=3D0xffffff0019933968) at ../../../../sys/netinet/ip_output.c
>>>> #13 0xffffffff804423dc in tcp_output (tp=3D0xffffff0019de2370) at
>>>> ../../../../sys/netinet/tcp_output.c
>>>> #14 0xffffffff8043ef5d in tcp_do_segment (m=3D0xffffff0019af1200,
>>>> th=3D0x100200, so=3D0xffffff011ac59570, tp=3D0xffffff0019de2370,
>>>> drop_hdrlen=3D52, tlen=3D0, iptos=3D0 '\000', ti_locked=3D3) at
>>>> ../../../../sys/netinet/tcp_input.c
>>>> #15 0xffffffff80440311 in tcp_input (m=3D0xffffff0019af1200,
>>>> off0=3D<optimized out>) at ../../../../sys/netinet/tcp_input.c
>>>> #16 0xffffffff8043530b in ip_input (m=3D0xffffff0019af1200) at
>>>> ../../../../sys/netinet/ip_input.c
>>>> #17 0xffffffff8040889f in netisr_process_workstream_proto
>>>> (proto=3D<optimized out>, nwsp=3D<optimized out>) at
>>>> ../../../../sys/net/netisr.c
>>>> #18 swi_net (arg=3D0xffffffff80f59800) at =
../../../../sys/net/netisr.c
>>>>=20
>>>> swi_net() just continues in this loop, ad nauseam:
>>>>=20
>>>> 759         while ((bits =3D nwsp->nws_pendingbits) !=3D 0) {
>>>> 760                 while ((prot =3D ffs(bits)) !=3D 0) {
>>>> 761                         prot--;
>>>> 762                         bits &=3D ~(1 << prot);
>>>> 763                         =
(void)netisr_process_workstream_proto(nwsp,
>>>> prot);
>>>> 764                 }
>>>> 765         }
>>>>=20
>>>> The tcp_output() being triggered in tcp_do_segment() in the case is
>>>> the one show on line 2303 below:
>>>>=20
>>>> 2212         /*
>>>> 2213          * In ESTABLISHED state: drop duplicate ACKs; ACK out =
of range
>>>> 2214          * ACKs.  If the ack is in the range
>>>> 2215          *      tp->snd_una < th->th_ack <=3D tp->snd_max
>>>> 2216          * then advance tp->snd_una to th->th_ack and drop
>>>> 2217          * data from the retransmission queue.  If this ACK =
reflects
>>>> 2218          * more up to date window information we update our
>>>> window information.
>>>> 2219          */
>>>> 2220         case TCPS_ESTABLISHED:
>>>> 2221         case TCPS_FIN_WAIT_1:
>>>> 2222         case TCPS_FIN_WAIT_2:
>>>> 2223         case TCPS_CLOSE_WAIT:
>>>> 2224         case TCPS_CLOSING:
>>>> 2225         case TCPS_LAST_ACK:
>>>> 2226                 if (SEQ_GT(th->th_ack, tp->snd_max)) {
>>>> 2227                         TCPSTAT_INC(tcps_rcvacktoomuch);
>>>> 2228                         goto dropafterack;
>>>> 2229                 }
>>>> ...
>>>> 2234                 if (SEQ_LEQ(th->th_ack, tp->snd_una)) {
>>>> ...
>>>> 2248                         if (tlen =3D=3D 0 && tiwin =3D=3D =
tp->snd_wnd) {
>>>> 2249                                 TCPSTAT_INC(tcps_rcvdupack);
>>>> ...
>>>> 2277                                 if (!tcp_timer_active(tp, =
TT_REXMT) ||
>>>> 2278                                     th->th_ack !=3D =
tp->snd_una)
>>>> 2279                                         tp->t_dupacks =3D 0;
>>>> 2280                                 else if (++tp->t_dupacks >
>>>> tcprexmtthresh ||
>>>> 2281                                     ((V_tcp_do_newreno ||
>>>> 2282                                       (tp->t_flags &
>>>> TF_SACK_PERMIT)) &&
>>>> 2283                                      IN_FASTRECOVERY(tp))) {
>>>> 2284                                         if ((tp->t_flags &
>>>> TF_SACK_PERMIT) &&
>>>> 2285                                             =
IN_FASTRECOVERY(tp)) {
>>>> 2286                                                 int awnd;
>>>> 2287
>>>> 2288                                                 /*
>>>> 2289                                                  * Compute the
>>>> amount of data in flight first.
>>>> 2290                                                  * We can =
inject
>>>> new data into the pipe iff
>>>> 2291                                                  * we have =
less
>>>> than 1/2 the original window's
>>>> 2292                                                  * worth of =
data in
>>>> flight.
>>>> 2293                                                  */
>>>> 2294                                                 awnd =3D
>>>> (tp->snd_nxt - tp->snd_fack) +
>>>> 2295
>>>> tp->sackhint.sack_bytes_rexmit;
>>>> 2296                                                 if (awnd <
>>>> tp->snd_ssthresh) {
>>>> 2297
>>>> tp->snd_cwnd +=3D tp->t_maxseg;
>>>> 2298                                                         if
>>>> (tp->snd_cwnd > tp->snd_ssthresh)
>>>> 2299
>>>> tp->snd_cwnd =3D tp->snd_ssthresh;
>>>> 2300                                                 }
>>>> 2301                                         } else
>>>> 2302                                                 tp->snd_cwnd =
+=3D
>>>> tp->t_maxseg;
>>>> 2303                                         (void) tcp_output(tp);
>>>> 2304                                         goto drop;
>>>>=20
>>>> I've noticed that we don't yet have this patch in our code:
>>>>=20
>>>> http://svnweb.freebsd.org/base?view=3Drevision&revision=3D239672
>>>>=20
>>>> Which seems like it could be relevant here to the general case of =
both
>>>> ends of the connection entering FIN_WAIT_1 at the same time and
>>>> sending FIN/ACKs repeatedly (though our connections are a bizarre =
case
>>>> of this where both ends of the connection are actually the same
>>>> connection).
>>>>=20
>>>> Thanks,
>>>>=20
>>>> Matt
>>>> _______________________________________________
>>>> freebsd-net@freebsd.org mailing list
>>>> http://lists.freebsd.org/mailman/listinfo/freebsd-net
>>>> To unsubscribe, send any mail to =
"freebsd-net-unsubscribe@freebsd.org"
>>>>=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


From touch@isi.edu  Mon Jul 22 11:03:02 2013
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On 7/22/2013 10:49 AM, David Borman wrote:
>> You'd have to prove that every protocol could handle simultaneous cases and state.
 >
> For every protocol used by sockets?  No, this is only about TCP
> sockets, not sockets in general.  Different protocols have different
> semantics, it doesn't matter what other protocols will or will not
> allow.

I meant for every TCP extension, modification, and option.

I've never ignored such cases, but I do recall others doing so (I'm not 
fully online today and can't easily grep for that, though).

Joe

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On 7/22/2013 10:49 AM, David Borman wrote:
> P.S. A use case?  Besides testing, the only semi-useful case that
> pops into my mind would be to pass data across an exec() call in lieu
> of using a tmp file, but you can also do that with a pair of sockets
> and a little more work.

If there are two processes, then there ought to be two different 
processes (or subprocesses if you prefer).

I.e., this is fine reason for 127.0.0.1:600 talking to 127.0.0.1.599, or 
for 127.0.0.1:600 talking to 127.0.0.2:600.

Joe

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On 7/22/2013 11:07 AM, Joe Touch wrote:
>
>
> On 7/22/2013 10:49 AM, David Borman wrote:
>> P.S. A use case?  Besides testing, the only semi-useful case that
>> pops into my mind would be to pass data across an exec() call in lieu
>> of using a tmp file, but you can also do that with a pair of sockets
>> and a little more work.
>
> If there are two processes, then there ought to be two different
> processes (or subprocesses if you prefer).

Let me try typing that again:

If there are two different processes (or subprocesses if you prefer), 
then there ought to be two *different* sockets.

> I.e., this is fine reason for 127.0.0.1:600 talking to 127.0.0.1.599, or
> for 127.0.0.1:600 talking to 127.0.0.2:600.
>
> Joe

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On Jul 22, 2013, at 1:16 PM, Joe Touch <touch@ISI.EDU> wrote:

>=20
>=20
> On 7/22/2013 11:07 AM, Joe Touch wrote:
>>=20
>>=20
>> On 7/22/2013 10:49 AM, David Borman wrote:
>>> P.S. A use case?  Besides testing, the only semi-useful case that
>>> pops into my mind would be to pass data across an exec() call in =
lieu
>>> of using a tmp file, but you can also do that with a pair of sockets
>>> and a little more work.
>>=20
>> If there are two processes, then there ought to be two different
>> processes (or subprocesses if you prefer).
>=20
> Let me try typing that again:
>=20
> If there are two different processes (or subprocesses if you prefer), =
then there ought to be two *different* sockets.

No, only one process, doing an exec(), no fork() involved.  Since all =
your data gets lost across an exec(), a typical way to pass data across =
an exec() is either as command line arguments to the exec() call or via =
a tmp file, depending on how much data you need to pass.
			-David

>=20
>> I.e., this is fine reason for 127.0.0.1:600 talking to 127.0.0.1.599, =
or
>> for 127.0.0.1:600 talking to 127.0.0.2:600.
>>=20
>> Joe
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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On 7/22/2013 11:27 AM, David Borman wrote:
>> >If there are two different processes (or subprocesses if you prefer), then there ought to be two*different*  sockets.
> No, only one process, doing an exec(), no fork() involved. Since all
> your data gets lost across an exec(), a typical way to pass data across
> an exec() is either as command line arguments to the exec() call or via
> a tmp file, depending on how much data you need to pass.

That's a potential argument for a self-(un*x)socket, but not a self-TCP 
socket.

There's no argument here why TCP needs to be involved.

Joe

From touch@isi.edu  Mon Jul 22 12:02:18 2013
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Subject: Re: [tcpm] TCP Loopback Connections with the Same Src/Dest Port
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Further, FWIW, I'd argue:

	- if you are the same process, a socket is not an appropriate
	place to "store" anything
		local variables for small things
		external storage for large things

		there's nothing you can store in a socket that you
		can't more efficiently store in local memory, within
		a single process

	- every doc I've read on exec() refers to the transition
	as either crossing processes (terminating the caller,
	starting a new process for the callee) or 'subprocesses'
	(when the caller doesn't terminate)
		that's not the "same process"

Not all corners are worth considering.

Joe

On 7/22/2013 11:52 AM, Joe Touch wrote:
>
>
> On 7/22/2013 11:27 AM, David Borman wrote:
>>> >If there are two different processes (or subprocesses if you
>>> prefer), then there ought to be two*different*  sockets.
>> No, only one process, doing an exec(), no fork() involved. Since all
>> your data gets lost across an exec(), a typical way to pass data across
>> an exec() is either as command line arguments to the exec() call or via
>> a tmp file, depending on how much data you need to pass.
>
> That's a potential argument for a self-(un*x)socket, but not a self-TCP
> socket.
>
> There's no argument here why TCP needs to be involved.
>
> Joe

From shep@xplot.org  Mon Jul 22 12:21:53 2013
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From: Tim Shepard <shep@alum.mit.edu>
To: Joe Touch <touch@isi.edu>
In-reply-to: Your message of Mon, 22 Jul 2013 12:00:53 -0700. <51ED8165.9020005@isi.edu> 
Date: Mon, 22 Jul 2013 15:21:19 -0400
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Subject: Re: [tcpm] TCP Loopback Connections with the Same Src/Dest Port
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> 	- every doc I've read on exec() refers to the transition
> 	as either crossing processes (terminating the caller,
> 	starting a new process for the callee) or 'subprocesses'
> 	(when the caller doesn't terminate)
> 		that's not the "same process"

No, exec() replaces the program (the executable) in the current
process.  It has always been that way as long as I can remember in
all the various flavors of unix and unix-like systems I've used.  I
just looked and the wikipedia article on exec seems to describe it
correctly.



And I see no reason to prohibit self-connection of a TCP, and I like
David Bormon's observation about it being an important test case in
the test suite.

The semantics of self-connection seem straightforward to me.  Works
the same as a self-connected TCP implemention via a mirror host.
(Mirror host is a testing hack where the mirror host at some IP
address out there simply swaps source and destination ports and IP
addresses in every packet it receives and sends the packet back out
which, I'm told, was a very useful testing hack in the early days
of TCP).

			-Tim Shepard
			 shep@alum.mit.edu

From Michael.Tuexen@lurchi.franken.de  Mon Jul 22 12:50:34 2013
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On Jul 22, 2013, at 9:21 PM, Tim Shepard <shep@alum.mit.edu> wrote:

> 
> 
>> 	- every doc I've read on exec() refers to the transition
>> 	as either crossing processes (terminating the caller,
>> 	starting a new process for the callee) or 'subprocesses'
>> 	(when the caller doesn't terminate)
>> 		that's not the "same process"
> 
> No, exec() replaces the program (the executable) in the current
> process.  It has always been that way as long as I can remember in
> all the various flavors of unix and unix-like systems I've used.  I
> just looked and the wikipedia article on exec seems to describe it
> correctly.
> 
> 
> 
> And I see no reason to prohibit self-connection of a TCP, and I like
> David Bormon's observation about it being an important test case in
> the test suite.
While testing the SCTP stack of FreeBSD we also used this. It is
a very useful testcase...

Best regards
Michael
> 
> The semantics of self-connection seem straightforward to me.  Works
> the same as a self-connected TCP implemention via a mirror host.
> (Mirror host is a testing hack where the mirror host at some IP
> address out there simply swaps source and destination ports and IP
> addresses in every packet it receives and sends the packet back out
> which, I'm told, was a very useful testing hack in the early days
> of TCP).
> 
> 			-Tim Shepard
> 			 shep@alum.mit.edu
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 


From pasi.sarolahti@iki.fi  Mon Jul 22 12:56:14 2013
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References: <CAO249yfdHnz0vd2rBQXpQ4RJqo2R09r0X7_-b03fkMC_csV=Yw@mail.gmail.com> <012C3117EDDB3C4781FD802A8C27DD4F24BC11A2@SACEXCMBX02-PRD.hq.netapp.com> <CAO249ycGrS3HpqJvy1DUQV=_sqpmVHEk529SW+Z0cLEhxOARiQ@mail.gmail.com> <16682_1370353034_51ADED8A_16682_97_1_012C3117EDDB3C4781FD802A8C27DD4F24BC8FA7@SACEXCMBX02-PRD.hq.netapp.com> <DE23153F-9697-48FC-BFA1-9032739ACBA0@iki.fi> <012C3117EDDB3C4781FD802A8C27DD4F25C057ED@SACEXCMBX02-PRD.hq.netapp.com> <51EC79B0.5090903@isi.edu>
To: Joe Touch <touch@ISI.EDU>
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Subject: Re: [tcpm] some questions related to PAWS
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On Jul 22, 2013, at 3:15 AM, Joe Touch <touch@ISI.EDU> wrote:

> If PAWS is a "best effort" assurance, then I accept your conclusion.
>=20
> If PAWS is something we think provides a guarantee (and I do), then it =
is a mistake, and the previous MUST and MUST NOTs should be retained.

I think this is a good summary, and something we can hopefully conclude =
by the meeting (knowing that some people cannot be there, in which case =
you should make yourselves heard on the list).

> If you keep the SHOULD and SHOULD NOT below, you're effectively =
killing PAWS capability. You need to at least admit that in the text.

I agree that the text needs to be clear about the consequences of not =
following the receiver-side SHOULD (or MUST). Even if it was MUST, I =
think it would be good to have that text, because it seems that there =
are implementations that accept segments without timestamps.

But I think "killing PAWS capability" is too strongly said. SHOULD would =
open a window of unreliability to PAWS, making it best effort instead of =
guarantee, but it would still work in majority of cases.

- Pasi


From michael.scharf@alcatel-lucent.com  Tue Jul 23 01:42:28 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Subject: [tcpm] Slides please!
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Dear Presenters,

TCPM is scheduled for the Tuesday morning session. Please submit your slide=
s to the chairs by Monday 17:00 CEST. This is a hard deadline.

Given the full TCPM agenda, it would help a lot if an earlier draft version=
 of the slides could be shared with the chairs until Sunday 17:00 CEST.

Thanks

Michael, Pasi, Yoshifumi

From touch@isi.edu  Tue Jul 23 08:49:38 2013
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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Hi, all,

OK, I've been convinced that supporting such loopback isn't prohibited 
by 793, but I remain convinced that supporting it within the protocol 
isn't strictly required.

There are other cases where TCP implementations take shortcuts when it 
knows better - e.g., ignoring congestion control when both ends are on 
the same subnet. TCP could just as easily ignore the entire state 
machine when both sockets are identical and just reflect everything back 
in one step, or the Un*x socket layer could do so directly.

IMO, an app that wants to talk to itself ought to either use different 
ports or different looback addresses (or both) - though, given my 
experience with Linux of late (65K total connection limit), I'd be more 
surprised if that works too.

However, the case below also hints at a bug where both ends enter 
FIN-WAIT-1 at the same time - which can happen whether reflexive or not, 
and should be fixed anyway.

Joe

On 7/21/2013 10:02 PM, Joe Touch wrote:
> Same src/dst for the same IP makes no sense; there need to be two ends
> to a connection, and each end is supposed to be uniquely determined by
> the socket (as defined in 793, not Un*x).
>
> IMO, it ought to be rejected by the API, just as would be one that was
> otherwise incompletely or incorrectly specified (picking a source
> address not on a local interface, picking a port range you don't have
> privilege to access, etc.).
>
> However, loopback is a subnet (127.0.0.0/8), not just a single address.
> It ought to be feasible and correct to open a connection to yourself on
> the same port on different loopback addresses.
>
> Joe
>
> On 7/21/2013 9:47 AM, Fernando Gont wrote:
>> Folks,
>>
>> Found this by chance -- probably a datapoint that advice is needed in
>> this area (that is, draft-gont-tcpm-tcp-seq-validation).
>>
>> P.S.: Will present results from real-world testing at the next tcpm
>> meeting.
>>
>> Cheers,
>> Fernando
>>
>>
>>
>>
>> -------- Original Message --------
>> From: Matt Miller <matt@matthewjmiller.net>
>> Date: Wed, 17 Jul 2013 07:08:26 -0400
>> X-Google-Sender-Auth: ba5SYKKiksigElenhewyH0EffCs
>> Message-ID:
>> <CAFc6gu_q1X10EzsHrvnmYuQ0ZnKz9uNXbfJJe-guva6J-QKAow@mail.gmail.com>
>> Subject: TCP Loopback Connections with the Same Src/Dest Port
>> To: FreeBSD Net <freebsd-net@freebsd.org>
>>
>> Our system is based on FreeBSD 8.1.  In some tests, we were having
>> issues caused by connections of this form (more details below):
>>
>>   TCP4      0      0      0/   0/   0    127.0.0.1.665   127.0.0.1.665
>>   FIN_WAIT_1
>>   TCP4      0      0      0/   0/   0    127.0.0.1.637   127.0.0.1.637
>>   FIN_WAIT_1
>>   TCP4      0      0      0/   0/   0    127.0.0.1.648   127.0.0.1.648
>>   FIN_WAIT_1
>>
>> Some questions we had:
>>
>> - Has anyone else ever seen these same src/dest address/port TCP
>> connections created?  Does anyone know of a legitimate reason why they
>> should be allowed?
>>
>> - If there are no known use cases for this type of connection, does
>> anyone have more context/insight on the design here: should this type
>> of inpcb creation be prevented in the kernel or is it the
>> application's responsibility to ensure it never creates this type of
>> socket?
>>
>> For those interested, more details of the issue seen follow.  The
>> connection seems to get stuck in swi_net sending and receiving pure
>> FIN/ACKs to itself:
>>
>> #12 0xffffffff804372ce in ip_output (m=0xffffff0003ccf300,
>> opt=<optimized out>, ro=0xffffff8020c2b6a0, flags=0, imo=0x0,
>> inp=0xffffff0019933968) at ../../../../sys/netinet/ip_output.c
>> #13 0xffffffff804423dc in tcp_output (tp=0xffffff0019de2370) at
>> ../../../../sys/netinet/tcp_output.c
>> #14 0xffffffff8043ef5d in tcp_do_segment (m=0xffffff0019af1200,
>> th=0x100200, so=0xffffff011ac59570, tp=0xffffff0019de2370,
>> drop_hdrlen=52, tlen=0, iptos=0 '\000', ti_locked=3) at
>> ../../../../sys/netinet/tcp_input.c
>> #15 0xffffffff80440311 in tcp_input (m=0xffffff0019af1200,
>> off0=<optimized out>) at ../../../../sys/netinet/tcp_input.c
>> #16 0xffffffff8043530b in ip_input (m=0xffffff0019af1200) at
>> ../../../../sys/netinet/ip_input.c
>> #17 0xffffffff8040889f in netisr_process_workstream_proto
>> (proto=<optimized out>, nwsp=<optimized out>) at
>> ../../../../sys/net/netisr.c
>> #18 swi_net (arg=0xffffffff80f59800) at ../../../../sys/net/netisr.c
>>
>> swi_net() just continues in this loop, ad nauseam:
>>
>>   759         while ((bits = nwsp->nws_pendingbits) != 0) {
>>   760                 while ((prot = ffs(bits)) != 0) {
>>   761                         prot--;
>>   762                         bits &= ~(1 << prot);
>>   763                         (void)netisr_process_workstream_proto(nwsp,
>> prot);
>>   764                 }
>>   765         }
>>
>> The tcp_output() being triggered in tcp_do_segment() in the case is
>> the one show on line 2303 below:
>>
>> 2212         /*
>> 2213          * In ESTABLISHED state: drop duplicate ACKs; ACK out of
>> range
>> 2214          * ACKs.  If the ack is in the range
>> 2215          *      tp->snd_una < th->th_ack <= tp->snd_max
>> 2216          * then advance tp->snd_una to th->th_ack and drop
>> 2217          * data from the retransmission queue.  If this ACK reflects
>> 2218          * more up to date window information we update our
>> window information.
>> 2219          */
>> 2220         case TCPS_ESTABLISHED:
>> 2221         case TCPS_FIN_WAIT_1:
>> 2222         case TCPS_FIN_WAIT_2:
>> 2223         case TCPS_CLOSE_WAIT:
>> 2224         case TCPS_CLOSING:
>> 2225         case TCPS_LAST_ACK:
>> 2226                 if (SEQ_GT(th->th_ack, tp->snd_max)) {
>> 2227                         TCPSTAT_INC(tcps_rcvacktoomuch);
>> 2228                         goto dropafterack;
>> 2229                 }
>> ...
>> 2234                 if (SEQ_LEQ(th->th_ack, tp->snd_una)) {
>> ...
>> 2248                         if (tlen == 0 && tiwin == tp->snd_wnd) {
>> 2249                                 TCPSTAT_INC(tcps_rcvdupack);
>> ...
>> 2277                                 if (!tcp_timer_active(tp,
>> TT_REXMT) ||
>> 2278                                     th->th_ack != tp->snd_una)
>> 2279                                         tp->t_dupacks = 0;
>> 2280                                 else if (++tp->t_dupacks >
>> tcprexmtthresh ||
>> 2281                                     ((V_tcp_do_newreno ||
>> 2282                                       (tp->t_flags &
>> TF_SACK_PERMIT)) &&
>> 2283                                      IN_FASTRECOVERY(tp))) {
>> 2284                                         if ((tp->t_flags &
>> TF_SACK_PERMIT) &&
>> 2285                                             IN_FASTRECOVERY(tp)) {
>> 2286                                                 int awnd;
>> 2287
>> 2288                                                 /*
>> 2289                                                  * Compute the
>> amount of data in flight first.
>> 2290                                                  * We can inject
>> new data into the pipe iff
>> 2291                                                  * we have less
>> than 1/2 the original window's
>> 2292                                                  * worth of data in
>> flight.
>> 2293                                                  */
>> 2294                                                 awnd =
>> (tp->snd_nxt - tp->snd_fack) +
>> 2295
>> tp->sackhint.sack_bytes_rexmit;
>> 2296                                                 if (awnd <
>> tp->snd_ssthresh) {
>> 2297
>> tp->snd_cwnd += tp->t_maxseg;
>> 2298                                                         if
>> (tp->snd_cwnd > tp->snd_ssthresh)
>> 2299
>> tp->snd_cwnd = tp->snd_ssthresh;
>> 2300                                                 }
>> 2301                                         } else
>> 2302                                                 tp->snd_cwnd +=
>> tp->t_maxseg;
>> 2303                                         (void) tcp_output(tp);
>> 2304                                         goto drop;
>>
>> I've noticed that we don't yet have this patch in our code:
>>
>> http://svnweb.freebsd.org/base?view=revision&revision=239672
>>
>> Which seems like it could be relevant here to the general case of both
>> ends of the connection entering FIN_WAIT_1 at the same time and
>> sending FIN/ACKs repeatedly (though our connections are a bizarre case
>> of this where both ends of the connection are actually the same
>> connection).
>>
>> Thanks,
>>
>> Matt
>> _______________________________________________
>> freebsd-net@freebsd.org mailing list
>> http://lists.freebsd.org/mailman/listinfo/freebsd-net
>> To unsubscribe, send any mail to "freebsd-net-unsubscribe@freebsd.org"
>>
>>

From trammell@tik.ee.ethz.ch  Tue Jul 23 09:20:24 2013
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From: Brian Trammell <trammell@tik.ee.ethz.ch>
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To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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hi Richard, all,

On 22 Jul 2013, at 18:58 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:

> Hi Brian,
>=20
>>> Including the optional ECN Nonce probing (4th segment with ECT1; 4th =
ACK
>> with NS=3D1) in the algorithm might be good...
>>=20
>> Good suggestion; we were originally thinking of something a little =
more
>> elaborate, and I do think we need to think about this a bit more, =
though,
>> since an actual CE mark would obliterate the nonce. I know that's =
unlikely
>> _now_, but one of the goals of the work is making it less so. :)
>=20
>=20
> Indeed... You would want to send the ECT(1) first, and the artificial =
CEs afterwards; if there is an ACK,ECE for the segment with ECT(1), this =
would also prove the path is ECN-capable, right?

