
From nobody Fri May  2 04:18:50 2014
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On 26 Apr 2014, at 01:19, Joe Touch <touch@ISI.EDU> wrote:

> I'd like to ask the chairs if we can consider a call to make this a WG =
doc?
>=20
> (given the work of the group is supposed to happen on the list - and I =
don't attend meetings in person these days -this doesn't seem like it =
needs to wait for a meeting to proceed)

We should discuss the possible WG adoption on the list before the =
meeting, and some people have already voiced their support. But there is =
value in discussing this in the f2f meeting, too, before confirming WG =
adoption (hum). Perhaps Wes can present this, or as the last resort the =
presentation could be proxied by chairs. I don't see any reason to hurry =
this so that we couldn't wait until the Toronto meeting.

- Pasi


From nobody Fri May  2 07:45:51 2014
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Date: Fri, 02 May 2014 10:45:24 -0400
From: Wesley Eddy <wes@mti-systems.com>
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On 5/2/2014 7:18 AM, Pasi Sarolahti wrote:
> 
> On 26 Apr 2014, at 01:19, Joe Touch <touch@ISI.EDU> wrote:
> 
>> I'd like to ask the chairs if we can consider a call to make this a
>> WG doc?
>> 
>> (given the work of the group is supposed to happen on the list -
>> and I don't attend meetings in person these days -this doesn't seem
>> like it needs to wait for a meeting to proceed)
> 
> We should discuss the possible WG adoption on the list before the
> meeting, and some people have already voiced their support. But there
> is value in discussing this in the f2f meeting, too, before
> confirming WG adoption (hum). Perhaps Wes can present this, or as the
> last resort the presentation could be proxied by chairs. I don't see
> any reason to hurry this so that we couldn't wait until the Toronto
> meeting.
> 


That would be fine with me.  IMO, it would be nice to know what
the perceived mailing list consensus is, and use the meeting to
confirm that.  It's only a couple months away.

I think Joe would agree that once adopted, this should be able to
proceed relatively rapidly.

-- 
Wes Eddy
MTI Systems


From nobody Fri May  2 08:03:33 2014
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On 5/2/2014 7:45 AM, Wesley Eddy wrote:
> On 5/2/2014 7:18 AM, Pasi Sarolahti wrote:
>>
>> On 26 Apr 2014, at 01:19, Joe Touch <touch@ISI.EDU> wrote:
>>
>>> I'd like to ask the chairs if we can consider a call to make this a
>>> WG doc?
>>>
>>> (given the work of the group is supposed to happen on the list -
>>> and I don't attend meetings in person these days -this doesn't seem
>>> like it needs to wait for a meeting to proceed)
>>
>> We should discuss the possible WG adoption on the list before the
>> meeting, and some people have already voiced their support. But there
>> is value in discussing this in the f2f meeting, too, before
>> confirming WG adoption (hum). Perhaps Wes can present this, or as the
>> last resort the presentation could be proxied by chairs. I don't see
>> any reason to hurry this so that we couldn't wait until the Toronto
>> meeting.
>
> That would be fine with me.  IMO, it would be nice to know what
> the perceived mailing list consensus is, and use the meeting to
> confirm that.  It's only a couple months away.

I'm not in a rush, but as a point of order:

	- work happens on the mailing list, NOT at the meetings

	- decisions at the meeting are confirmed on the list,
	not the other way around

That's why I don't see why we need to wait for the meeting. Meetings are 
places where we typically hash out issues that arise on the list, e.g.:
	a. post to the list
	b. need for interactive clarification arises
	c. meeting supports interactive clarification
	d. decision happens back on the list

Somewhere along the line this all got reversed. I would like to 
encourage the chairs to use case as an opportunity to try the intended 
process.

Joe


From nobody Fri May  2 08:27:52 2014
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From: John Leslie <john@jlc.net>
To: Joe Touch <touch@isi.edu>
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Joe Touch <touch@isi.edu> wrote:
> 
> Meetings are places where we typically hash out issues that arise
> on the list, e.g.:
> 	a. post to the list
> 	b. need for interactive clarification arises
> 	c. meeting supports interactive clarification
> 	d. decision happens back on the list
> 
> Somewhere along the line this all got reversed. I would like to 
> encourage the chairs to use case as an opportunity to try the
> intended process.

   +1

--
John Leslie <john@jlc.net>


From nobody Fri May  2 13:27:24 2014
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--001a11c3d9d467452704f8709b68
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Hi Joe,

On Fri, May 2, 2014 at 8:02 AM, Joe Touch <touch@isi.edu> wrote:

>
>
> On 5/2/2014 7:45 AM, Wesley Eddy wrote:
>
>> On 5/2/2014 7:18 AM, Pasi Sarolahti wrote:
>>
>>>
>>> On 26 Apr 2014, at 01:19, Joe Touch <touch@ISI.EDU> wrote:
>>>
>>>  I'd like to ask the chairs if we can consider a call to make this a
>>>> WG doc?
>>>>
>>>> (given the work of the group is supposed to happen on the list -
>>>> and I don't attend meetings in person these days -this doesn't seem
>>>> like it needs to wait for a meeting to proceed)
>>>>
>>>
>>> We should discuss the possible WG adoption on the list before the
>>> meeting, and some people have already voiced their support. But there
>>> is value in discussing this in the f2f meeting, too, before
>>> confirming WG adoption (hum). Perhaps Wes can present this, or as the
>>> last resort the presentation could be proxied by chairs. I don't see
>>> any reason to hurry this so that we couldn't wait until the Toronto
>>> meeting.
>>>
>>
>> That would be fine with me.  IMO, it would be nice to know what
>> the perceived mailing list consensus is, and use the meeting to
>> confirm that.  It's only a couple months away.
>>
>
> I'm not in a rush, but as a point of order:
>
>         - work happens on the mailing list, NOT at the meetings
>
>         - decisions at the meeting are confirmed on the list,
>         not the other way around
>
> That's why I don't see why we need to wait for the meeting. Meetings are
> places where we typically hash out issues that arise on the list, e.g.:
>         a. post to the list
>         b. need for interactive clarification arises
>         c. meeting supports interactive clarification
>         d. decision happens back on the list
>

I personally would like to put a line between c. and d. like:
           c1. gauge consensus from both the list and the meetings
I think having two ways to gauge consensus is good thing.
--
Yoshi

--001a11c3d9d467452704f8709b68
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<div dir=3D"ltr">Hi Joe,<div class=3D"gmail_extra"><br><div class=3D"gmail_=
quote">On Fri, May 2, 2014 at 8:02 AM, Joe Touch <span dir=3D"ltr">&lt;<a h=
ref=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"><div class=3D""><br>
<br>
On 5/2/2014 7:45 AM, Wesley Eddy wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 5/2/2014 7:18 AM, Pasi Sarolahti wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
On 26 Apr 2014, at 01:19, Joe Touch &lt;<a href=3D"mailto:touch@ISI.EDU" ta=
rget=3D"_blank">touch@ISI.EDU</a>&gt; wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I&#39;d like to ask the chairs if we can consider a call to make this a<br>
WG doc?<br>
<br>
(given the work of the group is supposed to happen on the list -<br>
and I don&#39;t attend meetings in person these days -this doesn&#39;t seem=
<br>
like it needs to wait for a meeting to proceed)<br>
</blockquote>
<br>
We should discuss the possible WG adoption on the list before the<br>
meeting, and some people have already voiced their support. But there<br>
is value in discussing this in the f2f meeting, too, before<br>
confirming WG adoption (hum). Perhaps Wes can present this, or as the<br>
last resort the presentation could be proxied by chairs. I don&#39;t see<br=
>
any reason to hurry this so that we couldn&#39;t wait until the Toronto<br>
meeting.<br>
</blockquote>
<br>
That would be fine with me. =C2=A0IMO, it would be nice to know what<br>
the perceived mailing list consensus is, and use the meeting to<br>
confirm that. =C2=A0It&#39;s only a couple months away.<br>
</blockquote>
<br></div>
I&#39;m not in a rush, but as a point of order:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 - work happens on the mailing list, NOT at the =
meetings<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 - decisions at the meeting are confirmed on the=
 list,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 not the other way around<br>
<br>
That&#39;s why I don&#39;t see why we need to wait for the meeting. Meeting=
s are places where we typically hash out issues that arise on the list, e.g=
.:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 a. post to the list<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 b. need for interactive clarification arises<br=
>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 c. meeting supports interactive clarification<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 d. decision happens back on the list<br></block=
quote><div><br></div><div>I personally would like to put a line between c. =
and d. like:</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0c1. gauge c=
onsensus from both the list and the meetings</div>
<div>I think having two ways to gauge consensus is good thing.</div><div>--=
</div><div>Yoshi</div><div><br></div></div></div></div>

--001a11c3d9d467452704f8709b68--


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From: Joe Touch <touch@isi.edu>
Date: Fri, 2 May 2014 13:33:46 -0700
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> On May 2, 2014, at 1:27 PM, Yoshifumi Nishida <nishida@sfc.wide.ad.jp> wro=
te:
>=20
> Hi Joe,
>=20
>> On Fri, May 2, 2014 at 8:02 AM, Joe Touch <touch@isi.edu> wrote:
>>=20
>>=20
>>> On 5/2/2014 7:45 AM, Wesley Eddy wrote:
>>>> On 5/2/2014 7:18 AM, Pasi Sarolahti wrote:
>>>>=20
>>>> On 26 Apr 2014, at 01:19, Joe Touch <touch@ISI.EDU> wrote:
>>>>=20
>>>>> I'd like to ask the chairs if we can consider a call to make this a
>>>>> WG doc?
>>>>>=20
>>>>> (given the work of the group is supposed to happen on the list -
>>>>> and I don't attend meetings in person these days -this doesn't seem
>>>>> like it needs to wait for a meeting to proceed)
>>>>=20
>>>> We should discuss the possible WG adoption on the list before the
>>>> meeting, and some people have already voiced their support. But there
>>>> is value in discussing this in the f2f meeting, too, before
>>>> confirming WG adoption (hum). Perhaps Wes can present this, or as the
>>>> last resort the presentation could be proxied by chairs. I don't see
>>>> any reason to hurry this so that we couldn't wait until the Toronto
>>>> meeting.
>>>=20
>>> That would be fine with me.  IMO, it would be nice to know what
>>> the perceived mailing list consensus is, and use the meeting to
>>> confirm that.  It's only a couple months away.
>>=20
>> I'm not in a rush, but as a point of order:
>>=20
>>         - work happens on the mailing list, NOT at the meetings
>>=20
>>         - decisions at the meeting are confirmed on the list,
>>         not the other way around
>>=20
>> That's why I don't see why we need to wait for the meeting. Meetings are p=
laces where we typically hash out issues that arise on the list, e.g.:
>>         a. post to the list
>>         b. need for interactive clarification arises
>>         c. meeting supports interactive clarification
>>         d. decision happens back on the list
>=20
> I personally would like to put a line between c. and d. like:
>            c1. gauge consensus from both the list and the meetings
> I think having two ways to gauge consensus is good thing.

It's good but should not be required.  Meeting attendance isn't required and=
 officially all IETF business is supposed to happen on the list. So holding u=
p work to involve meetings should not ever happen IMO.=20

Joe


> --
> Yoshi
>=20

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	charset=utf-8
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<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div><br></div><div><br>On May 2, 2014, at 1:27 PM, Yoshifumi Nishida &lt;<a href="mailto:nishida@sfc.wide.ad.jp">nishida@sfc.wide.ad.jp</a>&gt; wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr">Hi Joe,<div class="gmail_extra"><br><div class="gmail_quote">On Fri, May 2, 2014 at 8:02 AM, Joe Touch <span dir="ltr">&lt;<a href="mailto:touch@isi.edu" target="_blank">touch@isi.edu</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=""><br>
<br>
On 5/2/2014 7:45 AM, Wesley Eddy wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On 5/2/2014 7:18 AM, Pasi Sarolahti wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
On 26 Apr 2014, at 01:19, Joe Touch &lt;<a href="mailto:touch@ISI.EDU" target="_blank">touch@ISI.EDU</a>&gt; wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I'd like to ask the chairs if we can consider a call to make this a<br>
WG doc?<br>
<br>
(given the work of the group is supposed to happen on the list -<br>
and I don't attend meetings in person these days -this doesn't seem<br>
like it needs to wait for a meeting to proceed)<br>
</blockquote>
<br>
We should discuss the possible WG adoption on the list before the<br>
meeting, and some people have already voiced their support. But there<br>
is value in discussing this in the f2f meeting, too, before<br>
confirming WG adoption (hum). Perhaps Wes can present this, or as the<br>
last resort the presentation could be proxied by chairs. I don't see<br>
any reason to hurry this so that we couldn't wait until the Toronto<br>
meeting.<br>
</blockquote>
<br>
That would be fine with me. &nbsp;IMO, it would be nice to know what<br>
the perceived mailing list consensus is, and use the meeting to<br>
confirm that. &nbsp;It's only a couple months away.<br>
</blockquote>
<br></div>
I'm not in a rush, but as a point of order:<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; - work happens on the mailing list, NOT at the meetings<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; - decisions at the meeting are confirmed on the list,<br>
&nbsp; &nbsp; &nbsp; &nbsp; not the other way around<br>
<br>
That's why I don't see why we need to wait for the meeting. Meetings are places where we typically hash out issues that arise on the list, e.g.:<br>
&nbsp; &nbsp; &nbsp; &nbsp; a. post to the list<br>
&nbsp; &nbsp; &nbsp; &nbsp; b. need for interactive clarification arises<br>
&nbsp; &nbsp; &nbsp; &nbsp; c. meeting supports interactive clarification<br>
&nbsp; &nbsp; &nbsp; &nbsp; d. decision happens back on the list<br></blockquote><div><br></div><div>I personally would like to put a line between c. and d. like:</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;c1. gauge consensus from both the list and the meetings</div>
<div>I think having two ways to gauge consensus is good thing.</div></div></div></div></div></blockquote><div><br></div><div>It's good but should not be required. &nbsp;Meeting attendance isn't required and officially all IETF business is supposed to happen on the list. So holding up work to involve meetings should not ever happen IMO.&nbsp;</div><div><br></div><div>Joe</div><div><br></div><br><blockquote type="cite"><div><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><div>--</div><div>Yoshi</div><div><br></div></div></div></div>
</div></blockquote></body></html>
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Joe Touch <touch@isi.edu>
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From: John Leslie <john@jlc.net>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
Message-ID: <20140503122950.GM44329@verdi>
References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com>
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Cc: Martin Stiemerling <mls.ietf@gmail.com>, tcpm@ietf.org
Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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Scharf, Michael (Michael) <michael.scharf@alcatel-lucent.com> wrote:
> 
> Meeting attendance is not required. Still, I don't understand why not
> running an adoption call about two weeks after posting -00 really
> holds up work.

   (I don't understand how you're measuring "two weeks".)

   To me, it's more a question of how putting this question off until
the rush of IETF week can possibly help.

   We need to determine whether this work can get traction it TCPM; and
we need to get IETF support for doing this work in TCPM -- whose charter,
lest we forget, is quite clear about us doing "small" TCP changes.
  
> TCPM is not the only working group that has a high bar for adopting
> new work,

   A 30-second hum after a ten-minute presentation isn't my idea of
what "a high bar" should mean.

> and I think being conservative regarding std track TCP changes has
> always been a community consensus (e.g., our previous charter even
> told us that every new TCPM milestone requires IESG approval -
> that was certainly too conservative).

   The IESG today doesn't require "milestone" changes to come before
a formal telechat; but I can imagine some concern about whether this
particular change (invalidating the "Data offset" field) fits within
our charter or requires a change to the charter wording. I'd much
prefer that such a question not be squeezed into the IETF week rush.

> Also, in TCPM review cycles are unfortunately a rather scarce resource,
> and the chairs have to consider this as well when adopting new work.

   Absolutely: new work should not be started without commitments for
enough review cycles!

> Regarding draft-touch-tcpm-tcp-edo, I am particularly interested in
> more feedback from implementers and users before starting the formal
> adoption call.

   Ah, but _what_ feedback?

   Commitment to review, certainly.

   Do you need data on how crowded the option space is?

   Do you need security considerations? Does that include what happens
as the IP-layer path changes to include more or less middleboxes?

   Do you need analysis of how this sort of change interacts with
MPTCP?

   Do you need analysis of which middleboxes can't be modified to
recognize this proposed option?

   Please be specific.

> We usually get very valuable feedback from implementers during
> meetings, and that would be one reason not to rush right now -

   I really don't follow why adoption as a Working Group Draft would
get in the way of considering such feedback.

> but it is the feedback I am interested in, not the meeting. If
> corresponding aspects are discussed on the list before the next
> meeting, I'll be more than happy as well.

   Fair enough!

> Thus, instead of further arguing about process aspects, I am more
> interested in thoughts on:
> 
> * From designers of TCP extensions requiring more option space: Is
>   draft-touch-tcpm-tcp-edo a good solution for your needs and thus
>   the right way to move forward?

   We're desperate! We'll get behind anything!

> * From implementers: Are there deployment issues beyond what is
>   described in the draft already?

   Good lord, yes! (That deserves a separate thread -- but again,
surely that can be discussed for a WG Draft as easily as for an
individual draft, can't it?)

> And, would this mechanism be implemented in the Internet? 

   I'd be astounded if it wouldn't be implemented in Linux!

   What beyond that do you need?

--
John Leslie <john@jlc.net>


From nobody Sat May  3 12:52:06 2014
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Date: Sat, 03 May 2014 21:51:51 +0200
From: Olivier Bonaventure <Olivier.Bonaventure@uclouvain.be>
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To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>,  Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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Michael,


> Thus, instead of further arguing about process aspects, I am more interested in thoughts on:
>
> * From designers of TCP extensions requiring more option space: Is draft-touch-tcpm-tcp-edo a good solution for your needs and thus the right way to move forward?
>
> * From implementers: Are there deployment issues beyond what is described in the draft already? And, would this mechanism be implemented in the Internet?

My main concern with this draft are the possible interactions with 
segment splitting/coalescing middleboxes.

The draft considers that these middleboxes are rare and should not 
constitute a problem. Unfortunately, I have to disagree with this 
statement. An important concern are the recent NICs that support TSO or 
GRO. These NICs are very widely used today and there are environments 
where it is very difficult for a TCP stack to determine whether it runs 
with such a NIC a not. A typical case is a virtual machine running on a 
host that exposes a driver that does not correspond to the physical NIC 
that is actually used.

The EDO option proposed in this draft does not unfortunately work 
correctly through a middlebox that splits segments. Concerning the 
following example.

A host that has negotiated EDO during the three way handshake sends a 
segment that contains :
- TCP header indicates 4 bytes of option, followed by EDO option 
indicating an option of 100 bytes, 100 bytes of option (e.g. a very long 
SACK), 1000 bytes of data

The middlebox does not understand the EDO option but needs to split the 
segment. Some middleboxes will copy all options in the two segments. 
This results in  :

segment 1 : TCP header indicates 4 bytes of option, followed by EDO 
option indicating an option of 100 bytes, 100 bytes of option (e.g. a 
very long SACK), 500 bytes of data

segment 2 : TCP header indicates 4 bytes of option, followed by EDO 
option indicating an option of 100 bytes, 500 bytes of data

The receiver will try to parse 100 bytes of option in the second segment...

In their IMC paper entitled "Is it Still Possible to Extend TCP?", 
Michio Honda et al. provide the following recommendation
http://conferences.sigcomm.org/imc/2011/docs/p181.pdf

"All the NICs we tested correctly copied the options to all the split 
segments. TSO is now sufficiently commonplace so that designers of 
extensions to TCP should assume it. The implication is that TCP options 
must be designed so that when they are duplicated on consecutive 
segments, this does not adversely affect correctness or performance."

EDO should be modified to take this recommendation into account to be 
deployable in today's Internet.

One possibility could be to include a checksum inside the EDO option.


   +--------+--------+--------+--------+----------+----------+
   |  Kind  | Length |  Header_length  |       Checksum      |
   +--------+--------+--------+--------+----------+----------+

The checksum should be computed over the entire extended option by the 
sender. When a receiver receives a segment with the EDO option, it MUST 
verify the EDO checksum. If the checksum is correct, then the extended 
option can be parsed. Otherwise, the entire segment should be silently 
dropped.

The checksum could solve the problem mentionned above with the NIC that 
segment data, but additional work is required to check whether such 
modification would solve all possible problems with middleboxes.

With Multipath TCP, we managed to extend TCP while preserving 
connectivity. Multipath TCP can cope with various types of middleboxes 
but not all possible behaviours. For some of these behaviours, Multipath 
TCP fallsback to regular TCP to ensure that the application continues to 
operate without any loss of connectivity. The EDO option should also 
have this goal in mind.

Best regards,


Olivier


-- 
INL, ICTEAM, UCLouvain, Belgium, http://inl.info.ucl.ac.be


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: John Leslie <john@jlc.net>
Thread-Topic: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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John,=0A=
=0A=
I am convinced that TCPM is the right place to discuss draft-touch-tcpm-tcp=
-edo. If we have strong support and commitments regarding the technical app=
roach, we'll sort out the question whether we need a charter update or not.=
=0A=
=0A=
Regarding implementer and user feedback, I'd really be interested in a stud=
y like the IMC 2011 paper (or a verification that those insights still appl=
y). I might be wrong, but my assumption is that due to deployments of MPTCP=
, fast open, etc. it may perhaps become simpler to deploy new TCP extension=
s in parts of the Internet. That would be important to know.=0A=
=0A=
But this kind of work may require help from the TCPM community, and it will=
 require time to complete. For instance, just as a thought from an individu=
al community member, the outcome of such a study could become an informatio=
nal companion document like RFC 2416, RFC 2884, etc.=0A=
=0A=
Michael=0A=
=0A=
=0A=
________________________________________=0A=
Von: John Leslie [john@jlc.net]=0A=
Gesendet: Samstag, 3. Mai 2014 14:29=0A=
An: Scharf, Michael (Michael)=0A=
Cc: Martin Stiemerling; tcpm@ietf.org=0A=
Betreff: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-0=
1.txt=0A=
=0A=
Scharf, Michael (Michael) <michael.scharf@alcatel-lucent.com> wrote:=0A=
>=0A=
> Meeting attendance is not required. Still, I don't understand why not=0A=
> running an adoption call about two weeks after posting -00 really=0A=
> holds up work.=0A=
=0A=
   (I don't understand how you're measuring "two weeks".)=0A=
=0A=
   To me, it's more a question of how putting this question off until=0A=
the rush of IETF week can possibly help.=0A=
=0A=
   We need to determine whether this work can get traction it TCPM; and=0A=
we need to get IETF support for doing this work in TCPM -- whose charter,=
=0A=
lest we forget, is quite clear about us doing "small" TCP changes.=0A=
=0A=
> TCPM is not the only working group that has a high bar for adopting=0A=
> new work,=0A=
=0A=
   A 30-second hum after a ten-minute presentation isn't my idea of=0A=
what "a high bar" should mean.=0A=
=0A=
> and I think being conservative regarding std track TCP changes has=0A=
> always been a community consensus (e.g., our previous charter even=0A=
> told us that every new TCPM milestone requires IESG approval -=0A=
> that was certainly too conservative).=0A=
=0A=
   The IESG today doesn't require "milestone" changes to come before=0A=
a formal telechat; but I can imagine some concern about whether this=0A=
particular change (invalidating the "Data offset" field) fits within=0A=
our charter or requires a change to the charter wording. I'd much=0A=
prefer that such a question not be squeezed into the IETF week rush.=0A=
=0A=
> Also, in TCPM review cycles are unfortunately a rather scarce resource,=
=0A=
> and the chairs have to consider this as well when adopting new work.=0A=
=0A=
   Absolutely: new work should not be started without commitments for=0A=
enough review cycles!=0A=
=0A=
> Regarding draft-touch-tcpm-tcp-edo, I am particularly interested in=0A=
> more feedback from implementers and users before starting the formal=0A=
> adoption call.=0A=
=0A=
   Ah, but _what_ feedback?=0A=
=0A=
   Commitment to review, certainly.=0A=
=0A=
   Do you need data on how crowded the option space is?=0A=
=0A=
   Do you need security considerations? Does that include what happens=0A=
as the IP-layer path changes to include more or less middleboxes?=0A=
=0A=
   Do you need analysis of how this sort of change interacts with=0A=
MPTCP?=0A=
=0A=
   Do you need analysis of which middleboxes can't be modified to=0A=
recognize this proposed option?=0A=
=0A=
   Please be specific.=0A=
=0A=
> We usually get very valuable feedback from implementers during=0A=
> meetings, and that would be one reason not to rush right now -=0A=
=0A=
   I really don't follow why adoption as a Working Group Draft would=0A=
get in the way of considering such feedback.=0A=
=0A=
> but it is the feedback I am interested in, not the meeting. If=0A=
> corresponding aspects are discussed on the list before the next=0A=
> meeting, I'll be more than happy as well.=0A=
=0A=
   Fair enough!=0A=
=0A=
> Thus, instead of further arguing about process aspects, I am more=0A=
> interested in thoughts on:=0A=
>=0A=
> * From designers of TCP extensions requiring more option space: Is=0A=
>   draft-touch-tcpm-tcp-edo a good solution for your needs and thus=0A=
>   the right way to move forward?=0A=
=0A=
   We're desperate! We'll get behind anything!=0A=
=0A=
> * From implementers: Are there deployment issues beyond what is=0A=
>   described in the draft already?=0A=
=0A=
   Good lord, yes! (That deserves a separate thread -- but again,=0A=
surely that can be discussed for a WG Draft as easily as for an=0A=
individual draft, can't it?)=0A=
=0A=
> And, would this mechanism be implemented in the Internet?=0A=
=0A=
   I'd be astounded if it wouldn't be implemented in Linux!=0A=
=0A=
   What beyond that do you need?=0A=
=0A=
--=0A=
John Leslie <john@jlc.net>=0A=


From nobody Mon May  5 00:49:29 2014
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To: Olivier.Bonaventure@uclouvain.be
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I'd like to add to what Olivier has said.

One question I had about this draft is whether the options included in =
the EDO are supposed to be carrying sequence numbers or not. Two =
options:
1. they carry sequence numbers: in this case we are changing the =
semantics of options in TCP, sending them reliably, orderly and subject =
to flow control. Which may be just fine for most types of options, but =
its not conforming to the spec and I suspect it opens a can of worms =
when we consider retransmissions of old options etc. For instance, if =
one would send MPTCP's Data ACKs in EDO reliably and subject to flow =
control we can end up with deadlocks.

2. they don't carry sequence numbers. In this case, a box that does not =
know EDO and looks at packets it will see more data sent sent the =
combination of sequence number / length allows. Many boxes will just =
drop the extra bytes, others may reset the connection. Based on our IMC =
11 experience, I expect that at least 1/3 of access networks (e.g. DSL / =
Cellular / Hotspot  / etc) will break EDO. So this option is a no go, in =
my opinion.

Another thing: the draft says that NAT will work with EDO. However, most =
NATs have a passive FTP mode - which means they will look through the =
TCP payload for the host's IP address and rewrite with their own.=20
With EDO, the NAT could be messing up options placed in EDO space in the =
payload. To avoid this you need a checksum, as Olivier says.=20

I suspect that the solution has to be something in the direction of (1), =
where the EDO space is quoted explicitly  (have an SACK like options =
that says bytes XX-YY from the payload carry in fact  options). On top =
of that, we need a review of what options can be sent over EDO and what =
options cannot be sent over it. It's not going to be easy, in my =
opinion.

Cheers.
Costin




=20
On 3 May 2014, at 22:51, Olivier Bonaventure wrote:

> Michael,
>=20
>=20
>> Thus, instead of further arguing about process aspects, I am more =
interested in thoughts on:
>>=20
>> * =46rom designers of TCP extensions requiring more option space: Is =
draft-touch-tcpm-tcp-edo a good solution for your needs and thus the =
right way to move forward?
>>=20
>> * =46rom implementers: Are there deployment issues beyond what is =
described in the draft already? And, would this mechanism be implemented =
in the Internet?
>=20
> My main concern with this draft are the possible interactions with =
segment splitting/coalescing middleboxes.
>=20
> The draft considers that these middleboxes are rare and should not =
constitute a problem. Unfortunately, I have to disagree with this =
statement. An important concern are the recent NICs that support TSO or =
GRO. These NICs are very widely used today and there are environments =
where it is very difficult for a TCP stack to determine whether it runs =
with such a NIC a not. A typical case is a virtual machine running on a =
host that exposes a driver that does not correspond to the physical NIC =
that is actually used.
>=20
> The EDO option proposed in this draft does not unfortunately work =
correctly through a middlebox that splits segments. Concerning the =
following example.
>=20
> A host that has negotiated EDO during the three way handshake sends a =
segment that contains :
> - TCP header indicates 4 bytes of option, followed by EDO option =
indicating an option of 100 bytes, 100 bytes of option (e.g. a very long =
SACK), 1000 bytes of data
>=20
> The middlebox does not understand the EDO option but needs to split =
the segment. Some middleboxes will copy all options in the two segments. =
This results in  :
>=20
> segment 1 : TCP header indicates 4 bytes of option, followed by EDO =
option indicating an option of 100 bytes, 100 bytes of option (e.g. a =
very long SACK), 500 bytes of data
>=20
> segment 2 : TCP header indicates 4 bytes of option, followed by EDO =
option indicating an option of 100 bytes, 500 bytes of data
>=20
> The receiver will try to parse 100 bytes of option in the second =
segment...
>=20
> In their IMC paper entitled "Is it Still Possible to Extend TCP?", =
Michio Honda et al. provide the following recommendation
> http://conferences.sigcomm.org/imc/2011/docs/p181.pdf
>=20
> "All the NICs we tested correctly copied the options to all the split =
segments. TSO is now sufficiently commonplace so that designers of =
extensions to TCP should assume it. The implication is that TCP options =
must be designed so that when they are duplicated on consecutive =
segments, this does not adversely affect correctness or performance."
>=20
> EDO should be modified to take this recommendation into account to be =
deployable in today's Internet.
>=20
> One possibility could be to include a checksum inside the EDO option.
>=20
>=20
>  +--------+--------+--------+--------+----------+----------+
>  |  Kind  | Length |  Header_length  |       Checksum      |
>  +--------+--------+--------+--------+----------+----------+
>=20
> The checksum should be computed over the entire extended option by the =
sender. When a receiver receives a segment with the EDO option, it MUST =
verify the EDO checksum. If the checksum is correct, then the extended =
option can be parsed. Otherwise, the entire segment should be silently =
dropped.
>=20
> The checksum could solve the problem mentionned above with the NIC =
that segment data, but additional work is required to check whether such =
modification would solve all possible problems with middleboxes.
>=20
> With Multipath TCP, we managed to extend TCP while preserving =
connectivity. Multipath TCP can cope with various types of middleboxes =
but not all possible behaviours. For some of these behaviours, Multipath =
TCP fallsback to regular TCP to ensure that the application continues to =
operate without any loss of connectivity. The EDO option should also =
have this goal in mind.
>=20
> Best regards,
>=20
>=20
> Olivier
>=20
>=20
> --=20
> INL, ICTEAM, UCLouvain, Belgium, http://inl.info.ucl.ac.be
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon May  5 07:16:05 2014
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Scharf, Michael (Michael) <michael.scharf@alcatel-lucent.com> wrote:
> 
> I am convinced that TCPM is the right place to discuss
> draft-touch-tcpm-tcp-edo. If we have strong support and commitments
> regarding the technical approach, we'll sort out the question whether
> we need a charter update or not.

   :^)

> Regarding implementer and user feedback, I'd really be interested
> in a study like the IMC 2011 paper

   I assume you mean "Is it still possible to extend TCP?"

http://conferences.sigcomm.org/imc/2011/docs/p181.pdf

> (or a verification that those insights still apply).

   This paper reported research on 135 paths using 3 port numbers:
HTTP, HTTPS, and "middle-of-nowhere", and showed at best 96% passage
of unknown options. It discussed the operation of middleboxes (including
HTTP proxies), and concluded that MPTCP could detect damage and fallback,
TcpCrypt is fragile (and can't fallback after being enabled), and TCP
Extended Options (draft-eddy-tcp-loo) was unsafe.

   I take it on faith that those insights are still valid.

> I might be wrong, but my assumption is that due to deployments of
> MPTCP, fast open, etc. it may perhaps become simpler to deploy new
> TCP extensions in parts of the Internet. That would be important
> to know.

   I think it is a given that deployment will depend upon successful
negotiation end-to-end.

   Given the (still-growing!) extent of middlebox damage, I suggest
it will also be necessary to detect middlebox damage throughout the
life of the connection.

   Nonetheless, I see large portions of the Internet where expanding
the TCP Options field could function today. I predict that offending
middleboxes will be replaced over time (perhaps five years). So long
as on offending middlebox can be identified and folks understand the
damage it causes, replacing them need not be expensive.

> But this kind of work may require help from the TCPM community,
> and it will require time to complete. For instance, just as a thought
> from an individual community member, the outcome of such a study
> could become an informational companion document like RFC 2416,
> RFC 2884, etc.

   Repeating the IMC-2011/p181 study sounds more like an IRTF task;
but I certainly agree it would be worthwhile.

====

   I do expect a standardized EDO to be more complex than tcp-loo;
but there are a number of approaches which are (IMHO) sufficiently
safe. I believe touch-tcpm-tcp-edo to be a better starting point
than eddy-tcp-loo was; but it will need significant work before
submitting to the IESG.

--
John Leslie <john@jlc.net>


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Costin Raiciu <c.raiciu@cs.ucl.ac.uk> wrote:
> 
> One question I had about this draft is whether the options included
> in the EDO are supposed to be carrying sequence numbers...

   That's probably a bad idea: while we can depend upon the endpoints
to negotiate a common understanding or visibly fail, the same is not
true of middleboxes. A middlebox _can_ cause negotiation to fail, but
it has no way to signal exactly how it _does_ interpret an option.

> 1. they carry sequence numbers: in this case we are changing the
>    semantics of options in TCP... I suspect it opens a can of worms...

   Agreed.

> 2. they don't carry sequence numbers. In this case, a box that does not
>    know EDO and looks at packets it will see more data sent sent...
>    Many boxes will just drop the extra bytes...

   Which would be a disaster.

>    So this option is a no go, in my opinion.

   I don't agree that adding sequence numbers can be made safe; so I
don't agree with Costin's division into two parts.

> Another thing: the draft says that NAT will work with EDO.

   I'd prefer not to argue that yet...

> However, most NATs have a passive FTP mode - which means they will
> look through the TCP payload for the host's IP address and rewrite
> with their own. 

   (That's one of the reasons I prefer to have "Data Offset" be a clearly
invalid value whenever it is overridden by EDO: any middlebox will then
have fair warning that it is clueless.)

> With EDO, the NAT could be messing up options placed in EDO space in
> the payload. To avoid this you need a checksum, as Olivier says. 

   Checksum isn't the _only_ way; but some sort of checksum is probably
a good idea.

> I suspect that the solution has to be something in the direction of
> (1), where the EDO space is quoted explicitly (have an SACK like
> options that says bytes XX-YY from the payload carry in fact  options).

   I don't quite follow; but that may be a workable approach... (I think
of somehow overlaying the "Urgent pointer".)

> On top of that, we need a review of what options can be sent over EDO
> and what options cannot be sent over it.

   I don't think _anything_ should be "cannot" there; but we'll have
some work in how to ration what options go in the SYN (which, alas,
we're going to have to give up on expanding).

> It's not going to be easy, in my opinion.

   It never was easy. :^(

--
John Leslie <john@jlc.net>


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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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On 5/3/2014 12:51 PM, Olivier Bonaventure wrote:
> Michael,
>
>
>> Thus, instead of further arguing about process aspects, I am more
>> interested in thoughts on:
>>
>> * From designers of TCP extensions requiring more option space: Is
>> draft-touch-tcpm-tcp-edo a good solution for your needs and thus the
>> right way to move forward?
>>
>> * From implementers: Are there deployment issues beyond what is
>> described in the draft already? And, would this mechanism be
>> implemented in the Internet?
>
> My main concern with this draft are the possible interactions with
> segment splitting/coalescing middleboxes.

These already affect a few things - splitting replicates timestamps and 
repeats ACKs, and coalescing might need to drop SACK blocks and drops 
some of the timestamp info.

A key question for the group is the extent to which we want to support 
these or even detect them in our core mechanisms, or whether we should 
suggest that they be detected (and replaced) by existing mechanisms 
(e.g., TCP-AO or IPsec).

> The draft considers that these middleboxes are rare and should not
> constitute a problem. Unfortunately, I have to disagree with this
> statement. An important concern are the recent NICs that support TSO or
> GRO.

These are not merely middleboxes that resegment. They ought to be 
integrated with the endsystem sufficient to support TCP.

So we should consider the impact of network rewriting boxes vs. TSO/GRO 
NICs as separate issues, IMO.

 > These NICs are very widely used today and there are environments
> where it is very difficult for a TCP stack to determine whether it runs
> with such a NIC a not. A typical case is a virtual machine running on a
> host that exposes a driver that does not correspond to the physical NIC
> that is actually used.
>
> The EDO option proposed in this draft does not unfortunately work
> correctly through a middlebox that splits segments.

Again, we're back to the middlebox example:

> Concerning the
> following example.
>
> A host that has negotiated EDO during the three way handshake sends a
> segment that contains :
> - TCP header indicates 4 bytes of option, followed by EDO option
> indicating an option of 100 bytes, 100 bytes of option (e.g. a very long
> SACK), 1000 bytes of data
>
> The middlebox does not understand the EDO option but needs to split the
> segment. Some middleboxes will copy all options in the two segments.

A middlebox that examines the contents of TCP traffic - or worse, 
modifies it - needs to follow all of RFC1122 host requirements. It's 
interpreting content - if it sees a SYN with an unknown option, it ought 
to have removed it or dropped the SYN.

If it doesn't, you have a device that is broken and needs to be replaced.

We should design TCP to be liberal in what it accepts, but no Internet 
protocol should be designed to overcome deliberate implementation errors.

> This results in  :
>
> segment 1 : TCP header indicates 4 bytes of option, followed by EDO
> option indicating an option of 100 bytes, 100 bytes of option (e.g. a
> very long SACK), 500 bytes of data
>
> segment 2 : TCP header indicates 4 bytes of option, followed by EDO
> option indicating an option of 100 bytes, 500 bytes of data
>
> The receiver will try to parse 100 bytes of option in the second segment...

Agreed.

However, now you're back to the NIC issue (which ought not be related):

> In their IMC paper entitled "Is it Still Possible to Extend TCP?",
> Michio Honda et al. provide the following recommendation
> http://conferences.sigcomm.org/imc/2011/docs/p181.pdf
>
> "All the NICs we tested correctly copied the options to all the split
> segments. TSO is now sufficiently commonplace so that designers of
> extensions to TCP should assume it. The implication is that TCP options
> must be designed so that when they are duplicated on consecutive
> segments, this does not adversely affect correctness or performance."

This tells me your TCP interface was implemented incorrectly.

The correct interface passes options (including the extended option *and 
its associated contents*) ***SEPARATELY*** from the user data.

If that is done all the way down to the TSO, the TSO will correctly copy 
the option into individual segments correctly.

If, however, you implement this badly - and pass TCP segments as a whole 
to the offload engine, then the offload engine *has no business* 
splitting or coalescing segments.

> EDO should be modified to take this recommendation into account to be
> deployable in today's Internet.
>
> One possibility could be to include a checksum inside the EDO option.
>
>
>    +--------+--------+--------+--------+----------+----------+
>    |  Kind  | Length |  Header_length  |       Checksum      |
>    +--------+--------+--------+--------+----------+----------+

That logic would apply too EVERY option, and now we have a TCP that 
wastes space primarily to locate and debug implementation errors.

I don't agree that TCP should be designed to overcome deliberate 
implementation errors.

Joe


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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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On 5/5/2014 10:41 AM, John Leslie wrote:
>> However, most NATs have a passive FTP mode - which means they will
>> > look through the TCP payload for the host's IP address and rewrite
>> > with their own. 
>
>    (That's one of the reasons I prefer to have "Data Offset" be a clearly
> invalid value whenever it is overridden by EDO: any middlebox will then
> have fair warning that it is clueless.)
> 


I think I agree.  IMHO, the tradeoff comes down to whether it's more
desirable for the middlebox to end up doing something violent like:

- dropping the segment
- resetting the TCP connection
- generating an ICMP
- crashing

versus just doing something to the segments which the receiver can
detect and possibly work-around with greater size and complexity in
the mechanism itself.

It ends up being a matter of transition and whether we really
expect offending nodes to be found and fixed over time, and can
stand some deployment hassles and complaints in the meantime.
If not, then the design needs to be a bit more complicated forever
in order to accommodate today's legacy.

If there are significant advantages to being able to use long
options that relate to real business problems, then I think the
legacy won't be as much of a problem, but don't really know.

-- 
Wes Eddy
MTI Systems


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On 5/5/2014 12:49 AM, Costin Raiciu wrote:
> I'd like to add to what Olivier has said.
>
> One question I had about this draft is whether the options included
> in the EDO are supposed to be carrying sequence numbers or not. Two
> options:
>
> 1. they carry sequence numbers: in this case we are changing the
> semantics of options in TCP, sending them reliably, orderly and
> subject to flow control.

We're not doing that any more than the options in existing segments 
within the existing Data Offset are so covered.

The option isn't buried in the data stream - this was considered for 
MPTCP, but rejected, for some of the reasons you mention.

So EDO option space ought to be used the same way - with the same 
limitations (or not) of existing option space.

...
> Another thing: the draft says that NAT will work with EDO. However,
> most NATs have a passive FTP mode - which means they will look
> through the TCP payload for the host's IP address and rewrite with
> their own. With EDO, the NAT could be messing up options placed in
> EDO space in the payload. To avoid this you need a checksum, as
> Olivier says.

See my response to Oliver's issue on this. Such a NAT ought to have 
either dropped the EDO in the initial SYN or dropped the SYN completely.

Joe


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On 5/5/2014 7:41 AM, John Leslie wrote:
>> However, most NATs have a passive FTP mode - which means they will
>> >look through the TCP payload for the host's IP address and rewrite
>> >with their own.
 >
>     (That's one of the reasons I prefer to have "Data Offset" be a clearly
> invalid value whenever it is overridden by EDO: any middlebox will then
> have fair warning that it is clueless.)

It had that warning when it saw the EDO option and decided that it could 
ignore it quietly.

Joe


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PS - one other alternative would be to allow the header checksum to be 
optional - users can turn it on or off as needed.

It could be part of EDO (determined by the length of the EDO option 
itself) or as a separate option (which would then be useful for all 
other options as a "bad NAT traversal detection option".)

Joe


On 5/5/2014 8:50 AM, Wesley Eddy wrote:
> On 5/5/2014 10:41 AM, John Leslie wrote:
>>> However, most NATs have a passive FTP mode - which means they will
>>>> look through the TCP payload for the host's IP address and rewrite
>>>> with their own.
>>
>>     (That's one of the reasons I prefer to have "Data Offset" be a clearly
>> invalid value whenever it is overridden by EDO: any middlebox will then
>> have fair warning that it is clueless.)
>>
>
>
> I think I agree.  IMHO, the tradeoff comes down to whether it's more
> desirable for the middlebox to end up doing something violent like:
>
> - dropping the segment
> - resetting the TCP connection
> - generating an ICMP
> - crashing
>
> versus just doing something to the segments which the receiver can
> detect and possibly work-around with greater size and complexity in
> the mechanism itself.
>
> It ends up being a matter of transition and whether we really
> expect offending nodes to be found and fixed over time, and can
> stand some deployment hassles and complaints in the meantime.
> If not, then the design needs to be a bit more complicated forever
> in order to accommodate today's legacy.
>
> If there are significant advantages to being able to use long
> options that relate to real business problems, then I think the
> legacy won't be as much of a problem, but don't really know.
>


From nobody Mon May  5 09:04:24 2014
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> A middlebox that examines the contents of TCP traffic - or worse, =
modifies it - needs to follow all of RFC1122 host requirements. It's =
interpreting content - if it sees a SYN with an unknown option, it ought =
to have removed it or dropped the SYN.
>=20
> If it doesn't, you have a device that is broken and needs to be =
replaced.
>=20
> We should design TCP to be liberal in what it accepts, but no Internet =
protocol should be designed to overcome deliberate implementation =
errors.

As much as I agree with this purist view of the Internet, I also believe =
it is unworkable if we want stuff deployed.=20

Say we actually stuck to the plan and designed EDO for a world with =
reasonable middleboxes. Then we implement it in Linux, then we turn it =
on by default. Then, all of a sudden the Internet stops working, and you =
have disgruntled users cursing Linux right and left....