Yep. Would have to make sure we do the right thing if that one ECT(1) =
packet gets _legitimately_ lost though.

> But I'd like to listen to your elaborate scheme :)

It's basically a generalization of what we submitted (and which covers =
the case above): the general idea is to send J ECT codepoints (checking =
the nonce sum) followed by K CE codepoints, falling back any time a =
problem is detected (as described in the draft). We thought about =
various combinations before settling on (J,K) =3D (0,3) for this rev of =
the draft. This admittedly makes the assumption that middleboxes that =
would cause connectivity problems when seeing ECT would also do so on =
CE. On the other hand, we want to keep J+K as small as possible: the =
more data segments you're sending to probe, the more time you're not =
getting the full benefit of ECN-signaled CC.=20

It seems (J,K) =3D (1,3) would work for testing nonce functionality, =
though. It's probably worth thinking a bit about other values, too.

Cheers,

Brian

>> -----Original Message-----
>> From: Brian Trammell [mailto:trammell@tik.ee.ethz.ch]
>> Sent: Montag, 22. Juli 2013 17:03
>> To: Scheffenegger, Richard
>> Cc: tcpm@ietf.org; Mirja Kuehlewind
>> Subject: Re: [tcpm] New Version Notification for =
draft-kuehlewind-tcpm-
>> ecn-fallback-00.txt
>>=20
>> hi Richard,
>>=20
>> Thanks for the comments! Inline...
>>=20
>> On 16 Jul 2013, at 14:55 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:
>>=20
>>> Brian, Mirja,
>>>=20
>>>=20
>>> Having read this draft (I think the rationale for not probing in the =
IW
>> is a good one, but perhaps a little more text in the draft would do =
good -
>> i.e. a short flow will be effectively non-reactive anyway, thus any =
ECN
>> signal would be superfluous), I like it.
>>>=20
>>>=20
>>> I have one (probably academic) point though:
>>>=20
>>>=20
>>> (I would like to see a diagram visualizing the probing process).
>>=20
>> Good suggestion; we'll add one in the next rev.
>>=20
>>> In Step 4, the algo sends the next 3 segments after IW with "CE". In
>> addition to the "first hop must not have experienced congestion" =
scenario,
>> a middlebox might also clear CE if these three segments are pure =
ACKs, as
>> per 3168, pure ACKs should not be marked ECN-capable... also, you =
expect
>> an ACK for these. Thus you need to send 3 "data segments" not simply
>> segments, right?
>>=20
>> Yep, this is what we meant, but we need to make this more clear.
>>=20
>>> In Step 6, you mention an interaction with ECN Nonce... Perhaps =
having
>> an explicit example for an ECN-Nonce enabled session would do well =
(which
>> can detect the removal of the CE-flag in probing data segments by =
sending
>> one ECT(1) data segment).
>>>=20
>>> In Step 7, CWR should be sent with the segment (not necessarily data
>> segment) ACKing any of the 1st up to the 3rd probing CE data segment. =
This
>> could be 1, 2 or 3 segments (the text only hints on a single such
>> segment).
>>=20
>> hm, that wasn't the intention, but on review that could be clearer, =
as
>> well. We'll make it so.
>>=20
>>> Including the optional ECN Nonce probing (4th segment with ECT1; 4th =
ACK
>> with NS=3D1) in the algorithm might be good...
>>=20
>> Good suggestion; we were originally thinking of something a little =
more
>> elaborate, and I do think we need to think about this a bit more, =
though,
>> since an actual CE mark would obliterate the nonce. I know that's =
unlikely
>> _now_, but one of the goals of the work is making it less so. :)
>>=20
>> Cheers,
>>=20
>> Brian
>>=20
>>=20
>>>> -----Original Message-----
>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On =
Behalf
>>>> Of Brian Trammell
>>>> Sent: Donnerstag, 04. Juli 2013 14:14
>>>> To: tcpm@ietf.org
>>>> Subject: [tcpm] Fwd: New Version Notification for
>>>> draft-kuehlewind-tcpm- ecn-fallback-00.txt
>>>>=20
>>>> Greetings, all,
>>>>=20
>>>> We've posted a draft on ECN path probing and fallback, which we'd
>>>> like to discuss at the TCPM meeting in Berlin. In a recent work =
[1],
>>>> we found that though the ECN-readiness of popular webservers has
>>>> significantly increased even in the last couple of years, there =
still
>>>> exist paths on which attempts to use ECN after successful ECN
>>>> negotiation will cause connection disruption.
>>>>=20
>>>> This draft proposes an experimental approach to determine at =
runtime
>>>> whether a path is usable, by sending segments after connection
>>>> startup and ECN negotiation with the CE codepoint set, and =
disabling
>>>> ECN if all probe segments were lost, on a per-flow or =
per-destination
>> basis.
>>>>=20
>>>> Experiments with an implementation of the approach within the Linux
>>>> kernel are underway; we plan to be able to present initial results =
in
>> Berlin.
>>>>=20
>>>> Best regards,
>>>>=20
>>>> Mirja and Brian
>>>>=20
>>>> [1] Kuehlewind, M., Neuner, S., and Trammell, B., "On the state of
>>>> ECN and TCP Options on the Internet", in Proceedings of the the =
2013
>>>> Passive and Active Measurement Conference, Hong Kong SAR, China, 19
>> March 2013.
>>>>=20
>>>> Begin forwarded message:
>>>>=20
>>>>> From: internet-drafts@ietf.org
>>>>> Subject: New Version Notification for
>>>>> draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>>>> Date: 3 July 2013 17:40:32 GMT+02:00
>>>>> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, =
Brian
>>>>> Trammell <trammell@tik.ee.ethz.ch>
>>>>>=20
>>>>>=20
>>>>> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>>>> has been successfully submitted by Mirja Kuehlewind and posted to
>>>>> the IETF repository.
>>>>>=20
>>>>> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
>>>>> Revision:	 00
>>>>> Title:		 A Mechanism for ECN Path Probing and Fallback
>>>>> Creation date:	 2013-07-02
>>>>> Group:		 Individual Submission
>>>>> Number of pages: 5
>>>>> URL:             =
http://www.ietf.org/internet-drafts/draft-kuehlewind-
>>>> tcpm-ecn-fallback-00.txt
>>>>> Status:          http://datatracker.ietf.org/doc/draft-kuehlewind-
>> tcpm-
>>>> ecn-fallback
>>>>> Htmlized:        =
http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-
>>>> fallback-00
>>>>>=20
>>>>>=20
>>>>> Abstract:
>>>>> Explicit Congestion Notification (ECN) is a TCP/IP extension that
>>>>> is  widely implemented but hardly used due to the perceived
>>>>> unusablilty  of ECN on many paths through the Internet caused by
>>>>> ECN-ignorant  routers and middleboxes.  This document specifies an
>>>>> ECN probing and  fall-back mechanism in case ECN has be =
successfully
>>>>> negotiated  between two connection endpoints, but might not be
>>>>> usable on the  path.
>>>>>=20
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> The IETF Secretariat
>>>>=20
>>>> _______________________________________________
>>>> tcpm mailing list
>>>> tcpm@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/tcpm


From michael.scharf@alcatel-lucent.com  Wed Jul 24 07:31:58 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: Slides please!
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Small update: Please add slide numbers in order to facilitate remote partic=
ipation.

Thanks

Michael


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
> Behalf Of Scharf, Michael (Michael)
> Sent: Tuesday, July 23, 2013 10:42 AM
> To: tcpm@ietf.org Extensions
> Cc: tcpm-chairs@tools.ietf.org
> Subject: [tcpm] Slides please!
>=20
> Dear Presenters,
>=20
> TCPM is scheduled for the Tuesday morning session. Please=20
> submit your slides to the chairs by Monday 17:00 CEST. This=20
> is a hard deadline.
>=20
> Given the full TCPM agenda, it would help a lot if an earlier=20
> draft version of the slides could be shared with the chairs=20
> until Sunday 17:00 CEST.
>=20
> Thanks
>=20
> Michael, Pasi, Yoshifumi
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> =

From ietf-secretariat-reply@ietf.org  Wed Jul 24 09:19:45 2013
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Subject: [tcpm] Milestones changed for tcpm WG
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Changed milestone "Submit FRTO draft to IESG for publication as an
Experimental RFC", set due date to March 2004 from March 2004.

Changed milestone "Submit TCP Roadmap document to IESG for publication
as a  Best Current Practices RFC", set due date to May 2004 from May
2004.

Changed milestone "Submit SYN flooding document to the IESG for
publication as an Informational RFC", set due date to January 2007
from January 2007.

Changed milestone "Submit soft errors document to the IESG for
publication as an Informational RFC", set due date to January 2007
from January 2007.

Changed milestone "Submit In-Window Attack draft to IESG for
publication as a Proposed Standard RFC", set due date to March 2009
from March 2009.

Changed milestone "Submit ECN-SYN document to the IESG for publication
as a Proposed Standard RFC", set due date to March 2009 from March
2009.

Changed milestone "Submit revision of RFC 2581 to the IESG for
publication as a Draft Standard", set due date to March 2009 from
March 2009.

Changed milestone "Submit ICMP attack document to the IESG for
publication as an Informational RFC", set due date to July 2009 from
July 2009.

Changed milestone "Submit TCP Early-Retransmit document to the IESG
for Experimental RFC", set due date to July 2009 from July 2009.

Changed milestone "Submit update to RFC 1323 to the IESG for Proposed
Standard RFC", set due date to July 2009 from July 2009.

Changed milestone "Submit MSS text revision originally from RFC 1323
appendix to the IESG for Proposed Standard RFC", set due date to July
2009 from July 2009.

Changed milestone "Submit TCP Urgent Pointer draft to IESG for
publication as a Proposed Standard RFC", set due date to January 2010
from January 2010.

Changed milestone "Submit document on the use of SACK data to trigger
loss recovery to the IESG for Proposed Standard", set due date to
October 2010 from October 2010.

Changed milestone "Submit document on mitigation of 'Long Connectivity
Disruptions' to the IESG for Experimental", set due date to October
2010 from October 2010.

Changed milestone "Submit document on moving undeployed TCP extensions
to Historic status to the IESG for publication as an Informational
RFC", set due date to October 2010 from October 2010.

Changed milestone "Submit RFC2988bis document to the IESG for
publication as a Proposed Standard", set due date to March 2011 from
March 2011.

Changed milestone "Submit document on increasing the initial window to
IESG as Experimental", resolved as "Done".

Changed milestone "Submit document on a proportional rate reduction
mechanism to the IESG as Experimental", set due date to May 2012 from
May 2012, resolved as "Done".

Changed milestone "Submit document on restarting the RTO timer to the
IESG for publication as an Experimental RFC", set due date to August
2013 from August 2013.

Changed milestone " Decide the intended status of the document on TCP
support for rate-limited traffic", set due date to August 2013 from
August 2013.

Changed milestone "Submit document on TCP support for rate-limited
traffic for publication (status decided as earlier milestone)", set
due date to November 2013 from November 2013.

Changed milestone "Submit document on an analysis of more detailed ECN
feedback in TCP to the IESG for publication as an Informational RFC",
set due date to November 2013 from November 2013.

URL: http://datatracker.ietf.org/wg/tcpm/charter/

From nishida@sfc.wide.ad.jp  Thu Jul 25 22:53:01 2013
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: Nandita Dukkipati <nanditad@google.com>
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Cc: "tcpm-chairs@tools.ietf.org" <tcpm-chairs@tools.ietf.org>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] Tail loss probe talk [was Re: Draft agenda for IETF 87 in Berlin]
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Hi Nantida, Yuchung,

Just out from my curiosity, I have a very minor question on the draft.
In the draft, 1-byte probing is somehow problematic to implement in
certain TCP stacks.
I'm wondering how it will be problematic.. Could you elaborate a bit?
--
Yoshifumi


On Wed, Jul 17, 2013 at 10:54 AM, Nandita Dukkipati <nanditad@google.com> wrote:
>
>> > 10:25
>> > draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)
>> > Yuchung Cheng
>> > 10min
>> TCP Loss probe (TLP) is not updated the draft because we weren't sure
>> (yet) what to update, and the new data shows similar improvement that
>> we've presented
>>
>> http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en//pubs/archive/41217.pdf
>>
>> Based on the recent WG and private discussions and the SIGCOMM
>> reviews, people are not sure the performance improvement justifies
>> this new feature, as the loss recovery is feature rich and complicated
>> with RFC3517, RFC6937, RFC5682, RFC5827, etc.
>>
>> To simplify, I'd like to present TLP as a simple conceptual change to
>> RTO recovery: On first timeout, retransmit the last packet instead of
>> the first unacked packet, and don't change but reduce first timeout to
>> 2 SRTT, as a "quick loss probe".
>>
>> As an alternative we've tried to only crank down the RTO but the
>> result is not good, because reducing cwnd to 1 is catastrophic, and
>> TCP today has to live sudden delay spikes due to buffer bloat and
>> mobile delays. It's better to be very conservative before declaring
>> the network is busted and use LW=1.
>>
>> We are hopeful to get more WG interests for TLP :)
>
>
> That's right, it would be great to have a simplified TLP adopted as WG item.
> FWIW, if anyone would like to give the TLP algorithm a spin in their tests,
> here are the two commits:
>
> Basic TLP algorithm (Section 2 of
> http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01)
> Commit:
> http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=6ba8a3b19e764b6a65e4030ab0999be50c291e6c
>
> Detecting recovered losses in TLP (Section 3 of I-D)
> Commit:
> http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=9b717a8d245075ffb8e95a2dfb4ee97ce4747457
>
> Nandita
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From prvs=0919a81c5e=anna.brunstrom@kau.se  Fri Jul 26 16:19:52 2013
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Subject: [tcpm] Update on rto restart
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--------------050007080800000204060100
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Hi,

As we have not submitted a new draft for rto restart here is a short 
update on the status.

We have not seen the need to make any changes to the algorithm from the 
last version.

After the last version was posted, there were some suggestions on the 
list to apply rto restart for all segments. The reasons why this more 
aggressive behavior is not used was also discussed on the list. There is 
no reason to tune the rto for situations where fast retransmit can be 
used for recovery.

The main discussion point from earlier mailing list discussions has 
otherwise been the increased risk of spurious rtos. We will present some 
results on this from the Linux implementation at the meeting.

BR,
Anna

PS. Sorry for the late update, due to Swedish vacation time.

Skickat frÃ¥n min Samsung Mobil

--------------050007080800000204060100
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: 8bit

<html>
  <head>
    <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div>Hi,</div>
    <div><br>
    </div>
    <div>As we have not submitted a new draft for rto restart here is a
      short update on the status. <br>
      <br>
      We have not seen the need to make any changes to the algorithm
      from the last version.</div>
    <div><br>
    </div>
    <div>After the last version was posted, there were some suggestions
      on the list to apply rto restart for all segments. The reasons why
      this more aggressive behavior is not used was also discussed on
      the list. There is no reason to tune the rto for situations where
      fast retransmit can be used for recovery.</div>
    <div><br>
    </div>
    <div>The main discussion point from earlier mailing list discussions
      has otherwise been the increased risk of spurious rtos. We will
      present some results on this from the Linux implementation at the
      meeting.</div>
    <div><br>
    </div>
    <div>BR,</div>
    <div>Anna</div>
    <div><br>
      PS. Sorry for the late update, due to Swedish vacation time.<br>
    </div>
    <div><br>
    </div>
    <div>
      <div style="font-size:75%;color:#575757">Skickat frÃ¥n min Samsung
        Mobil</div>
    </div>
  </body>
</html>

--------------050007080800000204060100--


From michael.scharf@alcatel-lucent.com  Sun Jul 28 13:17:16 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: Heads-up: draft-bonica-6man-frag-deprecate 
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Date: Sun, 28 Jul 2013 20:17:00 +0000
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Subject: [tcpm] Heads-up: draft-bonica-6man-frag-deprecate
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Hi,

In case you missed this (like me...): TSVWG has tomorrow draft-bonica-6man-=
frag-deprecate on its agenda, which discusses TCP mechanisms in Section 2.2=
.

Link: http://tools.ietf.org/html/draft-bonica-6man-frag-deprecate-02

Michael

From nishida@sfc.wide.ad.jp  Sun Jul 28 19:11:40 2013
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: [tcpm] Fwd: I-D Action: draft-nishida-tcpm-apaws-00.txt
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Hello folks,
While seeing lots of nice discussions on 1323bis, I start wondering
that it might be possible to have PAWS like feature without using
timestamp at least in some cases. So, I've decided to submit an I-D to
think about it more.

The draft does not affect rfc1323 and 1323bis as it's an independent
approach and I have no intention to present or discuss this in Berlin
meeting.
But, I will be grateful if you can give some comments or feedback on
this so that I can think how this kind of thing is feasible.

Thanks,
--
Yoshifumi (as individual)


---------- Forwarded message ----------
From:  <internet-drafts@ietf.org>
Date: Sun, Jul 28, 2013 at 6:35 PM
Subject: I-D Action: draft-nishida-tcpm-apaws-00.txt
To: i-d-announce@ietf.org



A New Internet-Draft is available from the on-line Internet-Drafts directories.


        Title           : A-PAWS: Alternative Approach for PAWS
        Author(s)       : Yoshifumi Nishida
        Filename        : draft-nishida-tcpm-apaws-00.txt
        Pages           : 15
        Date            : 2013-07-28

Abstract:
   This documents describe a technique called A-PAWS which can provide
   protection against old duplicates segments like PAWS.  While PAWS
   requires TCP to set timestamp options in all segments in a TCP
   connection, A-PAWS supports the same feature without using
   timestamps.  A-PAWS is designed to be used complementary with PAWS.
   TCP needs to use PAWS when it is necessary and activates A-PAWS only
   when it is safe to use.  Without impairing the reliability and the
   robustness of TCP, A-PAWS can provide more option space to other TCP
   extensions.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-nishida-tcpm-apaws-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/

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From michael.scharf@alcatel-lucent.com  Mon Jul 29 02:21:35 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: draft-ietf-tcpm-accecn-reqs-03
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Hi all,

Quickly scanning through draft-ietf-tcpm-accecn-reqs-03...=20

The main diff in -03 is:

     "Overhead=09
			A more accurate ecn feedback signal should limit the=09
			additional network load. As feedback information has to be=09
			provided timely and frequently, potentially all or a large=09
			fraction of TCP acknowledgments will carry this information.=09
			Ideally, no additional segments are exchanged compared to a=09
			standard RFC3168 TCP session, while the overhead in each=09
			segment is kept minimal. Further, a feedback mechanism=09
			should be prepared to proved a method to fall-back to well=09
			known RFC3168 signaling, if the new signal is suppressed by=09
			middleboxes."

For what it is worth, I don't think that the sentence is about overhead. It=
 probably justifies a separate category such as "middlebox-compliance" (may=
be somebody has a better work for broken devices between the TCP endpoints?=
).

As a side node, I wonder if it has been discussed somewhere if the SACK opt=
ion could be used for accurate ECN feedback. E.g., something like SACK bloc=
ks below the cumulative ACK (or combined with some other signaling). I don'=
t claim this to be a good idea, but I don't recall a discussion on that out=
 of my head. Do I miss something obvious?

Thanks

Michael

From michawe@ifi.uio.no  Mon Jul 29 02:22:49 2013
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References: <20130703154032.15474.58799.idtracker@ietfa.amsl.com> <82B14E2A-C7B8-4899-8954-80C66BACFB76@tik.ee.ethz.ch> <012C3117EDDB3C4781FD802A8C27DD4F25C03F89@SACEXCMBX02-PRD.hq.netapp.com> <6AE1ADEB-CADA-4BC5-AE3B-75001296C635@tik.ee.ethz.ch> <012C3117EDDB3C4781FD802A8C27DD4F25C2C2ED@SACEXCMBX02-PRD.hq.netapp.com> <E84E1F28-5DA4-4D68-90E6-B248C76FAAE9@tik.ee.ethz.ch>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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Hi,

I also finally managed to really read this draft. As I said before, I =
like the idea a lot!

As for the actual algorithm, I'm not so convinced about it - here are a =
few comments:
- step 4: in addition to Richard's suggestion that these should be data =
segments, I'd also suggest to require there to be at least 3 more =
segment to be available in the sender buffer for sending. In case the =
probe packets are lost, it could be useful to have sent enough extra =
packets to be avoid an RTO (or at least reduce the risk for an RTO).
- step 6: "In this case, the sender may continue using ECN, because =
while it may not work for detecting congestion, the use of ECN does not =
negatively affect connectivity." =3D> I'd rather not have it used if =
it's not working - if a router marks ECN to CE but the next downstream =
router clears the codepoint, we ignore congestion, and we'd like to =
avoid that, no?

The paragraph just after the 8 steps suggests to put ECN probing in the =
API. I think this is quite bad - if applications need to ask for this to =
happen, you need to update all applications to move ECN deployment =
ahead=85 but this kind of deadlock situation is what we're trying to =
solve here?  I don't understand why the necessary information about the =
number of data that are available to send "is only available at the =
higher layer", can't TCP look into its send buffer to know how much it =
has available? Surely that must be possible?

Finally, I don't think it's worth including nonce probing. As it stands =
now, the nonce is not terribly useful anyway, so why probe for it? =
Anyway nobody uses it AFAIK=85

Cheers,
Michael


On Jul 23, 2013, at 6:20 PM, Brian Trammell <trammell@tik.ee.ethz.ch> =
wrote:

> hi Richard, all,
>=20
> On 22 Jul 2013, at 18:58 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:
>=20
>> Hi Brian,
>>=20
>>>> Including the optional ECN Nonce probing (4th segment with ECT1; =
4th ACK
>>> with NS=3D1) in the algorithm might be good...
>>>=20
>>> Good suggestion; we were originally thinking of something a little =
more
>>> elaborate, and I do think we need to think about this a bit more, =
though,
>>> since an actual CE mark would obliterate the nonce. I know that's =
unlikely
>>> _now_, but one of the goals of the work is making it less so. :)
>>=20
>>=20
>> Indeed... You would want to send the ECT(1) first, and the artificial =
CEs afterwards; if there is an ACK,ECE for the segment with ECT(1), this =
would also prove the path is ECN-capable, right?
>=20
> Yep. Would have to make sure we do the right thing if that one ECT(1) =
packet gets _legitimately_ lost though.
>=20
>> But I'd like to listen to your elaborate scheme :)
>=20
> It's basically a generalization of what we submitted (and which covers =
the case above): the general idea is to send J ECT codepoints (checking =
the nonce sum) followed by K CE codepoints, falling back any time a =
problem is detected (as described in the draft). We thought about =
various combinations before settling on (J,K) =3D (0,3) for this rev of =
the draft. This admittedly makes the assumption that middleboxes that =
would cause connectivity problems when seeing ECT would also do so on =
CE. On the other hand, we want to keep J+K as small as possible: the =
more data segments you're sending to probe, the more time you're not =
getting the full benefit of ECN-signaled CC.=20
>=20
> It seems (J,K) =3D (1,3) would work for testing nonce functionality, =
though. It's probably worth thinking a bit about other values, too.
>=20
> Cheers,
>=20
> Brian
>=20
>>> -----Original Message-----
>>> From: Brian Trammell [mailto:trammell@tik.ee.ethz.ch]
>>> Sent: Montag, 22. Juli 2013 17:03
>>> To: Scheffenegger, Richard
>>> Cc: tcpm@ietf.org; Mirja Kuehlewind
>>> Subject: Re: [tcpm] New Version Notification for =
draft-kuehlewind-tcpm-
>>> ecn-fallback-00.txt
>>>=20
>>> hi Richard,
>>>=20
>>> Thanks for the comments! Inline...
>>>=20
>>> On 16 Jul 2013, at 14:55 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:
>>>=20
>>>> Brian, Mirja,
>>>>=20
>>>>=20
>>>> Having read this draft (I think the rationale for not probing in =
the IW
>>> is a good one, but perhaps a little more text in the draft would do =
good -
>>> i.e. a short flow will be effectively non-reactive anyway, thus any =
ECN
>>> signal would be superfluous), I like it.
>>>>=20
>>>>=20
>>>> I have one (probably academic) point though:
>>>>=20
>>>>=20
>>>> (I would like to see a diagram visualizing the probing process).
>>>=20
>>> Good suggestion; we'll add one in the next rev.
>>>=20
>>>> In Step 4, the algo sends the next 3 segments after IW with "CE". =
In
>>> addition to the "first hop must not have experienced congestion" =
scenario,
>>> a middlebox might also clear CE if these three segments are pure =
ACKs, as
>>> per 3168, pure ACKs should not be marked ECN-capable... also, you =
expect
>>> an ACK for these. Thus you need to send 3 "data segments" not simply
>>> segments, right?
>>>=20
>>> Yep, this is what we meant, but we need to make this more clear.
>>>=20
>>>> In Step 6, you mention an interaction with ECN Nonce... Perhaps =
having
>>> an explicit example for an ECN-Nonce enabled session would do well =
(which
>>> can detect the removal of the CE-flag in probing data segments by =
sending
>>> one ECT(1) data segment).
>>>>=20
>>>> In Step 7, CWR should be sent with the segment (not necessarily =
data
>>> segment) ACKing any of the 1st up to the 3rd probing CE data =
segment. This
>>> could be 1, 2 or 3 segments (the text only hints on a single such
>>> segment).
>>>=20
>>> hm, that wasn't the intention, but on review that could be clearer, =
as
>>> well. We'll make it so.
>>>=20
>>>> Including the optional ECN Nonce probing (4th segment with ECT1; =
4th ACK
>>> with NS=3D1) in the algorithm might be good...
>>>=20
>>> Good suggestion; we were originally thinking of something a little =
more
>>> elaborate, and I do think we need to think about this a bit more, =
though,
>>> since an actual CE mark would obliterate the nonce. I know that's =
unlikely
>>> _now_, but one of the goals of the work is making it less so. :)
>>>=20
>>> Cheers,
>>>=20
>>> Brian
>>>=20
>>>=20
>>>>> -----Original Message-----
>>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On =
Behalf
>>>>> Of Brian Trammell
>>>>> Sent: Donnerstag, 04. Juli 2013 14:14
>>>>> To: tcpm@ietf.org
>>>>> Subject: [tcpm] Fwd: New Version Notification for
>>>>> draft-kuehlewind-tcpm- ecn-fallback-00.txt
>>>>>=20
>>>>> Greetings, all,
>>>>>=20
>>>>> We've posted a draft on ECN path probing and fallback, which we'd
>>>>> like to discuss at the TCPM meeting in Berlin. In a recent work =
[1],
>>>>> we found that though the ECN-readiness of popular webservers has
>>>>> significantly increased even in the last couple of years, there =
still
>>>>> exist paths on which attempts to use ECN after successful ECN
>>>>> negotiation will cause connection disruption.
>>>>>=20
>>>>> This draft proposes an experimental approach to determine at =
runtime
>>>>> whether a path is usable, by sending segments after connection
>>>>> startup and ECN negotiation with the CE codepoint set, and =
disabling
>>>>> ECN if all probe segments were lost, on a per-flow or =
per-destination
>>> basis.
>>>>>=20
>>>>> Experiments with an implementation of the approach within the =
Linux
>>>>> kernel are underway; we plan to be able to present initial results =
in
>>> Berlin.
>>>>>=20
>>>>> Best regards,
>>>>>=20
>>>>> Mirja and Brian
>>>>>=20
>>>>> [1] Kuehlewind, M., Neuner, S., and Trammell, B., "On the state of
>>>>> ECN and TCP Options on the Internet", in Proceedings of the the =
2013
>>>>> Passive and Active Measurement Conference, Hong Kong SAR, China, =
19
>>> March 2013.
>>>>>=20
>>>>> Begin forwarded message:
>>>>>=20
>>>>>> From: internet-drafts@ietf.org
>>>>>> Subject: New Version Notification for
>>>>>> draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>>>>> Date: 3 July 2013 17:40:32 GMT+02:00
>>>>>> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, =
Brian
>>>>>> Trammell <trammell@tik.ee.ethz.ch>
>>>>>>=20
>>>>>>=20
>>>>>> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>>>>> has been successfully submitted by Mirja Kuehlewind and posted to
>>>>>> the IETF repository.
>>>>>>=20
>>>>>> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
>>>>>> Revision:	 00
>>>>>> Title:		 A Mechanism for ECN Path Probing and Fallback
>>>>>> Creation date:	 2013-07-02
>>>>>> Group:		 Individual Submission
>>>>>> Number of pages: 5
>>>>>> URL:             =
http://www.ietf.org/internet-drafts/draft-kuehlewind-
>>>>> tcpm-ecn-fallback-00.txt
>>>>>> Status:          =
http://datatracker.ietf.org/doc/draft-kuehlewind-
>>> tcpm-
>>>>> ecn-fallback
>>>>>> Htmlized:        =
http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-
>>>>> fallback-00
>>>>>>=20
>>>>>>=20
>>>>>> Abstract:
>>>>>> Explicit Congestion Notification (ECN) is a TCP/IP extension that
>>>>>> is  widely implemented but hardly used due to the perceived
>>>>>> unusablilty  of ECN on many paths through the Internet caused by
>>>>>> ECN-ignorant  routers and middleboxes.  This document specifies =
an
>>>>>> ECN probing and  fall-back mechanism in case ECN has be =
successfully
>>>>>> negotiated  between two connection endpoints, but might not be
>>>>>> usable on the  path.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> The IETF Secretariat
>>>>>=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