Trouble is, for middleboxes to change you need to have a working =
protocol (that is maybe suboptimal) and a strong need for the change, =
and push from OS/app vendors etc.=20

Cheers.
Costin




>=20
>> This results in  :
>>=20
>> segment 1 : TCP header indicates 4 bytes of option, followed by EDO
>> option indicating an option of 100 bytes, 100 bytes of option (e.g. a
>> very long SACK), 500 bytes of data
>>=20
>> segment 2 : TCP header indicates 4 bytes of option, followed by EDO
>> option indicating an option of 100 bytes, 500 bytes of data
>>=20
>> The receiver will try to parse 100 bytes of option in the second =
segment...
>=20
> Agreed.
>=20
> However, now you're back to the NIC issue (which ought not be =
related):
>=20
>> In their IMC paper entitled "Is it Still Possible to Extend TCP?",
>> Michio Honda et al. provide the following recommendation
>> http://conferences.sigcomm.org/imc/2011/docs/p181.pdf
>>=20
>> "All the NICs we tested correctly copied the options to all the split
>> segments. TSO is now sufficiently commonplace so that designers of
>> extensions to TCP should assume it. The implication is that TCP =
options
>> must be designed so that when they are duplicated on consecutive
>> segments, this does not adversely affect correctness or performance."
>=20
> This tells me your TCP interface was implemented incorrectly.
>=20
> The correct interface passes options (including the extended option =
*and its associated contents*) ***SEPARATELY*** from the user data.
>=20
> If that is done all the way down to the TSO, the TSO will correctly =
copy the option into individual segments correctly.
>=20
> If, however, you implement this badly - and pass TCP segments as a =
whole to the offload engine, then the offload engine *has no business* =
splitting or coalescing segments.
>=20
>> EDO should be modified to take this recommendation into account to be
>> deployable in today's Internet.
>>=20
>> One possibility could be to include a checksum inside the EDO option.
>>=20
>>=20
>>   +--------+--------+--------+--------+----------+----------+
>>   |  Kind  | Length |  Header_length  |       Checksum      |
>>   +--------+--------+--------+--------+----------+----------+
>=20
> That logic would apply too EVERY option, and now we have a TCP that =
wastes space primarily to locate and debug implementation errors.
>=20
> I don't agree that TCP should be designed to overcome deliberate =
implementation errors.
>=20
> Joe
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon May  5 09:15:27 2014
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On 5/5/2014 9:04 AM, Costin Raiciu wrote:
>> A middlebox that examines the contents of TCP traffic - or worse, modifies it - needs to follow all of RFC1122 host requirements. It's interpreting content - if it sees a SYN with an unknown option, it ought to have removed it or dropped the SYN.
>>
>> If it doesn't, you have a device that is broken and needs to be
>> replaced.
>>
>> We should design TCP to be liberal in what it accepts, but no
>> Internet protocol should be designed to overcome deliberate
>> implementation errors.
>>
> As much as I agree with this purist view of the Internet, I also
> believe it is unworkable if we want stuff deployed.
>
> Say we actually stuck to the plan and designed EDO for a world with
> reasonable middleboxes. Then we implement it in Linux, then we turn it
> on by default. Then, all of a sudden the Internet stops working, and you
> have disgruntled users cursing Linux right and left....

Linux already runs lots of things we didn't approve, and never seems to 
take any blame. I don't see avoiding Linux user ire as a viable approach.

> Trouble is, for middleboxes to change you need to have a working
> protocol (that is maybe suboptimal) and a strong need for the change,
> and push from OS/app vendors etc.

For middleboxes to change, they have to start breaking things. As long 
as we adapt to the errors they make, we're helping create their market.

I agree with your overall point that we do need to consider how/whether 
this can avoid being held hostage by middleboxes, and a *temporary* 
header checksum might be a way.

But ultimately, that still means the new option will not work through 
middleboxes, and lots of people will not consider that the fault of the 
middlebox.

Joe


From nobody Mon May  5 09:30:36 2014
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Wesley Eddy <wes@mti-systems.com> wrote:
> On 5/5/2014 10:41 AM, John Leslie wrote:
>>
>> (That's one of the reasons I prefer to have "Data Offset" be a clearly
>> invalid value whenever it is overridden by EDO: any middlebox will then
>> have fair warning that it is clueless.)
> 
> I think I agree.  IMHO, the tradeoff comes down to whether it's more
> desirable for the middlebox to end up doing something violent like:
> 
> - dropping the segment

   This should always be "legal" for a middlebox. (I certainly prefer
it to damaging some unrelated part of the packet.) It's easy enough
to detect.

> - resetting the TCP connection

   This isn't quite "legal", but is certainly easy to detect.

> - generating an ICMP

   Would anyone actually do this? (Hopefully, it would be in combination
with dropping the packet.)

> - crashing

   That's my most-preferred one. There's no excuse to blame anyone but
the middlebox vendor. ;^)

> versus just doing something to the segments which the receiver can
> detect and possibly work-around with greater size and complexity in
> the mechanism itself.

   I'd _much_ rather not start down the road of guessing what
unrelated part of the packet has been damaged!

> It ends up being a matter of transition and whether we really
> expect offending nodes to be found and fixed over time, and can
> stand some deployment hassles and complaints in the meantime.

   IMHO, that's always the case -- it just is, sometimes the transition
takes decades. :^(

> If not, then the design needs to be a bit more complicated forever
> in order to accommodate today's legacy.

   Now we're arguing over price: how much is "a bit"?

> If there are significant advantages to being able to use long
> options that relate to real business problems, then I think the
> legacy won't be as much of a problem, but don't really know.

   We've already over-packed the Options field: there are already
_multiple_ business cases for supporting an expanded field.

   How much of a problem the legacy proves to be depends a lot on
how we design EDO...

--
John Leslie <john@jlc.net>


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Joe Touch <touch@isi.edu> wrote:
> On 5/5/2014 7:41 AM, John Leslie wrote:
>>
>> (That's one of the reasons I prefer to have "Data Offset" be a clearly
>> invalid value whenever it is overridden by EDO: any middlebox will then
>> have fair warning that it is clueless.)
> 
> It had that warning when it saw the EDO option and decided that it could 
> ignore it quietly.

   This points out another failure of the TCP-Options design: for such
a basic building block, Options should have carried an indicator of
what to do with an unrecognized option -- e.g. ignore, drop the packet,
etc.

   Joe is right that "conservative in what you send" sort-of implies
that you don't pass on an option when you have no idea what it might
actually mean...

   (But that horse left the barn many years ago.)

--
John Leslie <john@jlc.net>


From nobody Mon May  5 09:52:30 2014
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On 5/5/2014 9:36 AM, John Leslie wrote:
> Joe Touch <touch@isi.edu> wrote:
>> On 5/5/2014 7:41 AM, John Leslie wrote:
>>>
>>> (That's one of the reasons I prefer to have "Data Offset" be a clearly
>>> invalid value whenever it is overridden by EDO: any middlebox will then
>>> have fair warning that it is clueless.)
>>
>> It had that warning when it saw the EDO option and decided that it could
>> ignore it quietly.
>
>     This points out another failure of the TCP-Options design: for such
> a basic building block, Options should have carried an indicator of
> what to do with an unrecognized option -- e.g. ignore, drop the packet,
> etc.

Right - we built that into IPv6, but TCP missed that boat.

However, it still requires that middleboxes would do the right thing; 
their incentive to "keep working" suggests they would treat this like 
all other Internet standards - something to violate if it makes them money.

So even if the flag were there (it sort of is already - but "ignore" on 
pass-through should have been interpreted as "drop"), it's not clear 
middlebox vendors would do as we proscribe.

Joe


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Hi, all,

I've updated the doc to include some of the ideas suggested (checksum, 
DO=0) as issues to be discussed, including some ratinonale, as well as 
implementation issues like keeping data and options separate in the TCP API.

My view is that these details can be considered and consensus reached in 
the WG, but there seems to be a reasonable path (or paths) forward 
either way, AFAICT.

Joe

-------- Original Message --------
Subject: New Version Notification for draft-touch-tcpm-tcp-edo-02.txt
Date: Mon, 05 May 2014 13:47:27 -0700
From: internet-drafts@ietf.org
To: Wesley M. Eddy <wes@mti-systems.com>,        Dr. Joseph D. Touch 
<touch@isi.edu>, Joe Touch <touch@isi.edu>,        Wesley Eddy 
<wes@mti-systems.com>


A new version of I-D, draft-touch-tcpm-tcp-edo-02.txt
has been successfully submitted by Joe Touch and posted to the
IETF repository.

Name:		draft-touch-tcpm-tcp-edo
Revision:	02
Title:		TCP Extended Data Offset Option
Document date:	2014-05-05
Group:		Individual Submission
Pages:		12
URL: 
http://www.ietf.org/internet-drafts/draft-touch-tcpm-tcp-edo-02.txt
Status:         https://datatracker.ietf.org/doc/draft-touch-tcpm-tcp-edo/
Htmlized:       http://tools.ietf.org/html/draft-touch-tcpm-tcp-edo-02
Diff:           http://www.ietf.org/rfcdiff?url2=draft-touch-tcpm-tcp-edo-02

Abstract:
    TCP segments include a Data Offset field to indicate space for TCP
    options, but the size of the field can limit the space available for
    complex options that have evolved. This document updates RFC 793
    with an optional TCP extension to that space and explains why such
    extensions cannot be used in the initial SYN of a connection.

 



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.

The IETF Secretariat



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Some comments on the approaches for handling the connection:

On 5/5/2014 9:30 AM, John Leslie wrote:
> Wesley Eddy <wes@mti-systems.com> wrote:
...
>> I think I agree.  IMHO, the tradeoff comes down to whether it's more
>> desirable for the middlebox to end up doing something violent like:
>>
>> - dropping the segment
>
>     This should always be "legal" for a middlebox. (I certainly prefer
> it to damaging some unrelated part of the packet.) It's easy enough
> to detect.

Agreed.

>> - resetting the TCP connection
>
>     This isn't quite "legal", but is certainly easy to detect.

Agreed.

>> - generating an ICMP
>
>     Would anyone actually do this? (Hopefully, it would be in combination
> with dropping the packet.)

This ought to be legal only when the midbox sends a router-like ICMP 
error. The only one that makes sense to me is "protocol unreachable".

However, this could be interpreted by the source as the fact that TCP 
isn't reachable at all, not just TCP-with-EDO.

So I don't think this is viable at all.

>> - crashing
>
>     That's my most-preferred one. There's no excuse to blame anyone but
> the middlebox vendor. ;^)

I like that one too.

Joe


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To: Joe Touch <touch@isi.edu>, "Olivier.Bonaventure@uclouvain.be" <Olivier.Bonaventure@uclouvain.be>, "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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>=20
> However, now you're back to the NIC issue (which ought not be related):
>=20
> > In their IMC paper entitled "Is it Still Possible to Extend TCP?",
> > Michio Honda et al. provide the following recommendation
> > http://conferences.sigcomm.org/imc/2011/docs/p181.pdf
> >
> > "All the NICs we tested correctly copied the options to all the split
> > segments. TSO is now sufficiently commonplace so that designers of
> > extensions to TCP should assume it. The implication is that TCP
> > options must be designed so that when they are duplicated on
> > consecutive segments, this does not adversely affect correctness or
> performance."
>=20
> This tells me your TCP interface was implemented incorrectly.
>=20
> The correct interface passes options (including the extended option *and
> its associated contents*) ***SEPARATELY*** from the user data.
>=20
> If that is done all the way down to the TSO, the TSO will correctly copy
> the option into individual segments correctly.
>=20
> If, however, you implement this badly - and pass TCP segments as a whole
> to the offload engine, then the offload engine *has no business* splittin=
g
> or coalescing segments.


I can agree with the assessment, that TSO is typically done wrong, and that=
 the API was not properly designed.

Most TSO NICs only deal with "typical" header flags - from my ECN research =
I know that non-ECT (CE; ECT(0); ECT(1)) IP flags are not necessarily deliv=
ered correctly by LRO (only the IP bits of the first segment that was recei=
ved are usually handed up to the software TCP stack, even when they change;=
 thus ECN information in up to 200 segments may never reach the TCP layer f=
or handling (and recent measurements also indicate that for ECN, the gating=
 issue is not TCP ECN, but the IP ECN flags getting forwarded end-to-end pr=
operly...

On the LRO TCP side, any "unexpected" TCP header bits (EDO wouldn't account=
 for this; setting DO to an invalid value might) would typically stop the f=
astpath processing, and get individual segments delivered - thus the RFC316=
8 ECN TCP signaling usually works (as ECE and/or CWR bits are set in those =
segments).


As for a TSO sender: if the NIC driver is broken, the sending TCP can ensur=
e that the hardware doesn't perform invalid splits on segments containing E=
DO - it has to be provided as a NIC driver workaround, sure;=20


Richard Scheffenegger
=20


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In my group at the Radboud University Nijmegen, the Netherlands, we do 
research on automata learning. We develop algorithms that allow us to 
learn state diagrams of black box systems by providing inputs and 
observing outputs.

Together with my students Paul Fiterau and Ramon Janssen, we just 
completed a paper in which we describe how we learned models of small 
fragments of TCP for Windows8 and Ubuntu 13.10 implementations: 
http://www.mbsd.cs.ru.nl/publications/papers/fvaan/TCP/.

Based on these results, we conclude that both Ubuntu 13.10 and Windows8 
do not conform to the
TCP standard. On page 37 RFC793 says:

3.  If the connection is in a synchronized state (ESTABLISHED,
     FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
     any unacceptable segment (out of window sequence number or
     unacceptible acknowledgment number) must elicit only an empty
     acknowledgment segment containing the current send-sequence number
     and an acknowledgment indicating the next sequence number expected
     to be received, and the connection remains in the same state.

We discovered that when an Ubuntu server is in CLOSE_WAIT state and the client sends an ACK or a FIN+ACK with a valid seq number but an invalid ack number, the server
sometimes does not return any message. In Windows 8 the required acknowledgment segment is never sent.

We would be very interested to hear your opinion on these findings.
Is this indeed a violation of the standard? Is this a known issue?

Thanks!

Frits Vaandrager


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<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    In my group at the Radboud University Nijmegen, the Netherlands, we
    do research on automata learning. We develop algorithms that allow
    us to learn state diagrams of black box systems by providing inputs
    and observing outputs.<br>
    <br>
    Together with my students Paul Fiterau and Ramon Janssen, we just
    completed a paper in which we describe how we learned models of
    small fragments of TCP for Windows8 and Ubuntu 13.10
    implementations: <a class="moz-txt-link-freetext"
      href="http://www.mbsd.cs.ru.nl/publications/papers/fvaan/TCP/">http://www.mbsd.cs.ru.nl/publications/papers/fvaan/TCP/</a>.<br>
    <br>
    Based on these results, we conclude that both Ubuntu 13.10 and
    Windows8 do not conform to the<br>
    TCP standard. On page 37 RFC793 says:
    <pre class="" style="font-size:1em" id="yiv3183956486yui_3_13_0_1_1398878592264_2813">3.  If the connection is in a synchronized state (ESTABLISHED,
    FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
    any unacceptable segment (out of window sequence number or
    unacceptible acknowledgment number) must elicit only an empty
    acknowledgment segment containing the current send-sequence number
    and an acknowledgment indicating the next sequence number expected
    to be received, and the connection remains in the same state.
<font face="sans-serif">
We discovered that when an Ubuntu server is in CLOSE_WAIT state and the client sends an ACK or a FIN+ACK with a valid seq number but an invalid ack number, the server
sometimes does not return any message. In Windows 8 the required acknowledgment segment is never sent.</font></pre>
    <pre class="" style="font-size:1em" id="yiv3183956486yui_3_13_0_1_1398878592264_2813"><font face="sans-serif">We would be very interested to hear your opinion on these findings. 
Is this indeed a violation of the standard? Is this a known issue?

Thanks!

Frits Vaandrager
</font></pre>
  </body>
</html>

--------------050806030900030309030107--


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From: Hagen Paul Pfeifer <hagen@jauu.net>
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Subject: Re: [tcpm] TCP implementations in Windows8 and Ubuntu violate standard?
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The scenario that you spotted MAY violate 793, but it has no real practical
relevance. I.e. compared to 793 I see no performance or other negative
impact where the behavior is counterproductive. Except it differs from 793.

2) why the Linux implementation differs should be asked in netdev.
Sometimes standards are not fully implemented because of performance &
security considerations or simple bugs

Anyway: good catch!

Hagen

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<p dir="ltr">The scenario that you spotted MAY violate 793, but it has no real practical relevance. I.e. compared to 793 I see no performance or other negative impact where the behavior is counterproductive. Except it differs from 793.<br>

 <br>
2) why the Linux implementation differs should be asked in netdev. Sometimes standards are not fully implemented because of performance &amp; security considerations or simple bugs</p>
<p dir="ltr">Anyway: good catch!</p>
<p dir="ltr">Hagen</p>

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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: [tcpm] Definition of flow control
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Hi

I am writing up a paper that is fairly TCP-ish. I have a problem regarding =
the definition of the part of TCP that does the job to keep control over th=
e packet transmission i.e the "simple" part that transmits segments wheneve=
r the bytes in flight is less than the congestion window. I call this part =
flow control but this is perhaps wrong ?. What is then the correct term, "r=
ate control" or "rate limiting" or ?

/Ingemar
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

"Those are my principles, and if you don't like them...
well, I have others."  Groucho Marx<http://www.brainyquote.com/quotes/autho=
rs/g/groucho_marx.html>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D



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<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I am writing up a paper that is fairly TCP-ish. I ha=
ve a problem regarding the definition of the part of TCP that does the job =
to keep control over the packet transmission i.e the &#8220;simple&#8221; p=
art that transmits segments whenever the bytes
 in flight is less than the congestion window. I call this part flow contro=
l but this is perhaps wrong ?. What is then the correct term, &#8220;rate c=
ontrol&#8221; or &#8220;rate limiting&#8221; or ?
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">/Ingemar<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Joh=
ansson&nbsp; M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
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<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Wireless Ac=
cess Networks<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Labratorieg=
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>
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Date: Thu, 8 May 2014 16:35:58 +0000
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Hi Ingemar,

if it really is only the function of "when to send" rather than "what to se=
nd", that would probably be called scheduler (unfortunately, the "what"/"wh=
en" are mingled together inseparable in most stacks - perhaps not in Lamina=
r:

https://developers.google.com/speed/protocols/tcp-laminar

There, the term "transmission scheduling algorithm" is used...


Richard Scheffenegger


From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Ingemar Johansson S
Sent: Donnerstag, 08. Mai 2014 17:55
To: tcpm@ietf.org
Subject: [tcpm] Definition of flow control

Hi

I am writing up a paper that is fairly TCP-ish. I have a problem regarding =
the definition of the part of TCP that does the job to keep control over th=
e packet transmission i.e the "simple" part that transmits segments wheneve=
r the bytes in flight is less than the congestion window. I call this part =
flow control but this is perhaps wrong ?. What is then the correct term, "r=
ate control" or "rate limiting" or ?

/Ingemar
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

"Those are my principles, and if you don't like them...
well, I have others."  Groucho Marx<http://www.brainyquote.com/quotes/autho=
rs/g/groucho_marx.html>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Ingemar,<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">if it r=
eally is only the function of &#8222;when to send&#8220; rather than &#8222=
;what to send&#8220;, that would probably be called scheduler (unfortunatel=
y, the &#8222;what&#8220;/&#8220;when&#8220; are mingled together inseparab=
le in most
 stacks &#8211; perhaps not in Laminar:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><a href=
=3D"https://developers.google.com/speed/protocols/tcp-laminar">https://deve=
lopers.google.com/speed/protocols/tcp-laminar</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">There, =
the term &#8220;</span><span lang=3D"EN-US">transmission scheduling algorit=
hm&#8221; is used&#8230;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">Richard=
 Scheffenegger</span><span lang=3D"EN-US" style=3D"font-size:10.0pt;color:#=
1F497D"><br>
<br>
</span><span lang=3D"EN-US" style=3D"font-size:8.0pt;font-family:&quot;Luci=
da Console&quot;;color:#1F497D;mso-fareast-language:EN-US"><o:p></o:p></spa=
n></p>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
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<div>
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<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> tcpm [mailto:tcpm-bounces@ietf.org]
<b>On Behalf Of </b>Ingemar Johansson S<br>
<b>Sent:</b> Donnerstag, 08. Mai 2014 17:55<br>
<b>To:</b> tcpm@ietf.org<br>
<b>Subject:</b> [tcpm] Definition of flow control<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I am writing up a paper that is=
 fairly TCP-ish. I have a problem regarding the definition of the part of T=
CP that does the job to keep control over the packet transmission i.e the &=
#8220;simple&#8221; part that transmits segments
 whenever the bytes in flight is less than the congestion window. I call th=
is part flow control but this is perhaps wrong ?. What is then the correct =
term, &#8220;rate control&#8221; or &#8220;rate limiting&#8221; or ?
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">/Ingemar<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
t;">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
t;">Ingemar Johansson&nbsp; M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
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<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
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e:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p></o:p><=
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tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
t;;color:black">&#8220;Those are my principles, and if you don't like them.=
..
<br>
well, I have others.&#8221;&nbsp; </span><span lang=3D"EN-US" style=3D"font=
-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><a href=
=3D"http://www.brainyquote.com/quotes/authors/g/groucho_marx.html"><span st=
yle=3D"color:windowtext;text-decoration:none">Groucho Marx</span></a><br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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From nobody Thu May  8 10:21:01 2014
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To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: Re: [tcpm] Definition of flow control
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AFAICT, you answer your own question...

On 5/8/2014 8:54 AM, Ingemar Johansson S wrote:
> Hi
>
> I am writing up a paper that is fairly TCP-ish. I have a problem
> regarding the definition of the part of TCP that does the job to keep
> control over the packet transmission i.e the “simple” part that
> transmits segments whenever the bytes in flight is less than the
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> congestion window. I call this part flow control but this is perhaps
^^^^^^^^^^^^^^^^^^^
> wrong ?. What is then the correct term, “rate control” or “rate
> limiting” or ?

Congestion control.

Flow control is managed by the receive window.

TCP doesn't have rate control or rate limiting - it'll burst just fine, 
e.g., after the source goes idle during transactions like HTTP. Some 
variants have added rate management, but it's not part of the core 
required specification.

Joe

>
> /Ingemar
>
> =================================
>
> Ingemar Johansson  M.Sc.
>
> Senior Researcher
>
> Ericsson AB
>
> Wireless Access Networks
>
> Labratoriegränd 11
>
> 971 28, Luleå, Sweden
>
> Phone +46-1071 43042
>
> SMS/MMS +46-73 078 3289
>
> ingemar.s.johansson@ericsson.com <mailto:ingemar.s.johansson@ericsson.com>
>
> www.ericsson.com
>
> “Those are my principles, and if you don't like them...
> well, I have others.” Groucho Marx
> <http://www.brainyquote.com/quotes/authors/g/groucho_marx.html>
> =================================
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: Definition of flow control
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Subject: Re: [tcpm] Definition of flow control
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Keep in mind that there is not only the congestion window, but also the rec=
eive window.

>From RFC 5681: At any given time, a TCP MUST NOT send data with a sequence =
number higher than the sum of the highest acknowledged sequence number and =
the minimum of cwnd and rwnd.