From rs@netapp.com  Mon Jul 29 03:28:34 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: draft-ietf-tcpm-accecn-reqs-03
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Date: Mon, 29 Jul 2013 10:28:28 +0000
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Subject: Re: [tcpm] draft-ietf-tcpm-accecn-reqs-03
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Hi Michael,

the obvious objection to

> SACK blocks below the cumulative ACK

Is D-SACK, which does something similar; however, D-SACK only specifies the=
 signaling, not the mechanism. Re-using that has the potential to confuse s=
enders (a receiver indicating two segments arrived, which weren't sent by t=
he sender; similar to network segment duplication), and overloading this wi=
th an additional signal has the potential to make clear distinctions more h=
ard (network duplication, spurious retransmission and ECN signal).

Best regards,


Richard Scheffenegger


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Scharf, Michael (Michael)
> Sent: Montag, 29. Juli 2013 11:21
> To: tcpm@ietf.org Extensions
> Subject: [tcpm] draft-ietf-tcpm-accecn-reqs-03
>=20
> Hi all,
>=20
> Quickly scanning through draft-ietf-tcpm-accecn-reqs-03...
>=20
> The main diff in -03 is:
>=20
>      "Overhead
> 			A more accurate ecn feedback signal should limit the
> 			additional network load. As feedback information has to
> be
> 			provided timely and frequently, potentially all or a
> large
> 			fraction of TCP acknowledgments will carry this
> information.
> 			Ideally, no additional segments are exchanged compared
> to a
> 			standard RFC3168 TCP session, while the overhead in each
>=20
> 			segment is kept minimal. Further, a feedback mechanism
>=20
> 			should be prepared to proved a method to fall-back to
> well
> 			known RFC3168 signaling, if the new signal is suppressed
> by
> 			middleboxes."
>=20
> For what it is worth, I don't think that the sentence is about overhead.
> It probably justifies a separate category such as "middlebox-compliance"
> (maybe somebody has a better work for broken devices between the TCP
> endpoints?).
>=20
> As a side node, I wonder if it has been discussed somewhere if the SACK
> option could be used for accurate ECN feedback. E.g., something like SACK
> blocks below the cumulative ACK (or combined with some other signaling). =
I
> don't claim this to be a good idea, but I don't recall a discussion on
> that out of my head. Do I miss something obvious?
>=20
> Thanks
>=20
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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Date: Mon, 29 Jul 2013 13:12:21 +0200
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Subject: Re: [tcpm] draft-ietf-tcpm-1323bis-14.txt
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I only reviewed section 3 and it looks good to me.

Some minor suggestions on wordings.
1.    RTTM Rule: A TSecr value received in a segment MAY be used to update
              the averaged RTT measurement only if the segment advances
              the left edge of the send window, i.e.  SND.UNA is
              increased.
s/segment/ACK

2. "An <ACK> for an out-of-order segment
        SHOULD therefore contain the timestamp from the most recent
        segment that advanced the window."
s/advanced the window/advanced RCV.NXT


FWIW Linux recently changed to only use TS for RTTM if normal timings
aren't available. It now also measure RTT on new SACK for better RTO
during loss recovery
http://patchwork.ozlabs.org/patch/260836/
http://patchwork.ozlabs.org/patch/260838/
http://patchwork.ozlabs.org/patch/260837/


On Sun, Jul 21, 2013 at 10:39 PM, Pasi Sarolahti <pasi.sarolahti@iki.fi> wr=
ote:
> Lots of mails have been exchanged on this since the WGLC, but here is an =
attempt to shortly summarize the main points:
>
> * Mark Allman made a thorough set of comments during the WGLC. My underst=
anding is that these comments are now addressed. Some paragraphs have chang=
ed quite a bit as a result, so it would be good if people can check if the =
current text looks good.
>
> * After the WGLC there was an update on the text regarding handling of se=
gments without timestamp option after they have been successfully negotiate=
d (MUST drop in ver-14), and a follow-up discussion about possible interact=
ions with middleboxes. In a recent mail Richard proposes changing this to S=
HOULD. This change is not yet visible in the I-D repository.
>
> If I'm forgetting something important, please remind me (and the list).
>
> I think in the meeting we should go through the main recent changes, and =
check if people are ok with the current wordings. Additional comments on th=
e list are much appreciated.
>
> - Pasi
>
>
> On Jul 16, 2013, at 2:03 PM, "Scheffenegger, Richard" <rs@netapp.com> wro=
te:
>
>> Hi,
>>
>> I would like to take this opportunity to stir up the recent discussions =
(from April/May timeframe) again.
>>
>> There have been some comments past the posting of this version on the li=
st, which I wasn't able to fully qualify how they would impact this latest =
text.
>>
>> I would like to encourage the participants in these discussions to revie=
w the latest version, and point out if they cannot agree with any of the wo=
rding.
>>
>>
>>
>> There will probably be a slot at the upcoming IETF to try to find out wh=
at the consensus is on this document, too...
>>
>> Best regards,
>>
>> Richard Scheffenegger
>>
>>
>>> -----Original Message-----
>>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>>> Sent: Donnerstag, 23. Mai 2013 16:10
>>> To: Bob Braden; Scheffenegger, Richard; Robert T. Braden; David Borman;
>>> Van Jacobson
>>> Subject: New Version Notification for draft-ietf-tcpm-1323bis-14.txt
>>>
>>>
>>> A new version of I-D, draft-ietf-tcpm-1323bis-14.txt has been successfu=
lly
>>> submitted by David Borman and posted to the IETF repository.
>>>
>>> Filename:     draft-ietf-tcpm-1323bis
>>> Revision:     14
>>> Title:                TCP Extensions for High Performance
>>> Creation date:        2013-05-23
>>> Group:                tcpm
>>> Number of pages: 46
>>> URL:             http://www.ietf.org/internet-drafts/draft-ietf-tcpm-
>>> 1323bis-14.txt
>>> Status:          http://datatracker.ietf.org/doc/draft-ietf-tcpm-1323bi=
s
>>> Htmlized:        http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-14
>>> Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-132=
3bis-
>>> 14
>>>
>>> Abstract:
>>>   This document specifies a set of TCP extensions to improve
>>>   performance over paths with a large bandwidth * delay product and to
>>>   provide reliable operation over very high-speed paths.  It defines
>>>   TCP options for scaled windows and timestamps.  The timestamps are
>>>   used for two distinct mechanisms, RTTM (Round Trip Time Measurement)
>>>   and PAWS (Protection Against Wrapped Sequences).
>>>
>>>   This document updates and obsoletes RFC 1323.
>>>
>>>
>>>
>>>
>>> The IETF Secretariat
>>
>> _______________________________________________
>> 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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To: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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hi Michael,

Thanks! Comments on comments inline...

On 29 Jul 2013, at 11:22, Michael Welzl <michawe@ifi.uio.no> wrote:

> Hi,
>=20
> I also finally managed to really read this draft. As I said before, I =
like the idea a lot!
>=20
> As for the actual algorithm, I'm not so convinced about it - here are =
a few comments:
> - step 4: in addition to Richard's suggestion that these should be =
data segments, I'd also suggest to require there to be at least 3 more =
segment to be available in the sender buffer for sending.

This is a good point, yes. We're assuming that we have information about =
how long the flow will be, and we're not really specific about how we =
know that yet...

> In case the probe packets are lost, it could be useful to have sent =
enough extra packets to be avoid an RTO (or at least reduce the risk for =
an RTO).

...and requiring segments in the buffer would be a good way of doing =
this. We didn't want to get into buffer details, in on account of corner =
cases (in the timing of writes to the buffer) about which we didn't want =
to think about yet. But if we can ignore those,=20

> - step 6: "In this case, the sender may continue using ECN, because =
while it may not work for detecting congestion, the use of ECN does not =
negatively affect connectivity." =3D> I'd rather not have it used if =
it's not working - if a router marks ECN to CE but the next downstream =
router clears the codepoint, we ignore congestion, and we'd like to =
avoid that, no?

The problem in this case is that we don't know if we have an ECN-aware =
middlebox that is clearing a CE from the end system, on the assumption =
that it is closer to the end-system than one would expect -- we did see =
some of this activity in the study detailed in the PAM paper. In this =
case the CE-clearing is arguably correct (or at least acceptable).=20

>=20
> The paragraph just after the 8 steps suggests to put ECN probing in =
the API. I think this is quite bad - if applications need to ask for =
this to happen, you need to update all applications to move ECN =
deployment ahead=85 but this kind of deadlock situation is what we're =
trying to solve here?

Hm, good point. This probably shouldn't be a per-connection; however, it =
should be per-endpoint configurable via sysctl or equivalent (as, =
indeed, ECN is on most end systems today)

>  I don't understand why the necessary information about the number of =
data that are available to send "is only available at the higher layer", =
can't TCP look into its send buffer to know how much it has available? =
Surely that must be possible?

Again, we didn't want to make assumptions about how applications write =
into the buffer, but yeah, requiring app changes at the API level is =
much, much worse. So good point, I think I'm convinced, at least, that =
an approach based on buffer occupancy is worth pursuing.

> Finally, I don't think it's worth including nonce probing. As it =
stands now, the nonce is not terribly useful anyway, so why probe for =
it? Anyway nobody uses it AFAIK=85

We didn't look at nonce in our measurement study, so it's in there as =
future work in case it's useful. If it's not, we can certainly leave it =
out but (measurement geek hat on) I'd like data to support that =
decision. :)

Again, many thanks, and best regards,

Brian


> On Jul 23, 2013, at 6:20 PM, Brian Trammell <trammell@tik.ee.ethz.ch> =
wrote:
>=20
>> hi Richard, all,
>>=20
>> On 22 Jul 2013, at 18:58 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:
>>=20
>>> Hi Brian,
>>>=20
>>>>> Including the optional ECN Nonce probing (4th segment with ECT1; =
4th ACK
>>>> with NS=3D1) in the algorithm might be good...
>>>>=20
>>>> Good suggestion; we were originally thinking of something a little =
more
>>>> elaborate, and I do think we need to think about this a bit more, =
though,
>>>> since an actual CE mark would obliterate the nonce. I know that's =
unlikely
>>>> _now_, but one of the goals of the work is making it less so. :)
>>>=20
>>>=20
>>> Indeed... You would want to send the ECT(1) first, and the =
artificial CEs afterwards; if there is an ACK,ECE for the segment with =
ECT(1), this would also prove the path is ECN-capable, right?
>>=20
>> Yep. Would have to make sure we do the right thing if that one ECT(1) =
packet gets _legitimately_ lost though.
>>=20
>>> But I'd like to listen to your elaborate scheme :)
>>=20
>> It's basically a generalization of what we submitted (and which =
covers the case above): the general idea is to send J ECT codepoints =
(checking the nonce sum) followed by K CE codepoints, falling back any =
time a problem is detected (as described in the draft). We thought about =
various combinations before settling on (J,K) =3D (0,3) for this rev of =
the draft. This admittedly makes the assumption that middleboxes that =
would cause connectivity problems when seeing ECT would also do so on =
CE. On the other hand, we want to keep J+K as small as possible: the =
more data segments you're sending to probe, the more time you're not =
getting the full benefit of ECN-signaled CC.=20
>>=20
>> It seems (J,K) =3D (1,3) would work for testing nonce functionality, =
though. It's probably worth thinking a bit about other values, too.
>>=20
>> Cheers,
>>=20
>> Brian
>>=20
>>>> -----Original Message-----
>>>> From: Brian Trammell [mailto:trammell@tik.ee.ethz.ch]
>>>> Sent: Montag, 22. Juli 2013 17:03
>>>> To: Scheffenegger, Richard
>>>> Cc: tcpm@ietf.org; Mirja Kuehlewind
>>>> Subject: Re: [tcpm] New Version Notification for =
draft-kuehlewind-tcpm-
>>>> ecn-fallback-00.txt
>>>>=20
>>>> hi Richard,
>>>>=20
>>>> Thanks for the comments! Inline...
>>>>=20
>>>> On 16 Jul 2013, at 14:55 , "Scheffenegger, Richard" <rs@netapp.com> =
wrote:
>>>>=20
>>>>> Brian, Mirja,
>>>>>=20
>>>>>=20
>>>>> Having read this draft (I think the rationale for not probing in =
the IW
>>>> is a good one, but perhaps a little more text in the draft would do =
good -
>>>> i.e. a short flow will be effectively non-reactive anyway, thus any =
ECN
>>>> signal would be superfluous), I like it.
>>>>>=20
>>>>>=20
>>>>> I have one (probably academic) point though:
>>>>>=20
>>>>>=20
>>>>> (I would like to see a diagram visualizing the probing process).
>>>>=20
>>>> Good suggestion; we'll add one in the next rev.
>>>>=20
>>>>> In Step 4, the algo sends the next 3 segments after IW with "CE". =
In
>>>> addition to the "first hop must not have experienced congestion" =
scenario,
>>>> a middlebox might also clear CE if these three segments are pure =
ACKs, as
>>>> per 3168, pure ACKs should not be marked ECN-capable... also, you =
expect
>>>> an ACK for these. Thus you need to send 3 "data segments" not =
simply
>>>> segments, right?
>>>>=20
>>>> Yep, this is what we meant, but we need to make this more clear.
>>>>=20
>>>>> In Step 6, you mention an interaction with ECN Nonce... Perhaps =
having
>>>> an explicit example for an ECN-Nonce enabled session would do well =
(which
>>>> can detect the removal of the CE-flag in probing data segments by =
sending
>>>> one ECT(1) data segment).
>>>>>=20
>>>>> In Step 7, CWR should be sent with the segment (not necessarily =
data
>>>> segment) ACKing any of the 1st up to the 3rd probing CE data =
segment. This
>>>> could be 1, 2 or 3 segments (the text only hints on a single such
>>>> segment).
>>>>=20
>>>> hm, that wasn't the intention, but on review that could be clearer, =
as
>>>> well. We'll make it so.
>>>>=20
>>>>> Including the optional ECN Nonce probing (4th segment with ECT1; =
4th ACK
>>>> with NS=3D1) in the algorithm might be good...
>>>>=20
>>>> Good suggestion; we were originally thinking of something a little =
more
>>>> elaborate, and I do think we need to think about this a bit more, =
though,
>>>> since an actual CE mark would obliterate the nonce. I know that's =
unlikely
>>>> _now_, but one of the goals of the work is making it less so. :)
>>>>=20
>>>> Cheers,
>>>>=20
>>>> Brian
>>>>=20
>>>>=20
>>>>>> -----Original Message-----
>>>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On =
Behalf
>>>>>> Of Brian Trammell
>>>>>> Sent: Donnerstag, 04. Juli 2013 14:14
>>>>>> To: tcpm@ietf.org
>>>>>> Subject: [tcpm] Fwd: New Version Notification for
>>>>>> draft-kuehlewind-tcpm- ecn-fallback-00.txt
>>>>>>=20
>>>>>> Greetings, all,
>>>>>>=20
>>>>>> We've posted a draft on ECN path probing and fallback, which we'd
>>>>>> like to discuss at the TCPM meeting in Berlin. In a recent work =
[1],
>>>>>> we found that though the ECN-readiness of popular webservers has
>>>>>> significantly increased even in the last couple of years, there =
still
>>>>>> exist paths on which attempts to use ECN after successful ECN
>>>>>> negotiation will cause connection disruption.
>>>>>>=20
>>>>>> This draft proposes an experimental approach to determine at =
runtime
>>>>>> whether a path is usable, by sending segments after connection
>>>>>> startup and ECN negotiation with the CE codepoint set, and =
disabling
>>>>>> ECN if all probe segments were lost, on a per-flow or =
per-destination
>>>> basis.
>>>>>>=20
>>>>>> Experiments with an implementation of the approach within the =
Linux
>>>>>> kernel are underway; we plan to be able to present initial =
results in
>>>> Berlin.
>>>>>>=20
>>>>>> Best regards,
>>>>>>=20
>>>>>> Mirja and Brian
>>>>>>=20
>>>>>> [1] Kuehlewind, M., Neuner, S., and Trammell, B., "On the state =
of
>>>>>> ECN and TCP Options on the Internet", in Proceedings of the the =
2013
>>>>>> Passive and Active Measurement Conference, Hong Kong SAR, China, =
19
>>>> March 2013.
>>>>>>=20
>>>>>> Begin forwarded message:
>>>>>>=20
>>>>>>> From: internet-drafts@ietf.org
>>>>>>> Subject: New Version Notification for
>>>>>>> draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>>>>>> Date: 3 July 2013 17:40:32 GMT+02:00
>>>>>>> To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, =
Brian
>>>>>>> Trammell <trammell@tik.ee.ethz.ch>
>>>>>>>=20
>>>>>>>=20
>>>>>>> A new version of I-D, draft-kuehlewind-tcpm-ecn-fallback-00.txt
>>>>>>> has been successfully submitted by Mirja Kuehlewind and posted =
to
>>>>>>> the IETF repository.
>>>>>>>=20
>>>>>>> Filename:	 draft-kuehlewind-tcpm-ecn-fallback
>>>>>>> Revision:	 00
>>>>>>> Title:		 A Mechanism for ECN Path Probing and Fallback
>>>>>>> Creation date:	 2013-07-02
>>>>>>> Group:		 Individual Submission
>>>>>>> Number of pages: 5
>>>>>>> URL:             =
http://www.ietf.org/internet-drafts/draft-kuehlewind-
>>>>>> tcpm-ecn-fallback-00.txt
>>>>>>> Status:          =
http://datatracker.ietf.org/doc/draft-kuehlewind-
>>>> tcpm-
>>>>>> ecn-fallback
>>>>>>> Htmlized:        =
http://tools.ietf.org/html/draft-kuehlewind-tcpm-ecn-
>>>>>> fallback-00
>>>>>>>=20
>>>>>>>=20
>>>>>>> Abstract:
>>>>>>> Explicit Congestion Notification (ECN) is a TCP/IP extension =
that
>>>>>>> is  widely implemented but hardly used due to the perceived
>>>>>>> unusablilty  of ECN on many paths through the Internet caused by
>>>>>>> ECN-ignorant  routers and middleboxes.  This document specifies =
an
>>>>>>> ECN probing and  fall-back mechanism in case ECN has be =
successfully
>>>>>>> negotiated  between two connection endpoints, but might not be
>>>>>>> usable on the  path.
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> The IETF Secretariat
>>>>>>=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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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 29 Jul 2013 13:56:30 +0200
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
> We've just uploaded a new version of our draft. The congestion window
> management algorithm has not changed since -00, but in -01 and -02 drafts
> we introduce a more robust way to measure the pipeACK that attempts to
> account for the wide variety of interactive TCP application behaviour.
> This uses a sampling method to sample pipeACK over the last second.
>
> Clearly the cwnd poorly reflects the capacity in the non-validated phase
> and I think it would be wrong to use this for TCP control block sharing.
> We therefore also  introduce text that proposes that TCB sharing should
> use the pipeACK in place of cwnd when a TCP sender is in the Nonvalidated
> phase.  We think this value better reflects the capacity that the flow has
> utilised in the network path - but would welcome more thought on this.
> Thoughts on this would be most welcome.

Hi Gorry,

I am interested in comparing this draft with this solution:

1. Do not reduce cwnd on idle (of any length)
2. From idle to active, send max(cwnd, data) over RTT in small bursts.
    E.g., cwnd=20, RTT= 100ms, send 4 packets every 20ms
    This emulates the ACK clocking.

TCP ack clocking offers smooth send. But it does not work well with
modern (structured) communication patterns. Often TCP does NOT have
data to send when it receives acks as the application uses persistent
connection or is thinking. This leads to bursts up to cwnd size.

If the idle interval is within an RTO, nothing can save the burst
including this draft. If it's longer than an RTO, this draft mitigates
it. But I would argue cutting cwnd is not addressing the fundamental
problem, because TCP still sends a cwnd of burst!

Burst induced losses are bad b/c it has not correlation to utilization
other than tricking TCP into congestion avoidance and low throughput.
Unfortunately it's hard to distinguish whether a loss is caused by
burst or congestion.

My sense is that any artificial idle length or cwnd/ssthresh cut
factor helps but does not scale well with time and bandwidth, but
maybe we can simply emulate ack clocking to really reduce the burst.
In some terms, it's either "always slow-start if ack clock is lost" or
"trust and send the old cwnd/rtt rate but not at line-rate burst". The
latter might be favored by Web transfers as the former is known to
kill Web performance.

Thoughts?

>
> Gorry
>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>  This draft is a work item of the TCP Maintenance and Minor Extensions
>> Working Group of the IETF.
>>
>>       Title           : Updating TCP to support Rate-Limited Traffic
>>       Author(s)       : Godred Fairhurst
>>                           Arjuna Sathiaseelan
>>                           Raffaello Secchi
>>       Filename        : draft-ietf-tcpm-newcwv-02.txt
>>       Pages           : 19
>>       Date            : 2013-07-01
>>
>> Abstract:
>>    This document proposes an update to RFC 5681 to address issues that
>>    arise when TCP is used to support traffic that exhibits periods where
>>    the sending rate is limited by the application rather than the
>>    congestion window.  It updates TCP to allow a TCP sender to restart
>>    quickly following either an idle or rate-limited interval.  This
>>    method is expected to benefit applications that send rate-limited
>>    traffic using TCP, while also providing an appropriate response if
>>    congestion is experienced.
>>
>>    It also evaluates the Experimental specification of TCP Congestion
>>    Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>>    2861 sought to address important issues, but failed to deliver a
>>    widely used solution.  This document therefore recommends that the
>>    status of RFC 2861 is moved from Experimental to Historic, and that
>>    it is replaced by the current specification.
>>
>>    NOTE: The standards status of this WG document is under review for
>>    consideration as either Experimental (EXP) or Proposed Standard (PS).
>>    This decision will be made later as the document is finalised.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>>
>>
>> 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
>>
>
>
> _______________________________________________
> 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, 29 Jul 2013 14:08:49 +0200
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Subject: Re: [tcpm] Tail loss probe talk [was Re: Draft agenda for IETF 87 in Berlin]
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On Fri, Jul 26, 2013 at 7:52 AM, Yoshifumi Nishida
<nishida@sfc.wide.ad.jp> wrote:
> Hi Nantida, Yuchung,
>
> Just out from my curiosity, I have a very minor question on the draft.
> In the draft, 1-byte probing is somehow problematic to implement in
> certain TCP stacks.
> I'm wondering how it will be problematic.. Could you elaborate a bit?
It refers more or less to our implementation experience in Linux,
which uses a packet-based CC. the problem is the original packet is
counted as 1, and now the 1-byte probe should be counted as 1/MSS
packet, or another 1 packet?
The solution was not trivial and was to now count the split the
original packet into 2 packets (MSS-1, 1 bytes). But this makes SACK
scoreboard accounting complicated and we always had some strange bugs,
therefore we ditched the approach.

But I agree the sentence is vague and needs to be removed or
clarified. Thanks for pointing that out.


> --
> Yoshifumi
>
>
> On Wed, Jul 17, 2013 at 10:54 AM, Nandita Dukkipati <nanditad@google.com> wrote:
>>
>>> > 10:25
>>> > draft-dukkipati-tcpm-tcp-loss-probe-01 (no new draft)
>>> > Yuchung Cheng
>>> > 10min
>>> TCP Loss probe (TLP) is not updated the draft because we weren't sure
>>> (yet) what to update, and the new data shows similar improvement that
>>> we've presented
>>>
>>> http://static.googleusercontent.com/external_content/untrusted_dlcp/research.google.com/en//pubs/archive/41217.pdf
>>>
>>> Based on the recent WG and private discussions and the SIGCOMM
>>> reviews, people are not sure the performance improvement justifies
>>> this new feature, as the loss recovery is feature rich and complicated
>>> with RFC3517, RFC6937, RFC5682, RFC5827, etc.
>>>
>>> To simplify, I'd like to present TLP as a simple conceptual change to
>>> RTO recovery: On first timeout, retransmit the last packet instead of
>>> the first unacked packet, and don't change but reduce first timeout to
>>> 2 SRTT, as a "quick loss probe".
>>>
>>> As an alternative we've tried to only crank down the RTO but the
>>> result is not good, because reducing cwnd to 1 is catastrophic, and
>>> TCP today has to live sudden delay spikes due to buffer bloat and
>>> mobile delays. It's better to be very conservative before declaring
>>> the network is busted and use LW=1.
>>>
>>> We are hopeful to get more WG interests for TLP :)
>>
>>
>> That's right, it would be great to have a simplified TLP adopted as WG item.
>> FWIW, if anyone would like to give the TLP algorithm a spin in their tests,
>> here are the two commits:
>>
>> Basic TLP algorithm (Section 2 of
>> http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01)
>> Commit:
>> http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=6ba8a3b19e764b6a65e4030ab0999be50c291e6c
>>
>> Detecting recovered losses in TLP (Section 3 of I-D)
>> Commit:
>> http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=9b717a8d245075ffb8e95a2dfb4ee97ce4747457
>>
>> Nandita
>>
>> _______________________________________________
>> 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 michawe@ifi.uio.no  Mon Jul 29 05:15:15 2013
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Cutting, cutting=85 inline:


>> In case the probe packets are lost, it could be useful to have sent =
enough extra packets to be avoid an RTO (or at least reduce the risk for =
an RTO).
>=20
> ...and requiring segments in the buffer would be a good way of doing =
this. We didn't want to get into buffer details, in on account of corner =
cases (in the timing of writes to the buffer) about which we didn't want =
to think about yet. But if we can ignore those,=20

I think you can't avoid that anyway because of this:


>> I don't understand why the necessary information about the number of =
data that are available to send "is only available at the higher layer", =
can't TCP look into its send buffer to know how much it has available? =
Surely that must be possible?
>=20
> Again, we didn't want to make assumptions about how applications write =
into the buffer, but yeah, requiring app changes at the API level is =
much, much worse. So good point, I think I'm convinced, at least, that =
an approach based on buffer occupancy is worth pursuing.

=85 seems like we agree now (wow, that was fast  :-)  ).


>> - step 6: "In this case, the sender may continue using ECN, because =
while it may not work for detecting congestion, the use of ECN does not =
negatively affect connectivity." =3D> I'd rather not have it used if =
it's not working - if a router marks ECN to CE but the next downstream =
router clears the codepoint, we ignore congestion, and we'd like to =
avoid that, no?
>=20
> The problem in this case is that we don't know if we have an ECN-aware =
middlebox that is clearing a CE from the end system, on the assumption =
that it is closer to the end-system than one would expect -- we did see =
some of this activity in the study detailed in the PAM paper. In this =
case the CE-clearing is arguably correct (or at least acceptable).=20

I'm beginning to understand your point. It's about this sentence in the =
draft:
"If the ECE flag is not set on the ACK segment(s) sent by the
       receiver acknowledging the CE probe segments, the path may or may
       not be usable, as that there might be middleboxes/gateways that
       (arguably correctly) clear CE on segments from end hosts, because
       they assume that congestion can not have occurred up to this
       point on the path."

But here you're assuming that this is the only possible reason for a CE =
to be cleared, i.e. CE cannot be cleared by a middlebox for any other =
reason. It might well be that a middlebox makes a mistake when it clears =
CE=85  well, but if that wasn't seen in any of the measurement studies =
(the IMC one and yours), I'm inclined to accept that it's okay to leave =
ECN on then.

Cheers,
Michael


From michael.scharf@alcatel-lucent.com  Mon Jul 29 05:38:09 2013
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> the obvious objection to
>=20
> > SACK blocks below the cumulative ACK
>=20
> Is D-SACK, which does something similar; however, D-SACK only=20
> specifies the signaling, not the mechanism. Re-using that has=20
> the potential to confuse senders (a receiver indicating two=20
> segments arrived, which weren't sent by the sender; similar=20
> to network segment duplication), and overloading this with an=20
> additional signal has the potential to make clear=20
> distinctions more hard (network duplication, spurious=20
> retransmission and ECN signal).

Well, obviously I do not suggest *only* using SACK blocks. I was thinking o=
f something like

- an "ECN-SACK block" (semantics: "This byte range got ECN marked.")

or

- a flag in the TCP header that makes it clear how the receiver generated t=
he D-SACK based on ECN marking.

The point is that instead of returning counters (cf. draft-kuehlewind-tcpm-=
accurate-ecn-option), one could return byte ranges. This has obvious disadv=
antages (option space consumption etc.), but since this document briefly su=
rveys the solution space, I wonder whether you could comment on non-counter=
-based feedback schemes.

Michael (naive TCP researcher)



> > -----Original Message-----
> > From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org]=20
> On Behalf=20
> > Of Scharf, Michael (Michael)
> > Sent: Montag, 29. Juli 2013 11:21
> > To: tcpm@ietf.org Extensions
> > Subject: [tcpm] draft-ietf-tcpm-accecn-reqs-03
> >=20
> > Hi all,
> >=20
> > Quickly scanning through draft-ietf-tcpm-accecn-reqs-03...
> >=20
> > The main diff in -03 is:
> >=20
> >      "Overhead
> > 			A more accurate ecn feedback signal=20
> should limit the
> > 			additional network load. As feedback=20
> information has to be
> > 			provided timely and frequently,=20
> potentially all or a large
> > 			fraction of TCP acknowledgments will=20
> carry this information.
> > 			Ideally, no additional segments are=20
> exchanged compared to a
> > 			standard RFC3168 TCP session, while the=20
> overhead in each
> >=20
> > 			segment is kept minimal. Further, a=20
> feedback mechanism
> >=20
> > 			should be prepared to proved a method=20
> to fall-back to well
> > 			known RFC3168 signaling, if the new=20
> signal is suppressed by
> > 			middleboxes."
> >=20
> > For what it is worth, I don't think that the sentence is=20
> about overhead.
> > It probably justifies a separate category such as=20
> "middlebox-compliance"
> > (maybe somebody has a better work for broken devices=20
> between the TCP=20
> > endpoints?).
> >=20
> > As a side node, I wonder if it has been discussed somewhere if the=20
> > SACK option could be used for accurate ECN feedback. E.g.,=20
> something=20
> > like SACK blocks below the cumulative ACK (or combined with=20
> some other=20
> > signaling). I don't claim this to be a good idea, but I=20
> don't recall a=20
> > discussion on that out of my head. Do I miss something obvious?
> >=20
> > Thanks
> >=20
> > Michael
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> =

From Alexander.Zimmermann@netapp.com  Mon Jul 29 05:44:36 2013
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Yuchung Cheng <ycheng@google.com>
Thread-Topic: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Hi Yuchung,

Am 29.07.2013 um 07:56 schrieb Yuchung Cheng <ycheng@google.com>:

> On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
>> We've just uploaded a new version of our draft. The congestion window
>> management algorithm has not changed since -00, but in -01 and -02 drafts
>> we introduce a more robust way to measure the pipeACK that attempts to
>> account for the wide variety of interactive TCP application behaviour.
>> This uses a sampling method to sample pipeACK over the last second.
>> 
>> Clearly the cwnd poorly reflects the capacity in the non-validated phase
>> and I think it would be wrong to use this for TCP control block sharing.
>> We therefore also  introduce text that proposes that TCB sharing should
>> use the pipeACK in place of cwnd when a TCP sender is in the Nonvalidated
>> phase.  We think this value better reflects the capacity that the flow has
>> utilised in the network path - but would welcome more thought on this.
>> Thoughts on this would be most welcome.
> 
> Hi Gorry,
> 
> I am interested in comparing this draft with this solution:
> 
> 1. Do not reduce cwnd on idle (of any length)
> 2. From idle to active, send max(cwnd, data) over RTT in small bursts.
>    E.g., cwnd=20, RTT= 100ms, send 4 packets every 20ms
>    This emulates the ACK clocking.

Smells like you propose a "IW=oo" here (with a straight forward burst
protection), isn't it?

> 
> TCP ack clocking offers smooth send. But it does not work well with
> modern (structured) communication patterns. Often TCP does NOT have
> data to send when it receives acks as the application uses persistent
> connection or is thinking. This leads to bursts up to cwnd size.

True, but this is a general problem, not only TCP CWV related.

> 
> If the idle interval is within an RTO, nothing can save the burst
> including this draft. If it's longer than an RTO, this draft mitigates
> it. But I would argue cutting cwnd is not addressing the fundamental
> problem, because TCP still sends a cwnd of burst!

See above. Maybe it's worthwhile to tackle the burst problem in general.

> 
> Burst induced losses are bad b/c it has not correlation to utilization
> other than tricking TCP into congestion avoidance and low throughput.
> Unfortunately it's hard to distinguish whether a loss is caused by
> burst or congestion.
> 
> My sense is that any artificial idle length or cwnd/ssthresh cut
> factor helps but does not scale well with time and bandwidth, but
> maybe we can simply emulate ack clocking to really reduce the burst.
> In some terms, it's either "always slow-start if ack clock is lost" or
> "trust and send the old cwnd/rtt rate but not at line-rate burst". The
> latter might be favored by Web transfers as the former is known to
> kill Web performance.
> 
> Thoughts?
> 
>> 
>> Gorry
>> 
>>> 
>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>> directories.
>>> This draft is a work item of the TCP Maintenance and Minor Extensions
>>> Working Group of the IETF.
>>> 
>>>      Title           : Updating TCP to support Rate-Limited Traffic
>>>      Author(s)       : Godred Fairhurst
>>>                          Arjuna Sathiaseelan
>>>                          Raffaello Secchi
>>>      Filename        : draft-ietf-tcpm-newcwv-02.txt
>>>      Pages           : 19
>>>      Date            : 2013-07-01
>>> 
>>> Abstract:
>>>   This document proposes an update to RFC 5681 to address issues that
>>>   arise when TCP is used to support traffic that exhibits periods where
>>>   the sending rate is limited by the application rather than the
>>>   congestion window.  It updates TCP to allow a TCP sender to restart
>>>   quickly following either an idle or rate-limited interval.  This
>>>   method is expected to benefit applications that send rate-limited
>>>   traffic using TCP, while also providing an appropriate response if
>>>   congestion is experienced.
>>> 
>>>   It also evaluates the Experimental specification of TCP Congestion
>>>   Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>>>   2861 sought to address important issues, but failed to deliver a
>>>   widely used solution.  This document therefore recommends that the
>>>   status of RFC 2861 is moved from Experimental to Historic, and that
>>>   it is replaced by the current specification.
>>> 
>>>   NOTE: The standards status of this WG document is under review for
>>>   consideration as either Experimental (EXP) or Proposed Standard (PS).
>>>   This decision will be made later as the document is finalised.
>>> 
>>> 
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>> 
>>> There's also a htmlized version available at:
>>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>> 
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>>> 
>>> 
>>> 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
>>> 
>> 
>> 
>> _______________________________________________
>> 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: Mon, 29 Jul 2013 15:05:51 +0200
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On Mon, Jul 29, 2013 at 2:44 PM, Zimmermann, Alexander
<Alexander.Zimmermann@netapp.com> wrote:
> Hi Yuchung,
>
> Am 29.07.2013 um 07:56 schrieb Yuchung Cheng <ycheng@google.com>:
>
>> On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
>>> We've just uploaded a new version of our draft. The congestion window
>>> management algorithm has not changed since -00, but in -01 and -02 drafts
>>> we introduce a more robust way to measure the pipeACK that attempts to
>>> account for the wide variety of interactive TCP application behaviour.
>>> This uses a sampling method to sample pipeACK over the last second.
>>>
>>> Clearly the cwnd poorly reflects the capacity in the non-validated phase
>>> and I think it would be wrong to use this for TCP control block sharing.
>>> We therefore also  introduce text that proposes that TCB sharing should
>>> use the pipeACK in place of cwnd when a TCP sender is in the Nonvalidated
>>> phase.  We think this value better reflects the capacity that the flow has
>>> utilised in the network path - but would welcome more thought on this.
>>> Thoughts on this would be most welcome.
>>
>> Hi Gorry,
>>
>> I am interested in comparing this draft with this solution:
>>
>> 1. Do not reduce cwnd on idle (of any length)
>> 2. From idle to active, send max(cwnd, data) over RTT in small bursts.
>>    E.g., cwnd=20, RTT= 100ms, send 4 packets every 20ms
>>    This emulates the ACK clocking.
>
> Smells like you propose a "IW=oo" here (with a straight forward burst
> protection), isn't it?
No. I am proposing: when idle keep cwnd as is.

>
>>
>> TCP ack clocking offers smooth send. But it does not work well with
>> modern (structured) communication patterns. Often TCP does NOT have
>> data to send when it receives acks as the application uses persistent
>> connection or is thinking. This leads to bursts up to cwnd size.
>
> True, but this is a general problem, not only TCP CWV related.

Yes. I am arguing that keep cwnd as-is is less a problem than the
burst generated w/o ack clocking in TCP. After all, there is a reason
cwnd grown to that so it's not a blind guess.

But when ack clock stops, you either trust the old rate (cwnd/rtt) or
conservatively always slow start with IW. What's the theory to do this
half way (e.g., reduce cwnd by X% after Y minute)?

Some anecdotal data: when we try the approach above (pace), the loss
rate is reduced and is as good as always slow-start after rto
(RFC5681), with better latency by keeping cwnd.

An expired draft has a great discussion on this.
http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt


>
>>
>> If the idle interval is within an RTO, nothing can save the burst
>> including this draft. If it's longer than an RTO, this draft mitigates
>> it. But I would argue cutting cwnd is not addressing the fundamental
>> problem, because TCP still sends a cwnd of burst!
>
> See above. Maybe it's worthwhile to tackle the burst problem in general.
>
>>
>> Burst induced losses are bad b/c it has not correlation to utilization
>> other than tricking TCP into congestion avoidance and low throughput.
>> Unfortunately it's hard to distinguish whether a loss is caused by
>> burst or congestion.
>>
>> My sense is that any artificial idle length or cwnd/ssthresh cut
>> factor helps but does not scale well with time and bandwidth, but
>> maybe we can simply emulate ack clocking to really reduce the burst.
>> In some terms, it's either "always slow-start if ack clock is lost" or
>> "trust and send the old cwnd/rtt rate but not at line-rate burst". The
>> latter might be favored by Web transfers as the former is known to
>> kill Web performance.
>>
>> Thoughts?
>>
>>>
>>> Gorry
>>>
>>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>> directories.
>>>> This draft is a work item of the TCP Maintenance and Minor Extensions
>>>> Working Group of the IETF.
>>>>
>>>>      Title           : Updating TCP to support Rate-Limited Traffic
>>>>      Author(s)       : Godred Fairhurst
>>>>                          Arjuna Sathiaseelan
>>>>                          Raffaello Secchi
>>>>      Filename        : draft-ietf-tcpm-newcwv-02.txt
>>>>      Pages           : 19
>>>>      Date            : 2013-07-01
>>>>
>>>> Abstract:
>>>>   This document proposes an update to RFC 5681 to address issues that
>>>>   arise when TCP is used to support traffic that exhibits periods where
>>>>   the sending rate is limited by the application rather than the
>>>>   congestion window.  It updates TCP to allow a TCP sender to restart
>>>>   quickly following either an idle or rate-limited interval.  This
>>>>   method is expected to benefit applications that send rate-limited
>>>>   traffic using TCP, while also providing an appropriate response if
>>>>   congestion is experienced.
>>>>
>>>>   It also evaluates the Experimental specification of TCP Congestion
>>>>   Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>>>>   2861 sought to address important issues, but failed to deliver a
>>>>   widely used solution.  This document therefore recommends that the
>>>>   status of RFC 2861 is moved from Experimental to Historic, and that
>>>>   it is replaced by the current specification.
>>>>
>>>>   NOTE: The standards status of this WG document is under review for
>>>>   consideration as either Experimental (EXP) or Proposed Standard (PS).
>>>>   This decision will be made later as the document is finalised.
>>>>
>>>>
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>>>
>>>> There's also a htmlized version available at:
>>>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>>>
>>>> A diff from the previous version is available at:
>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>>>>
>>>>
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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 Alexander.Zimmermann@netapp.com  Mon Jul 29 06:26:52 2013
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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Am 29.07.2013 um 09:05 schrieb Yuchung Cheng <ycheng@google.com>:

> On Mon, Jul 29, 2013 at 2:44 PM, Zimmermann, Alexander
> <Alexander.Zimmermann@netapp.com> wrote:
>> Hi Yuchung,
>>=20
>> Am 29.07.2013 um 07:56 schrieb Yuchung Cheng <ycheng@google.com>:
>>=20
>>> On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
>>>> We've just uploaded a new version of our draft. The congestion =
window
>>>> management algorithm has not changed since -00, but in -01 and -02 =
drafts
>>>> we introduce a more robust way to measure the pipeACK that attempts =
to
>>>> account for the wide variety of interactive TCP application =
behaviour.
>>>> This uses a sampling method to sample pipeACK over the last second.
>>>>=20
>>>> Clearly the cwnd poorly reflects the capacity in the non-validated =
phase
>>>> and I think it would be wrong to use this for TCP control block =
sharing.
>>>> We therefore also  introduce text that proposes that TCB sharing =
should
>>>> use the pipeACK in place of cwnd when a TCP sender is in the =
Nonvalidated
>>>> phase.  We think this value better reflects the capacity that the =
flow has
>>>> utilised in the network path - but would welcome more thought on =
this.
>>>> Thoughts on this would be most welcome.
>>>=20
>>> Hi Gorry,
>>>=20
>>> I am interested in comparing this draft with this solution:
>>>=20
>>> 1. Do not reduce cwnd on idle (of any length)
>>> 2. =46rom idle to active, send max(cwnd, data) over RTT in small =
bursts.
>>>   E.g., cwnd=3D20, RTT=3D 100ms, send 4 packets every 20ms
>>>   This emulates the ACK clocking.
>>=20
>> Smells like you propose a "IW=3Doo" here (with a straight forward =
burst
>> protection), isn't it?
> No. I am proposing: when idle keep cwnd as is.

In a extreme case - let's say CWND =3D 100 and 7 days old - it's like =
IW=3D100
with burst protection.

The propose of CWND is to reflect the BDP. After 7 days the CWND is =
everything
but not the BDP.

>=20
>>=20
>>>=20
>>> TCP ack clocking offers smooth send. But it does not work well with
>>> modern (structured) communication patterns. Often TCP does NOT have
>>> data to send when it receives acks as the application uses =
persistent
>>> connection or is thinking. This leads to bursts up to cwnd size.
>>=20
>> True, but this is a general problem, not only TCP CWV related.
>=20
> Yes. I am arguing that keep cwnd as-is is less a problem than the
> burst generated w/o ack clocking in TCP. After all, there is a reason
> cwnd grown to that so it's not a blind guess.
>=20
> But when ack clock stops, you either trust the old rate (cwnd/rtt) or
> conservatively always slow start with IW. What's the theory to do this
> half way (e.g., reduce cwnd by X% after Y minute)?
>=20
> Some anecdotal data: when we try the approach above (pace), the loss
> rate is reduced and is as good as always slow-start after rto
> (RFC5681), with better latency by keeping cwnd.
>=20
> An expired draft has a great discussion on this.
> http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
>=20
>=20
>>=20
>>>=20
>>> If the idle interval is within an RTO, nothing can save the burst
>>> including this draft. If it's longer than an RTO, this draft =
mitigates
>>> it. But I would argue cutting cwnd is not addressing the fundamental
>>> problem, because TCP still sends a cwnd of burst!
>>=20
>> See above. Maybe it's worthwhile to tackle the burst problem in =
general.
>>=20
>>>=20
>>> Burst induced losses are bad b/c it has not correlation to =
utilization
>>> other than tricking TCP into congestion avoidance and low =
throughput.
>>> Unfortunately it's hard to distinguish whether a loss is caused by
>>> burst or congestion.
>>>=20
>>> My sense is that any artificial idle length or cwnd/ssthresh cut
>>> factor helps but does not scale well with time and bandwidth, but
>>> maybe we can simply emulate ack clocking to really reduce the burst.
>>> In some terms, it's either "always slow-start if ack clock is lost" =
or
>>> "trust and send the old cwnd/rtt rate but not at line-rate burst". =
The
>>> latter might be favored by Web transfers as the former is known to
>>> kill Web performance.
>>>=20
>>> Thoughts?
>>>=20
>>>>=20
>>>> Gorry
>>>>=20
>>>>>=20
>>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>>> directories.
>>>>> This draft is a work item of the TCP Maintenance and Minor =
Extensions
>>>>> Working Group of the IETF.
>>>>>=20
>>>>>     Title           : Updating TCP to support Rate-Limited Traffic
>>>>>     Author(s)       : Godred Fairhurst
>>>>>                         Arjuna Sathiaseelan
>>>>>                         Raffaello Secchi
>>>>>     Filename        : draft-ietf-tcpm-newcwv-02.txt
>>>>>     Pages           : 19
>>>>>     Date            : 2013-07-01
>>>>>=20
>>>>> Abstract:
>>>>>  This document proposes an update to RFC 5681 to address issues =
that
>>>>>  arise when TCP is used to support traffic that exhibits periods =
where
>>>>>  the sending rate is limited by the application rather than the
>>>>>  congestion window.  It updates TCP to allow a TCP sender to =
restart
>>>>>  quickly following either an idle or rate-limited interval.  This
>>>>>  method is expected to benefit applications that send rate-limited
>>>>>  traffic using TCP, while also providing an appropriate response =
if
>>>>>  congestion is experienced.
>>>>>=20
>>>>>  It also evaluates the Experimental specification of TCP =
Congestion
>>>>>  Window Validation, CWV, defined in RFC 2861, and concludes that =
RFC
>>>>>  2861 sought to address important issues, but failed to deliver a
>>>>>  widely used solution.  This document therefore recommends that =
the
>>>>>  status of RFC 2861 is moved from Experimental to Historic, and =
that
>>>>>  it is replaced by the current specification.
>>>>>=20
>>>>>  NOTE: The standards status of this WG document is under review =
for
>>>>>  consideration as either Experimental (EXP) or Proposed Standard =
(PS).
>>>>>  This decision will be made later as the document is finalised.
>>>>>=20
>>>>>=20
>>>>> The IETF datatracker status page for this draft is:
>>>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>>>>=20
>>>>> There's also a htmlized version available at:
>>>>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>>>>=20
>>>>> A diff from the previous version is available at:
>>>>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-newcwv-02
>>>>>=20
>>>>>=20
>>>>> Internet-Drafts are also available by anonymous FTP at:
>>>>> ftp://ftp.ietf.org/internet-drafts/
>>>>>=20
>>>>> _______________________________________________
>>>>> tcpm mailing list
>>>>> tcpm@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>>>=20
>>>>=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


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References: <20130701194925.25967.70174.idtracker@ietfa.amsl.com> <f9c8b57462006f1dd53408c0c8cc5ff8.squirrel@www.erg.abdn.ac.uk> <CAK6E8=cifNh1Cf1Nd_N7k+z=K=tB4vBHMdXCem+F_++STg_Ysg@mail.gmail.com> <618CDE13-6926-4088-B318-1DA15E38985E@netapp.com> <CAK6E8=dyyde9epedO0wXD79b57tGtomRoAocAPja+JR3kU0D0g@mail.gmail.com> <BCF5A773-DB68-4AF6-887C-9F0B1ACD7884@netapp.com>
From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 29 Jul 2013 15:50:38 +0200
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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On Mon, Jul 29, 2013 at 3:26 PM, Zimmermann, Alexander
<Alexander.Zimmermann@netapp.com> wrote:
>
> Am 29.07.2013 um 09:05 schrieb Yuchung Cheng <ycheng@google.com>:
>
>> On Mon, Jul 29, 2013 at 2:44 PM, Zimmermann, Alexander
>> <Alexander.Zimmermann@netapp.com> wrote:
>>> Hi Yuchung,
>>>
>>> Am 29.07.2013 um 07:56 schrieb Yuchung Cheng <ycheng@google.com>:
>>>
>>>> On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
>>>>> We've just uploaded a new version of our draft. The congestion window
>>>>> management algorithm has not changed since -00, but in -01 and -02 drafts
>>>>> we introduce a more robust way to measure the pipeACK that attempts to
>>>>> account for the wide variety of interactive TCP application behaviour.
>>>>> This uses a sampling method to sample pipeACK over the last second.
>>>>>
>>>>> Clearly the cwnd poorly reflects the capacity in the non-validated phase
>>>>> and I think it would be wrong to use this for TCP control block sharing.
>>>>> We therefore also  introduce text that proposes that TCB sharing should
>>>>> use the pipeACK in place of cwnd when a TCP sender is in the Nonvalidated
>>>>> phase.  We think this value better reflects the capacity that the flow has
>>>>> utilised in the network path - but would welcome more thought on this.
>>>>> Thoughts on this would be most welcome.
>>>>
>>>> Hi Gorry,
>>>>
>>>> I am interested in comparing this draft with this solution:
>>>>
>>>> 1. Do not reduce cwnd on idle (of any length)
>>>> 2. From idle to active, send max(cwnd, data) over RTT in small bursts.
>>>>   E.g., cwnd=20, RTT= 100ms, send 4 packets every 20ms
>>>>   This emulates the ACK clocking.
>>>
>>> Smells like you propose a "IW=oo" here (with a straight forward burst
>>> protection), isn't it?
>> No. I am proposing: when idle keep cwnd as is.
>
> In a extreme case - let's say CWND = 100 and 7 days old - it's like IW=100
> with burst protection.
>
> The propose of CWND is to reflect the BDP. After 7 days the CWND is everything
> but not the BDP.
In the not-so-extreme case, cutting cwnd to half after x minutes and
send 50 packets are not a good idea either.