In TCP contect, the term "flow control" IMHO usually refers to the handling=
 of rwnd, while "congestion control" refers to the algorithms manipuiating =
cwnd. However, the term "flow control" is not always used consistently, in =
particular outside the IETF.

Michael



________________________________
Von: tcpm [tcpm-bounces@ietf.org]" im Auftrag von "Ingemar Johansson S [ing=
emar.s.johansson@ericsson.com]
Gesendet: Donnerstag, 8. Mai 2014 17:54
Bis: tcpm@ietf.org
Betreff: [tcpm] Definition of flow control

Hi

I am writing up a paper that is fairly TCP-ish. I have a problem regarding =
the definition of the part of TCP that does the job to keep control over th=
e packet transmission i.e the =93simple=94 part that transmits segments whe=
never the bytes in flight is less than the congestion window. I call this p=
art flow control but this is perhaps wrong ?. What is then the correct term=
, =93rate control=94 or =93rate limiting=94 or ?

/Ingemar
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com<UrlBlockedError.aspx>

=93Those are my principles, and if you don't like them...
well, I have others.=94  Groucho Marx<http://www.brainyquote.com/quotes/aut=
hors/g/groucho_marx.html>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D



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">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">Keep in mind that there is not only the congestion window, but also =
the receive window.<br>
<br>
>From RFC 5681: At any given time, a TCP MUST NOT send data with a sequence =
number higher than the sum of the highest acknowledged sequence number and =
the minimum of cwnd and rwnd.<br>
<br>
In TCP contect, the term &quot;flow control&quot; IMHO usually refers to th=
e handling of rwnd, while &quot;congestion control&quot; refers to the algo=
rithms manipuiating cwnd. However, the term &quot;flow control&quot; is not=
 always used consistently, in particular outside the IETF.<br>
<br>
Michael<br>
<br>
<br>
<br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<hr tabindex=3D"-1">
<div style=3D"direction: ltr;" id=3D"divRpF55981"><font color=3D"#000000" f=
ace=3D"Tahoma" size=3D"2"><b>Von:</b> tcpm [tcpm-bounces@ietf.org]&quot; im=
 Auftrag von &quot;Ingemar Johansson S [ingemar.s.johansson@ericsson.com]<b=
r>
<b>Gesendet:</b> Donnerstag, 8. Mai 2014 17:54<br>
<b>Bis:</b> tcpm@ietf.org<br>
<b>Betreff:</b> [tcpm] Definition of flow control<br>
</font><br>
</div>
<div></div>
<div>
<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">I am writing up a paper that is fairly TCP-ish. I ha=
ve a problem regarding the definition of the part of TCP that does the job =
to keep control over the packet transmission i.e the =93simple=94 part that=
 transmits segments whenever the bytes
 in flight is less than the congestion window. I call this part flow contro=
l but this is perhaps wrong ?. What is then the correct term, =93rate contr=
ol=94 or =93rate limiting=94 or ?
</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">/Ingemar</p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Jo=
hansson&nbsp; M.Sc.
</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Senior Res=
earcher</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">&nbsp;</sp=
an></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ericsson A=
B</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Wireless A=
ccess Networks</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Labratorie=
gr=E4nd 11</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">971 28, Lu=
le=E5, Sweden</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Phone &#43=
;46-1071 43042</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">SMS/MMS &#=
43;46-73 078 3289</span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;" lang=3D"SV=
"><a href=3D"mailto:ingemar.s.johansson@ericsson.com" target=3D"_blank"><sp=
an style=3D"color:windowtext" lang=3D"EN-US">ingemar.s.johansson@ericsson.c=
om</span></a></span><span style=3D"font-size:10.0pt; font-family:&quot;Aria=
l&quot;,&quot;sans-serif&quot;"></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;" lang=3D"SV=
"><a href=3D"UrlBlockedError.aspx" target=3D"_blank"><span style=3D"color:w=
indowtext" lang=3D"EN-US">www.ericsson.com</span></a></span><span style=3D"=
font-size:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"></s=
pan></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">&nbsp;</sp=
an></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-serif&quot;; color:blac=
k">=93Those are my principles, and if you don't like them...
<br>
well, I have others.=94&nbsp; </span><span style=3D"font-size:10.0pt; font-=
family:&quot;Arial&quot;,&quot;sans-serif&quot;"><a href=3D"http://www.brai=
nyquote.com/quotes/authors/g/groucho_marx.html" target=3D"_blank"><span sty=
le=3D"color:windowtext; text-decoration:none">Groucho Marx</span></a><br>
</span><span style=3D"font-size:10.0pt; font-family:&quot;Arial&quot;,&quot=
;sans-serif&quot;">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:7.5pt; line-height:15.0pt"><s=
pan style=3D"font-size:10.0pt; font-family:&quot;Arial&quot;,&quot;sans-ser=
if&quot;; color:black">&nbsp;</span></p>
<p class=3D"MsoNormal">&nbsp;</p>
</div>
</div>
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From nobody Thu May  8 18:32:09 2014
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Date: Thu, 08 May 2014 18:31:17 -0700
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP implementations in Windows8 and Ubuntu violate standard?
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On 5/7/2014 6:11 AM, Frits Vaandrager wrote:
> In my group at the Radboud University Nijmegen, the Netherlands, we do
> research on automata learning. We develop algorithms that allow us to
> learn state diagrams of black box systems by providing inputs and
> observing outputs.
>
> Together with my students Paul Fiterau and Ramon Janssen, we just
> completed a paper in which we describe how we learned models of small
> fragments of TCP for Windows8 and Ubuntu 13.10 implementations:
> http://www.mbsd.cs.ru.nl/publications/papers/fvaan/TCP/.
>
> Based on these results, we conclude that both Ubuntu 13.10 and Windows8
> do not conform to the
> TCP standard. On page 37 RFC793 says:
>
> 3.  If the connection is in a synchronized state (ESTABLISHED,
>      FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
>      any unacceptable segment (out of window sequence number or
>      unacceptible acknowledgment number) must elicit only an empty
>      acknowledgment segment containing the current send-sequence number
>      and an acknowledgment indicating the next sequence number expected
>      to be received, and the connection remains in the same state.
>
> We discovered that when an Ubuntu server is in CLOSE_WAIT state and
> the client sends an ACK or a FIN+ACK with a valid seq number but an
> invalid ack number, the server sometimes does not return any message.
> In Windows 8 the required acknowledgment segment is never sent.
>
> We would be very interested to hear your opinion on these findings.
> Is this indeed a violation of the standard? Is this a known issue?

It seems so (we'd have to check in detail). I don't recall anyone 
raising it before.

The key question is why this requirement was given. It seems like it's 
intended to help the sender of the unacceptable segment to determine 
that it had outstanding data to resend, or that it had exceeded the 
receiver's window.

The behavior you saw might cause connections to not recover properly, to 
either retransmit data it already had or to drop connections that could 
have been recovered after a loss or receiver change (e.g., if the 
receive window were made smaller during a connection).

The "practical" relevance depends on how often you expect this to occur, 
and whether the current behavior is acceptable vs. full recovery.

Joe





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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "Scheffenegger, Richard" <rs@netapp.com>, Joe Touch <touch@isi.edu>
Thread-Topic: Definition of flow control
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Subject: Re: [tcpm] Definition of flow control
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Thanks all, the TCP laminar definition "transmission scheduling algorithm" =
helped me a lot, most important I did not need to rewrite the whole documen=
t

/Ingemar

From: Scheffenegger, Richard [mailto:rs@netapp.com]
Sent: den 8 maj 2014 18:36
To: Ingemar Johansson S; tcpm@ietf.org
Subject: RE: Definition of flow control


Hi Ingemar,

if it really is only the function of "when to send" rather than "what to se=
nd", that would probably be called scheduler (unfortunately, the "what"/"wh=
en" are mingled together inseparable in most stacks - perhaps not in Lamina=
r:

https://developers.google.com/speed/protocols/tcp-laminar

There, the term "transmission scheduling algorithm" is used...


Richard Scheffenegger

From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Ingemar Johansson S
Sent: Donnerstag, 08. Mai 2014 17:55
To: tcpm@ietf.org<mailto:tcpm@ietf.org>
Subject: [tcpm] Definition of flow control

Hi

I am writing up a paper that is fairly TCP-ish. I have a problem regarding =
the definition of the part of TCP that does the job to keep control over th=
e packet transmission i.e the "simple" part that transmits segments wheneve=
r the bytes in flight is less than the congestion window. I call this part =
flow control but this is perhaps wrong ?. What is then the correct term, "r=
ate control" or "rate limiting" or ?

/Ingemar
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

"Those are my principles, and if you don't like them...
well, I have others."  Groucho Marx<http://www.brainyquote.com/quotes/autho=
rs/g/groucho_marx.html>
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks all, the TCP la=
minar definition &#8220;</span>transmission scheduling algorithm<span style=
=3D"color:#1F497D">&#8221; helped me a lot, most important I did not need t=
o rewrite the whole document<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">/Ingemar<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Scheffen=
egger, Richard [mailto:rs@netapp.com]
<br>
<b>Sent:</b> den 8 maj 2014 18:36<br>
<b>To:</b> Ingemar Johansson S; tcpm@ietf.org<br>
<b>Subject:</b> RE: Definition of flow control<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"DE-AT" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"DE-AT" style=3D"color:#1F497D">Hi Inge=
mar,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"DE-AT" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">if it really is only t=
he function of &#8222;when to send&#8220; rather than &#8222;what to send&#=
8220;, that would probably be called scheduler (unfortunately, the &#8222;w=
hat&#8220;/&#8220;when&#8220; are mingled together inseparable in most stac=
ks &#8211; perhaps
 not in Laminar:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><a href=3D"https://dev=
elopers.google.com/speed/protocols/tcp-laminar">https://developers.google.c=
om/speed/protocols/tcp-laminar</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">There, the term &#8220=
;</span>transmission scheduling algorithm&#8221; is used&#8230;<o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"color:=
#1F497D">Richard Scheffenegger</span><span style=3D"font-size:8.0pt;font-fa=
mily:&quot;Lucida Console&quot;;color:#1F497D"><o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
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0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> tcpm [<a=
 href=3D"mailto:tcpm-bounces@ietf.org">mailto:tcpm-bounces@ietf.org</a>]
<b>On Behalf Of </b>Ingemar Johansson S<br>
<b>Sent:</b> Donnerstag, 08. Mai 2014 17:55<br>
<b>To:</b> <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<b>Subject:</b> [tcpm] Definition of flow control<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span lang=3D"DE-AT"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I am writing up a paper that is fairly TCP-ish. I ha=
ve a problem regarding the definition of the part of TCP that does the job =
to keep control over the packet transmission i.e the &#8220;simple&#8221; p=
art that transmits segments whenever the bytes
 in flight is less than the congestion window. I call this part flow contro=
l but this is perhaps wrong ?. What is then the correct term, &#8220;rate c=
ontrol&#8221; or &#8220;rate limiting&#8221; or ?
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">/Ingemar<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">=3D=3D=3D=
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=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Joh=
ansson&nbsp; M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Senior Rese=
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Date: Fri, 09 May 2014 10:45:33 -0400
From: Brandon Williams <brandon.williams@akamai.com>
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Subject: Re: [tcpm] New Version Notification for draft-williams-exp-tcp-host-id-opt-03.txt
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Lars and Wes,

I understand and share the objections to adding options in-transit that 
have been raised on the list. It's bad and should be avoided if there is 
any other technically feasible option for solving a problem. On the 
other hand, I also recognize that NATs do exist and they frequently 
cause significant problems, including load-balancing, policy control, 
and attack-mitigation. Is there something specific in our analysis of 
the problem and possible solution sets that you consider to be flawed? 
Or is it simply your opinion that the negative side-affects of adding 
options in-transit are worse than the negative consequences of treating 
all hosts behind a NAT as if they were the same entity?

Considering the fact that some day (fingers crossed) IPv6 will become 
the norm and NATs will fall out of use, I'm more interested in focusing 
on the proxy use case. I have a harder time understanding your 
objections in the context of this use case (esp. for transport layer 
proxies), since the option is applied to the packet by the TCP stack 
itself, rather than being applied in-transit. In reference to the 
application proxy use case, Wes has said previously "[It] is better done 
at the application layer. ... I wouldn't feel strongly against [this 
case], it's just not good architecture." But what about the case of 
transport layer proxies that aren't involved at the application layer? 
Or receiver-side use cases where the information must be applied before 
the data stream even shows up to be delivered? Is it that you just don't 
consider these to be legitimate use cases?

In addition to the above, an argument was made on the list (by Wes) that 
there is value to driving vendors to converge on a common option rather 
than continuing with the fragmented solution space that exists today. It 
is already the case that multiple vendors (those the authors represent) 
are attempting to use the draft for this purpose, and we have every 
reason to believe, based on our interactions with other vendors (see for 
example a thread on the list re: the dante socks proxy), that some other 
vendors would move to the common format if the draft is adopted and 
published. What number of vendors moving to a common format would be 
considered large enough for this argument to be persuasive?

If you have the time, any further elaboration from you on the above 
questions would be greatly appreciated.

Thanks,
--Brandon

On 04/30/2014 03:18 PM, Wesley Eddy wrote:
> On 4/30/2014 10:51 AM, Eggert, Lars wrote:
>> For the record, I am opposed to the WG taking this document on, and I
>> am opposed to it being published as an RFC on any stream.
>>
>
>
> My opinion is similar.
>
> An RFC isn't necessary to use an ExID.  It doesn't require IETF
> approval, and is unlikely to get IETF approval, in order to do
> this in a system, if you really want to discount the problems
> and do it anyways.
>
>

-- 
Brandon Williams; Senior Principal Software Engineer
Emerging Products Engineering; Akamai Technologies Inc.


From nobody Mon May 12 11:48:45 2014
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Subject: [tcpm] Fwd: New Version Notification for draft-touch-tcp-ao-encrypt-01.txt
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FYI - this may be of interest to TCPM, though it's primarily aimed at 
the TCPCRYPT discussions.

Feedback welcome on either list.

Joe


-------- Original Message --------
Subject: New Version Notification for draft-touch-tcp-ao-encrypt-01.txt
Date: Mon, 12 May 2014 11:43:57 -0700
From: internet-drafts@ietf.org
To: Dr. Joseph D. Touch <touch@isi.edu>, Joe Touch <touch@isi.edu>


A new version of I-D, draft-touch-tcp-ao-encrypt-01.txt
has been successfully submitted by Joe Touch and posted to the
IETF repository.

Name:		draft-touch-tcp-ao-encrypt
Revision:	01
Title:		A TCP Authentication Option Extension for Payload Encryption
Document date:	2014-05-12
Group:		Individual Submission
Pages:		9
URL: 
http://www.ietf.org/internet-drafts/draft-touch-tcp-ao-encrypt-01.txt
Status:         https://datatracker.ietf.org/doc/draft-touch-tcp-ao-encrypt/
Htmlized:       http://tools.ietf.org/html/draft-touch-tcp-ao-encrypt-01
Diff: 
http://www.ietf.org/rfcdiff?url2=draft-touch-tcp-ao-encrypt-01

Abstract:
    This document describes an extension to the TCP Authentication
    Option (TCP-AO) to encrypt the TCP segment payload in addition to
    providing TCP-AO's authentication of the payload, TCP header, and IP
    pseudoheader. This extension augments how the packet contents and
    headers are processed and which keys are derived, and adds a
    capability for in-band coordination of unauthenticated Diffie-
    Hellman key exchange at connection establishment. The extension
    preserves key rollover coordination and protection of long-lived
    connections.

 



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.

The IETF Secretariat



From nobody Mon May 12 14:34:17 2014
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Hi, all,

Although I appreciate some concerns about this doc, I don't quite share 
them.

A key concern is whether this encourages middleboxes to alter TCP options.

IMO, a middlebox that alters an IP address and TCP port is taking on the 
role of "host" to the public side. In that case, it is responsible for 
any options that are related to those values - including this one.

I.e., so long as we restrict this to devices that set IP addresses 
and/or TCP ports, I see no problem. If the issue is that "middleboxes 
shouldn't alter TCP options", then the real issue is that such 
middleboxes shouldn't alter IP addresses or ports.

Joe

On 5/9/2014 7:45 AM, Brandon Williams wrote:
> Lars and Wes,
>
> I understand and share the objections to adding options in-transit that
> have been raised on the list. It's bad and should be avoided if there is
> any other technically feasible option for solving a problem. On the
> other hand, I also recognize that NATs do exist and they frequently
> cause significant problems, including load-balancing, policy control,
> and attack-mitigation. Is there something specific in our analysis of
> the problem and possible solution sets that you consider to be flawed?
> Or is it simply your opinion that the negative side-affects of adding
> options in-transit are worse than the negative consequences of treating
> all hosts behind a NAT as if they were the same entity?
>
> Considering the fact that some day (fingers crossed) IPv6 will become
> the norm and NATs will fall out of use, I'm more interested in focusing
> on the proxy use case. I have a harder time understanding your
> objections in the context of this use case (esp. for transport layer
> proxies), since the option is applied to the packet by the TCP stack
> itself, rather than being applied in-transit. In reference to the
> application proxy use case, Wes has said previously "[It] is better done
> at the application layer. ... I wouldn't feel strongly against [this
> case], it's just not good architecture." But what about the case of
> transport layer proxies that aren't involved at the application layer?
> Or receiver-side use cases where the information must be applied before
> the data stream even shows up to be delivered? Is it that you just don't
> consider these to be legitimate use cases?
>
> In addition to the above, an argument was made on the list (by Wes) that
> there is value to driving vendors to converge on a common option rather
> than continuing with the fragmented solution space that exists today. It
> is already the case that multiple vendors (those the authors represent)
> are attempting to use the draft for this purpose, and we have every
> reason to believe, based on our interactions with other vendors (see for
> example a thread on the list re: the dante socks proxy), that some other
> vendors would move to the common format if the draft is adopted and
> published. What number of vendors moving to a common format would be
> considered large enough for this argument to be persuasive?
>
> If you have the time, any further elaboration from you on the above
> questions would be greatly appreciated.
>
> Thanks,
> --Brandon
>
> On 04/30/2014 03:18 PM, Wesley Eddy wrote:
>> On 4/30/2014 10:51 AM, Eggert, Lars wrote:
>>> For the record, I am opposed to the WG taking this document on, and I
>>> am opposed to it being published as an RFC on any stream.
>>>
>>
>>
>> My opinion is similar.
>>
>> An RFC isn't necessary to use an ExID.  It doesn't require IETF
>> approval, and is unlikely to get IETF approval, in order to do
>> this in a system, if you really want to discount the problems
>> and do it anyways.
>>
>>
>


From nobody Tue May 13 00:15:25 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: Brandon Williams <brandon.williams@akamai.com>
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Hi,

apart from the technical disagreement I have with the document, there is =
a broader political issue here.

The IETF has come to broad consensus that pervasive surveillance =
represents an attack on the Internet, and that we are taking engineering =
steps to protect privacy. (See http://www.ietf.org/blog/2013/11/ for =
example.)

Having middleboxes tag flows with options that make identifying =
individuals easier goes directly against that stated goal.

Lars

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From nobody Tue May 13 01:15:32 2014
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Date: Tue, 13 May 2014 08:15:19 +0000
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Subject: [tcpm] Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Hi

Karen pointed out this thread to me
http://www.ietf.org/mail-archive/web/tcpm/current/msg08315.html
I started to look closer at this issue quite recently, the problem I have i=
s that SSthresh can drop to very low values after only a few lost packets. =
I have seen this odd effect earlier but have not bothered with it.


The experiment is to run a large FTP transfer over a 1Mbps bottleneck with =
min RTT =3D 40ms. No AQM or tail drop queue enabled i.e a buffer bloated sc=
enario.

TCP NewReno. After 10s I "pull the plug" for 100ms (100% packet drop), this=
 leads to 5 lost segments. at T =3D10.1s packets are forwarded as usual.

What I have seen is that a retransmission timeout is immediately followed b=
y a loss event, the effect of which is that SSThresh goes down to 2 MSS.  I=
t seems to me that the RTO timer value is too low, I have not understood th=
e effect completely though. Could the RTO timer be the culprit or is there =
some other effect ?.

I am running these experiments in a proprietary LTE system simulator, we tr=
y to keep it up to date to match the Linux TCP stack reasonably well, it ca=
nnot be ruled out however that our implementation miss some important featu=
re.

/Ingemar

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

"Those are my principles, and if you don't like them...
well, I have others."  Groucho Marx<http://www.brainyquote.com/quotes/autho=
rs/g/groucho_marx.html>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D



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<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Karen pointed out this thread to me <o:p></o:p></p>
<p class=3D"MsoNormal"><a href=3D"http://www.ietf.org/mail-archive/web/tcpm=
/current/msg08315.html">http://www.ietf.org/mail-archive/web/tcpm/current/m=
sg08315.html</a>
<o:p></o:p></p>
<p class=3D"MsoNormal">I started to look closer at this issue quite recentl=
y, the problem I have is that SSthresh can drop to very low values after on=
ly a few lost packets. I have seen this odd effect earlier but have not bot=
hered with it.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">The experiment is to run a large FTP transfer ove=
r a 1Mbps bottleneck with min RTT =3D 40ms. No AQM or tail drop queue enabl=
ed i.e a buffer bloated scenario.<o:p></o:p></p>
<p class=3D"MsoPlainText">TCP NewReno. After 10s I &#8220;pull the plug&#82=
21; for 100ms (100% packet drop), this leads to 5 lost segments. at T =3D10=
.1s packets are forwarded as usual.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">What I have seen is that a retransmission timeout is=
 immediately followed by a loss event, the effect of which is that SSThresh=
 goes down to 2 MSS. &nbsp;It seems to me that the RTO timer value is too l=
ow, I have not understood the effect completely
 though. Could the RTO timer be the culprit or is there some other effect ?=
.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I am running these experiments in a proprietary LTE =
system simulator, we try to keep it up to date to match the Linux TCP stack=
 reasonably well, it cannot be ruled out however that our implementation mi=
ss some important feature.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">/Ingemar<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Joh=
ansson&nbsp; M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Senior Rese=
archer<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ericsson AB=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Wireless Ac=
cess Networks<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Labratorieg=
r=E4nd 11<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">971 28, Lul=
e=E5, Sweden<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Phone &#43;=
46-1071 43042<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">SMS/MMS &#4=
3;46-73 078 3289<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"=
><a href=3D"mailto:ingemar.s.johansson@ericsson.com"><span lang=3D"EN-US" s=
tyle=3D"color:windowtext">ingemar.s.johansson@ericsson.com</span></a></span=
><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-s=
erif&quot;"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"=
><a href=3D"www.ericsson.com"><span lang=3D"EN-US" style=3D"color:windowtex=
t">www.ericsson.com</span></a></span><span style=3D"font-size:10.0pt;font-f=
amily:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:black"=
>&#8220;Those are my principles, and if you don't like them...
<br>
well, I have others.&#8221;&nbsp; </span><span style=3D"font-size:10.0pt;fo=
nt-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><a href=3D"http://www.b=
rainyquote.com/quotes/authors/g/groucho_marx.html"><span style=3D"color:win=
dowtext;text-decoration:none">Groucho Marx</span></a><br>
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:7.5pt;line-height:15.0pt"><sp=
an style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif=
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From: "Eggert, Lars" <lars@netapp.com>
To: Brandon Williams <brandon.williams@akamai.com>
Thread-Topic: [tcpm] New Version Notification for draft-williams-exp-tcp-host-id-opt-03.txt
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PS: http://tools.ietf.org/html/rfc7258 came out today

On 2014-5-13, at 9:15, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>=20
> apart from the technical disagreement I have with the document, there =
is a broader political issue here.
>=20
> The IETF has come to broad consensus that pervasive surveillance =
represents an attack on the Internet, and that we are taking engineering =
steps to protect privacy. (See http://www.ietf.org/blog/2013/11/ for =
example.)
>=20
> Having middleboxes tag flows with options that make identifying =
individuals easier goes directly against that stated goal.
>=20
> Lars
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From nobody Tue May 13 02:58:04 2014
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: "draft-ietf-tcpm-rtorestart@tools.ietf.org" <draft-ietf-tcpm-rtorestart@tools.ietf.org>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Hi folks,

first of all thank you for writing this draft. The draft is clearly =
written: the
problem is well described, the doc is self-contained (it=92s not =
necessary to read
dozens of other RFCs to understand the problem), and the authors state =
why the
doc is experimental and what further experiments are needed for being a =
STD.
However, on the technical side I see some open points. In detail:

* Sec 1, 4. para: =84a considerable number of flows have such =
properties=93
  Can you backup this with some numbers? This is exactly the point I =
raised at the
  (last?) IETF. I would like to see some data on how severe the problem =
is we would
  like to fix.