I did say "OR slow start after IW". What I object is another cwnd
reduction after an artificial idle period. Slow start is fast bw probe
clocked by acks with an initial burst (=IW), I don't see a reason to
bump the burst size just b/c a sender has transmited something X
seconds ago, because it does not mean cwnd/N burst size works.

>
>>
>>>
>>>>
>>>> TCP ack clocking offers smooth send. But it does not work well with
>>>> modern (structured) communication patterns. Often TCP does NOT have
>>>> data to send when it receives acks as the application uses persistent
>>>> connection or is thinking. This leads to bursts up to cwnd size.
>>>
>>> True, but this is a general problem, not only TCP CWV related.
>>
>> Yes. I am arguing that keep cwnd as-is is less a problem than the
>> burst generated w/o ack clocking in TCP. After all, there is a reason
>> cwnd grown to that so it's not a blind guess.
>>
>> But when ack clock stops, you either trust the old rate (cwnd/rtt) or
>> conservatively always slow start with IW. What's the theory to do this
>> half way (e.g., reduce cwnd by X% after Y minute)?
>>
>> Some anecdotal data: when we try the approach above (pace), the loss
>> rate is reduced and is as good as always slow-start after rto
>> (RFC5681), with better latency by keeping cwnd.
>>
>> An expired draft has a great discussion on this.
>> http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
>>
>>
>>>
>>>>
>>>> If the idle interval is within an RTO, nothing can save the burst
>>>> including this draft. If it's longer than an RTO, this draft mitigates
>>>> it. But I would argue cutting cwnd is not addressing the fundamental
>>>> problem, because TCP still sends a cwnd of burst!
>>>
>>> See above. Maybe it's worthwhile to tackle the burst problem in general.
>>>
>>>>
>>>> Burst induced losses are bad b/c it has not correlation to utilization
>>>> other than tricking TCP into congestion avoidance and low throughput.
>>>> Unfortunately it's hard to distinguish whether a loss is caused by
>>>> burst or congestion.
>>>>
>>>> My sense is that any artificial idle length or cwnd/ssthresh cut
>>>> factor helps but does not scale well with time and bandwidth, but
>>>> maybe we can simply emulate ack clocking to really reduce the burst.
>>>> In some terms, it's either "always slow-start if ack clock is lost" or
>>>> "trust and send the old cwnd/rtt rate but not at line-rate burst". The
>>>> latter might be favored by Web transfers as the former is known to
>>>> kill Web performance.
>>>>
>>>> Thoughts?
>>>>
>>>>>
>>>>> Gorry
>>>>>
>>>>>>
>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>>>> directories.
>>>>>> This draft is a work item of the TCP Maintenance and Minor Extensions
>>>>>> Working Group of the IETF.
>>>>>>
>>>>>>     Title           : Updating TCP to support Rate-Limited Traffic
>>>>>>     Author(s)       : Godred Fairhurst
>>>>>>                         Arjuna Sathiaseelan
>>>>>>                         Raffaello Secchi
>>>>>>     Filename        : draft-ietf-tcpm-newcwv-02.txt
>>>>>>     Pages           : 19
>>>>>>     Date            : 2013-07-01
>>>>>>
>>>>>> Abstract:
>>>>>>  This document proposes an update to RFC 5681 to address issues that
>>>>>>  arise when TCP is used to support traffic that exhibits periods where
>>>>>>  the sending rate is limited by the application rather than the
>>>>>>  congestion window.  It updates TCP to allow a TCP sender to restart
>>>>>>  quickly following either an idle or rate-limited interval.  This
>>>>>>  method is expected to benefit applications that send rate-limited
>>>>>>  traffic using TCP, while also providing an appropriate response if
>>>>>>  congestion is experienced.
>>>>>>
>>>>>>  It also evaluates the Experimental specification of TCP Congestion
>>>>>>  Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>>>>>>  2861 sought to address important issues, but failed to deliver a
>>>>>>  widely used solution.  This document therefore recommends that the
>>>>>>  status of RFC 2861 is moved from Experimental to Historic, and that
>>>>>>  it is replaced by the current specification.
>>>>>>
>>>>>>  NOTE: The standards status of this WG document is under review for
>>>>>>  consideration as either Experimental (EXP) or Proposed Standard (PS).
>>>>>>  This decision will be made later as the document is finalised.
>>>>>>
>>>>>>
>>>>>> The IETF datatracker status page for this draft is:
>>>>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>>>>>
>>>>>> There's also a htmlized version available at:
>>>>>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>>>>>
>>>>>> A diff from the previous version is available at:
>>>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>>>>>>
>>>>>>
>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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 hagen@jauu.net  Mon Jul 29 07:29:02 2013
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Date: Mon, 29 Jul 2013 16:31:00 +0200
From: Hagen Paul Pfeifer <hagen@jauu.net>
To: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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* Yuchung Cheng | 2013-07-29 15:50:38 [+0200]:

>>>> Smells like you propose a "IW=oo" here (with a straight forward burst
>>>> protection), isn't it?
>>> No. I am proposing: when idle keep cwnd as is.
>>
>> In a extreme case - let's say CWND = 100 and 7 days old - it's like IW=100
>> with burst protection.
>>
>> The propose of CWND is to reflect the BDP. After 7 days the CWND is everything
>> but not the BDP.
>
>In the not-so-extreme case, cutting cwnd to half after x minutes and
>send 50 packets are not a good idea either.

The cwnd after idle could be a function of time - cwnd is already a function
of RTT (time) we could wide the mechanism for idle periods too.

Shortly after an active period the BDP is likely to be similar.
(slowstart/congestion avoidance is based on this assumption of time/BDP
correlation). After 10 minutes the BDP can vary significant as Alexander
points out. For some networks the assumption of a similar BDP is just wrong
(e.g TDMA based networks).

Hagen

From gorry@erg.abdn.ac.uk  Mon Jul 29 07:39:48 2013
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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This is a good question.

The "pacing" case is worth thinking through, but I'm not sure it is at 
all specific to cwv, see below.

I can see that a well-paced flow can in some cases experience lower 
loss. This is good for an app when there is a small amount of capacity 
available and good for the network also in this case. It does however 
reshape the app traffic, under all cases, even when there is no capacity 
limit - which suggests an incentive to build a large cwnd before you do 
this or to pad a flow to avoid this being triggered.

Pacing: First, if we pace bursts then we create a traffic shaper. That 
isn't much of an issue for a bulk sender, where the timing of individual 
packets is not important. If a resuming app runs for > RTT this 
recreates the ACK-Clock, but there again there are other Apps where this 
simply creates additional delay by attempting to create a well-paced 
stream. If the app is bursty in nature this impacts the latency - and 
there is the possibility that better performance can be got from the App 
perspective by padding the flow (not going silent), bursting before 
going idle (to inflate cwnd)  etc, to avoid the need to pace. This to me 
seems a bad incentive.

Pacing Interval: If we consider an app that can wait for some time to 
send, and therefore it is happy with it's bursts being smoothed, the 
case you chose cwnd=20, RTT = 100ms seems like it works out as a 
reasonable pacing interval. However a flow with a smaller cwnd and 
larger RTT would then get very shaped, and one with a smaller RTT e.g. 
10ms delay would imply a short pacing interval that would compete with 
times for transmit, checksum offloading, etc. I question whether your 
proposal is practical.

The "cwn-cut" after minutes is a mechanism to deal with long-term 
reduction of rate. It's not the main mechanism, but I think it is 
essential on longer timescales to avoid the problems of apps 
individually all building large cwnds and becoming idle for extended 
periods which then effectively means they have the ability to send 
whatever they like - for ever.

My fundamental question is do we need to reshape by pacing the app data 
bursts, to use TCP safely?

Gorry

On 29/07/2013 15:50, Yuchung Cheng wrote:
> On Mon, Jul 29, 2013 at 3:26 PM, Zimmermann, Alexander
> <Alexander.Zimmermann@netapp.com> wrote:
>>
>> Am 29.07.2013 um 09:05 schrieb Yuchung Cheng <ycheng@google.com>:
>>
>>> On Mon, Jul 29, 2013 at 2:44 PM, Zimmermann, Alexander
>>> <Alexander.Zimmermann@netapp.com> wrote:
>>>> Hi Yuchung,
>>>>
>>>> Am 29.07.2013 um 07:56 schrieb Yuchung Cheng <ycheng@google.com>:
>>>>
>>>>> On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
>>>>>> We've just uploaded a new version of our draft. The congestion window
>>>>>> management algorithm has not changed since -00, but in -01 and -02 drafts
>>>>>> we introduce a more robust way to measure the pipeACK that attempts to
>>>>>> account for the wide variety of interactive TCP application behaviour.
>>>>>> This uses a sampling method to sample pipeACK over the last second.
>>>>>>
>>>>>> Clearly the cwnd poorly reflects the capacity in the non-validated phase
>>>>>> and I think it would be wrong to use this for TCP control block sharing.
>>>>>> We therefore also  introduce text that proposes that TCB sharing should
>>>>>> use the pipeACK in place of cwnd when a TCP sender is in the Nonvalidated
>>>>>> phase.  We think this value better reflects the capacity that the flow has
>>>>>> utilised in the network path - but would welcome more thought on this.
>>>>>> Thoughts on this would be most welcome.
>>>>>
>>>>> Hi Gorry,
>>>>>
>>>>> I am interested in comparing this draft with this solution:
>>>>>
>>>>> 1. Do not reduce cwnd on idle (of any length)
>>>>> 2. From idle to active, send max(cwnd, data) over RTT in small bursts.
>>>>>    E.g., cwnd=20, RTT= 100ms, send 4 packets every 20ms
>>>>>    This emulates the ACK clocking.
>>>>
>>>> Smells like you propose a "IW=oo" here (with a straight forward burst
>>>> protection), isn't it?
>>> No. I am proposing: when idle keep cwnd as is.
>>
>> In a extreme case - let's say CWND = 100 and 7 days old - it's like IW=100
>> with burst protection.
>>
>> The propose of CWND is to reflect the BDP. After 7 days the CWND is everything
>> but not the BDP.
> In the not-so-extreme case, cutting cwnd to half after x minutes and
> send 50 packets are not a good idea either.
>
> I did say "OR slow start after IW". What I object is another cwnd
> reduction after an artificial idle period. Slow start is fast bw probe
> clocked by acks with an initial burst (=IW), I don't see a reason to
> bump the burst size just b/c a sender has transmited something X
> seconds ago, because it does not mean cwnd/N burst size works.
>
>>
>>>
>>>>
>>>>>
>>>>> TCP ack clocking offers smooth send. But it does not work well with
>>>>> modern (structured) communication patterns. Often TCP does NOT have
>>>>> data to send when it receives acks as the application uses persistent
>>>>> connection or is thinking. This leads to bursts up to cwnd size.
>>>>
>>>> True, but this is a general problem, not only TCP CWV related.
>>>
>>> Yes. I am arguing that keep cwnd as-is is less a problem than the
>>> burst generated w/o ack clocking in TCP. After all, there is a reason
>>> cwnd grown to that so it's not a blind guess.
>>>
>>> But when ack clock stops, you either trust the old rate (cwnd/rtt) or
>>> conservatively always slow start with IW. What's the theory to do this
>>> half way (e.g., reduce cwnd by X% after Y minute)?
>>>
>>> Some anecdotal data: when we try the approach above (pace), the loss
>>> rate is reduced and is as good as always slow-start after rto
>>> (RFC5681), with better latency by keeping cwnd.
>>>
>>> An expired draft has a great discussion on this.
>>> http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
>>>
>>>
>>>>
>>>>>
>>>>> If the idle interval is within an RTO, nothing can save the burst
>>>>> including this draft. If it's longer than an RTO, this draft mitigates
>>>>> it. But I would argue cutting cwnd is not addressing the fundamental
>>>>> problem, because TCP still sends a cwnd of burst!
>>>>
>>>> See above. Maybe it's worthwhile to tackle the burst problem in general.
>>>>
>>>>>
>>>>> Burst induced losses are bad b/c it has not correlation to utilization
>>>>> other than tricking TCP into congestion avoidance and low throughput.
>>>>> Unfortunately it's hard to distinguish whether a loss is caused by
>>>>> burst or congestion.
>>>>>
>>>>> My sense is that any artificial idle length or cwnd/ssthresh cut
>>>>> factor helps but does not scale well with time and bandwidth, but
>>>>> maybe we can simply emulate ack clocking to really reduce the burst.
>>>>> In some terms, it's either "always slow-start if ack clock is lost" or
>>>>> "trust and send the old cwnd/rtt rate but not at line-rate burst". The
>>>>> latter might be favored by Web transfers as the former is known to
>>>>> kill Web performance.
>>>>>
>>>>> Thoughts?
>>>>>
>>>>>>
>>>>>> Gorry
>>>>>>
>>>>>>>
>>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>>>>> directories.
>>>>>>> This draft is a work item of the TCP Maintenance and Minor Extensions
>>>>>>> Working Group of the IETF.
>>>>>>>
>>>>>>>      Title           : Updating TCP to support Rate-Limited Traffic
>>>>>>>      Author(s)       : Godred Fairhurst
>>>>>>>                          Arjuna Sathiaseelan
>>>>>>>                          Raffaello Secchi
>>>>>>>      Filename        : draft-ietf-tcpm-newcwv-02.txt
>>>>>>>      Pages           : 19
>>>>>>>      Date            : 2013-07-01
>>>>>>>
>>>>>>> Abstract:
>>>>>>>   This document proposes an update to RFC 5681 to address issues that
>>>>>>>   arise when TCP is used to support traffic that exhibits periods where
>>>>>>>   the sending rate is limited by the application rather than the
>>>>>>>   congestion window.  It updates TCP to allow a TCP sender to restart
>>>>>>>   quickly following either an idle or rate-limited interval.  This
>>>>>>>   method is expected to benefit applications that send rate-limited
>>>>>>>   traffic using TCP, while also providing an appropriate response if
>>>>>>>   congestion is experienced.
>>>>>>>
>>>>>>>   It also evaluates the Experimental specification of TCP Congestion
>>>>>>>   Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>>>>>>>   2861 sought to address important issues, but failed to deliver a
>>>>>>>   widely used solution.  This document therefore recommends that the
>>>>>>>   status of RFC 2861 is moved from Experimental to Historic, and that
>>>>>>>   it is replaced by the current specification.
>>>>>>>
>>>>>>>   NOTE: The standards status of this WG document is under review for
>>>>>>>   consideration as either Experimental (EXP) or Proposed Standard (PS).
>>>>>>>   This decision will be made later as the document is finalised.
>>>>>>>
>>>>>>>
>>>>>>> The IETF datatracker status page for this draft is:
>>>>>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>>>>>>
>>>>>>> There's also a htmlized version available at:
>>>>>>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>>>>>>
>>>>>>> A diff from the previous version is available at:
>>>>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>>>>>>>
>>>>>>>
>>>>>>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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 trammell@tik.ee.ethz.ch  Mon Jul 29 07:58:18 2013
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To: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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hi Michael,

inline below...

On 29 Jul 2013, at 14:14, Michael Welzl <michawe@ifi.uio.no> wrote:

<snip>

>>> - step 6: "In this case, the sender may continue using ECN, because =
while it may not work for detecting congestion, the use of ECN does not =
negatively affect connectivity." =3D> I'd rather not have it used if =
it's not working - if a router marks ECN to CE but the next downstream =
router clears the codepoint, we ignore congestion, and we'd like to =
avoid that, no?
>>=20
>> The problem in this case is that we don't know if we have an =
ECN-aware middlebox that is clearing a CE from the end system, on the =
assumption that it is closer to the end-system than one would expect -- =
we did see some of this activity in the study detailed in the PAM paper. =
In this case the CE-clearing is arguably correct (or at least =
acceptable).=20
>=20
> I'm beginning to understand your point. It's about this sentence in =
the draft:
> "If the ECE flag is not set on the ACK segment(s) sent by the
>       receiver acknowledging the CE probe segments, the path may or =
may
>       not be usable, as that there might be middleboxes/gateways that
>       (arguably correctly) clear CE on segments from end hosts, =
because
>       they assume that congestion can not have occurred up to this
>       point on the path."
>=20
> But here you're assuming that this is the only possible reason for a =
CE to be cleared, i.e. CE cannot be cleared by a middlebox for any other =
reason. It might well be that a middlebox makes a mistake when it clears =
CE=85  well, but if that wasn't seen in any of the measurement studies =
(the IMC one and yours), I'm inclined to accept that it's okay to leave =
ECN on then.

The question is one of requirements (and therefore one of philosophy): =
are we trying to guarantee (1) that ECN works, or (2) that attempts to =
activate ECN do not lead to connectivity problems. Given where we are =
now (increasing capability in endpoints, but not much activation in =
routers and clients) we think (2) is the more useful goal: we want to =
build confidence that TCP stacks could activate ECN negotiation on =
active open by default without causing connectivity issues that are hard =
for end-users and admins to debug.

We're not assuming that CE clearing is benign (though it does read that =
way; we'll need to be more explicit about a few things that aren't as =
clear as they should be in the -01 rev) -- rather, we're assuming that =
CE clearing that doesn't result in a dropped packet indicates that ECN =
can be activated on a path, even if you'll in effect never receive an =
ECE.

Cheers,

Brian=

From gorry@erg.abdn.ac.uk  Mon Jul 29 09:07:50 2013
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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On 29/07/2013 16:31, Hagen Paul Pfeifer wrote:
> * Yuchung Cheng | 2013-07-29 15:50:38 [+0200]:
>
>>>>> Smells like you propose a "IW=oo" here (with a straight forward burst
>>>>> protection), isn't it?
>>>> No. I am proposing: when idle keep cwnd as is.
>>>
>>> In a extreme case - let's say CWND = 100 and 7 days old - it's like IW=100
>>> with burst protection.
>>>
>>> The propose of CWND is to reflect the BDP. After 7 days the CWND is everything
>>> but not the BDP.
>>
>> In the not-so-extreme case, cutting cwnd to half after x minutes and
>> send 50 packets are not a good idea either.
>
> The cwnd after idle could be a function of time - cwnd is already a function
> of RTT (time) we could wide the mechanism for idle periods too.
>
> Shortly after an active period the BDP is likely to be similar.
> (slowstart/congestion avoidance is based on this assumption of time/BDP
> correlation). After 10 minutes the BDP can vary significant as Alexander
> points out. For some networks the assumption of a similar BDP is just wrong
> (e.g TDMA based networks).
>
Your proposal is similar to the RFC 2861 approach.
draft-ietf-tcpm-newcwv adopts a different approach.

Gorry

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


From michawe@ifi.uio.no  Mon Jul 29 09:11:54 2013
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Subject: Re: [tcpm] New Version Notification for draft-kuehlewind-tcpm-ecn-fallback-00.txt
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On Jul 29, 2013, at 10:58 AM, Brian Trammell <trammell@tik.ee.ethz.ch> =
wrote:

> hi Michael,
>=20
> inline below...
>=20
> On 29 Jul 2013, at 14:14, Michael Welzl <michawe@ifi.uio.no> wrote:
>=20
> <snip>
>=20
>>>> - step 6: "In this case, the sender may continue using ECN, because =
while it may not work for detecting congestion, the use of ECN does not =
negatively affect connectivity." =3D> I'd rather not have it used if =
it's not working - if a router marks ECN to CE but the next downstream =
router clears the codepoint, we ignore congestion, and we'd like to =
avoid that, no?
>>>=20
>>> The problem in this case is that we don't know if we have an =
ECN-aware middlebox that is clearing a CE from the end system, on the =
assumption that it is closer to the end-system than one would expect -- =
we did see some of this activity in the study detailed in the PAM paper. =
In this case the CE-clearing is arguably correct (or at least =
acceptable).=20
>>=20
>> I'm beginning to understand your point. It's about this sentence in =
the draft:
>> "If the ECE flag is not set on the ACK segment(s) sent by the
>>      receiver acknowledging the CE probe segments, the path may or =
may
>>      not be usable, as that there might be middleboxes/gateways that
>>      (arguably correctly) clear CE on segments from end hosts, =
because
>>      they assume that congestion can not have occurred up to this
>>      point on the path."
>>=20
>> But here you're assuming that this is the only possible reason for a =
CE to be cleared, i.e. CE cannot be cleared by a middlebox for any other =
reason. It might well be that a middlebox makes a mistake when it clears =
CE=85  well, but if that wasn't seen in any of the measurement studies =
(the IMC one and yours), I'm inclined to accept that it's okay to leave =
ECN on then.
>=20
> The question is one of requirements (and therefore one of philosophy): =
are we trying to guarantee (1) that ECN works, or (2) that attempts to =
activate ECN do not lead to connectivity problems. Given where we are =
now (increasing capability in endpoints, but not much activation in =
routers and clients) we think (2) is the more useful goal: we want to =
build confidence that TCP stacks could activate ECN negotiation on =
active open by default without causing connectivity issues that are hard =
for end-users and admins to debug.

I agree,


> We're not assuming that CE clearing is benign (though it does read =
that way; we'll need to be more explicit about a few things that aren't =
as clear as they should be in the -01 rev) -- rather, we're assuming =
that CE clearing that doesn't result in a dropped packet indicates that =
ECN can be activated on a path, even if you'll in effect never receive =
an ECE.

I agree.

Cheers,
Michael



From andrew.knutsen@bluecoat.com  Mon Jul 29 12:35:43 2013
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Subject: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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    I'd like to ask one question again, before we implement something to 
avoid issues with a customer.

    Most of the discussion here seems to be about avoiding bursts after 
idle while at the same time avoiding slow-start.

    The problem we have is not slow-start after idle; its going into 
congestion-avoidance after idle, due to an obsolete ssthresh.

    The current draft, like RFC2861, suggests setting ssthresh to 3/4 
the old cwnd value after idle.  We've tested this, and found that 
although it is an improvement over not doing anything (ie, rfc5681),  
cwnd does not recover with this factor when presented with a series of 
bursts if ssthresh is low.  We've also tested with setting it to  full 
cwnd, and to the advertised window (which is the receive window when 
idle).  Note that all of these are more conservative than closing and 
re-opening the connection, which sets ssthresh to infinity (if it isn't 
cached).

    We've found that setting cwnd to IW after idle, and setting ssthresh 
to awnd, allows acceptable recovery from a congestion event while 
running burst traffic.  In other words, slow-start works well enough 
from IW (it's fast compared to CA).

    We've tried to start discussion about this, but it looks like the 
consensus is a lack of concern.  While we would prefer to see a spec 
which addresses our concerns, we do have pressing need to fix this issue.

Andrew


From ycheng@google.com  Mon Jul 29 13:07:56 2013
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 29 Jul 2013 22:07:28 +0200
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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On Mon, Jul 29, 2013 at 9:34 PM, Andrew Knutsen
<andrew.knutsen@bluecoat.com> wrote:
>
>    I'd like to ask one question again, before we implement something to
> avoid issues with a customer.
>
>    Most of the discussion here seems to be about avoiding bursts after idle
> while at the same time avoiding slow-start.
>
>    The problem we have is not slow-start after idle; its going into
> congestion-avoidance after idle, due to an obsolete ssthresh.
>
>    The current draft, like RFC2861, suggests setting ssthresh to 3/4 the old
> cwnd value after idle.  We've tested this, and found that although it is an
> improvement over not doing anything (ie, rfc5681),  cwnd does not recover
> with this factor when presented with a series of bursts if ssthresh is low.
> We've also tested with setting it to  full cwnd, and to the advertised
> window (which is the receive window when idle).  Note that all of these are
> more conservative than closing and re-opening the connection, which sets
> ssthresh to infinity (if it isn't cached).
>
>    We've found that setting cwnd to IW after idle, and setting ssthresh to
> awnd, allows acceptable recovery from a congestion event while running burst
> traffic.  In other words, slow-start works well enough from IW (it's fast
> compared to CA).

Ah I understand your problem now. Since RFC5681 section 4.1 does not
specify what to do with ssthresh. Why not set to infinity?