* Sec 1.1: please change this subsection to a section (1.1 =3D> 2) and =
also
  introduce your new state variable rrthresh here

* Sec 1, 5. para: =84Spurious timeouts typically degrade the performance =
of flows
  with multiple bursts of data, as a burst following a spurious timeout =
might
  not fit within the reduced congestion window (cwnd)=93

  This is (only) true with respect to your algo, not in general. The =
general
  problem of a spurious timeout is the cwnd reduction, the go-back-N
  retransmissions, =85 See RFC 3522. After reading section 4.2 of your =
draft I
  understand what you want to see here. Please rephrase the section and =
maybe
  add the spurious RTO RFC as reference.

* Sec 3: Suppose FlightSize is 2 and you have exactly one segment to =
send,
  your algo doesn=92t trigger since step 2.b isn=92t true. Bug? I would =
say yes.

* Sec 3: Why the condition 2.b is different from the early =
retransmission
  condition 2.b or 3.b? Is there any specific reason why we exclude the
  advertised receive window part from the condition?

* Sec 3: IMO the algo/doc is too much Linux driven. I would like to see =
a
  segment-based *and* byte-based version of the algo, like RFC 5827.

* General: IMO I would be a little bit easier to read the doc if you =
give the
  algo a proper name. By reading =84RTO restart=93 I had sometimes =
trouble to
  know if mean your algo or the =84action=93 of restarting the RTO.


Alex




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From nobody Tue May 13 05:56:47 2014
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Hi Lars,

Regarding the privacy concern, is there something that we have missed in 
the section about privacy contained in the document? If the guidance 
provided in the document is followed, then I don't see how the use of 
this option makes identifying individuals easier.

In addition, I argue that the privacy concern is an argument in favor of 
adopting this draft. TCP options of this nature are in use on the 
Internet today and will continue to be used going forward. As long as 
there is no move toward consensus on the option format, number, etc. and 
as long as people active in the WG actively discourage authors from 
publishing documents related to this use of the option space, it will be 
difficult for anonymizing systems to recognize such options and protect 
anonymity. This is part of the reason why we give direct guidance on 
this topic in draft.

--Brandon

On 05/13/2014 03:15 AM, Eggert, Lars wrote:
> Hi,
>
> apart from the technical disagreement I have with the document, there is a broader political issue here.
>
> The IETF has come to broad consensus that pervasive surveillance represents an attack on the Internet, and that we are taking engineering steps to protect privacy. (See http://www.ietf.org/blog/2013/11/ for example.)
>
> Having middleboxes tag flows with options that make identifying individuals easier goes directly against that stated goal.
>
> Lars
>

-- 
Brandon Williams; Senior Principal Software Engineer
Emerging Products Engineering; Akamai Technologies Inc.


From nobody Tue May 13 08:16:28 2014
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On 5/13/2014 12:15 AM, Eggert, Lars wrote:
> Hi,
>
> apart from the technical disagreement I have with the document, there is a broader political issue here.

I certainly hope not, but the IESG Chair warned us about this (see 
below), so we shouldn't be surprised.

> The IETF has come to broad consensus that pervasive surveillance represents an attack on the Internet,

It would be more useful for the iETF to take positions on devices that 
undermine its architecture (e.g., NATs) and protocols that are deployed 
against current technical advice (CUBIC).

> and that we are taking engineering steps to protect privacy. (See http://www.ietf.org/blog/2013/11/ for example.)

The IESG Chair's report on the conclusion of community discussion 
specifically addressed the use of this document to block other docs:

---
3) The document, particularly as a BCP, could be used to block documents 
by ADs in an unreasonable fashion. There was considerable concern about 
this. However, the IESG process for document approval is most commonly 
not based on specific rules in existing RFCs, but rather specific 
technical concerns of the individual AD reviewers. ADs occasionally 
raise invalid issues on documents, and we expect (and get) pushback when 
that happens. I have certainly made mistakes when filing a Discuss. 
These are resolved through discussion. In more severe cases, we have 
override procedures, appeals, discussions with the Nomcom, and recall 
procedures to deal with these issues. In other words, if we start from 
the assumption of a misbehaving AD, no new documents are needed to file 
an inappropriate Discuss. Any AD can raise concerns about pervasive 
monitoring under the current Discuss criteria, and this document doesn't 
change that. We don't think this document will significantly change 
things for the worse. However, the IESG felt that there is an 
opportunity to clarify our procedures with regards to this specific case:

The IESG intends to make a change to its Discuss criteria statement [3]. 
While the general thrust of the statement already indicates that AD 
disagreement with informed WG consensus is a NOT an acceptable basis for 
a discuss, we wanted to say that more explicitly in the specific context 
of adequate consideration for pervasive monitoring issues. Section 3.2 
of the Discuss criteria statement is about criteria that are not 
appropriate basis for a Discuss. We plan to say that while it is 
generally necessary that IETF work considers the impact of specific 
threats such as pervasive surveillance, informed and rational community 
decisions about the particular protocols and the possible need for 
mitigating mechanisms will be respected.

> Having middleboxes tag flows with options that make identifying individuals easier goes directly against that stated goal.

First, we're talking about use of this option by middleboxes and 
endpoints that already change host IDs (not personal IDs).

Second, this mechanism can easily be used to obfuscate the 
identification hosts, to re-group them in alternate ways.

Finally, I sincerely hope this draft's process doesn't become the poster 
child of the process fears raised by the IESG Chair in approving perpass 
and recommending it as BCP.

Joe


From nobody Tue May 13 09:30:36 2014
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Hi,

On 2014-5-13, at 17:14, Joe Touch <touch@isi.edu> wrote:
> It would be more useful for the iETF to take positions on devices that =
undermine its architecture (e.g., NATs) and protocols that are deployed =
against current technical advice (CUBIC).

We're getting off-topic here, but what "current technical advice" is =
Cubic violating?=20

It's not an IETF standard, and it's on by default in Linux and that's =
not so nice, but it doesn't seem to be have serious flaws (anymore). So =
while I wish there was an ID we could progress (and I have it on my list =
to talk to Injong and get his OK for someone else to carry the torch), I =
think you're overstating things.

>> and that we are taking engineering steps to protect privacy. (See =
http://www.ietf.org/blog/2013/11/ for example.)
>=20
> The IESG Chair's report on the conclusion of community discussion =
specifically addressed the use of this document to block other docs:

The text you quoted is entirely about what ADs should and should not do =
*during IESG processing*. I'm not an AD (anymore), and I am certainly =
free to make this argument as an individual *during a call for WG =
adoption*.

>> Having middleboxes tag flows with options that make identifying =
individuals easier goes directly against that stated goal.
>=20
> First, we're talking about use of this option by middleboxes and =
endpoints that already change host IDs (not personal IDs).

Sure. But I don't see why this is relevant.

In some sense CGNs - as problematic as they are - actually increase =
privacy, because more users share an IP address. This document is about =
counteracting this increased privacy. And since we don't know by what =
mechanism these host IDs get set, and given that the document talks =
about potentially quite large ID namespace (32bits), you could actually =
identify things at a finer granularity than hosts.

> Second, this mechanism can easily be used to obfuscate the =
identification hosts, to re-group them in alternate ways.

How and by whom? Even if hosts end up setting it, it'll get stripped out =
or rewritten by an ISP middlebox that can set it based on subscriber =
information and current connection context.

> Finally, I sincerely hope this draft's process doesn't become the =
poster child of the process fears raised by the IESG Chair in approving =
perpass and recommending it as BCP.

So I actually hope it does. In my view, the reason that the text you =
quoted was entirely about the IESG process was to not have late =
surprises happen to stuff that are already WG documents or in IETF last =
call.

But that's not the case here. This is not a WG document, and using the =
BCP - that we have IETF consensus on - as an argument to not *start* =
problematic work is entirely reasonable. If not, what *can* we use that =
consensus BCP for? Feeling good about ourselves without actually doing =
anything?

Lars

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From nobody Tue May 13 09:56:49 2014
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On 5/13/2014 9:30 AM, Eggert, Lars wrote:
> Hi,
>
> On 2014-5-13, at 17:14, Joe Touch <touch@isi.edu> wrote:
>> It would be more useful for the iETF to take positions on devices that undermine its architecture (e.g., NATs) and protocols that are deployed against current technical advice (CUBIC).
>
> We're getting off-topic here, but what "current technical advice" is Cubic violating?
>
> It's not an IETF standard,

RFC5681

>>> and that we are taking engineering steps to protect privacy. (See http://www.ietf.org/blog/2013/11/ for example.)
>>
>> The IESG Chair's report on the conclusion of community discussion specifically addressed the use of this document to block other docs:
>
> The text you quoted is entirely about what ADs should and should not
> do *during IESG processing*. I'm not an AD (anymore), and I am certainly
> free to make this argument as an individual *during a call for WG adoption*.

Agreed. However, the entirety of the message indicated basically that 
the Chair was taking very rough - at best - consensus about a draft and 
making it BCP even though the community is widely split on the issue.

>>> Having middleboxes tag flows with options that make identifying individuals easier goes directly against that stated goal.
>>
>> First, we're talking about use of this option by middleboxes and endpoints that already change host IDs (not personal IDs).
>
> Sure. But I don't see why this is relevant.

Middleboxes create a problem that this is intended to let *them* fix.

> In some sense CGNs - as problematic as they are - actually increase
> privacy, because more users share an IP address. This document is about
> counteracting this increased privacy.

It's about letting *them* not provide that privacy where privacy wasn't 
intended.

> And since we don't know by what
> mechanism these host IDs get set, and given that the document talks
> about potentially quite large ID namespace (32bits), you could actually
> identify things at a finer granularity than hosts.

You can also use ARP to find out the manufacturer of a NIC. We aren't 
deprecating ARP because of privacy concerns.

You can also have existing NATs and CGNs expose host privacy in the 
source IP and port number selected, as well as in other fields that 
could easily be modified for their nefarious uses (e.g., ISNs).

I don't see this new political statement as requiring WGs to scrub their 
protocols for all *potential* or *perceived* issues with privacy.

>> Second, this mechanism can easily be used to obfuscate the identification hosts, to re-group them in alternate ways.
>
> How and by whom?
> Even if hosts end up setting it, it'll get stripped
> out or rewritten by an ISP middlebox that can set it based on subscriber
> information and current connection context.

First, that depends on whether they're protected by the endpoint or not 
- e.g., IPsec or TCP-AO could allow the source endpoint to set this 
information usefully, and prevent a middlebox from obfuscating it.

>> Finally, I sincerely hope this draft's process doesn't become the
>> poster child of the process fears raised by the IESG Chair in approving
>> perpass and recommending it as BCP.
>
> So I actually hope it does.

If so, I predict it will be the death of the IETF as a technical body. I 
can't speak for others, but I don't participate here to express my 
political positions.

> But that's not the case here. This is not a WG document, and using
> theBCP - that we have IETF consensus on

Every rough consensus, and an IESG Chair that set the document type 
*after* last call.

> - as an argument to not *start*
> problematic work is entirely reasonable. If not, what *can* we use that
> consensus BCP for? Feeling good about ourselves without actually doing
> anything?

In this case, it can and should drive the use cases discussed and 
security considerations presented.

However, unless this mechanism provides more opportunity for tracking 
than existing protocols and mechanisms, I don't think it should be used 
to block it.

Joe


From nobody Thu May 15 03:00:48 2014
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Hi Alexander

thanks for the review, please see my comments inline=85


On 13 May 2014, at 11:57, Zimmermann, Alexander =
<Alexander.Zimmermann@netapp.com> wrote:

> Hi folks,
>=20
> first of all thank you for writing this draft. The draft is clearly =
written: the
> problem is well described, the doc is self-contained (it=92s not =
necessary to read
> dozens of other RFCs to understand the problem), and the authors state =
why the
> doc is experimental and what further experiments are needed for being =
a STD.
> However, on the technical side I see some open points. In detail:
>=20
> * Sec 1, 4. para: =84a considerable number of flows have such =
properties=93
>  Can you backup this with some numbers? This is exactly the point I =
raised at the
>  (last?) IETF. I would like to see some data on how severe the problem =
is we would
>  like to fix.
>=20

For short flows (e.g. web) there are a number of references in the =
draft. These are
the primary target for the mechanism. We=92ll elaborate some more on the =
importance
for other types of traffic.=20


> * Sec 1.1: please change this subsection to a section (1.1 =3D> 2) and =
also
>  introduce your new state variable rrthresh here
>=20

I don=92t get why the section should be renumbered. The =93terminology=94 =
is often the
last subsection of the introduction of most drafts and RFCs. =
Furthermore, it seems=20
slightly overkill to introduce the variable there. RFCs that manages =
many more=20
variables, e.g. RFC6298 does not explicitly introduce any variable in =
this section.

> * Sec 1, 5. para: =84Spurious timeouts typically degrade the =
performance of flows
>  with multiple bursts of data, as a burst following a spurious timeout =
might
>  not fit within the reduced congestion window (cwnd)=93
>=20
>  This is (only) true with respect to your algo, not in general. The =
general
>  problem of a spurious timeout is the cwnd reduction, the go-back-N
>  retransmissions, =85 See RFC 3522. After reading section 4.2 of your =
draft I
>  understand what you want to see here. Please rephrase the section and =
maybe
>  add the spurious RTO RFC as reference.
>=20
What is meant is the actual cwnd reduction. We should clarify this.


> * Sec 3: Suppose FlightSize is 2 and you have exactly one segment to =
send,
>  your algo doesn=92t trigger since step 2.b isn=92t true. Bug? I would =
say yes.
>=20
Good catch! The question is if it=92s worth fixing since the algorithm =
will
become more complex and the situation you mention is really a corner =
case that
requires either (i) the cwnd to be exactly 3 segments large; or (ii) =
having a
packet written to the socket just between previous data transmission and =
the
arrival of the acknowledgment.

Furthermore, this mechanism is only an optimization to the standard =
timer. So
if it doesn=92t work in this particular scenario it won=92t break =
anything. It will
just not be triggered.=20

> * Sec 3: Why the condition 2.b is different from the early =
retransmission
>  condition 2.b or 3.b? Is there any specific reason why we exclude the
>  advertised receive window part from the condition?
>=20
Because the advertised window can be small in situations where it=92s =
not
preferable to use RTO restart. For instance, in the middle of a transfer =
where
it=92s better to wait for fast/early retransmit to kick in.

> * Sec 3: IMO the algo/doc is too much Linux driven. I would like to =
see a
>  segment-based *and* byte-based version of the algo, like RFC 5827.
>=20
Yes, this comment was given by several people at the last IETF. We will =
update
the draft to address this.

> * General: IMO I would be a little bit easier to read the doc if you =
give the
>  algo a proper name. By reading =84RTO restart=93 I had sometimes =
trouble to
>  know if mean your algo or the =84action=93 of restarting the RTO.
>=20
Agreed, this will be fixed.



Thank you,
Per Hurtig

>=20
> Alex
>=20
>=20
>=20



From nobody Thu May 15 03:58:37 2014
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Date: Thu, 15 May 2014 11:58:06 +0100
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Subject: Re: [tcpm] Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Hi

We believe that low ssthresh resetting is a real problem especially in
data-limited
conditions when few packets are in flight.

We've also tried to see what happens when we drop a SYN in Linux.
In this case TCP follows the standard and drop the ssthresh to 2 MSS being
the SYN the only packet in flight.

This is a real harm for a starting connection if the SYN was dropped for
non-congestion related reasons (eg due to unsupported TCP options).


Raffaello


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Subject: Re: [tcpm] Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Hi Ingemar,

I'm not sure what the rational was for dropping ssthresh to 2*MSS, but it s=
eems to me that there are too many non-congestion-related events that can c=
ause this extreme setting of ssthresh. Once ssthresh gets set so low, the r=
eal problem we have seen is recovery on long-lived connections or where an =
ssthresh-host-cache is in use. In these cases, the current congestion RFCs =
don't provide a specific mechanism for ssthresh to recover to a level that =
represents the actual congestion level. So what we have seen are connection=
s between a particular client and server go to very low throughput and not =
recover for a very long time. In fact, the traffic between the client and s=
erver may be such that it can never recover until the connection is dropped=
 (in the case of a long-lived connections) or the cached ssthresh is reset.

To allow our customers to avoid this problem, we have provided two mechanis=
ms in our software:


1.       They can disable ssthresh-host-cache so that new connections will =
always restart ssthresh.

2.       Since RFC 5681 paragraph 4.1 is silent on what to do with ssthresh=
 when restarting idle connections, we assume ssthresh is no longer valid af=
ter a sufficiently long idle period. Given the next transfer is going to pe=
rform a slow-start that will (essentially) search for the appropriate cwnd =
level and restart the ack clock,  we also restart ssthresh so that the appr=
opriate cwnd level can be found.

These mechanisms seem to work quite well and we haven't seen any cases wher=
e they have caused other problems, but I would be much happier if the conge=
stion RFCs were not so silent on what to do with ssthresh to recover from c=
ases where ssthresh gets set to an inappropriately low level.

Thanks,
Gary



From: Ingemar Johansson S [mailto:ingemar.s.johansson@ericsson.com]
Sent: Tuesday, May 13, 2014 1:15 AM
To: tcpm@ietf.org
Cc: Karen E. E. Nielsen; gorry@erg.abdn.ac.uk; McAlpine, Gary
Subject: Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-=
03.txt)

Hi

Karen pointed out this thread to me
http://www.ietf.org/mail-archive/web/tcpm/current/msg08315.html
I started to look closer at this issue quite recently, the problem I have i=
s that SSthresh can drop to very low values after only a few lost packets. =
I have seen this odd effect earlier but have not bothered with it.


The experiment is to run a large FTP transfer over a 1Mbps bottleneck with =
min RTT =3D 40ms. No AQM or tail drop queue enabled i.e a buffer bloated sc=
enario.

TCP NewReno. After 10s I "pull the plug" for 100ms (100% packet drop), this=
 leads to 5 lost segments. at T =3D10.1s packets are forwarded as usual.

What I have seen is that a retransmission timeout is immediately followed b=
y a loss event, the effect of which is that SSThresh goes down to 2 MSS.  I=
t seems to me that the RTO timer value is too low, I have not understood th=
e effect completely though. Could the RTO timer be the culprit or is there =
some other effect ?.

I am running these experiments in a proprietary LTE system simulator, we tr=
y to keep it up to date to match the Linux TCP stack reasonably well, it ca=
nnot be ruled out however that our implementation miss some important featu=
re.