I would like to point out that fundamentally initial cwnd (IW) == idle
cwnd (RW) because ack clock is stopped. If RW can be > IW, I argue IW
should be further increased.

But there is the idea of cwnd (or rate) caching which enhances
performance. So I think the middle-ground is retain cwnd but pace over
an RTT.


"" 4.1.  Restarting Idle Connections

   A known problem with the TCP congestion control algorithms described
   above is that they allow a potentially inappropriate burst of traffic thi
   to be transmitted after TCP has been idle for a relatively long
   period of time.  After an idle period, TCP cannot use the ACK clock
   to strobe new segments into the network, as all the ACKs have drained
   from the network.  Therefore, as specified above, TCP can potentially
   send a cwnd-size line-rate burst into the network after an idle
   period.  In addition, changing network conditions may have rendered
   TCP's notion of the available end-to-end network capacity between two
   endpoints, as estimated by cwnd, inaccurate during the course of a
   long idle period.
""


>
>    We've tried to start discussion about this, but it looks like the
> consensus is a lack of concern.  While we would prefer to see a spec which
> addresses our concerns, we do have pressing need to fix this issue.
>
> Andrew
>
> _______________________________________________
> 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, 29 Jul 2013 22:25:16 +0200
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt
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On Mon, Jul 29, 2013 at 4:39 PM, Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:
> This is a good question.
>
> The "pacing" case is worth thinking through, but I'm not sure it is at all
> specific to cwv, see below.
>
> I can see that a well-paced flow can in some cases experience lower loss.
> This is good for an app when there is a small amount of capacity available
> and good for the network also in this case. It does however reshape the app
> traffic, under all cases, even when there is no capacity limit - which
> suggests an incentive to build a large cwnd before you do this or to pad a
> flow to avoid this being triggered.
>
> Pacing: First, if we pace bursts then we create a traffic shaper. That isn't
> much of an issue for a bulk sender, where the timing of individual packets
> is not important. If a resuming app runs for > RTT this recreates the
> ACK-Clock, but there again there are other Apps where this simply creates
> additional delay by attempting to create a well-paced stream. If the app is
> bursty in nature this impacts the latency - and there is the possibility
> that better performance can be got from the App perspective by padding the
> flow (not going silent), bursting before going idle (to inflate cwnd)  etc,
> to avoid the need to pace. This to me seems a bad incentive.
See my reply to Andrew's question. The original reason of RW in
RFC5681 is to avoid burst.

>
> Pacing Interval: If we consider an app that can wait for some time to send,
> and therefore it is happy with it's bursts being smoothed, the case you
> chose cwnd=20, RTT = 100ms seems like it works out as a reasonable pacing
> interval. However a flow with a smaller cwnd and larger RTT would then get
> very shaped, and one with a smaller RTT e.g. 10ms delay would imply a short
> pacing interval that would compete with times for transmit, checksum
> offloading, etc. I question whether your proposal is practical.
Only running code can answer your question :)

>
> The "cwn-cut" after minutes is a mechanism to deal with long-term reduction
> of rate. It's not the main mechanism, but I think it is essential on longer
> timescales to avoid the problems of apps individually all building large
> cwnds and becoming idle for extended periods which then effectively means
> they have the ability to send whatever they like - for ever.
>
> My fundamental question is do we need to reshape by pacing the app data
> bursts, to use TCP safely?

We don't need to. The Internet is running w/o this today. It's just (a
lot of) line-rate burst induced losses :(

>
> Gorry
>
>
> On 29/07/2013 15:50, Yuchung Cheng wrote:
>>
>> On Mon, Jul 29, 2013 at 3:26 PM, Zimmermann, Alexander
>> <Alexander.Zimmermann@netapp.com> wrote:
>>>
>>>
>>> Am 29.07.2013 um 09:05 schrieb Yuchung Cheng <ycheng@google.com>:
>>>
>>>> On Mon, Jul 29, 2013 at 2:44 PM, Zimmermann, Alexander
>>>> <Alexander.Zimmermann@netapp.com> wrote:
>>>>>
>>>>> Hi Yuchung,
>>>>>
>>>>> Am 29.07.2013 um 07:56 schrieb Yuchung Cheng <ycheng@google.com>:
>>>>>
>>>>>> On Tue, Jul 2, 2013 at 9:20 AM,  <gorry@erg.abdn.ac.uk> wrote:
>>>>>>>
>>>>>>> We've just uploaded a new version of our draft. The congestion window
>>>>>>> management algorithm has not changed since -00, but in -01 and -02
>>>>>>> drafts
>>>>>>> we introduce a more robust way to measure the pipeACK that attempts
>>>>>>> to
>>>>>>> account for the wide variety of interactive TCP application
>>>>>>> behaviour.
>>>>>>> This uses a sampling method to sample pipeACK over the last second.
>>>>>>>
>>>>>>> Clearly the cwnd poorly reflects the capacity in the non-validated
>>>>>>> phase
>>>>>>> and I think it would be wrong to use this for TCP control block
>>>>>>> sharing.
>>>>>>> We therefore also  introduce text that proposes that TCB sharing
>>>>>>> should
>>>>>>> use the pipeACK in place of cwnd when a TCP sender is in the
>>>>>>> Nonvalidated
>>>>>>> phase.  We think this value better reflects the capacity that the
>>>>>>> flow has
>>>>>>> utilised in the network path - but would welcome more thought on
>>>>>>> this.
>>>>>>> Thoughts on this would be most welcome.
>>>>>>
>>>>>>
>>>>>> Hi Gorry,
>>>>>>
>>>>>> I am interested in comparing this draft with this solution:
>>>>>>
>>>>>> 1. Do not reduce cwnd on idle (of any length)
>>>>>> 2. From idle to active, send max(cwnd, data) over RTT in small bursts.
>>>>>>    E.g., cwnd=20, RTT= 100ms, send 4 packets every 20ms
>>>>>>    This emulates the ACK clocking.
>>>>>
>>>>>
>>>>> Smells like you propose a "IW=oo" here (with a straight forward burst
>>>>> protection), isn't it?
>>>>
>>>> No. I am proposing: when idle keep cwnd as is.
>>>
>>>
>>> In a extreme case - let's say CWND = 100 and 7 days old - it's like
>>> IW=100
>>> with burst protection.
>>>
>>> The propose of CWND is to reflect the BDP. After 7 days the CWND is
>>> everything
>>> but not the BDP.
>>
>> In the not-so-extreme case, cutting cwnd to half after x minutes and
>> send 50 packets are not a good idea either.
>>
>> I did say "OR slow start after IW". What I object is another cwnd
>> reduction after an artificial idle period. Slow start is fast bw probe
>> clocked by acks with an initial burst (=IW), I don't see a reason to
>> bump the burst size just b/c a sender has transmited something X
>> seconds ago, because it does not mean cwnd/N burst size works.
>>
>>>
>>>>
>>>>>
>>>>>>
>>>>>> TCP ack clocking offers smooth send. But it does not work well with
>>>>>> modern (structured) communication patterns. Often TCP does NOT have
>>>>>> data to send when it receives acks as the application uses persistent
>>>>>> connection or is thinking. This leads to bursts up to cwnd size.
>>>>>
>>>>>
>>>>> True, but this is a general problem, not only TCP CWV related.
>>>>
>>>>
>>>> Yes. I am arguing that keep cwnd as-is is less a problem than the
>>>> burst generated w/o ack clocking in TCP. After all, there is a reason
>>>> cwnd grown to that so it's not a blind guess.
>>>>
>>>> But when ack clock stops, you either trust the old rate (cwnd/rtt) or
>>>> conservatively always slow start with IW. What's the theory to do this
>>>> half way (e.g., reduce cwnd by X% after Y minute)?
>>>>
>>>> Some anecdotal data: when we try the approach above (pace), the loss
>>>> rate is reduced and is as good as always slow-start after rto
>>>> (RFC5681), with better latency by keeping cwnd.
>>>>
>>>> An expired draft has a great discussion on this.
>>>> http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
>>>>
>>>>
>>>>>
>>>>>>
>>>>>> If the idle interval is within an RTO, nothing can save the burst
>>>>>> including this draft. If it's longer than an RTO, this draft mitigates
>>>>>> it. But I would argue cutting cwnd is not addressing the fundamental
>>>>>> problem, because TCP still sends a cwnd of burst!
>>>>>
>>>>>
>>>>> See above. Maybe it's worthwhile to tackle the burst problem in
>>>>> general.
>>>>>
>>>>>>
>>>>>> Burst induced losses are bad b/c it has not correlation to utilization
>>>>>> other than tricking TCP into congestion avoidance and low throughput.
>>>>>> Unfortunately it's hard to distinguish whether a loss is caused by
>>>>>> burst or congestion.
>>>>>>
>>>>>> My sense is that any artificial idle length or cwnd/ssthresh cut
>>>>>> factor helps but does not scale well with time and bandwidth, but
>>>>>> maybe we can simply emulate ack clocking to really reduce the burst.
>>>>>> In some terms, it's either "always slow-start if ack clock is lost" or
>>>>>> "trust and send the old cwnd/rtt rate but not at line-rate burst". The
>>>>>> latter might be favored by Web transfers as the former is known to
>>>>>> kill Web performance.
>>>>>>
>>>>>> Thoughts?
>>>>>>
>>>>>>>
>>>>>>> Gorry
>>>>>>>
>>>>>>>>
>>>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>>>>>> directories.
>>>>>>>> This draft is a work item of the TCP Maintenance and Minor
>>>>>>>> Extensions
>>>>>>>> Working Group of the IETF.
>>>>>>>>
>>>>>>>>      Title           : Updating TCP to support Rate-Limited Traffic
>>>>>>>>      Author(s)       : Godred Fairhurst
>>>>>>>>                          Arjuna Sathiaseelan
>>>>>>>>                          Raffaello Secchi
>>>>>>>>      Filename        : draft-ietf-tcpm-newcwv-02.txt
>>>>>>>>      Pages           : 19
>>>>>>>>      Date            : 2013-07-01
>>>>>>>>
>>>>>>>> Abstract:
>>>>>>>>   This document proposes an update to RFC 5681 to address issues
>>>>>>>> that
>>>>>>>>   arise when TCP is used to support traffic that exhibits periods
>>>>>>>> where
>>>>>>>>   the sending rate is limited by the application rather than the
>>>>>>>>   congestion window.  It updates TCP to allow a TCP sender to
>>>>>>>> restart
>>>>>>>>   quickly following either an idle or rate-limited interval.  This
>>>>>>>>   method is expected to benefit applications that send rate-limited
>>>>>>>>   traffic using TCP, while also providing an appropriate response if
>>>>>>>>   congestion is experienced.
>>>>>>>>
>>>>>>>>   It also evaluates the Experimental specification of TCP Congestion
>>>>>>>>   Window Validation, CWV, defined in RFC 2861, and concludes that
>>>>>>>> RFC
>>>>>>>>   2861 sought to address important issues, but failed to deliver a
>>>>>>>>   widely used solution.  This document therefore recommends that the
>>>>>>>>   status of RFC 2861 is moved from Experimental to Historic, and
>>>>>>>> that
>>>>>>>>   it is replaced by the current specification.
>>>>>>>>
>>>>>>>>   NOTE: The standards status of this WG document is under review for
>>>>>>>>   consideration as either Experimental (EXP) or Proposed Standard
>>>>>>>> (PS).
>>>>>>>>   This decision will be made later as the document is finalised.
>>>>>>>>
>>>>>>>>
>>>>>>>> The IETF datatracker status page for this draft is:
>>>>>>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv
>>>>>>>>
>>>>>>>> There's also a htmlized version available at:
>>>>>>>> http://tools.ietf.org/html/draft-ietf-tcpm-newcwv-02
>>>>>>>>
>>>>>>>> A diff from the previous version is available at:
>>>>>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-02
>>>>>>>>
>>>>>>>>
>>>>>>>> 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
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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 andrew.knutsen@bluecoat.com  Mon Jul 29 13:36:50 2013
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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    It is possible that there are ways to retain cwnd after idle and use 
some other mechanism to avoid inappropriate bursts. However, in the 
cases we've seen, slow-start from IW works just fine for this.  Its CA 
from IW we have issues with, and it seems like there are simple solutions.

    I don't mean to say there isn't room for more complex solutions 
involving ramping cwnd and perhaps simulating ack clocking, but its 
seems like we'll still want to deal with ssthresh.

Andrew

On 7/29/2013 1:07 PM, Yuchung Cheng wrote:
> On Mon, Jul 29, 2013 at 9:34 PM, Andrew Knutsen
> <andrew.knutsen@bluecoat.com> wrote:
>>     I'd like to ask one question again, before we implement something to
>> avoid issues with a customer.
>>
>>     Most of the discussion here seems to be about avoiding bursts after idle
>> while at the same time avoiding slow-start.
>>
>>     The problem we have is not slow-start after idle; its going into
>> congestion-avoidance after idle, due to an obsolete ssthresh.
>>
>>     The current draft, like RFC2861, suggests setting ssthresh to 3/4 the old
>> cwnd value after idle.  We've tested this, and found that although it is an
>> improvement over not doing anything (ie, rfc5681),  cwnd does not recover
>> with this factor when presented with a series of bursts if ssthresh is low.
>> We've also tested with setting it to  full cwnd, and to the advertised
>> window (which is the receive window when idle).  Note that all of these are
>> more conservative than closing and re-opening the connection, which sets
>> ssthresh to infinity (if it isn't cached).
>>
>>     We've found that setting cwnd to IW after idle, and setting ssthresh to
>> awnd, allows acceptable recovery from a congestion event while running burst
>> traffic.  In other words, slow-start works well enough from IW (it's fast
>> compared to CA).
> Ah I understand your problem now. Since RFC5681 section 4.1 does not
> specify what to do with ssthresh. Why not set to infinity?
>
> I would like to point out that fundamentally initial cwnd (IW) == idle
> cwnd (RW) because ack clock is stopped. If RW can be > IW, I argue IW
> should be further increased.
>
> But there is the idea of cwnd (or rate) caching which enhances
> performance. So I think the middle-ground is retain cwnd but pace over
> an RTT.
>
>
> "" 4.1.  Restarting Idle Connections
>
>     A known problem with the TCP congestion control algorithms described
>     above is that they allow a potentially inappropriate burst of traffic thi
>     to be transmitted after TCP has been idle for a relatively long
>     period of time.  After an idle period, TCP cannot use the ACK clock
>     to strobe new segments into the network, as all the ACKs have drained
>     from the network.  Therefore, as specified above, TCP can potentially
>     send a cwnd-size line-rate burst into the network after an idle
>     period.  In addition, changing network conditions may have rendered
>     TCP's notion of the available end-to-end network capacity between two
>     endpoints, as estimated by cwnd, inaccurate during the course of a
>     long idle period.
> ""
>
>
>>     We've tried to start discussion about this, but it looks like the
>> consensus is a lack of concern.  While we would prefer to see a spec which
>> addresses our concerns, we do have pressing need to fix this issue.
>>
>> Andrew
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


From prvs=09226a86b1=anna.brunstrom@kau.se  Mon Jul 29 15:35:10 2013
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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Hi Andrew,

I agree with you that the value of ssthresh is important and that this 
part may need to be updated. If subsequent bursts are not long enough to 
take you out of an unlucky state, a random loss in the the first burst 
may hamper performance for all later bursts by invoking CA too early.

We have seen the equivalent problem when you have a sequence of short 
flows and the ssthresh value is cached. A random loss in one short flow 
can reduce performance for all the short flows that follow. See our ICC 
paper from last year for details 
http://dx.doi.org/10.1109/ICC.2012.6364516.

BR,
Anna


On 2013-07-29 21:34, Andrew Knutsen wrote:
>
>     I'd like to ask one question again, before we implement something to
> avoid issues with a customer.
>
>     Most of the discussion here seems to be about avoiding bursts after
> idle while at the same time avoiding slow-start.
>
>     The problem we have is not slow-start after idle; its going into
> congestion-avoidance after idle, due to an obsolete ssthresh.
>
>     The current draft, like RFC2861, suggests setting ssthresh to 3/4
> the old cwnd value after idle.  We've tested this, and found that
> although it is an improvement over not doing anything (ie, rfc5681),
> cwnd does not recover with this factor when presented with a series of
> bursts if ssthresh is low.  We've also tested with setting it to  full
> cwnd, and to the advertised window (which is the receive window when
> idle).  Note that all of these are more conservative than closing and
> re-opening the connection, which sets ssthresh to infinity (if it isn't
> cached).
>
>     We've found that setting cwnd to IW after idle, and setting ssthresh
> to awnd, allows acceptable recovery from a congestion event while
> running burst traffic.  In other words, slow-start works well enough
> from IW (it's fast compared to CA).
>
>     We've tried to start discussion about this, but it looks like the
> consensus is a lack of concern.  While we would prefer to see a spec
> which addresses our concerns, we do have pressing need to fix this issue.
>
> Andrew
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From fernando@gont.com.ar  Mon Jul 29 15:44:00 2013
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Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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HI, Joe,

On 07/23/2013 05:48 PM, Joe Touch wrote:
> 
> OK, I've been convinced that supporting such loopback isn't prohibited
> by 793, but I remain convinced that supporting it within the protocol
> isn't strictly required.

Well, there's nothing in RFC793 that relieves you from supportng that
corner case....



> There are other cases where TCP implementations take shortcuts when it
> knows better - e.g., ignoring congestion control when both ends are on
> the same subnet. TCP could just as easily ignore the entire state
> machine when both sockets are identical and just reflect everything back
> in one step, or the Un*x socket layer could do so directly.

In the same way it could do that when both endpoints are on the same node...



> IMO, an app that wants to talk to itself ought to either use different
> ports or different looback addresses (or both) - though, given my
> experience with Linux of late (65K total connection limit), I'd be more
> surprised if that works too.

What would be the rationale for that "ought to"?


> However, the case below also hints at a bug where both ends enter
> FIN-WAIT-1 at the same time - which can happen whether reflexive or not,
> and should be fixed anyway.

Agreed.

Thanks for the input, by the way!

Cheers,
Fernando




-- 
-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@si6networks.com
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1




From touch@isi.edu  Mon Jul 29 16:13:37 2013
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On 7/29/2013 3:43 PM, Fernando Gont wrote:
> HI, Joe,
>
> On 07/23/2013 05:48 PM, Joe Touch wrote:
>>
>> OK, I've been convinced that supporting such loopback isn't prohibited
>> by 793, but I remain convinced that supporting it within the protocol
>> isn't strictly required.
>
> Well, there's nothing in RFC793 that relieves you from supportng that
> corner case....

That depends on the interpretation of a few key sentences in that RFC.

>> There are other cases where TCP implementations take shortcuts when it
>> knows better - e.g., ignoring congestion control when both ends are on
>> the same subnet. TCP could just as easily ignore the entire state
>> machine when both sockets are identical and just reflect everything back
>> in one step, or the Un*x socket layer could do so directly.
>
> In the same way it could do that when both endpoints are on the same node...

Yes, but that would not require support by the actual exchange of 
reflected messages.

>> IMO, an app that wants to talk to itself ought to either use different
>> ports or different looback addresses (or both) - though, given my
>> experience with Linux of late (65K total connection limit), I'd be more
>> surprised if that works too.
>
> What would be the rationale for that "ought to"?

There seems little point in pushing a corner case of TCP when it can be 
avoided by implementing both ends independently. There are 250+ loopback 
addresses.

Joe

From fernando@gont.com.ar  Mon Jul 29 16:48:10 2013
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On 07/30/2013 01:13 AM, Joe Touch wrote:
>> On 07/23/2013 05:48 PM, Joe Touch wrote:
>>>
>>> OK, I've been convinced that supporting such loopback isn't prohibited
>>> by 793, but I remain convinced that supporting it within the protocol
>>> isn't strictly required.
>>
>> Well, there's nothing in RFC793 that relieves you from supportng that
>> corner case....
> 
> That depends on the interpretation of a few key sentences in that RFC.

May be. The way I rad it is that you need a local endpoint, and a remote
endpoint. But there's nothing in RFC 793 that requires both endpoints to
be different.



>>> There are other cases where TCP implementations take shortcuts when it
>>> knows better - e.g., ignoring congestion control when both ends are on
>>> the same subnet. TCP could just as easily ignore the entire state
>>> machine when both sockets are identical and just reflect everything back
>>> in one step, or the Un*x socket layer could do so directly.
>>
>> In the same way it could do that when both endpoints are on the same
>> node...
> 
> Yes, but that would not require support by the actual exchange of
> reflected messages.

Well, you actually need special code to *not* support reflected
endpoints -- at the end of the day, reflected endpoints are just a
specific case of "simultaneous open".



>>> IMO, an app that wants to talk to itself ought to either use different
>>> ports or different looback addresses (or both) - though, given my
>>> experience with Linux of late (65K total connection limit), I'd be more
>>> surprised if that works too.
>>
>> What would be the rationale for that "ought to"?
> 
> There seems little point in pushing a corner case of TCP when it can be
> avoided by implementing both ends independently. There are 250+ loopback
> addresses.

I see it exactly the other way around: you need additional code not to
support mirrored end-points, since you essentially have to disable a
corner case that *would* work if you already support simultaneous opens
(which you must support).

Cheers,
-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@si6networks.com
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1




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On 7/29/2013 4:47 PM, Fernando Gont wrote:
> at the end of the day, reflected endpoints are just a
> specific case of "simultaneous open".

They're more than that.

They're simultaneous everything, with the same ISN.

FWIW, an app needs special code to deal with the self-reflected case 
anyway - it needs to know that only one incarnation should close the 
connection, vs. giving each incarnation a unique socket pair.

I recall that there have also been extensions that don't handle the 
simultaneous case - that basically just punt, and say "try again" when 
it happens. When these are two different sockets, that's a reasonable 
approach; when it's a single reflexive socket, it would continually fail.

Joe

From touch@isi.edu  Mon Jul 29 17:36:45 2013
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On 7/29/2013 6:05 AM, Yuchung Cheng wrote:
> An expired draft has a great discussion on this.
> http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt

Yes, but it proposes solutions that are not being considered, AFAICT.

That draft was concerned *anytime* the window grew beyond what was being 
actively used, regardless of whether that happens all at once (idle 
period) or over time (e.g., a source that can't send as much as CWND 
allows, but the ACKs keep accumulating permission to burst).

We felt that the better solution was to limit the ability to accumulate 
that permission. Note that this was also an issue with Lixia's original 
virtual clock mechanism, FWIW.

I'm still not sure why this isn't relevant here; is it more useful to 
validate a large unused permission-to-send window, or just prohibit 
hoarding that as a possible resource?

Joe

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On 7/29/2013 7:39 AM, Gorry Fairhurst wrote:
> The "pacing" case is worth thinking through, but I'm not sure it is at
> all specific to cwv, see below.

In this doc, http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt, 
we considered pacing too, but it was suboptimal for a number of reasons 
(complexity, the need to determine additional parameters, and the lag in 
response time).

Joe

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On 07/30/2013 01:55 AM, Joe Touch wrote:
> On 7/29/2013 4:47 PM, Fernando Gont wrote:
>> at the end of the day, reflected endpoints are just a
>> specific case of "simultaneous open".
> 
> They're more than that.
> 
> They're simultaneous everything, with the same ISN.

Does it really matter that the ISNs are the same? (packets are not
actaully traveling over a network)



> FWIW, an app needs special code to deal with the self-reflected case
> anyway - it needs to know that only one incarnation should close the
> connection, vs. giving each incarnation a unique socket pair.

Well, if the connection is closed, you get a simultaneous close.

[Yes, if anything, I would only expect applications that "really know
what they're doing" to use this "mirrored-endpoints connections"...
However, when ti comes to kernel code, it takes more work to not support
them , than to do it (assuming that you support simultaneous opens,
simultaneous closes, etc.)]



> I recall that there have also been extensions that don't handle the
> simultaneous case - that basically just punt, and say "try again" when
> it happens. When these are two different sockets, that's a reasonable
> approach; when it's a single reflexive socket, it would continually fail.

Me, I'd argue that if e.g. simultaneous-opens are part of TCP, your
extension should work with them rather than rely on "let's retry and
hope that the SYNs don't cross on the network".

Cheers,
-- 
Fernando Gont
SI6 Networks
e-mail: fgont@si6networks.com
PGP Fingerprint: 6666 31C6 D484 63B2 8FB1 E3C4 AE25 0D55 1D4E 7492





From touch@isi.edu  Mon Jul 29 18:19:03 2013
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On 7/29/2013 5:58 PM, Fernando Gont wrote:
> On 07/30/2013 01:55 AM, Joe Touch wrote:
>> On 7/29/2013 4:47 PM, Fernando Gont wrote:
>>> at the end of the day, reflected endpoints are just a
>>> specific case of "simultaneous open".
>>
>> They're more than that.
>>
>> They're simultaneous everything, with the same ISN.
>
> Does it really matter that the ISNs are the same? (packets are not
> actaully traveling over a network)

It doesn't to me, but again this is a corner case that some systems 
don't expect.

>> FWIW, an app needs special code to deal with the self-reflected case
>> anyway - it needs to know that only one incarnation should close the
>> connection, vs. giving each incarnation a unique socket pair.
>
> Well, if the connection is closed, you get a simultaneous close.

So from an app-writer's viewpoint, why would all connections be 
simultaneous open and simultaneous close? What if that's a case that 
either the protocol or the app doesn't want - maybe it wants to try to 
de-sync by executing a timer and retrying.

It won't work on a reflexive socket. It will work everywhere else.

> [Yes, if anything, I would only expect applications that "really know
> what they're doing" to use this "mirrored-endpoints connections"...
> However, when ti comes to kernel code, it takes more work to not support
> them , than to do it (assuming that you support simultaneous opens,
> simultaneous closes, etc.)]

I disagree; the work to not do it is very simple (it's just a test). The 
work to ensure it always works could be a lot more.

The key question isn't whether this is a 'fun case' that the protocol 
ought to handle; it's whether we should bother to ensure it is always 
supported. IMO, the special case is a lot simpler than testing every 
possible alternative.

>> I recall that there have also been extensions that don't handle the
>> simultaneous case - that basically just punt, and say "try again" when
>> it happens. When these are two different sockets, that's a reasonable
>> approach; when it's a single reflexive socket, it would continually fail.
>
> Me, I'd argue that if e.g. simultaneous-opens are part of TCP, your
> extension should work with them rather than rely on "let's retry and
> hope that the SYNs don't cross on the network".

"handling simultaneous opens" can be done two ways:
	- support them
	- don't support them, and retry to de-sync

The latter will never work with reflexive connections, but it's just as 
valid an approach.

Joe

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Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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On 07/30/2013 03:18 AM, Joe Touch wrote:

>> Does it really matter that the ISNs are the same? (packets are not
>> actaully traveling over a network)
> 
> It doesn't to me, but again this is a corner case that some systems
> don't expect.

If it doesn't matter, then there's no problem if systems do not expect it.


> 
>>> FWIW, an app needs special code to deal with the self-reflected case
>>> anyway - it needs to know that only one incarnation should close the
>>> connection, vs. giving each incarnation a unique socket pair.
>>
>> Well, if the connection is closed, you get a simultaneous close.
> 
> So from an app-writer's viewpoint, why would all connections be
> simultaneous open and simultaneous close? What if that's a case that
> either the protocol or the app doesn't want 

You just don't use mirrored endpoints.



>> [Yes, if anything, I would only expect applications that "really know
>> what they're doing" to use this "mirrored-endpoints connections"...
>> However, when ti comes to kernel code, it takes more work to not support
>> them , than to do it (assuming that you support simultaneous opens,
>> simultaneous closes, etc.)]
> 
> I disagree; the work to not do it is very simple (it's just a test). 

THat's additional code -- no matter how simple it is.



> The
> work to ensure it always works could be a lot more.

That's assuming your TCP impleemntation is broken in terms of
simultaneous opens, simultaneous closes, etc. -- i.e., if your
implementation is buggy, fix it.



> The key question isn't whether this is a 'fun case' that the protocol
> ought to handle; it's whether we should bother to ensure it is always
> supported. IMO, the special case is a lot simpler than testing every
> possible alternative.

Not sure what you mean.




>> Me, I'd argue that if e.g. simultaneous-opens are part of TCP, your
>> extension should work with them rather than rely on "let's retry and
>> hope that the SYNs don't cross on the network".
> 
> "handling simultaneous opens" can be done two ways:
>     - support them
>     - don't support them, and retry to de-sync
> 
> The latter will never work with reflexive connections, but it's just as
> valid an approach.

So.. yu're arguing that a piece of of RFC 793 (that is part of TCP)
should not be supported?

Part of TCP is support for simultaneous opens. If you don't support
them, ether that's not TCP, or that's a buggy implementation.

Cheers,
-- 
Fernando Gont
SI6 Networks
e-mail: fgont@si6networks.com
PGP Fingerprint: 6666 31C6 D484 63B2 8FB1 E3C4 AE25 0D55 1D4E 7492





From touch@isi.edu  Mon Jul 29 19:56:13 2013
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From: Joe Touch <touch@isi.edu>
Date: Mon, 29 Jul 2013 19:55:32 -0700
To: Fernando Gont <fgont@si6networks.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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On Jul 29, 2013, at 6:28 PM, Fernando Gont <fgont@si6networks.com> wrote:

> On 07/30/2013 03:18 AM, Joe Touch wrote:
>=20
>>=20
>>>> FWIW, an app needs special code to deal with the self-reflected case
>>>> anyway - it needs to know that only one incarnation should close the
>>>> connection, vs. giving each incarnation a unique socket pair.
>>>=20
>>> Well, if the connection is closed, you get a simultaneous close.
>>=20
>> So from an app-writer's viewpoint, why would all connections be
>> simultaneous open and simultaneous close? What if that's a case that
>> either the protocol or the app doesn't want
>=20
> You just don't use mirrored endpoints.

But that's the point. If they behave differently they're not useful.=20

>> The
>> work to ensure it always works could be a lot more.
>=20
> That's assuming your TCP impleemntation is broken in terms of
> simultaneous opens, simultaneous closes, etc. -- i.e., if your
> implementation is buggy, fix it.

It's more than that. Simultaneous < reflexive.=20

>> The key question isn't whether this is a 'fun case' that the protocol
>> ought to handle; it's whether we should bother to ensure it is always
>> supported. IMO, the special case is a lot simpler than testing every
>> possible alternative.
>=20
> Not sure what you mean.

Not sue I can explain better.  This is the crux of my concern. It's cute ths=
t it could work but not necessary that it should. =20

>>> Me, I'd argue that if e.g. simultaneous-opens are part of TCP, your
>>> extension should work with them rather than rely on "let's retry and
>>> hope that the SYNs don't cross on the network".
>>=20
>> "handling simultaneous opens" can be done two ways:
>>    - support them
>>    - don't support them, and retry to de-sync
>>=20
>> The latter will never work with reflexive connections, but it's just as
>> valid an approach.
>=20
> So.. yu're arguing that a piece of of RFC 793 (that is part of TCP)
> should not be supported?
>=20
> Part of TCP is support for simultaneous opens. If you don't support
> them, ether that's not TCP, or that's a buggy implementation.

Again simultaneous < reflexive.=20

Joe

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From: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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On Mon, Jul 29, 2013 at 10:36 PM, Andrew Knutsen
<andrew.knutsen@bluecoat.com> wrote:
>
>    It is possible that there are ways to retain cwnd after idle and use some
> other mechanism to avoid inappropriate bursts. However, in the cases we've
> seen, slow-start from IW works just fine for this.  Its CA from IW we have
> issues with, and it seems like there are simple solutions.
+1

>
>    I don't mean to say there isn't room for more complex solutions involving
> ramping cwnd and perhaps simulating ack clocking, but its seems like we'll
> still want to deal with ssthresh.
yes. it will be nice get some empirical study of setting ssthresh to
inifinity after idle

>
> Andrew
>
>
> On 7/29/2013 1:07 PM, Yuchung Cheng wrote:
>>
>> On Mon, Jul 29, 2013 at 9:34 PM, Andrew Knutsen
>> <andrew.knutsen@bluecoat.com> wrote:
>>>
>>>     I'd like to ask one question again, before we implement something to
>>> avoid issues with a customer.
>>>
>>>     Most of the discussion here seems to be about avoiding bursts after
>>> idle
>>> while at the same time avoiding slow-start.
>>>
>>>     The problem we have is not slow-start after idle; its going into
>>> congestion-avoidance after idle, due to an obsolete ssthresh.
>>>
>>>     The current draft, like RFC2861, suggests setting ssthresh to 3/4 the
>>> old
>>> cwnd value after idle.  We've tested this, and found that although it is
>>> an
>>> improvement over not doing anything (ie, rfc5681),  cwnd does not recover
>>> with this factor when presented with a series of bursts if ssthresh is
>>> low.
>>> We've also tested with setting it to  full cwnd, and to the advertised
>>> window (which is the receive window when idle).  Note that all of these
>>> are
>>> more conservative than closing and re-opening the connection, which sets
>>> ssthresh to infinity (if it isn't cached).
>>>
>>>     We've found that setting cwnd to IW after idle, and setting ssthresh
>>> to
>>> awnd, allows acceptable recovery from a congestion event while running
>>> burst
>>> traffic.  In other words, slow-start works well enough from IW (it's fast
>>> compared to CA).
>>
>> Ah I understand your problem now. Since RFC5681 section 4.1 does not
>> specify what to do with ssthresh. Why not set to infinity?
>>
>> I would like to point out that fundamentally initial cwnd (IW) == idle
>> cwnd (RW) because ack clock is stopped. If RW can be > IW, I argue IW
>> should be further increased.
>>
>> But there is the idea of cwnd (or rate) caching which enhances
>> performance. So I think the middle-ground is retain cwnd but pace over
>> an RTT.
>>
>>
>> "" 4.1.  Restarting Idle Connections
>>
>>     A known problem with the TCP congestion control algorithms described
>>     above is that they allow a potentially inappropriate burst of traffic
>> thi
>>     to be transmitted after TCP has been idle for a relatively long
>>     period of time.  After an idle period, TCP cannot use the ACK clock
>>     to strobe new segments into the network, as all the ACKs have drained
>>     from the network.  Therefore, as specified above, TCP can potentially
>>     send a cwnd-size line-rate burst into the network after an idle
>>     period.  In addition, changing network conditions may have rendered
>>     TCP's notion of the available end-to-end network capacity between two
>>     endpoints, as estimated by cwnd, inaccurate during the course of a
>>     long idle period.
>> ""
>>
>>
>>>     We've tried to start discussion about this, but it looks like the
>>> consensus is a lack of concern.  While we would prefer to see a spec
>>> which
>>> addresses our concerns, we do have pressing need to fix this issue.
>>>
>>> Andrew
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>
>

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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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HI,

We also have observed issues with ssthresh setting after idle.
This regards both TCP and SCTP usecase scenarios.

Our thoughts have been to reset ssthresh to inifinity after idle.
Setting to awnd is not a good solution in SCTP (or for mptcp) as the awnd=20
may be small even if an idle period has been observed on a specific path on=
 which traffic now resumes.

Possible SCTP or multipath issues are not within cwv, but still I would lik=
e to point to this fact.

BR, Karen


>-----Original Message-----
>From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of An=
na
>Brunstrom
>Sent: Tuesday, July 30, 2013 12:35 AM
>To: tcpm@ietf.org
>Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
>
>Hi Andrew,
>
>I agree with you that the value of ssthresh is important and that this par=
t may need
>to be updated. If subsequent bursts are not long enough to take you out of=
 an
>unlucky state, a random loss in the the first burst may hamper performance=
 for all
>later bursts by invoking CA too early.
>
>We have seen the equivalent problem when you have a sequence of short flow=
s and
>the ssthresh value is cached. A random loss in one short flow can reduce
>performance for all the short flows that follow. See our ICC paper from la=
st year for
>details http://dx.doi.org/10.1109/ICC.2012.6364516.
>
>BR,
>Anna
>
>
>On 2013-07-29 21:34, Andrew Knutsen wrote:
>>
>>     I'd like to ask one question again, before we implement something to
>> avoid issues with a customer.
>>
>>     Most of the discussion here seems to be about avoiding bursts after
>> idle while at the same time avoiding slow-start.
>>
>>     The problem we have is not slow-start after idle; its going into
>> congestion-avoidance after idle, due to an obsolete ssthresh.
>>
>>     The current draft, like RFC2861, suggests setting ssthresh to 3/4
>> the old cwnd value after idle.  We've tested this, and found that
>> although it is an improvement over not doing anything (ie, rfc5681),
>> cwnd does not recover with this factor when presented with a series of
>> bursts if ssthresh is low.  We've also tested with setting it to  full
>> cwnd, and to the advertised window (which is the receive window when
>> idle).  Note that all of these are more conservative than closing and
>> re-opening the connection, which sets ssthresh to infinity (if it isn't
>> cached).
>>
>>     We've found that setting cwnd to IW after idle, and setting ssthresh
>> to awnd, allows acceptable recovery from a congestion event while
>> running burst traffic.  In other words, slow-start works well enough
>> from IW (it's fast compared to CA).
>>
>>     We've tried to start discussion about this, but it looks like the
>> consensus is a lack of concern.  While we would prefer to see a spec
>> which addresses our concerns, we do have pressing need to fix this issue=
.
>>
>> Andrew
>>
>> _______________________________________________
>> 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 michawe@ifi.uio.no  Tue Jul 30 01:13:06 2013
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Subject: [tcpm] draft-ietf-tcpm-newcwv-02.txt: new and old connections
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Hi,

In the discussion in TCPM just now, it seemed that people look at a =
connection that goes idle and later sends again as being roughly the =
same thing as a connection that stops and a new connection that starts. =
But: if there is a box in the path that does ECMP, this box may use the =
SYN and/or the use of a new port number (if only on the client side) as =
a basis to pick a path, and so a second connection starting after one =
that stopped may traverse a different path, right?

If I'm right then this is perhaps quite a different thing=85  (but note =
that I have no experience with boxes doing stuff like I'm writing here, =
this is just guesswork)

Cheers,
Michael


From fernando@gont.com.ar  Tue Jul 30 01:16:19 2013
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Subject: Re: [tcpm] Fwd: TCP Loopback Connections with the Same Src/Dest Port
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On 07/30/2013 04:55 AM, Joe Touch wrote:
>>
>> Part of TCP is support for simultaneous opens. If you don't support
>> them, ether that's not TCP, or that's a buggy implementation.
> 
> Again simultaneous < reflexive. 

There is no difference in the TCP code that is required for
mirrored-endpoint-connections than the code that is required for
simultaneous-{open, close} to work.

In any case, my take is that the high-order-bit of this I-D is not the
mirrered-endpoint connections, but rather all the other stuff: the
update that is needed for simultaneous opens, simultaneous closes, and
simultaneous probes to work as expected -- and it looks like everyone
(both of us, and other folks that have viced their opinion, plus the
implementations out there) agrees that that must be addressed.

The mirrored-endpoints corner case is something on which I have a
position, but, as noted, is a secondary issue (at the end of the day, I
could myself live with not supporting them).

Thanks,
-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@si6networks.com
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1




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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] draft-ietf-tcpm-newcwv-02.txt: new and old connections
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You're correct in that changing the port number on one end will probably
cause a different ECMP path to become active.  However, in most cases the
operator is trying very hard to keep the ECMP paths balanced, so I believe
this is OK.  If not, they're doing complex traffic engineering, and chances
are the path can change mid-connection anyway as the TE process reacts to
load.  So I don't believe this to be a serious issue in either case.


On 30 July 2013 01:12, Michael Welzl <michawe@ifi.uio.no> wrote:

> Hi,
>
> In the discussion in TCPM just now, it seemed that people look at a
> connection that goes idle and later sends again as being roughly the same
> thing as a connection that stops and a new connection that starts. But: i=
f
> there is a box in the path that does ECMP, this box may use the SYN and/o=
r
> the use of a new port number (if only on the client side) as a basis to
> pick a path, and so a second connection starting after one that stopped m=
ay
> traverse a different path, right?
>
> If I'm right then this is perhaps quite a different thing=85  (but note t=
hat
> I have no experience with boxes doing stuff like I'm writing here, this i=
s
> just guesswork)
>
> Cheers,
> Michael
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



--=20
Andrew McGregor | SRE | andrewmcgr@google.com | +61 4 8143 7128

--089e0158b34c98aa4804e2b6404b
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">You&#39;re correct in that changing the port number on one=
 end will probably cause a different ECMP path to become active. =A0However=
, in most cases the operator is trying very hard to keep the ECMP paths bal=
anced, so I believe this is OK. =A0If not, they&#39;re doing complex traffi=
c engineering, and chances are the path can change mid-connection anyway as=
 the TE process reacts to load. =A0So I don&#39;t believe this to be a seri=
ous issue in either case.</div>
<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On 30 July 20=
13 01:12, Michael Welzl <span dir=3D"ltr">&lt;<a href=3D"mailto:michawe@ifi=
.uio.no" target=3D"_blank">michawe@ifi.uio.no</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">
Hi,<br>
<br>
In the discussion in TCPM just now, it seemed that people look at a connect=
ion that goes idle and later sends again as being roughly the same thing as=
 a connection that stops and a new connection that starts. But: if there is=
 a box in the path that does ECMP, this box may use the SYN and/or the use =
of a new port number (if only on the client side) as a basis to pick a path=
, and so a second connection starting after one that stopped may traverse a=
 different path, right?<br>

<br>
If I&#39;m right then this is perhaps quite a different thing=85 =A0(but no=
te that I have no experience with boxes doing stuff like I&#39;m writing he=
re, this is just guesswork)<br>
<br>
Cheers,<br>
Michael<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" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><div dir=3D"=
ltr"><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:sm=
all;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-co=
lor:rgb(213,15,37);padding-top:2px;margin-top:2px">Andrew McGregor=A0|</spa=
n><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:small=
;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-color=
:rgb(51,105,232);padding-top:2px;margin-top:2px">=A0SRE=A0|</span><span sty=
le=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:small;line-heigh=
t:1.5em;border-width:2px 0px 0px;border-style:solid;border-color:rgb(0,153,=
57);padding-top:2px;margin-top:2px">=A0<a href=3D"mailto:andrewmcgr@google.=
com" target=3D"_blank">andrewmcgr@google.com</a>=A0|</span><span style=3D"c=
olor:rgb(85,85,85);font-family:sans-serif;font-size:small;line-height:1.5em=
;border-width:2px 0px 0px;border-style:solid;border-color:rgb(238,178,17);p=
adding-top:2px;margin-top:2px">=A0+61 4 8143 7128</span><br>
</div>
</div>

--089e0158b34c98aa4804e2b6404b--

From gorry@erg.abdn.ac.uk  Tue Jul 30 01:32:42 2013
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Date: Tue, 30 Jul 2013 10:30:33 +0200
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tcpm] draft-ietf-tcpm-newcwv-02.txt: new and old connections
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I agree. In theory a new port generates new entropy that *could* result 
in a new ECMP path. I'm not sure this is a new major issue for cwv.

Gorry

On 30/07/2013 10:18, Andrew Mcgregor wrote:
> You're correct in that changing the port number on one end will probably
> cause a different ECMP path to become active.  However, in most cases
> the operator is trying very hard to keep the ECMP paths balanced, so I
> believe this is OK.  If not, they're doing complex traffic engineering,
> and chances are the path can change mid-connection anyway as the TE
> process reacts to load.  So I don't believe this to be a serious issue
> in either case.
>
>
> On 30 July 2013 01:12, Michael Welzl <michawe@ifi.uio.no
> <mailto:michawe@ifi.uio.no>> wrote:
>
>     Hi,
>
>     In the discussion in TCPM just now, it seemed that people look at a
>     connection that goes idle and later sends again as being roughly the
>     same thing as a connection that stops and a new connection that
>     starts. But: if there is a box in the path that does ECMP, this box
>     may use the SYN and/or the use of a new port number (if only on the
>     client side) as a basis to pick a path, and so a second connection
>     starting after one that stopped may traverse a different path, right?
>
>     If I'm right then this is perhaps quite a different thing…  (but
>     note that I have no experience with boxes doing stuff like I'm
>     writing here, this is just guesswork)
>
>     Cheers,
>     Michael
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>
>
>
>
> --
> Andrew McGregor | SRE |andrewmcgr@google.com
> <mailto:andrewmcgr@google.com> | +61 4 8143 7128
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From michawe@ifi.uio.no  Tue Jul 30 02:14:30 2013
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Subject: Re: [tcpm] draft-ietf-tcpm-newcwv-02.txt: new and old connections
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It isn't an issue for cwv, but it may be a reason why the logic: "if we =
allow such a large burst for cwv, we could just as well allow an even =
larger IW" may not hold.

Cheers,
Michael


On Jul 30, 2013, at 10:30 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk> =
wrote:

> I agree. In theory a new port generates new entropy that *could* =
result in a new ECMP path. I'm not sure this is a new major issue for =
cwv.
>=20
> Gorry
>=20
> On 30/07/2013 10:18, Andrew Mcgregor wrote:
>> You're correct in that changing the port number on one end will =
probably
>> cause a different ECMP path to become active.  However, in most cases
>> the operator is trying very hard to keep the ECMP paths balanced, so =
I
>> believe this is OK.  If not, they're doing complex traffic =
engineering,
>> and chances are the path can change mid-connection anyway as the TE
>> process reacts to load.  So I don't believe this to be a serious =
issue
>> in either case.
>>=20
>>=20
>> On 30 July 2013 01:12, Michael Welzl <michawe@ifi.uio.no
>> <mailto:michawe@ifi.uio.no>> wrote:
>>=20
>>    Hi,
>>=20
>>    In the discussion in TCPM just now, it seemed that people look at =
a
>>    connection that goes idle and later sends again as being roughly =
the
>>    same thing as a connection that stops and a new connection that
>>    starts. But: if there is a box in the path that does ECMP, this =
box
>>    may use the SYN and/or the use of a new port number (if only on =
the
>>    client side) as a basis to pick a path, and so a second connection
>>    starting after one that stopped may traverse a different path, =
right?
>>=20
>>    If I'm right then this is perhaps quite a different thing=85  (but
>>    note that I have no experience with boxes doing stuff like I'm
>>    writing here, this is just guesswork)
>>=20
>>    Cheers,
>>    Michael
>>=20
>>    _______________________________________________
>>    tcpm mailing list
>>    tcpm@ietf.org <mailto:tcpm@ietf.org>
>>    https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>>=20
>>=20
>>=20
>> --
>> Andrew McGregor | SRE |andrewmcgr@google.com
>> <mailto:andrewmcgr@google.com> | +61 4 8143 7128
>>=20
>>=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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Subject: [tcpm] (reducing) tcp bursts
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There are a lot of discussion on bursts across talks.
1. newcwv: idle-restart
2. tlp: how often is tail drops be caused by (higher) initial burst/send
3. burst (loss) after recovery due to snd.una + rwin jump.
4. I can throw in another one: video player application throttle
sender by not reading the socket or clamp the receive buffer. But this
causes TCP to burst when rwin opens up.

I think the working group should work on a general solution to reduce
burst in the window-based, ack-clocked, TCP. I have heard solutions
like
1. BSD/randy's max-burst solution
2. pace cwnd/rtt but in max-burst chunks
3. more ideas in http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt

We all know TCP is very smooth in bulk transfer. Unfortunately modern
Apps are chatty even on video.

Thoughts?

From michawe@ifi.uio.no  Tue Jul 30 04:27:53 2013
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] (reducing) tcp bursts
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Great idea!

Cheers,
Michael

On Jul 30, 2013, at 1:20 PM, Yuchung Cheng <ycheng@google.com> wrote:

> There are a lot of discussion on bursts across talks.
> 1. newcwv: idle-restart
> 2. tlp: how often is tail drops be caused by (higher) initial burst/send
> 3. burst (loss) after recovery due to snd.una + rwin jump.
> 4. I can throw in another one: video player application throttle
> sender by not reading the socket or clamp the receive buffer. But this
> causes TCP to burst when rwin opens up.
> 
> I think the working group should work on a general solution to reduce
> burst in the window-based, ack-clocked, TCP. I have heard solutions
> like
> 1. BSD/randy's max-burst solution
> 2. pace cwnd/rtt but in max-burst chunks
> 3. more ideas in http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
> 
> We all know TCP is very smooth in bulk transfer. Unfortunately modern
> Apps are chatty even on video.
> 
> Thoughts?
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From Alexander.Zimmermann@netapp.com  Tue Jul 30 04:32:50 2013
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Am 30.07.2013 um 07:20 schrieb Yuchung Cheng <ycheng@google.com>:

> There are a lot of discussion on bursts across talks.
> 1. newcwv: idle-restart
> 2. tlp: how often is tail drops be caused by (higher) initial burst/send
> 3. burst (loss) after recovery due to snd.una + rwin jump.
> 4. I can throw in another one: video player application throttle
> sender by not reading the socket or clamp the receive buffer. But this
> causes TCP to burst when rwin opens up.
> 
> I think the working group should work on a general solution to reduce
> burst in the window-based, ack-clocked, TCP. I have heard solutions
> like
> 1. BSD/randy's max-burst solution
> 2. pace cwnd/rtt but in max-burst chunks
> 3. more ideas in http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt

4. http://www.icir.org/floyd/tcp_small.html#pacing

> 
> We all know TCP is very smooth in bulk transfer. Unfortunately modern
> Apps are chatty even on video.
> 
> Thoughts?
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From mallman@icir.org  Tue Jul 30 08:36:09 2013
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Subject: Re: [tcpm] (reducing) tcp bursts
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> I think the working group should work on a general solution to reduce
> burst in the window-based, ack-clocked, TCP. 

Is this a demonstrated problem?  When Ethan studied this a while back
the high order bit was that bursting wasn't a huge issue, as I recall.
The paper is:

  Ethan Blanton, Mark Allman.  On the Impact of Bursting on TCP
  Performance.  Proceedings of the Workshop for Passive and Active
  Measurement, March 2005.
  http://www.icir.org/mallman/papers/burst-pam05.ps

My current feeling is that I don't necessarily have a problem with a
burst mitigater as long as it mostly stays out of the way and only tries
to go after really egregious bursts.  But, maybe that notion is just
based on out-of-date information.  It'd be cool to see empirical data on
the subject.

allman




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From: Yuchung Cheng <ycheng@google.com>
Date: Tue, 30 Jul 2013 18:02:09 +0200
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On Tue, Jul 30, 2013 at 5:35 PM, Mark Allman <mallman@icir.org> wrote:
>
>> I think the working group should work on a general solution to reduce
>> burst in the window-based, ack-clocked, TCP.
>
> Is this a demonstrated problem?  When Ethan studied this a while back
> the high order bit was that bursting wasn't a huge issue, as I recall.
> The paper is:
>
>   Ethan Blanton, Mark Allman.  On the Impact of Bursting on TCP
>   Performance.  Proceedings of the Workshop for Passive and Active
>   Measurement, March 2005.
>   http://www.icir.org/mallman/papers/burst-pam05.ps
>
> My current feeling is that I don't necessarily have a problem with a
> burst mitigater as long as it mostly stays out of the way and only tries
stays out of whose way?

> to go after really egregious bursts.  But, maybe that notion is just
> based on out-of-date information.  It'd be cool to see empirical data on
> the subject.
Absolutely we should proceed with empirical data and scientific experiments.

A indirect measure is the trickle work we did: when we clamp cwnd to
smooth youtube video burst, the loss rate is reduced nearly 40% on
flows that are not bottle-necked by network.
https://www.usenix.org/system/files/conference/atc12/atc12-final236.pdf

I will post more data to show the loss & burst size correlation at
Google servers.