/Ingemar

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

"Those are my principles, and if you don't like them...
well, I have others."  Groucho Marx<http://www.brainyquote.com/quotes/autho=
rs/g/groucho_marx.html>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D



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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Ingemar,<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I&#8217;m not sure wha=
t the rational was for dropping ssthresh to 2*MSS, but it seems to me that =
there are too many non-congestion-related events that can cause this extrem=
e setting of ssthresh. Once ssthresh gets
 set so low, the real problem we have seen is recovery on long-lived connec=
tions or where an ssthresh-host-cache is in use. In these cases, the curren=
t congestion RFCs don&#8217;t provide a specific mechanism for ssthresh to =
recover to a level that represents the
 actual congestion level. So what we have seen are connections between a pa=
rticular client and server go to very low throughput and not recover for a =
very long time. In fact, the traffic between the client and server may be s=
uch that it can never recover until
 the connection is dropped (in the case of a long-lived connections) or the=
 cached ssthresh is reset.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">To allow our customers=
 to avoid this problem, we have provided two mechanisms in our software:<o:=
p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"color:#1F497D"><span style=3D"m=
so-list:Ignore">1.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"color:#1F497D">They can disab=
le ssthresh-host-cache so that new connections will always restart ssthresh=
.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"color:#1F497D"><span style=3D"m=
so-list:Ignore">2.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"color:#1F497D">Since RFC 5681=
 paragraph 4.1 is silent on what to do with ssthresh when restarting idle c=
onnections, we assume ssthresh is no longer valid after a sufficiently long=
 idle period. Given the next transfer
 is going to perform a slow-start that will (essentially) search for the ap=
propriate cwnd level and restart the ack clock, &nbsp;we also restart ssthr=
esh so that the appropriate cwnd level can be found.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">These mechanisms seem =
to work quite well and we haven&#8217;t seen any cases where they have caus=
ed other problems, but I would be much happier if the congestion RFCs were =
not so silent on what to do with ssthresh
 to recover from cases where ssthresh gets set to an inappropriately low le=
vel.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Gary<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Ingemar =
Johansson S [mailto:ingemar.s.johansson@ericsson.com]
<br>
<b>Sent:</b> Tuesday, May 13, 2014 1:15 AM<br>
<b>To:</b> tcpm@ietf.org<br>
<b>Cc:</b> Karen E. E. Nielsen; gorry@erg.abdn.ac.uk; McAlpine, Gary<br>
<b>Subject:</b> Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-=
newcwv-03.txt)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Karen pointed out this thread to me <o:p></o:p></p>
<p class=3D"MsoNormal"><a href=3D"http://www.ietf.org/mail-archive/web/tcpm=
/current/msg08315.html">http://www.ietf.org/mail-archive/web/tcpm/current/m=
sg08315.html</a>
<o:p></o:p></p>
<p class=3D"MsoNormal">I started to look closer at this issue quite recentl=
y, the problem I have is that SSthresh can drop to very low values after on=
ly a few lost packets. I have seen this odd effect earlier but have not bot=
hered with it.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">The experiment is to run a large FTP transfer ove=
r a 1Mbps bottleneck with min RTT =3D 40ms. No AQM or tail drop queue enabl=
ed i.e a buffer bloated scenario.<o:p></o:p></p>
<p class=3D"MsoPlainText">TCP NewReno. After 10s I &#8220;pull the plug&#82=
21; for 100ms (100% packet drop), this leads to 5 lost segments. at T =3D10=
.1s packets are forwarded as usual.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">What I have seen is that a retransmission timeout is=
 immediately followed by a loss event, the effect of which is that SSThresh=
 goes down to 2 MSS. &nbsp;It seems to me that the RTO timer value is too l=
ow, I have not understood the effect completely
 though. Could the RTO timer be the culprit or is there some other effect ?=
.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I am running these experiments in a proprietary LTE =
system simulator, we try to keep it up to date to match the Linux TCP stack=
 reasonably well, it cannot be ruled out however that our implementation mi=
ss some important feature.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">/Ingemar<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Joh=
ansson&nbsp; M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Senior Rese=
archer<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ericsson AB=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Wireless Ac=
cess Networks<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Labratorieg=
r=E4nd 11<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">971 28, Lul=
e=E5, Sweden<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Phone &#43;=
46-1071 43042<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">SMS/MMS &#4=
3;46-73 078 3289<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"=
><a href=3D"mailto:ingemar.s.johansson@ericsson.com"><span lang=3D"EN-US" s=
tyle=3D"color:windowtext">ingemar.s.johansson@ericsson.com</span></a></span=
><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-s=
erif&quot;"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"=
><a href=3D"www.ericsson.com"><span lang=3D"EN-US" style=3D"color:windowtex=
t">www.ericsson.com</span></a></span><span style=3D"font-size:10.0pt;font-f=
amily:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:black"=
>&#8220;Those are my principles, and if you don't like them...
<br>
well, I have others.&#8221;&nbsp; </span><span style=3D"font-size:10.0pt;fo=
nt-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><a href=3D"http://www.b=
rainyquote.com/quotes/authors/g/groucho_marx.html"><span style=3D"color:win=
dowtext;text-decoration:none">Groucho Marx</span></a><br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:7.5pt;line-height:15.0pt"><sp=
an style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif=
&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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To: "McAlpine, Gary" <gary.mcalpine@bluecoat.com>,  Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: Re: [tcpm] Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Yes - this seems like an oversight in the way ssthresh was conceived. 
The value is intended to capture congestion history, and hence it's nice 
to cache this for new flows to  prevent them overshooting the bottleneck 
capacity. I can think of 3 things where current ssthresh methods give 
unwarranted poor performance:

  - path changes, presumably uncommon.
  - historic information (an event hours ago has little bearing on a 
long-lived connection and is a real impediment when the RTT is large).
  - non-congestion loss (especially when FS was small)

Gorry

On 15/05/2014 18:21, McAlpine, Gary wrote:
> Hi Ingemar,
>
> I'm not sure what the rational was for dropping ssthresh to 2*MSS, but it seems to me that there are too many non-congestion-related events that can cause this extreme setting of ssthresh. Once ssthresh gets set so low, the real problem we have seen is recovery on long-lived connections or where an ssthresh-host-cache is in use. In these cases, the current congestion RFCs don't provide a specific mechanism for ssthresh to recover to a level that represents the actual congestion level. So what we have seen are connections between a particular client and server go to very low throughput and not recover for a very long time. In fact, the traffic between the client and server may be such that it can never recover until the connection is dropped (in the case of a long-lived connections) or the cached ssthresh is reset.
>
> To allow our customers to avoid this problem, we have provided two mechanisms in our software:
>
>
> 1.       They can disable ssthresh-host-cache so that new connections will always restart ssthresh.
>
> 2.       Since RFC 5681 paragraph 4.1 is silent on what to do with ssthresh when restarting idle connections, we assume ssthresh is no longer valid after a sufficiently long idle period. Given the next transfer is going to perform a slow-start that will (essentially) search for the appropriate cwnd level and restart the ack clock,  we also restart ssthresh so that the appropriate cwnd level can be found.
>
> These mechanisms seem to work quite well and we haven't seen any cases where they have caused other problems, but I would be much happier if the congestion RFCs were not so silent on what to do with ssthresh to recover from cases where ssthresh gets set to an inappropriately low level.
>
> Thanks,
> Gary
>
>
>
> From: Ingemar Johansson S [mailto:ingemar.s.johansson@ericsson.com]
> Sent: Tuesday, May 13, 2014 1:15 AM
> To: tcpm@ietf.org
> Cc: Karen E. E. Nielsen; gorry@erg.abdn.ac.uk; McAlpine, Gary
> Subject: Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
>
> Hi
>
> Karen pointed out this thread to me
> http://www.ietf.org/mail-archive/web/tcpm/current/msg08315.html
> I started to look closer at this issue quite recently, the problem I have is that SSthresh can drop to very low values after only a few lost packets. I have seen this odd effect earlier but have not bothered with it.
>
>
> The experiment is to run a large FTP transfer over a 1Mbps bottleneck with min RTT = 40ms. No AQM or tail drop queue enabled i.e a buffer bloated scenario.
>
> TCP NewReno. After 10s I "pull the plug" for 100ms (100% packet drop), this leads to 5 lost segments. at T =10.1s packets are forwarded as usual.
>
> What I have seen is that a retransmission timeout is immediately followed by a loss event, the effect of which is that SSThresh goes down to 2 MSS.  It seems to me that the RTO timer value is too low, I have not understood the effect completely though. Could the RTO timer be the culprit or is there some other effect ?.
>
> I am running these experiments in a proprietary LTE system simulator, we try to keep it up to date to match the Linux TCP stack reasonably well, it cannot be ruled out however that our implementation miss some important feature.
>
> /Ingemar
>
> =================================
> Ingemar Johansson  M.Sc.
> Senior Researcher
>
> Ericsson AB
> Wireless Access Networks
> Labratoriegränd 11
> 971 28, Luleå, Sweden
> Phone +46-1071 43042
> SMS/MMS +46-73 078 3289
> ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
> www.ericsson.com
>
> "Those are my principles, and if you don't like them...
> well, I have others."  Groucho Marx<http://www.brainyquote.com/quotes/authors/g/groucho_marx.html>
> =================================
>
>
>


From nobody Mon May 19 10:54:04 2014
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Brandon Williams <brandon.williams@akamai.com>
Thread-Topic: [tcpm] Fwd: New Version Notification for draft-williams-exp-tcp-host-id-opt-03.txt
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Hi Brandon, all,

Thanks a lot for this update. Indeed, the update takes into account TCPM fe=
edback, and this is excellent.

However, based on the list discussion so far, it seems to the chairs that t=
his document would be a better fit to an independent stream submission. The=
 TCP option described in this document actually does not modify any TCP log=
ic. Identifying the hosts that are hidden behind a shared address/prefix sh=
aring device or application-layer proxy seems like a new service that curre=
nt TCP does not offer. TCPM is only chartered for minor TCP extensions.

Also, in its current form, it seems that this document formally does not re=
quire TCPM adoption. While TCPM documents probably (hopefully!) have more w=
eight among end-system TCP stack implementers, it is much less clear whethe=
r a TCPM document on this specific topic will indeed foster implementation =
convergence among middleboxes. We have no empirical evidence for that.

TCPM is used to adopt new working group items based on rather strong consen=
sus. According to the list discussion so far, the TCPM chairs currently und=
erstand that there is no consensus in the TCPM community to adopt this docu=
ment at this point in time.=20

Obviously, as usual in the IETF, further list discussion is welcome.

Best regards

Michael
on behalf of the TCPM chairs=20


> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Brandon Williams
> Sent: Wednesday, April 30, 2014 4:44 PM
> To: tcpm@ietf.org
> Subject: [tcpm] Fwd: New Version Notification for draft-williams-exp-
> tcp-host-id-opt-03.txt
>=20
> We have updated the draft based on feedback on the list and in London.
> In particular, we have attempted to provide a clearer explanation of
> the
> use cases we are attempting to address and why solutions at other
> layers
> in the protocol stack do not work for these use cases. Previous updates
> had already attempted to provide stronger limiting guidance for option
> usage.
>=20
> As we have indicated previously on the list, we would prefer to see WG
> adoption of this I.D. rather than working toward publishing as an
> individual submission. We think WG adoption has a few benefits, even in
> the absence of clear consensus on the question of whether this is a
> good
> idea. First and foremost, WG adoption would mean that the limiting
> guidance on usage carries more weight. And second, WG adoption would
> more strongly encourage individual implementations that are already in
> use to converge on a single option format.
>=20
> We look forward to your feedback.
>=20
> Thanks,
> --Brandon
>=20
>=20
> -------- Original Message --------
> Subject: New Version Notification for
> draft-williams-exp-tcp-host-id-opt-03.txt
> Date: Wed, 30 Apr 2014 10:22:20 -0400
> From: internet-drafts@ietf.org <internet-drafts@ietf.org>
> To: Williams, Brandon <brandon.williams@akamai.com>, Dan Wing
> <dwing@cisco.com>, Mohamed Boucadair <mohamed.boucadair@orange.com>,
> Mohamed Boucadair <mohamed.boucadair@orange.com>, Dan Wing
> <dwing@cisco.com>, Williams, Brandon <brandon.williams@akamai.com>
>=20
>=20
> A new version of I-D, draft-williams-exp-tcp-host-id-opt-03.txt
> has been successfully submitted by Brandon Williams and posted to the
> IETF repository.
>=20
> Name:		draft-williams-exp-tcp-host-id-opt
> Revision:	03
> Title:		Experimental Option for TCP Host Identification
> Document date:	2014-04-29
> Group:		Individual Submission
> Pages:		11
> URL:
> http://www.ietf.org/internet-drafts/draft-williams-exp-tcp-host-id-opt-
> 03.txt
> Status:
> https://datatracker.ietf.org/doc/draft-williams-exp-tcp-host-id-opt/
> Htmlized:
> http://tools.ietf.org/html/draft-williams-exp-tcp-host-id-opt-03
> Diff:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-williams-exp-tcp-host-id-opt-03
>=20
> Abstract:
>     Recent IETF proposals have identified benefits to more distinctly
>     identifying the hosts that are hidden behind a shared
> address/prefix
>     sharing device or application-layer proxy.  Analysis indicates that
>     the use of a TCP option for this purpose can be successfully
> applied
>     to a broad range of use cases.  This document describes a common
>     experimental TCP option format for host identification.
>=20
>=20
>=20
>=20
>=20
> 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.
>=20
> The IETF Secretariat
>=20
>=20
> --
> Brandon Williams; Senior Principal Software Engineer
> Emerging Products Engineering; Akamai Technologies Inc.
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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--089e0160c156d84bdc04f9c56ce6
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FYI.
--
Yoshi

---------- Forwarded message ----------
From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Mon, May 19, 2014 at 10:54 AM
Subject: Publication has been requested for
draft-ietf-tcpm-tcp-rfc4614bis-05
To: mls.ietf@gmail.com
Cc: tcpm-chairs@tools.ietf.org, iesg-secretary@ietf.org,
draft-ietf-tcpm-tcp-rfc4614bis@tools.ietf.org


Yoshifumi Nishida has requested publication of
draft-ietf-tcpm-tcp-rfc4614bis-05 as Informational on behalf of the TCPM
working group.

Please verify the document's state at
http://datatracker.ietf.org/doc/draft-ietf-tcpm-tcp-rfc4614bis/

--089e0160c156d84bdc04f9c56ce6
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<div dir=3D"ltr">FYI.<div>--</div><div>Yoshi<br><br><div class=3D"gmail_quo=
te">---------- Forwarded message ----------<br>From: <b class=3D"gmail_send=
ername">Yoshifumi Nishida</b> <span dir=3D"ltr">&lt;<a href=3D"mailto:nishi=
da@sfc.wide.ad.jp">nishida@sfc.wide.ad.jp</a>&gt;</span><br>
Date: Mon, May 19, 2014 at 10:54 AM<br>Subject: Publication has been reques=
ted for draft-ietf-tcpm-tcp-rfc4614bis-05<br>To: <a href=3D"mailto:mls.ietf=
@gmail.com">mls.ietf@gmail.com</a><br>Cc: <a href=3D"mailto:tcpm-chairs@too=
ls.ietf.org">tcpm-chairs@tools.ietf.org</a>, <a href=3D"mailto:iesg-secreta=
ry@ietf.org">iesg-secretary@ietf.org</a>, <a href=3D"mailto:draft-ietf-tcpm=
-tcp-rfc4614bis@tools.ietf.org">draft-ietf-tcpm-tcp-rfc4614bis@tools.ietf.o=
rg</a><br>
<br><br>Yoshifumi Nishida has requested publication of draft-ietf-tcpm-tcp-=
rfc4614bis-05 as Informational on behalf of the TCPM working group.<br>
<br>
Please verify the document&#39;s state at <a href=3D"http://datatracker.iet=
f.org/doc/draft-ietf-tcpm-tcp-rfc4614bis/" target=3D"_blank">http://datatra=
cker.ietf.org/doc/draft-ietf-tcpm-tcp-rfc4614bis/</a><br>
<br>
</div><br></div></div>

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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "McAlpine, Gary" <gary.mcalpine@bluecoat.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Date: Tue, 20 May 2014 07:02:08 +0000
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Subject: Re: [tcpm] Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-newcwv-03.txt)
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Thanks for the response sofar

I run a TCP stack that is part of a larger LTE system simulator, I know tha=
t it is not a full match of e.g  a Linux TCP stack, I am trying to dig into=
 the Linux TCP code as well as in ns2 code to see if something important is=
 missing that makes my results unreliable, or if this is infact a flaw in h=
ow=20

The setup:
1Mbps bottleneck, RTT ~35ms
Large FTP transfer (100MB)
No AQM, infinite buffer =3D bufferbloated network
Fake path change after 10s, packets are dropped for 100ms
	- Note! dropper after bottleneck

The chain of events that I see are=20
1) DUPACKs received from from T~10.15 to T~14.2s
2) Fast retransmit at T=3D10.17s re-arms RTO timer (3.60s)
   2a) CWND restriction. Only one segment transmitted at T=3D17 the remaini=
ng segments transmitted at T =3D 12.18s due to CWND restriction, but RTO ti=
mer not restarted at T=3D12.18 because of loss recovery state
3. Retransmission timeout at T=3D13.78s  DUPACK counter and SACK scoreboard=
 is reset
4. DUPACK counter=3D3 at T=3D13.81s  fast retransmit, SSThresh set to 2MSS =
!

I have a few slides that show more detail, I can email them to anyone inter=
ested.=20
A questions:=20
In my code the DUPACK counter and the SACK scoreboard is reset  at retransm=
ission timeout and this is what causes SSThresh to drop to  2MSS in this ca=
se. Is this a bug in my code or that it is too simplistic ?, or is it just =
the way it should be ?.=20

I am thankful for any pointers to a more comprehensible documentation on ho=
w e.g a Linux TCP stack behaves.

Now I believe that bufferbloat is at least partly to blame for this, but I =
am not yet convinced that it is the full story.

Regards
/Ingemar



> -----Original Message-----
> From: Gorry Fairhurst [mailto:gorry@erg.abdn.ac.uk]
> Sent: den 16 maj 2014 15:49
> To: McAlpine, Gary; Ingemar Johansson S; tcpm@ietf.org
> Cc: Karen E. E. Nielsen
> Subject: Re: Problem with Low SSThresh (was I-D Action: draft-ietf-tcpm-
> newcwv-03.txt)
>=20
> Yes - this seems like an oversight in the way ssthresh was conceived.
> The value is intended to capture congestion history, and hence it's nice =
to
> cache this for new flows to  prevent them overshooting the bottleneck
> capacity. I can think of 3 things where current ssthresh methods give
> unwarranted poor performance:
>=20
>   - path changes, presumably uncommon.
>   - historic information (an event hours ago has little bearing on a long=
-lived
> connection and is a real impediment when the RTT is large).
>   - non-congestion loss (especially when FS was small)
>=20
> Gorry
>=20
> On 15/05/2014 18:21, McAlpine, Gary wrote:
> > Hi Ingemar,
> >
> > I'm not sure what the rational was for dropping ssthresh to 2*MSS, but =
it
> seems to me that there are too many non-congestion-related events that
> can cause this extreme setting of ssthresh. Once ssthresh gets set so low=
,
> the real problem we have seen is recovery on long-lived connections or
> where an ssthresh-host-cache is in use. In these cases, the current
> congestion RFCs don't provide a specific mechanism for ssthresh to recove=
r
> to a level that represents the actual congestion level. So what we have s=
een
> are connections between a particular client and server go to very low
> throughput and not recover for a very long time. In fact, the traffic bet=
ween
> the client and server may be such that it can never recover until the
> connection is dropped (in the case of a long-lived connections) or the ca=
ched
> ssthresh is reset.
> >
> > To allow our customers to avoid this problem, we have provided two
> mechanisms in our software:
> >
> >
> > 1.       They can disable ssthresh-host-cache so that new connections w=
ill
> always restart ssthresh.
> >
> > 2.       Since RFC 5681 paragraph 4.1 is silent on what to do with ssth=
resh
> when restarting idle connections, we assume ssthresh is no longer valid a=
fter
> a sufficiently long idle period. Given the next transfer is going to perf=
orm a
> slow-start that will (essentially) search for the appropriate cwnd level =
and
> restart the ack clock,  we also restart ssthresh so that the appropriate =
cwnd
> level can be found.
> >
> > These mechanisms seem to work quite well and we haven't seen any cases
> where they have caused other problems, but I would be much happier if the
> congestion RFCs were not so silent on what to do with ssthresh to recover
> from cases where ssthresh gets set to an inappropriately low level.
> >
> > Thanks,
> > Gary
> >
> >
> >
> > From: Ingemar Johansson S [mailto:ingemar.s.johansson@ericsson.com]
> > Sent: Tuesday, May 13, 2014 1:15 AM
> > To: tcpm@ietf.org
> > Cc: Karen E. E. Nielsen; gorry@erg.abdn.ac.uk; McAlpine, Gary
> > Subject: Problem with Low SSThresh (was I-D Action:
> > draft-ietf-tcpm-newcwv-03.txt)
> >
> > Hi
> >
> > Karen pointed out this thread to me
> > http://www.ietf.org/mail-archive/web/tcpm/current/msg08315.html
> > I started to look closer at this issue quite recently, the problem I ha=
ve is
> that SSthresh can drop to very low values after only a few lost packets. =
I have
> seen this odd effect earlier but have not bothered with it.
> >
> >
> > The experiment is to run a large FTP transfer over a 1Mbps bottleneck w=
ith
> min RTT =3D 40ms. No AQM or tail drop queue enabled i.e a buffer bloated
> scenario.
> >
> > TCP NewReno. After 10s I "pull the plug" for 100ms (100% packet drop), =
this
> leads to 5 lost segments. at T =3D10.1s packets are forwarded as usual.
> >
> > What I have seen is that a retransmission timeout is immediately follow=
ed
> by a loss event, the effect of which is that SSThresh goes down to 2 MSS.=
  It
> seems to me that the RTO timer value is too low, I have not understood th=
e
> effect completely though. Could the RTO timer be the culprit or is there
> some other effect ?.
> >
> > I am running these experiments in a proprietary LTE system simulator, w=
e
> try to keep it up to date to match the Linux TCP stack reasonably well, i=
t
> cannot be ruled out however that our implementation miss some important
> feature.
> >
> > /Ingemar
> >
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > Ingemar Johansson  M.Sc.
> > Senior Researcher
> >
> > Ericsson AB
> > Wireless Access Networks
> > Labratoriegr=E4nd 11
> > 971 28, Lule=E5, Sweden
> > Phone +46-1071 43042
> > SMS/MMS +46-73 078 3289
> >
> ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.c
> > om>
> > www.ericsson.com
> >
> > "Those are my principles, and if you don't like them...
> > well, I have others."  Groucho
> >
> Marx<http://www.brainyquote.com/quotes/authors/g/groucho_marx.html
> >
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >
> >
> >


From nobody Wed May 21 00:36:18 2014
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Cc: ingemar.s.johansson@ericsson.com, martin.isaksson@ericsson.com
Subject: [tcpm] RFC3465:TCP ABC and SACK'ed data
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TCP ABC (still in the experimental track) defines a method to increase the cwnd
based on the bytes being acknowledged, instead of the number of acks that
arrive. I am assuming that SACK'ed (RFC2018) data should be included in this
calculation, but i could find no mention of this in RFC3465.
Could someone confirm this?

What i'd like to achieve with this is to solve an issue that follows after a
TCP path change where a large portion of the window have been lost. On the
new path, the data that was successfully passed to the receiver is sacked
and the rest is being retransmitted. This prevents the cwnd from growing and the
throughput is very low.

Am i correct in that a SACK-aware ABC implementation would allow the cwnd
to grow and the "hole in the window" to be closed faster?

//E


From nobody Wed May 21 01:29:07 2014
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Per Hurtig <per.hurtig@kau.se>
Thread-Topic: Review of draft-ietf-tcpm-rtorestart-02.txt
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Subject: Re: [tcpm] Review of draft-ietf-tcpm-rtorestart-02.txt
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Hi Per,

Am 15.05.2014 um 12:00 schrieb Per Hurtig <per.hurtig@kau.se>:

> Hi Alexander
>=20
> thanks for the review, please see my comments inline=85
>=20
>=20
> On 13 May 2014, at 11:57, Zimmermann, Alexander =
<Alexander.Zimmermann@netapp.com> wrote:
>=20
>> Hi folks,
>>=20
>> first of all thank you for writing this draft. The draft is clearly =
written: the
>> problem is well described, the doc is self-contained (it=92s not =
necessary to read
>> dozens of other RFCs to understand the problem), and the authors =
state why the
>> doc is experimental and what further experiments are needed for being =
a STD.
>> However, on the technical side I see some open points. In detail:
>>=20
>> * Sec 1, 4. para: =84a considerable number of flows have such =
properties=93
>> Can you backup this with some numbers? This is exactly the point I =
raised at the
>> (last?) IETF. I would like to see some data on how severe the problem =
is we would
>> like to fix.
>>=20
>=20
> For short flows (e.g. web) there are a number of references in the =
draft. These are
> the primary target for the mechanism. We=92ll elaborate some more on =
the importance
> for other types of traffic.=20

Your draft focus on two different kinds of flows: a) on short flows and =
b) on flows
w/ low transmission rates. For the former you give two references [RJ10] =
and [FDT13],
but not for the latter. You say that =84a considerable number of flows =
have such
properties=93 and my question was, if you can backup this w/ some =
numbers. Additionally,
Yuchung reports that he wasn=92t able to see much improvements while =
implementing your
scheme. Do you have some data that you can share to see how much your =
scheme helps to
improve the latency?

Also I=92m not sure if we really focus on short flows. Is it not rather =
the case that we
concentrate us on tail losses? Your scheme will not only kick in for =
short flows, it
gets activated at the end of any stream. Or do I miss something here? =
(BTW [FDT13]
reports that 77% of the RXmits are RTO-based, it doesn=92t say that 77% =
are short flows.)

>=20
>=20
>> * Sec 1.1: please change this subsection to a section (1.1 =3D> 2) =
and also
>> introduce your new state variable rrthresh here
>>=20
>=20
> I don=92t get why the section should be renumbered. The =93terminology=94=
 is often the
> last subsection of the introduction of most drafts and RFCs.

This was only a nit. In German language we have the grammar rule that =
you cannot
introduce a subsection x.1 if you don=92t have a second subsection in =
the same section.
While searching a little bit in the web, it seems that is also valid in =
English
(http://www.cs.berkeley.edu/~pattrsn/talks/writingtips.html). Anyway, it =
was only
a nit :-)

> Furthermore, it seems=20
> slightly overkill to introduce the variable there. RFCs that manages =
many more=20
> variables, e.g. RFC6298 does not explicitly introduce any variable in =
this section.

and RFC 5681 or RFC 6675 are counterexamples :-) Anyway, the question is =
more whether
or not we need an additional state variable in the TCB. If you think =
that they are cases
possible where =82rrthresh=91 should not be initialized with =82dupthresh =
+ 1', then we should
introduce the state variable and explain why we need it. Otherwise, I =
recommend do
use =82dupthresh + 1=92 instead of 'rrthresh=91

>=20
>> * Sec 1, 5. para: =84Spurious timeouts typically degrade the =
performance of flows
>> with multiple bursts of data, as a burst following a spurious timeout =
might
>> not fit within the reduced congestion window (cwnd)=93
>>=20
>> This is (only) true with respect to your algo, not in general. The =
general
>> problem of a spurious timeout is the cwnd reduction, the go-back-N
>> retransmissions, =85 See RFC 3522. After reading section 4.2 of your =
draft I
>> understand what you want to see here. Please rephrase the section and =
maybe
>> add the spurious RTO RFC as reference.
>>=20
> What is meant is the actual cwnd reduction. We should clarify this.
>=20
>=20
>> * Sec 3: Suppose FlightSize is 2 and you have exactly one segment to =
send,
>> your algo doesn=92t trigger since step 2.b isn=92t true. Bug? I would =
say yes.
>>=20
> Good catch! The question is if it=92s worth fixing since the algorithm =
will
> become more complex

Not really. You can 1) restart your RTO (as usual), 2) transmit new =
data,
3) re-arm your RTO with RTO - T_earliest if FlightSize < 4. In terms of =
=84re-arming=93
we did more or less the same in RFC6069=85

> and the situation you mention is really a corner case that
> requires either (i) the cwnd to be exactly 3 segments large;

less than 4, no?

> or (ii) having a
> packet written to the socket just between previous data transmission =
and the
> arrival of the acknowledgment.

Yes

>=20
> Furthermore, this mechanism is only an optimization to the standard =
timer. So
> if it doesn=92t work in this particular scenario it won=92t break =
anything.

Sure, but if we exclude some traffic pattern which we can on the other =
hand
easily include by a small algo change, we should do that. Nevertheless, =
you
are right if we only speak here about rare corner cases (and I don=92t =
know
the answer) we can ignore them.

> It will
> just not be triggered.=20
>=20
>> * Sec 3: Why the condition 2.b is different from the early =
retransmission
>> condition 2.b or 3.b? Is there any specific reason why we exclude the
>> advertised receive window part from the condition?
>>=20
> Because the advertised window can be small in situations where it=92s =
not
> preferable to use RTO restart. For instance, in the middle of a =
transfer where
> it=92s better to wait for fast/early retransmit to kick in.

Ah, OK I see. Could you please explain this in the draft? Is it valid to =
say
that RTO restart try to cover all cases where early retransmit does=92t =
work?
I have the feeling that I not fully understand which cases should be =
covered by
which algo.

>=20
>> * Sec 3: IMO the algo/doc is too much Linux driven. I would like to =
see a
>> segment-based *and* byte-based version of the algo, like RFC 5827.
>>=20
> Yes, this comment was given by several people at the last IETF. We =
will update
> the draft to address this.
>=20
>> * General: IMO I would be a little bit easier to read the doc if you =
give the
>> algo a proper name. By reading =84RTO restart=93 I had sometimes =
trouble to
>> know if mean your algo or the =84action=93 of restarting the RTO.
>>=20
> Agreed, this will be fixed.
>=20
>=20
>=20
> Thank you,
> Per Hurtig
>=20
>>=20
>> Alex
>>=20
>>=20
>>=20
>=20
>=20


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From nobody Wed May 21 01:57:14 2014
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Erik Hugne <erik.hugne@ericsson.com>
Thread-Topic: [tcpm] RFC3465:TCP ABC and SACK'ed data
Thread-Index: AQHPdMddddHPwgB//UW60ClLS35c7ZtLMRYA
Date: Wed, 21 May 2014 08:57:09 +0000
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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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Hi,

Am 21.05.2014 um 09:36 schrieb Erik Hugne <erik.hugne@ericsson.com>:

> TCP ABC (still in the experimental track) defines a method to increase =
the cwnd
> based on the bytes being acknowledged, instead of the number of acks =
that
> arrive. I am assuming that SACK'ed (RFC2018) data should be included =
in this
> calculation, but i could find no mention of this in RFC3465.
> Could someone confirm this?

No, SACKed data is not included. RFC 5681 says that you increment the =
CWND for
each ACK received that *cumulatively* acknowledges new data. The same =
applies
for ABC.

SACKed data =84counts=93 as DUPACKs. C&P from RFC 6675: =84For the =
purposes of this
specification, we define a =84duplicate acknowledgment=93 as a segment =
that arrives
carrying a SACK block that identifies previously unacknowledged and =
un-SACKed
octets=85 =84

>=20
> What i'd like to achieve with this is to solve an issue that follows =
after a
> TCP path change where a large portion of the window have been lost. On =
the
> new path, the data that was successfully passed to the receiver is =
sacked
> and the rest is being retransmitted. This prevents the cwnd from =
growing and the
> throughput is very low.

Why? For each received SACK block you will later get a cumulative ACK =
(as soon as
the retransmission arrives) which includes the SACK block and which =
increases the
CWND. In other words: as soon as a full ACK arrives (an ACK which covers =
the
RecoveryPoint), the CWND will increased on the basis of all newly =
cumulatively
acknowledged data (if ABC is on)

Alex


>=20
> Am i correct in that a SACK-aware ABC implementation would allow the =
cwnd
> to grow and the "hole in the window" to be closed faster?
>=20
> //E
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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> TCP ABC (still in the experimental track) defines a method to increase
> the cwnd based on the bytes being acknowledged, instead of the number
> of acks that arrive. I am assuming that SACK'ed (RFC2018) data should
> be included in this calculation, but i could find no mention of this
> in RFC3465.  Could someone confirm this?

ABC does not include SACK information in its calculations and it cannot
include SACK information.  ABC applies to slow start---which ends when
loss occurs.  And, (essentially) SACKs only arrive during loss recovery.
So, there is no need for ABC to deal with SACKs.

allman




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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
> 
> > TCP ABC (still in the experimental track) defines a method to increase
> > the cwnd based on the bytes being acknowledged, instead of the number
> > of acks that arrive. I am assuming that SACK'ed (RFC2018) data should
> > be included in this calculation, but i could find no mention of this
> > in RFC3465.  Could someone confirm this?
> 
> ABC does not include SACK information in its calculations and it cannot
> include SACK information.  ABC applies to slow start---which ends when
> loss occurs.  And, (essentially) SACKs only arrive during loss recovery.
> So, there is no need for ABC to deal with SACKs.
>

Thanks for taking the time to explain this.

The problem (this is related to Ingemars posts around "Problem with low
SSthresh") is that following the path change, there is a large train of
segments being lost (say 150k out of a 500k window). Data at the end of the 
window did make it through and is SACK'ed. Upon receiving these SACK's the
sending side enters loss recovery. It will take a very long time before
recovering the 100k lost segments.

An aggressive recovery algorithm would mitigate the effects of this, but i was
hoping for a more elegant solution.

//E


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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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> The problem (this is related to Ingemars posts around "Problem with
> low SSthresh") is that following the path change, there is a large
> train of segments being lost (say 150k out of a 500k window). Data at
> the end of the window did make it through and is SACK'ed. Upon
> receiving these SACK's the sending side enters loss recovery. It will
> take a very long time before recovering the 100k lost segments.
> 
> An aggressive recovery algorithm would mitigate the effects of this,
> but i was hoping for a more elegant solution.

You may be able to define some solution to this problem, but out of the
box TCP is just going to see a loss event and going to use whatever
usual loss recovery procedure it uses in this case.  ABC is just a riff
on slow start.  And, not slow start that follows an RTO because the
cumulative ACKs are not to be believed in that case (plus, delayed ACKs
should not be in play anyway, making byte counting less needed).

allman





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From nobody Wed May 21 23:48:02 2014
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Erik Hugne <erik.hugne@ericsson.com>, Mark Allman <mallman@icir.org>
Thread-Topic: [tcpm] RFC3465:TCP ABC and SACK'ed data
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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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Are you perhaps observing an effect of the ACK clock loss, when you refer t=
o a huge number of consecutive lost segments?

Standard SACK does suffer from an issue that during loss recovery, there ma=
y be a gap (SACKs don't clock out new/retransmitted packets while flightsiz=
e > cwnd (which is now 1/2); only during the 2nd half, SACK starts to clock=
 out packets, but at the same rate as before (which may not that bright an =
idea if a path change with differnet characteristics is now there).

RateHalving (Linux has some kind of variant of this) / Proportional Rate Re=
duction RFC6937 may be the thing that you are after?


ABC really only comes into play during slowstart, to make connections which=
 use delayed ACKs not perform much less aggressive than sessions where the =
receiver features QuickAck (Linux non-standard).

Best regards,


Richard Scheffenegger


> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
> Sent: Mittwoch, 21. Mai 2014 14:55
> To: Mark Allman
> Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
> tcpm@ietf.org
> Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
>=20
> On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
> >
> > > TCP ABC (still in the experimental track) defines a method to
> > > increase the cwnd based on the bytes being acknowledged, instead of
> > > the number of acks that arrive. I am assuming that SACK'ed (RFC2018)
> > > data should be included in this calculation, but i could find no
> > > mention of this in RFC3465.  Could someone confirm this?
> >
> > ABC does not include SACK information in its calculations and it
> > cannot include SACK information.  ABC applies to slow start---which
> > ends when loss occurs.  And, (essentially) SACKs only arrive during los=
s
> recovery.
> > So, there is no need for ABC to deal with SACKs.
> >
>=20
> Thanks for taking the time to explain this.
>=20
> The problem (this is related to Ingemars posts around "Problem with low
> SSthresh") is that following the path change, there is a large train of
> segments being lost (say 150k out of a 500k window). Data at the end of
> the window did make it through and is SACK'ed. Upon receiving these SACK'=
s
> the sending side enters loss recovery. It will take a very long time
> before recovering the 100k lost segments.
>=20
> An aggressive recovery algorithm would mitigate the effects of this, but =
i
> was hoping for a more elegant solution.
>=20
> //E
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Joe,

Returning to the question of adoption, we have to=20
address the question of why previous attempts to=20
do this have failed. I don't believe it is as=20
simple as that they tried to include options on=20
SYNs, so all we have to do is avoid the SYN problem.

1) There is obviously the re-segmentation=20
problem, which Olivier/Costin have usefully=20
highlighted, and I agree an optional checksum would at least detect this.

2) However, I think the main problem is that many=20
important cases will need as large or larger TCP=20
option space on the SYN as on non-SYNs.

The option space pressure for all the following=20
(except SACK) is at least as critical for the SYN as for non-SYN segments:
* SACK          (SYN << non-SYN)
* MPTCP (SYN > non-SYN - typically)
* Timestamp     (SYN =3D non-SYN)
* Window scale  (SYN > non-SYN)
* TCP-AO        (SYN =3D non-SYN)
* TFO init      (SYN << non-SYN - but no use without TFO resume as well)
* TFO resume    (SYN >> non-SYN)

Given the above list, if bigger TCP options are=20
not available for SYNs, is a critical mass really=20
going to be persuaded that it is worth the effort=20
of implementing, deploying, debugging,=20
supporting, etc? And we need a critical mass,=20
because until EDO is deployed at both ends it=20
does nothing, so early implementers have to work on faith.

Admittedly, EDO is partly trying to make space=20
for future options and partly trying to solve a=20
problem we already have with existing options.=20
So, I admit that the relative size of existing=20
options is not the whole story. However, even new=20
options have to fit with the existing ones.

3) The EDO draft implies that it is provably=20
impossible to increase the option space on a SYN.=20
A couple of ways have been proposed to solve this problem:
* LOIC <draft-yourtchenko-tcp-loic-00> that sends=20
two parallel SYNs; a regular one and one with a=20
longer TCP option space AND a newly defined=20
checksum calculation, so that it will be discarded by legacy TCPs.
* An out-of-band control channel, e.g. <draft-paasch-mptcp-control-stream>

Much earlier in this thread, you dismissed the latter, wrongly I believe:

> >At 16:00 25/04/2014, Zimmermann, Alexander wrote:
> >> * Sec 4:
> >>    - Should we mention =84draft-paasch-mptcp-control-stream=93 here?
> >
> > I don't think so, any more than mentioning=20
> FTP's control channel. In both cases, a=20
> separate data channel is used for control info.=20
> The MPTCP approach isn't applicable to=20
> individual TCP connections - it only works in=20
> MPTCP because the group of connections is co-associated.

This seems to miss the point that there could be=20
a whole class of solutions where we create an=20
associated connection, precisely in order to add=20
a control channel of unlimited size to one (or=20
more) data channels. This brings its own=20
problems, not least it loses the intrinsic=20
security binding when control and data are in the=20
same segment. So, I wouldn't separate off a=20
control channel if we were starting from scratch.=20
But it's probably the most promising approach,=20
given we have to add a carbuncle to a wart.

In fact there are some similarities between=20
parallel SYNs and parallel channels.

4) Finally, the EDO draft cites=20
<draft-ananth-tcpm-tcpoptext-00> as if it is just=20
another solution. It's not. It's actually a very=20
useful survey of all the previous attempts to=20
solve this problem, including a useful=20
enumeration of the problems that have to be surmounted.

The arguments on this thread show that we don't=20
agree on the problem space. So, I suggest we=20
adopt Anatha's draft, and as we develop it, we=20
agree on the problem we are trying to solve=20
first. Boring, but apparently necessary.


You could precis this whole email as "Necessity=20
and impossibility are the mothers of invention".

Regards


Bob


________________________________________________________________
Bob Briscoe,                                                  BT=20


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Bob Briscoe <bob.briscoe@bt.com> wrote:
> 
> Returning to the question of adoption, we have to 
> address the question of why previous attempts to 
> do this have failed. I don't believe it is as 
> simple as that they tried to include options on 
> SYNs, so all we have to do is avoid the SYN problem.

   Of course, it's not that simple; but we've so far failed to convince
folks that the schemes for expanding option-space for SYNs are safe.

> 1) There is obviously the re-segmentation 
> problem, which Olivier/Costin have usefully 
> highlighted, and I agree an optional checksum would at least detect this.

   Yes.

> 2) However, I think the main problem is that many 
> important cases will need as large or larger TCP 
> option space on the SYN as on non-SYNs.

   No.

   We have not yet passed the point where the needed initial-negotiate
can't fit in an un-expanded SYN.

   We will, of course, if we continue to do nothing. :^(

> The option space pressure for all the following 
> (except SACK) is at least as critical for the SYN as for non-SYN segments:
> * SACK          (SYN << non-SYN)
> * MPTCP (SYN > non-SYN - typically)
> * Timestamp     (SYN = non-SYN)
> * Window scale  (SYN > non-SYN)
> * TCP-AO        (SYN = non-SYN)
> * TFO init      (SYN << non-SYN - but no use without TFO resume as well)
> * TFO resume    (SYN >> non-SYN)

   Of those, certainly MPTCP can adapt to a really-small SYN.

   The Timestamp/Window-Scale is scary, but not actually fatal.

   (We may, of course, need to eventually design a way to push the
initial-negotiate issue beyond the _initial_ SYN into the SYN-ACK, but
that's not something we need to solve in tcpm-tcp-edo.)

> Given the above list, if bigger TCP options are 
> not available for SYNs, is a critical mass really 
> going to be persuaded that it is worth the effort 
> of implementing, deploying, debugging, 
> supporting, etc? And we need a critical mass, 
> because until EDO is deployed at both ends it 
> does nothing, so early implementers have to work on faith.

   The critical-mass will be slow to develop anyway. What will drive
the speed of adoption, IMHO, is whether it is perceived as risk-free.

> Admittedly, EDO is partly trying to make space 
> for future options and partly trying to solve a 
> problem we already have with existing options. 
> So, I admit that the relative size of existing 
> options is not the whole story. However, even new 
> options have to fit with the existing ones.

   They can't, already. :^(

> 3) The EDO draft implies that it is provably 
> impossible to increase the option space on a SYN. 

   I don't read it that way.

> A couple of ways have been proposed to solve this problem:
> * LOIC <draft-yourtchenko-tcp-loic-00> that sends 
> two parallel SYNs; a regular one and one with a 
> longer TCP option space AND a newly defined 
> checksum calculation, so that it will be discarded by legacy TCPs.

   That's too big of a change to ask folks to believe it safe.

> * An out-of-band control channel, e.g. <draft-paasch-mptcp-control-stream>

   That's workable, but really doesn't belong as part of an option-space
expansion update to TCP.

> Much earlier in this thread, you dismissed the latter, wrongly I believe:

   I don't agree; but that question is not germane.

> This seems to miss the point that there could be 
> a whole class of solutions where we create an 
> associated connection, precisely in order to add 
> a control channel of unlimited size to one (or 
> more) data channels. This brings its own 
> problems, not least it loses the intrinsic 
> security binding when control and data are in the 
> same segment. So, I wouldn't separate off a 
> control channel if we were starting from scratch. 
> But it's probably the most promising approach, 
> given we have to add a carbuncle to a wart.

   Yes.

> 4) Finally, the EDO draft cites 
> <draft-ananth-tcpm-tcpoptext-00> as if it is just 
> another solution. It's not. It's actually a very 
> useful survey of all the previous attempts to 
> solve this problem, including a useful 
> enumeration of the problems that have to be surmounted.

   How would _you_ like it cited?

> The arguments on this thread show that we don't 
> agree on the problem space. So, I suggest we 
> adopt Anatha's draft, and as we develop it, we 
> agree on the problem we are trying to solve 
> first. Boring, but apparently necessary.

   I don't believe we'll be taken seriously trying to resurrect a
two-year-old draft.

   IMHO, we need to concentrate on a smaller piece of the problem.

   YMMV, of course...

--
John Leslie <john@jlc.net>


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Hi, Bob,

On 5/22/2014 10:10 AM, Bob Briscoe wrote:
> Joe,
>
> Returning to the question of adoption, we have to address the question
> of why previous attempts to do this have failed. I don't believe it is
> as simple as that they tried to include options on SYNs, so all we have
> to do is avoid the SYN problem.

There haven't been any I can recall that didn't either focus on SYN 
space extension or include it as a key component.

> 1) There is obviously the re-segmentation problem, which Olivier/Costin
> have usefully highlighted, and I agree an optional checksum would at
> least detect this.

An optional checksum would help protect TCP from such "attacks" - 
re-segmentation isn't supported, and already breaks a number of existing 
TCP features (TCP-AO, TCP MD5, ACK clocking requirements, etc.).

> 2) However, I think the main problem is that many important cases will
> need as large or larger TCP option space on the SYN as on non-SYNs.

Oh, I certainly agree with this. The point of this proposal is twofold:

	a) (primary) to put to bed the notion that 'there is a way'
	to extend SYN option space without contaminating connections to
	legacy hosts

	b) (secondary) to do the only extension possible - post-SYN.

> The option space pressure for all the following (except SACK) is at
> least as critical for the SYN as for non-SYN segments:
> * SACK          (SYN << non-SYN)
> * MPTCP (SYN > non-SYN - typically)
> * Timestamp     (SYN = non-SYN)
> * Window scale  (SYN > non-SYN)
> * TCP-AO        (SYN = non-SYN)
> * TFO init      (SYN << non-SYN - but no use without TFO resume as well)
> * TFO resume    (SYN >> non-SYN)
>
> Given the above list, if bigger TCP options are not available for SYNs,
> is a critical mass really going to be persuaded that it is worth the
> effort of implementing, deploying, debugging, supporting, etc?

Maybe never; I don't know. This isn't a "build it and they will come" 
proposal; it's intended to document the 'negative' about SYN extension 
primarily. The extension of non-SYN is intended primarily to show how it 
can trivially be done, not to argue that it's a great thing to do it.

> And we
> need a critical mass, because until EDO is deployed at both ends it does
> nothing, so early implementers have to work on faith.
>
> Admittedly, EDO is partly trying to make space for future options and
> partly trying to solve a problem we already have with existing options.
> So, I admit that the relative size of existing options is not the whole
> story. However, even new options have to fit with the existing ones.
>
> 3) The EDO draft implies that it is provably impossible to increase the
> option space on a SYN.

It states it, with the condition of not contaminating connections to 
legacy TCPs.

> A couple of ways have been proposed to solve this
> problem:
> * LOIC <draft-yourtchenko-tcp-loic-00> that sends two parallel SYNs; a
> regular one and one with a longer TCP option space AND a newly defined
> checksum calculation, so that it will be discarded by legacy TCPs.

The problem happens when the non-legacy endpoint gets the legacy SYN 
first - or only. There's no good way to do support dual-stack without 
incurring huge delay penalties waiting for a SYN with the new feature.