Or data wizard Mark can start processing his latest traces too.

>
> allman
>
>
>

From mallman@icir.org  Tue Jul 30 09:09:19 2013
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> On Tue, Jul 30, 2013 at 5:35 PM, Mark Allman <mallman@icir.org> wrote:
> >
> >> I think the working group should work on a general solution to reduce
> >> burst in the window-based, ack-clocked, TCP.
> >
> > Is this a demonstrated problem?  When Ethan studied this a while back
> > the high order bit was that bursting wasn't a huge issue, as I recall.
> > The paper is:
> >
> >   Ethan Blanton, Mark Allman.  On the Impact of Bursting on TCP
> >   Performance.  Proceedings of the Workshop for Passive and Active
> >   Measurement, March 2005.
> >   http://www.icir.org/mallman/papers/burst-pam05.ps
> >
> > My current feeling is that I don't necessarily have a problem with a
> > burst mitigater as long as it mostly stays out of the way and only tries
>
> stays out of whose way?

The connection's.  The above paper basically shows that small-ish bursts
cause little-to-no extra loss.  But, once a burst size hits some point
the loss is big (and probably not just for the given connection, as it
has filled a buffer).  So, probably, small-to-modest bursts are not
causing a whole big issue.  But, big bursts do.  So, a burst limiter
could clamp those big bursts, but why bother with the little ones?

> I will post more data to show the loss & burst size correlation at
> Google servers.

Cool!

> Or data wizard Mark can start processing his latest traces too.

Heh.  Well, I am no wizard.  But, we are working on something that might
speak to this indirectly.  We'll see.

allman




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From touch@isi.edu  Tue Jul 30 09:51:38 2013
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On 7/30/2013 1:16 AM, Fernando Gont wrote:
> On 07/30/2013 04:55 AM, Joe Touch wrote:
>>>
>>> Part of TCP is support for simultaneous opens. If you don't support
>>> them, ether that's not TCP, or that's a buggy implementation.
>>
>> Again simultaneous < reflexive.
>
> There is no difference in the TCP code that is required for
> mirrored-endpoint-connections than the code that is required for
> simultaneous-{open, close} to work.

First, there is code in TCP to deal with similar special cases (e.g., 
not negotiating the window size when on the local subnet). So there's 
already similar special-case code deployed.

However, as I noted, the simultaneous open/close is only a subset of 
what needs to be vetted to ensure reflexive connections work.

> In any case, my take is that the high-order-bit of this I-D is not the
> mirrered-endpoint connections, but rather all the other stuff: the
> update that is needed for simultaneous opens, simultaneous closes, and
> simultaneous probes to work as expected -- and it looks like everyone
> (both of us, and other folks that have viced their opinion, plus the
> implementations out there) agrees that that must be addressed.

I agree with that.

> The mirrored-endpoints corner case is something on which I have a
> position, but, as noted, is a secondary issue (at the end of the day, I
> could myself live with not supporting them).

I agree this is a separate issue.

Joe

From touch@isi.edu  Tue Jul 30 10:21:16 2013
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Seems like time to revive draft-hughes-restart, and include some of the 
more recent suggestions.

Joe

On 7/30/2013 4:20 AM, Yuchung Cheng wrote:
> There are a lot of discussion on bursts across talks.
> 1. newcwv: idle-restart
> 2. tlp: how often is tail drops be caused by (higher) initial burst/send
> 3. burst (loss) after recovery due to snd.una + rwin jump.
> 4. I can throw in another one: video player application throttle
> sender by not reading the socket or clamp the receive buffer. But this
> causes TCP to burst when rwin opens up.
>
> I think the working group should work on a general solution to reduce
> burst in the window-based, ack-clocked, TCP. I have heard solutions
> like
> 1. BSD/randy's max-burst solution
> 2. pace cwnd/rtt but in max-burst chunks
> 3. more ideas in http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
>
> We all know TCP is very smooth in bulk transfer. Unfortunately modern
> Apps are chatty even on video.
>
> Thoughts?
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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On 7/30/2013 1:12 AM, Michael Welzl wrote:
> Hi,
>
> In the discussion in TCPM just now, it seemed that people look at a
> connection that goes idle and later sends again as being roughly the
> same thing as a connection that stops and a new connection that starts.

That was the assumption when the current code was designed, but we know 
better now. We can optimize across successive connections (TCB control 
block sharing), and we can do better during even brief periods of idle.

Note that the burst after idle is a specific case of a more general 
condition that doesn't require true idle to occur - it just requires a 
source that doesn't use the accumulated "permission to send" implied by 
an increasing congestion window.

Joe

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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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    Given this observation, and the apparent general agreement that 
reducing bursts while avoiding slow-start is more important, I'm going 
to go ahead and bump ssthresh higher than recommended by the draft (and 
RFC2861) in our TCP.  Increasing the RW can wait for the draft to be 
finalized, if we need it.

    Thanks for your attention.

Andrew

On 7/29/2013 11:38 PM, Karen.Nielsen@tieto.com wrote:
> HI,
>
> We also have observed issues with ssthresh setting after idle.
> This regards both TCP and SCTP usecase scenarios.
>
> Our thoughts have been to reset ssthresh to inifinity after idle.
> Setting to awnd is not a good solution in SCTP (or for mptcp) as the awnd
> may be small even if an idle period has been observed on a specific path on which traffic now resumes.
>
> Possible SCTP or multipath issues are not within cwv, but still I would like to point to this fact.
>
> BR, Karen
>
>
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of Anna
>> Brunstrom
>> Sent: Tuesday, July 30, 2013 12:35 AM
>> To: tcpm@ietf.org
>> Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
>>
>> Hi Andrew,
>>
>> I agree with you that the value of ssthresh is important and that this part may need
>> to be updated. If subsequent bursts are not long enough to take you out of an
>> unlucky state, a random loss in the the first burst may hamper performance for all
>> later bursts by invoking CA too early.
>>
>> We have seen the equivalent problem when you have a sequence of short flows and
>> the ssthresh value is cached. A random loss in one short flow can reduce
>> performance for all the short flows that follow. See our ICC paper from last year for
>> details http://dx.doi.org/10.1109/ICC.2012.6364516.
>>
>> BR,
>> Anna
>>
>>
>> On 2013-07-29 21:34, Andrew Knutsen wrote:
>>>      I'd like to ask one question again, before we implement something to
>>> avoid issues with a customer.
>>>
>>>      Most of the discussion here seems to be about avoiding bursts after
>>> idle while at the same time avoiding slow-start.
>>>
>>>      The problem we have is not slow-start after idle; its going into
>>> congestion-avoidance after idle, due to an obsolete ssthresh.
>>>
>>>      The current draft, like RFC2861, suggests setting ssthresh to 3/4
>>> the old cwnd value after idle.  We've tested this, and found that
>>> although it is an improvement over not doing anything (ie, rfc5681),
>>> cwnd does not recover with this factor when presented with a series of
>>> bursts if ssthresh is low.  We've also tested with setting it to  full
>>> cwnd, and to the advertised window (which is the receive window when
>>> idle).  Note that all of these are more conservative than closing and
>>> re-opening the connection, which sets ssthresh to infinity (if it isn't
>>> cached).
>>>
>>>      We've found that setting cwnd to IW after idle, and setting ssthresh
>>> to awnd, allows acceptable recovery from a congestion event while
>>> running burst traffic.  In other words, slow-start works well enough
>>> from IW (it's fast compared to CA).
>>>
>>>      We've tried to start discussion about this, but it looks like the
>>> consensus is a lack of concern.  While we would prefer to see a spec
>>> which addresses our concerns, we do have pressing need to fix this issue.
>>>
>>> Andrew
>>>
>>> _______________________________________________
>>> 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


From michael.scharf@alcatel-lucent.com  Wed Jul 31 00:01:47 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Andrew Knutsen <andrew.knutsen@bluecoat.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
Thread-Index: AQHOjJL6m/ShjGKxDEicWZgddkmTepl8HQ0AgACG/oCAATTAgIAAfn0A
Date: Wed, 31 Jul 2013 07:01:23 +0000
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References: <20130701194925.25967.70174.idtracker@ietfa.amsl.com> <f9c8b57462006f1dd53408c0c8cc5ff8.squirrel@www.erg.abdn.ac.uk> <CAK6E8=cifNh1Cf1Nd_N7k+z=K=tB4vBHMdXCem+F_++STg_Ysg@mail.gmail.com> <618CDE13-6926-4088-B318-1DA15E38985E@netapp.com> <CAK6E8=dyyde9epedO0wXD79b57tGtomRoAocAPja+JR3kU0D0g@mail.gmail.com> <BCF5A773-DB68-4AF6-887C-9F0B1ACD7884@netapp.com> <CAK6E8=eaMp5ZDXhtxK5B0niJonA4-+zcz9MJoh_0nVdToy_FQQ@mail.gmail.com> <20130729143100.GA17380@localhost.localdomain> <51F6934B.3040808@erg.abdn.ac.uk>	<51F6C3E0.7040205@bluecoat.com> <51F6EE16.2020504@kau.se> <CF340E42AED0874C81947E18863DE77B29DBD5C95E@EXMB03.eu.tieto.com> <51F86253.4040009@bluecoat.com>
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Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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Hi Andrew,

I wonder if you have experimented with setting ssthresh after an idle perio=
d to a value that is of the order of the pipeACK parameter (cf. draft-ietf-=
tcpm-newcwv-02 Section 4.2 - not flightsize)?

(Yes, I am aware of several other alternatives.)

Michael (without blue dot)


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
> Behalf Of Andrew Knutsen
> Sent: Wednesday, July 31, 2013 3:03 AM
> To: tcpm@ietf.org
> Subject: Re: [tcpm] New revision:=20
> draft-ietf-tcpm-newcwv-02.txt: ssthresh
>=20
>=20
>     Given this observation, and the apparent general=20
> agreement that reducing bursts while avoiding slow-start is=20
> more important, I'm going to go ahead and bump ssthresh=20
> higher than recommended by the draft (and
> RFC2861) in our TCP.  Increasing the RW can wait for the=20
> draft to be finalized, if we need it.
>=20
>     Thanks for your attention.
>=20
> Andrew
>=20
> On 7/29/2013 11:38 PM, Karen.Nielsen@tieto.com wrote:
> > HI,
> >
> > We also have observed issues with ssthresh setting after idle.
> > This regards both TCP and SCTP usecase scenarios.
> >
> > Our thoughts have been to reset ssthresh to inifinity after idle.
> > Setting to awnd is not a good solution in SCTP (or for=20
> mptcp) as the=20
> > awnd may be small even if an idle period has been observed=20
> on a specific path on which traffic now resumes.
> >
> > Possible SCTP or multipath issues are not within cwv, but=20
> still I would like to point to this fact.
> >
> > BR, Karen
> >
> >
> >> -----Original Message-----
> >> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org]=20
> On Behalf=20
> >> Of Anna Brunstrom
> >> Sent: Tuesday, July 30, 2013 12:35 AM
> >> To: tcpm@ietf.org
> >> Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt:=20
> >> ssthresh
> >>
> >> Hi Andrew,
> >>
> >> I agree with you that the value of ssthresh is important and that=20
> >> this part may need to be updated. If subsequent bursts are=20
> not long=20
> >> enough to take you out of an unlucky state, a random loss=20
> in the the=20
> >> first burst may hamper performance for all later bursts by=20
> invoking CA too early.
> >>
> >> We have seen the equivalent problem when you have a=20
> sequence of short=20
> >> flows and the ssthresh value is cached. A random loss in one short=20
> >> flow can reduce performance for all the short flows that=20
> follow. See=20
> >> our ICC paper from last year for details=20
> http://dx.doi.org/10.1109/ICC.2012.6364516.
> >>
> >> BR,
> >> Anna
> >>
> >>
> >> On 2013-07-29 21:34, Andrew Knutsen wrote:
> >>>      I'd like to ask one question again, before we implement=20
> >>> something to avoid issues with a customer.
> >>>
> >>>      Most of the discussion here seems to be about=20
> avoiding bursts=20
> >>> after idle while at the same time avoiding slow-start.
> >>>
> >>>      The problem we have is not slow-start after idle; its going=20
> >>> into congestion-avoidance after idle, due to an obsolete ssthresh.
> >>>
> >>>      The current draft, like RFC2861, suggests setting=20
> ssthresh to=20
> >>> 3/4 the old cwnd value after idle.  We've tested this, and found=20
> >>> that although it is an improvement over not doing anything (ie,=20
> >>> rfc5681), cwnd does not recover with this factor when=20
> presented with=20
> >>> a series of bursts if ssthresh is low.  We've also tested with=20
> >>> setting it to  full cwnd, and to the advertised window=20
> (which is the=20
> >>> receive window when idle).  Note that all of these are more=20
> >>> conservative than closing and re-opening the connection,=20
> which sets=20
> >>> ssthresh to infinity (if it isn't cached).
> >>>
> >>>      We've found that setting cwnd to IW after idle, and setting=20
> >>> ssthresh to awnd, allows acceptable recovery from a=20
> congestion event=20
> >>> while running burst traffic.  In other words, slow-start=20
> works well=20
> >>> enough from IW (it's fast compared to CA).
> >>>
> >>>      We've tried to start discussion about this, but it=20
> looks like=20
> >>> the consensus is a lack of concern.  While we would=20
> prefer to see a=20
> >>> spec which addresses our concerns, we do have pressing=20
> need to fix this issue.
> >>>
> >>> Andrew
> >>>
> >>> _______________________________________________
> >>> 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
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> =

From michael.scharf@alcatel-lucent.com  Wed Jul 31 06:58:10 2013
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
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Date: Wed, 31 Jul 2013 13:57:46 +0000
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Subject: [tcpm] WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
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Hi all,

In yesterday's meeting there was support for WG acceptance of draft-zimmerm=
ann-tcpm-tcp-rfc4614bis-02. The chairs would like to confirm this on the ma=
iling list.

Please comment on the list if you have any thoughts or concerns regarding a=
doption of this draft until August 7, 2013.

Thanks

Michael

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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] (reducing) tcp bursts
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On Tue, Jul 30, 2013 at 6:07 PM, Mark Allman <mallman@icir.org> wrote:
>
>> On Tue, Jul 30, 2013 at 5:35 PM, Mark Allman <mallman@icir.org> wrote:
>> >
>> >> I think the working group should work on a general solution to reduce
>> >> burst in the window-based, ack-clocked, TCP.
>> >
>> > Is this a demonstrated problem?  When Ethan studied this a while back
>> > the high order bit was that bursting wasn't a huge issue, as I recall.
>> > The paper is:
>> >
>> >   Ethan Blanton, Mark Allman.  On the Impact of Bursting on TCP
>> >   Performance.  Proceedings of the Workshop for Passive and Active
>> >   Measurement, March 2005.
>> >   http://www.icir.org/mallman/papers/burst-pam05.ps
>> >
>> > My current feeling is that I don't necessarily have a problem with a
>> > burst mitigater as long as it mostly stays out of the way and only tries
>>
>> stays out of whose way?
>
> The connection's.  The above paper basically shows that small-ish bursts
> cause little-to-no extra loss.  But, once a burst size hits some point
> the loss is big (and probably not just for the given connection, as it
> has filled a buffer).  So, probably, small-to-modest bursts are not
> causing a whole big issue.  But, big bursts do.  So, a burst limiter
> could clamp those big bursts, but why bother with the little ones?
ack. TCP shouldn't. The research questions are how big is big and the
trade-off of complexity vs performance of different solutions.

AFAIK Randell implements the max-burst in hughes-restart draft in BSD,
so the section on "BSD implementation" definitely needs an update if
we want to revive it.

>
>> I will post more data to show the loss & burst size correlation at
>> Google servers.
>
> Cool!
>
>> Or data wizard Mark can start processing his latest traces too.
>
> Heh.  Well, I am no wizard.  But, we are working on something that might
> speak to this indirectly.  We'll see.
>
> allman
>
>
>

From rs@netapp.com  Wed Jul 31 09:03:50 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
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Subject: Re: [tcpm] WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
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Positive.

Richard Scheffenegger

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Scharf, Michael (Michael)
> Sent: Mittwoch, 31. Juli 2013 15:58
> To: tcpm@ietf.org
> Subject: [tcpm] WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
>=20
> Hi all,
>=20
> In yesterday's meeting there was support for WG acceptance of draft-
> zimmermann-tcpm-tcp-rfc4614bis-02. The chairs would like to confirm this
> on the mailing list.
>=20
> Please comment on the list if you have any thoughts or concerns regarding
> adoption of this draft until August 7, 2013.
>=20
> Thanks
>=20
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From touch@isi.edu  Wed Jul 31 15:20:41 2013
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Date: Wed, 31 Jul 2013 15:20:10 -0700
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
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Support.

FWIW, the experimental options doc should have an RFC number any day 
now, and should be included, as should RFC 4727 which defined the TCP 
experimental option numbers.

Joe

On 7/31/2013 6:57 AM, Scharf, Michael (Michael) wrote:
> Hi all,
>
> In yesterday's meeting there was support for WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02. The chairs would like to confirm this on the mailing list.
>
> Please comment on the list if you have any thoughts or concerns regarding adoption of this draft until August 7, 2013.
>
> Thanks
>
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From nishida@sfc.wide.ad.jp  Wed Jul 31 19:42:35 2013
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] (reducing) tcp bursts
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Hi Yuchung,

I think this looks a very interesting and important point, but I have
several questions.
What is the difference between 1 and 4? newcwv draft seems to address
the issue for application throttle.
Also, what is the difference between 2 and 3? It seems to me that both
are the issue for burst after recovery.
In my understanding, the issue for burst after recovery can be
mitigated by PRR. I'm wondering why you don't mention it..

I'm also wondering how much we can have a general solution.
The characteristic of idle-restart/app throttle and burst after
recovery seems to be very different, we might need different
solutions.

Thanks,
--
Yoshifumi


On Tue, Jul 30, 2013 at 4:20 AM, Yuchung Cheng <ycheng@google.com> wrote:
> There are a lot of discussion on bursts across talks.
> 1. newcwv: idle-restart
> 2. tlp: how often is tail drops be caused by (higher) initial burst/send
> 3. burst (loss) after recovery due to snd.una + rwin jump.
> 4. I can throw in another one: video player application throttle
> sender by not reading the socket or clamp the receive buffer. But this
> causes TCP to burst when rwin opens up.
>
> I think the working group should work on a general solution to reduce
> burst in the window-based, ack-clocked, TCP. I have heard solutions
> like
> 1. BSD/randy's max-burst solution
> 2. pace cwnd/rtt but in max-burst chunks
> 3. more ideas in http://www.isi.edu/touch/pubs/draft-hughes-restart-00.txt
>
> We all know TCP is very smooth in bulk transfer. Unfortunately modern
> Apps are chatty even on video.
>
> Thoughts?
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From kevin.lahey@oracle.com  Wed Jul 31 23:08:18 2013
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Subject: Re: [tcpm] WG acceptance of draft-zimmermann-tcpm-tcp-rfc4614bis-02
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On Jul 31, 2013, at 3:57 PM, Scharf, Michael (Michael) wrote:

> In yesterday's meeting there was support for WG acceptance of =
draft-zimmermann-tcpm-tcp-rfc4614bis-02. The chairs would like to =
confirm this on the mailing list.


This seems like a fine idea.

It would be nice if, instead of saying, "Read RFC793, then read the =
additions from RFC1122, then update it with info from RFC2873, RFC3390, =
etc.," we could say, "Read RFC8888 for the on-wire protocol details, and =
RFC8889 for the congestion control details."  It's tough to keep all =
those details in your head while trying to write code, and it's =
frustrating to explain all this to new engineers trying to work on the =
TCP stack (RFC4614 certainly helps a great deal!).

On the other hand, watching Richard work so hard to get consensus on the =
(relatively) simple updates to RFC1323, I'm not sure who's ready to bell =
the RFC793bis cat.

Certainly not me,

Kevin
kevin.lahey@oracle.com


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From: Gorry <gorry@erg.abdn.ac.uk>
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Date: Wed, 31 Jul 2013 20:35:40 +0200
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt: ssthresh
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So,

Here is my own current "idea" ....

we think that cwv needs one mechanism to reduce cwnd and at the same time in=
crease ssthresh to 3/4 of the value prior to reduction. This is a separate p=
roblem.

There is another case when there is a long time since the last congestion ev=
ent (define long) and we should then allow ssthresh to be reset to its initi=
al ssthresh. As you suggest, important for many use cases.

Another case, an idle app - which could also do an initial ssthresh reset. T=
his is a specific case of above, but one that is much easier to detect (for a=
n app that goes idle).

There is possibly a Small cwnd case, if the cwnd if less than a few IW, then=
 maybe there is a perhaps a case for a reset of ssthresh after a shorter tim=
e.

That is a lot of different options... We have many inputs and we think we ca=
n propose text on the list for new-cwv, and then see if this is ok with peop=
le.

Gorry


On 31 Jul 2013, at 09:01 AM, "Scharf, Michael (Michael)" <michael.scharf@alc=
atel-lucent.com> wrote:

> Hi Andrew,
>=20
> I wonder if you have experimented with setting ssthresh after an idle peri=
od to a value that is of the order of the pipeACK parameter (cf. draft-ietf-=
tcpm-newcwv-02 Section 4.2 - not flightsize)?
>=20
> (Yes, I am aware of several other alternatives.)
>=20
> Michael (without blue dot)
>=20
>=20
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
>> Behalf Of Andrew Knutsen
>> Sent: Wednesday, July 31, 2013 3:03 AM
>> To: tcpm@ietf.org
>> Subject: Re: [tcpm] New revision:=20
>> draft-ietf-tcpm-newcwv-02.txt: ssthresh
>>=20
>>=20
>>    Given this observation, and the apparent general=20
>> agreement that reducing bursts while avoiding slow-start is=20
>> more important, I'm going to go ahead and bump ssthresh=20
>> higher than recommended by the draft (and
>> RFC2861) in our TCP.  Increasing the RW can wait for the=20
>> draft to be finalized, if we need it.
>>=20
>>    Thanks for your attention.
>>=20
>> Andrew
>>=20
>> On 7/29/2013 11:38 PM, Karen.Nielsen@tieto.com wrote:
>>> HI,
>>>=20
>>> We also have observed issues with ssthresh setting after idle.
>>> This regards both TCP and SCTP usecase scenarios.
>>>=20
>>> Our thoughts have been to reset ssthresh to inifinity after idle.
>>> Setting to awnd is not a good solution in SCTP (or for=20
>> mptcp) as the=20
>>> awnd may be small even if an idle period has been observed=20
>> on a specific path on which traffic now resumes.
>>>=20
>>> Possible SCTP or multipath issues are not within cwv, but=20
>> still I would like to point to this fact.
>>>=20
>>> BR, Karen
>>>=20
>>>=20
>>>> -----Original Message-----
>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org]=20
>> On Behalf=20
>>>> Of Anna Brunstrom
>>>> Sent: Tuesday, July 30, 2013 12:35 AM
>>>> To: tcpm@ietf.org
>>>> Subject: Re: [tcpm] New revision: draft-ietf-tcpm-newcwv-02.txt:=20
>>>> ssthresh
>>>>=20
>>>> Hi Andrew,
>>>>=20
>>>> I agree with you that the value of ssthresh is important and that=20
>>>> this part may need to be updated. If subsequent bursts are=20
>> not long=20
>>>> enough to take you out of an unlucky state, a random loss=20
>> in the the=20
>>>> first burst may hamper performance for all later bursts by=20
>> invoking CA too early.
>>>>=20
>>>> We have seen the equivalent problem when you have a=20
>> sequence of short=20
>>>> flows and the ssthresh value is cached. A random loss in one short=20
>>>> flow can reduce performance for all the short flows that=20
>> follow. See=20
>>>> our ICC paper from last year for details=20
>> http://dx.doi.org/10.1109/ICC.2012.6364516.
>>>>=20
>>>> BR,
>>>> Anna
>>>>=20
>>>>=20
>>>> On 2013-07-29 21:34, Andrew Knutsen wrote:
>>>>>     I'd like to ask one question again, before we implement=20
>>>>> something to avoid issues with a customer.
>>>>>=20
>>>>>     Most of the discussion here seems to be about=20
>> avoiding bursts=20
>>>>> after idle while at the same time avoiding slow-start.
>>>>>=20
>>>>>     The problem we have is not slow-start after idle; its going=20
>>>>> into congestion-avoidance after idle, due to an obsolete ssthresh.
>>>>>=20
>>>>>     The current draft, like RFC2861, suggests setting=20
>> ssthresh to=20
>>>>> 3/4 the old cwnd value after idle.  We've tested this, and found=20
>>>>> that although it is an improvement over not doing anything (ie,=20
>>>>> rfc5681), cwnd does not recover with this factor when=20
>> presented with=20
>>>>> a series of bursts if ssthresh is low.  We've also tested with=20
>>>>> setting it to  full cwnd, and to the advertised window=20
>> (which is the=20
>>>>> receive window when idle).  Note that all of these are more=20
>>>>> conservative than closing and re-opening the connection,=20
>> which sets=20
>>>>> ssthresh to infinity (if it isn't cached).
>>>>>=20
>>>>>     We've found that setting cwnd to IW after idle, and setting=20
>>>>> ssthresh to awnd, allows acceptable recovery from a=20
>> congestion event=20
>>>>> while running burst traffic.  In other words, slow-start=20
>> works well=20
>>>>> enough from IW (it's fast compared to CA).
>>>>>=20
>>>>>     We've tried to start discussion about this, but it=20
>> looks like=20
>>>>> the consensus is a lack of concern.  While we would=20
>> prefer to see a=20
>>>>> spec which addresses our concerns, we do have pressing=20
>> need to fix this issue.
>>>>>=20
>>>>> Andrew
>>>>>=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
>>> _______________________________________________
>>> 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
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