> * An out-of-band control channel, e.g. <draft-paasch-mptcp-control-stream>

That won't work for the first SYN to a new endpoint, which is always 
where the problem exists. Subsequent SYNs can always be affected by 
persistent state, if desired.

> Much earlier in this thread, you dismissed the latter, wrongly I believe:
>
>> >At 16:00 25/04/2014, Zimmermann, Alexander wrote:
>> >> * Sec 4:
>> >>    - Should we mention „draft-paasch-mptcp-control-stream“ here?
>> >
>> > I don't think so, any more than mentioning FTP's control channel. In
>> both cases, a separate data channel is used for control info. The
>> MPTCP approach isn't applicable to individual TCP connections - it
>> only works in MPTCP because the group of connections is co-associated.
>
> This seems to miss the point that there could be a whole class of
> solutions where we create an associated connection, precisely in order
> to add a control channel of unlimited size to one (or more) data
> channels. This brings its own problems, not least it loses the intrinsic
> security binding when control and data are in the same segment. So, I
> wouldn't separate off a control channel if we were starting from
> scratch. But it's probably the most promising approach, given we have to
> add a carbuncle to a wart.
>
> In fact there are some similarities between parallel SYNs and parallel
> channels.

The problematic case is first-contact. All other contacts - either 
within a single connection or with subsequent connections can be handled 
in any number of ways - a control channel, state that's left-behind, etc.

> 4) Finally, the EDO draft cites <draft-ananth-tcpm-tcpoptext-00> as if
> it is just another solution. It's not. It's actually a very useful
> survey of all the previous attempts to solve this problem, including a
> useful enumeration of the problems that have to be surmounted.

I'll correct that.

> The arguments on this thread show that we don't agree on the problem
> space. So, I suggest we adopt Anatha's draft, and as we develop it, we
> agree on the problem we are trying to solve first. Boring, but
> apparently necessary.

I don't agree with much of the content of that doc regarding the 
enumeration of requirements. I don't agree that this is a useful place 
to start, nor did the WG when it was first proposed AFAIR.

Joe


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On 5/22/2014 1:58 PM, Joe Touch wrote:
>> 2) However, I think the main problem is that many important cases will
>> need as large or larger TCP option space on the SYN as on non-SYNs.
> 
> Oh, I certainly agree with this. The point of this proposal is twofold:
> 
>     a) (primary) to put to bed the notion that 'there is a way'
>     to extend SYN option space without contaminating connections to
>     legacy hosts
> 
>     b) (secondary) to do the only extension possible - post-SYN.


I agree with this.  In my view, we actually have had consensus
for a long time that extending the SYN is "really hard" (requiring
some creativity), whereas extending the options space on other
packets can be done quite a bit more trivially.

So, the problem is decomposed into 2 parts; the harder part and
the easier part.  EDO is leaving the harder part alone for now;
although it may ultimately be the more useful one to solve.  I
don't see this as a problem with EDO or reason not to do it ...
any future solution for SYNs may differ substantially, but would
seem like it should not impact how options are extended on other
segments.


-- 
Wes Eddy
MTI Systems


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On 5/22/2014 11:58 AM, Wesley Eddy wrote:
> On 5/22/2014 1:58 PM, Joe Touch wrote:
>>> 2) However, I think the main problem is that many important cases will
>>> need as large or larger TCP option space on the SYN as on non-SYNs.
>>
>> Oh, I certainly agree with this. The point of this proposal is twofold:
>>
>>      a) (primary) to put to bed the notion that 'there is a way'
>>      to extend SYN option space without contaminating connections to
>>      legacy hosts
>>
>>      b) (secondary) to do the only extension possible - post-SYN.
>
>
> I agree with this.  In my view, we actually have had consensus
> for a long time that extending the SYN is "really hard" (requiring
> some creativity),

Well, AFAICT we've also known for a long time that it's basically 
impossible when needed - i.e., to new endpoints, without impacting 
legacy endpoints (i.e., making their connections to new systems slow or 
misinterpreting option info as user data).

Yes, there are a few other potential solutions that need to be discussed 
as non-viable, but IMO that's an important part of this doc.

At the very least, even if it's not provably impossible, the obvious 
solutions don't work and it's harder than it looks.

Joe


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On May 22, 2014, at 12:58 PM, Joe Touch <touch@isi.edu> wrote:

>> 2) However, I think the main problem is that many important cases =
will
>> need as large or larger TCP option space on the SYN as on non-SYNs.
>=20
> Oh, I certainly agree with this. The point of this proposal is =
twofold:
>=20
> 	a) (primary) to put to bed the notion that 'there is a way'
> 	to extend SYN option space without contaminating connections to
> 	legacy hosts

Add to that "without adding any additional packet exchanges."  That=92s =
really the issue.  This can be done, within the existing TCP processing, =
but at the cost of an additional RTT, which everyone tries to avoid.  =
The receiver could respond to the initial SYN with another SYN, which =
*can* take advantage of an extended option space because it now knows =
that the other side supports EDO.  The originator incurs an additional =
RTT before it can send data (2 vs 1 RTT), the receiver has no delay for =
when it can send data (1.5 RTT).

Besides the additional RTT for the originator, the biggest problem with =
this would be all those blasted firewalls that would drop the returning =
SYN-only responses.  The other way to do this is that when the SYN/ACK =
comes back indicating that the other side supports EDO, then the =
originator sends another SYN with the expanded option space.  That has a =
cost of a full RTT in both directions.


But the issue still remains the same: for that initial SYN packet, with =
no a priori knowledge, there is no way to extend the option space and =
maintain 100% backwards compatibility.

			-David Borman=


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Joe, and everyone else who wants to work on this,

Just because it's easier to make a chocolate teapot than a cast-iron 
one, doesn't imply that there is any need for chocolate teapots.

IOW, we will be asking the IESG to spend reviewer time on EDO, so we 
need to give some plausible indication that someone might find it 
useful and it's not just an academic exercise. The current draft 
solely gives SACK + MPTCP + TCP-AO as an example, but is that really 
something that can't be done today?

Adding more complexity to the TCP stack (with the potential for more 
vulnerabilities) is only worthwhile if there's an identifiable 
benefit, otherwise few production stacks are going to implement it anyway.



Bob



At 22:05 22/05/2014, Joe Touch wrote:


>On 5/22/2014 11:58 AM, Wesley Eddy wrote:
>>On 5/22/2014 1:58 PM, Joe Touch wrote:
>>>>2) However, I think the main problem is that many important cases will
>>>>need as large or larger TCP option space on the SYN as on non-SYNs.
>>>
>>>Oh, I certainly agree with this. The point of this proposal is twofold:
>>>
>>>      a) (primary) to put to bed the notion that 'there is a way'
>>>      to extend SYN option space without contaminating connections to
>>>      legacy hosts
>>>
>>>      b) (secondary) to do the only extension possible - post-SYN.
>>
>>
>>I agree with this.  In my view, we actually have had consensus
>>for a long time that extending the SYN is "really hard" (requiring
>>some creativity),
>
>Well, AFAICT we've also known for a long time that it's basically 
>impossible when needed - i.e., to new endpoints, without impacting 
>legacy endpoints (i.e., making their connections to new systems slow 
>or misinterpreting option info as user data).
>
>Yes, there are a few other potential solutions that need to be 
>discussed as non-viable, but IMO that's an important part of this doc.
>
>At the very least, even if it's not provably impossible, the obvious 
>solutions don't work and it's harder than it looks.
>
>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 


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David,

There is certainly no need for an additional RTT.

Let me propose a couple of concrete strawmen (concrete men?) to show 
this. I'm not claiming these are novel. I know Joe and others think 
that we've seen enough of attempts to extend the TCP options on the 
SYN, but Joe should know that when he claims something is impossible, 
he wakes sleeping dragons in people like me.

1) Parallel control channel
___________________________
Client A sends two SYNs back-to-back to an existing well-known port (e.g. 80).
* SYN D, establishes a regular data connection, with sufficient TCP 
options to be workable but they still fit within the existing 40B option limit.
* SYN C establishes another parallel connection to the same 
well-known port that looks like regular data from the outside (it 
could even be an extension to HTTP to ensure middleboxes will let it 
pass), but it talks a new app-layer 'TCP control' protocol inside.

If there is no support for the new app-layer protocol on port 80 the 
control channel just shuts down with a suitable HTTP error, while SYN 
D has opened a data connection with sufficient TCP options to be workable.
If the new app-layer TCP control protocol is supported on port 80, 
the parallel control channel (C) adds unlimited additional control 
flexibility to the data channel (D) hardly any added latency.

Establishing a similar control channel in the opposite direction 
would be fairly trivial.

There are few, if any, middlebox problems with the above approach. 
However, there are certainly other problems, but no more 
insurmountable than all the problems that have already been discussed 
with taking the 'easy' route of EDO:
* A secure binding would have to be added to bind channel C to a 
secret known only to the originator of channel D, otherwise it would 
open up data channels to spoof control channel attacks. This binding 
could be built on a TCP-AO option in channel D.
* Channel C would need some way to refer to the segments of channel D 
that was robust against re-segmentation.
* The main problem is that the two channels don't share fate; a 
control packet can be delayed relative to the point in the data 
stream at which it is attempting to exert control, possibly for a RTT 
if it is lost and has to be retransmitted. However, this is not 
insurmountable. The control protocol could include a mode to 
"synthesise shared fate", by making the data channel buffer data 
until an associated control segment had arrived. This would duplicate 
the latency impact of a loss or delay on either channel, but one can 
imagine mitigations that would consign this latency impact to corner cases.
* It's a bit of a mess, but that comes with the territory when trying 
to fix legacy protocol problems.
* The internal stack architecture seems to require a trombone back 
down into the kernel from user-space, but that is not insurmountable 
- a shim within the kernel on port 80 (for example) could redirect 
control channel data across to the "TCP control channel module" in 
the kernel, while passing non-control channel connections to user-space.

2) Build on LOIC
______________________
Long option with invalid checksum <draft-yourtchenko-tcp-loic-00>

At 18:53 22/05/2014, John Leslie wrote:
>    That's too big of a change to ask folks to believe it safe.

When I read an idea, I don't take it as set in stone and just find a 
hole and dismiss it. I see it as a potential stepping stone to a 
solution and think about how it could be done better. In fact, Andrew 
Yourtchenko said that was the intention of his write-up of LOIC.

I believe that an approach worth further thought would be a mixture 
of the control channel idea and the invalid checksum idea. I'm thinking of:
* a pure control SYN (C) sent first, then a base SYN (D) sent 
back-to-back, both to the same port.
* SYN C would contain something invalid to cause a legacy TCP stack 
or legacy app to discard it (and hopefully less probability that a 
middlebox would), e.g. a payload that is invalid for the application 
protocol on the port.
* there would be additional TCP options in the payload of SYN C to be 
added to the TCP options that arrived separately on the base SYN
* The control SYN could be bound crytographically to the base SYN (as 
already described).
* It could use the shim-like control stack arangement described earlier.

By focusing solely on extending the SYN, this would avoid the ongoing 
shared fate problems that a separate control channel suffers 
throughout the connection. There would still be shared fate problems 
with 2 SYNs (e.g. the two SYNs get re-ordered), but the protocol 
would have to be designed to be robust to that (naively, SYN D could 
include a new TCP option that told a new stack to wait a few ticks 
for a SYN C, but that would be vulnerable to meddleboxes). Not insurmountable.


Bob

At 22:12 22/05/2014, David Borman wrote:

>On May 22, 2014, at 12:58 PM, Joe Touch <touch@isi.edu> wrote:
>
> >> 2) However, I think the main problem is that many important cases will
> >> need as large or larger TCP option space on the SYN as on non-SYNs.
> >
> > Oh, I certainly agree with this. The point of this proposal is twofold:
> >
> >       a) (primary) to put to bed the notion that 'there is a way'
> >       to extend SYN option space without contaminating connections to
> >       legacy hosts
>
>Add to that "without adding any additional packet 
>exchanges."  That's really the issue.  This can be done, within the 
>existing TCP processing, but at the cost of an additional RTT, which 
>everyone tries to avoid.  The receiver could respond to the initial 
>SYN with another SYN, which *can* take advantage of an extended 
>option space because it now knows that the other side supports 
>EDO.  The originator incurs an additional RTT before it can send 
>data (2 vs 1 RTT), the receiver has no delay for when it can send 
>data (1.5 RTT).
>
>Besides the additional RTT for the originator, the biggest problem 
>with this would be all those blasted firewalls that would drop the 
>returning SYN-only responses.  The other way to do this is that when 
>the SYN/ACK comes back indicating that the other side supports EDO, 
>then the originator sends another SYN with the expanded option 
>space.  That has a cost of a full RTT in both directions.
>
>
>But the issue still remains the same: for that initial SYN packet, 
>with no a priori knowledge, there is no way to extend the option 
>space and maintain 100% backwards compatibility.
>
>                         -David Borman

________________________________________________________________
Bob Briscoe,                                                  BT 


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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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Maybe..
This is a standard Linux 3.0 kernel with sack and frto enabled.
Additional lab experiments have shown that following the path change, a stream
of lost packets occurs (almost a full window, ~400kB). The receiver sends
dupacks on the new path to trigger fast retransmit, but then the retransmission
timer on the sender side fires and all unacked data will be resent. At the RTO,
we get the expected cwnd drop. While the receiver keeps kicking the the sender
by sending dupacks, the retransmission is reasonably fast but after a few ms it
grinds down to a stop-and-go behaviour.

I failed in my efforts to find some explanation to this in the RFC's.
What i did find was in VJ's paper from '88.

-When starting or restarting after a loss, set cwnd to one packet.
-On each ack for new data, increase cwnd by one packet

If retransmitted data are not considered "new", that would explain the behavior
i guess.

Scripted ss -ti for the connection every 3 sec.
In this test it wasn't even a path change, we just downed the interface on a
router in the middle for some 300ms to simulate the packetloss (This was done at 15:06:18)

Fri May 23 15:06:01 CEST 2014
	reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
Fri May 23 15:06:04 CEST 2014
	reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
Fri May 23 15:06:07 CEST 2014
	reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
Fri May 23 15:06:10 CEST 2014
	reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
Fri May 23 15:06:14 CEST 2014
	reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 cwnd:2215 send 16.0Mbps rcv_space:14480
Fri May 23 15:06:17 CEST 2014
	reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
Fri May 23 15:06:20 CEST 2014
	reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwnd:3 ssthresh:1107 send 53.4Kbps rcv_space:14480
Fri May 23 15:06:23 CEST 2014
	reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwnd:3 ssthresh:1107 send 132.1Kbps rcv_space:14480
Fri May 23 15:06:26 CEST 2014
	reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cwnd:3 ssthresh:1107 send 165.1Kbps rcv_space:14480
Fri May 23 15:06:29 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwnd:3 ssthresh:1107 send 170.8Kbps rcv_space:14480
Fri May 23 15:06:32 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:3 ssthresh:1107 send 170.8Kbps rcv_space:14480
Fri May 23 15:06:35 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:38 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:41 CEST 2014
	reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:6 ssthresh:1107 send 340.7Kbps rcv_space:14480
Fri May 23 15:06:44 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:47 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:50 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:53 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:56 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:06:59 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
Fri May 23 15:07:02 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 send 512.3Kbps rcv_space:14480
Fri May 23 15:07:05 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 send 512.3Kbps rcv_space:14480
Fri May 23 15:07:08 CEST 2014
	reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 send 512.3Kbps rcv_space:14480
Fri May 23 15:07:11 CEST 2014
	reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwnd:960 ssthresh:1107 send 22.8Mbps rcv_space:14480
Fri May 23 15:07:14 CEST 2014
	reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwnd:1111 ssthresh:1107 send 14.9Mbps rcv_space:14480
Fri May 23 15:07:17 CEST 2014
	reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:1114 ssthresh:1107 send 14.9Mbps rcv_space:14480
Fri May 23 15:07:20 CEST 2014
	reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:1118 ssthresh:1107 send 14.9Mbps rcv_space:14480

On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:
> 
> Are you perhaps observing an effect of the ACK clock loss, when you refer to a huge number of consecutive lost segments?
> 
> Standard SACK does suffer from an issue that during loss recovery, there may be a gap (SACKs don't clock out new/retransmitted packets while flightsize > cwnd (which is now 1/2); only during the 2nd half, SACK starts to clock out packets, but at the same rate as before (which may not that bright an idea if a path change with differnet characteristics is now there).
> 
> RateHalving (Linux has some kind of variant of this) / Proportional Rate Reduction RFC6937 may be the thing that you are after?
> 
> 
> ABC really only comes into play during slowstart, to make connections which use delayed ACKs not perform much less aggressive than sessions where the receiver features QuickAck (Linux non-standard).
> 
> Best regards,
> 
> 
> Richard Scheffenegger
> 
> 
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
> > Sent: Mittwoch, 21. Mai 2014 14:55
> > To: Mark Allman
> > Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
> > tcpm@ietf.org
> > Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
> > 
> > On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
> > >
> > > > TCP ABC (still in the experimental track) defines a method to
> > > > increase the cwnd based on the bytes being acknowledged, instead of
> > > > the number of acks that arrive. I am assuming that SACK'ed (RFC2018)
> > > > data should be included in this calculation, but i could find no
> > > > mention of this in RFC3465.  Could someone confirm this?
> > >
> > > ABC does not include SACK information in its calculations and it
> > > cannot include SACK information.  ABC applies to slow start---which
> > > ends when loss occurs.  And, (essentially) SACKs only arrive during loss
> > recovery.
> > > So, there is no need for ABC to deal with SACKs.
> > >
> > 
> > Thanks for taking the time to explain this.
> > 
> > The problem (this is related to Ingemars posts around "Problem with low
> > SSthresh") is that following the path change, there is a large train of
> > segments being lost (say 150k out of a 500k window). Data at the end of
> > the window did make it through and is SACK'ed. Upon receiving these SACK's
> > the sending side enters loss recovery. It will take a very long time
> > before recovering the 100k lost segments.
> > 
> > An aggressive recovery algorithm would mitigate the effects of this, but i
> > was hoping for a more elegant solution.
> > 
> > //E
> > 
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri May 23 09:33:54 2014
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References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <1AD79820-22C1-4500-84D1-1383F264D68C@weston.borman.com> <201405231213.s4NCDa5P005525@bagheera.jungle.bt.co.uk>
To: Bob Briscoe <bob.briscoe@bt.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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Hi Bob,

On May 23, 2014, at 7:13 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:

> David,
>=20
> There is certainly no need for an additional RTT.

Oh, there will always be proposals for ways to avoid the additional RTT, =
but at what cost?  A parallel control channel is a lot of complexity if =
its only purpose is to get more option space for a single, initial SYN =
packet.


> Let me propose a couple of concrete strawmen (concrete men?) to show =
this. I'm not claiming these are novel. I know Joe and others think that =
we've seen enough of attempts to extend the TCP options on the SYN, but =
Joe should know that when he claims something is impossible, he wakes =
sleeping dragons in people like me.

What is impossible is to expand the option space in the initial SYN =
packet and maintain 100% backwards compatibility with the following =
restrictions:
   1) No additional packets are sent, just the initial SYN
   2) There is no a priori knowledge as to whether or not the remote =
host understands expanded option space.

Every proposal that maintains 100% backwards compatibility has to send =
additional data in other packets in some form.  A parallel data channel, =
a second SYN, whatever, they all require additional data in separate =
packets.

			-David
=20=


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Hi, Bob,

On 5/23/2014 3:03 AM, Bob Briscoe wrote:
> Joe, and everyone else who wants to work on this,
>
> Just because it's easier to make a chocolate teapot than a cast-iron
> one, doesn't imply that there is any need for chocolate teapots.

You don't get a cast iron teapot just because you want one either ;-)

> IOW, we will be asking the IESG to spend reviewer time on EDO, so we
> need to give some plausible indication that someone might find it useful
> and it's not just an academic exercise.

Sometimes the answer "you can't have A, but at least here's B" is more 
than an exercise; it educates the community. By not providing either 
answer, we have continued to drag this issue around the block for far 
too long -- and spent far too many cycles in this and other WGs seeking 
solutions.

 > The current draft solely gives
> SACK + MPTCP + TCP-AO as an example, but is that really something that
> can't be done today?

Current total for SYN options in widespread concurrent use (as already 
described in sec 6.4):

	2	SACK permitted
	10	timestamp
	3	window scale
	4	MSS
	------------------
	11 bytes

The current DO field is 4 bits, with a max value of 15 = 60 bytes for 
the total header, less 20 for the base TCP header which leaves 40 for 
options.

So let's see what happens when we add:

	11	widespread basic options
	16	TCP-AO
	20	MPTCP
	--------------------
	47

That's more than 40.

> Adding more complexity to the TCP stack (with the potential for more
> vulnerabilities) is only worthwhile if there's an identifiable benefit,
> otherwise few production stacks are going to implement it anyway.

There are two identifiable benefits:

	1) explain the ways we already know we can't extend the SYN
	so we stop wasting time trying them repeatedly (i.e., education)

	2) provide a solution for the other segments, so that can be
	used - e.g., for large SACK responses

	3) educate the community

Joe


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From: Joe Touch <touch@isi.edu>
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On 5/23/2014 9:33 AM, David Borman wrote:
> Hi Bob,
>
> On May 23, 2014, at 7:13 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
>
>> David,
>>
>> There is certainly no need for an additional RTT.
>
> Oh, there will always be proposals for ways to avoid the additional RTT, but at what cost?  A parallel control channel is a lot of complexity if its only purpose is to get more option space for a single, initial SYN packet.
>
>
>> Let me propose a couple of concrete strawmen (concrete men?) to show this. I'm not claiming these are novel. I know Joe and others think that we've seen enough of attempts to extend the TCP options on the SYN, but Joe should know that when he claims something is impossible, he wakes sleeping dragons in people like me.
>
> What is impossible is to expand the option space in the initial SYN packet and maintain 100% backwards compatibility with the following restrictions:
>     1) No additional packets are sent, just the initial SYN
>     2) There is no a priori knowledge as to whether or not the remote host understands expanded option space.

I would pose the conditions as follows:

	1) is robust to packet loss, reordering, and alternate paths
		any assumption of "back to back" packets isn't

	2) does not negatively impact legacy hosts
		e.g., dual-stack/multiple SYN approaches fail here when
		a legacy system is trying to connect to an updated one
		(which has to wait to see if the updated SYN will show)

	3) succeeds through basic NAT/NAPTs
		anything that uses a nonstandard checksum will fail
		to support the extended space

	4) addresses new connections to hosts not previously contacted
		it's trivial to retain state across sequences of
		connections, as per RFC 2140.

> Every proposal that maintains 100% backwards compatibility has to
> sendadditional data in other packets in some form. A parallel data channel,
 > a second SYN, whatever, they all require additional data in separate
 > packets.

+1


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On 5/23/2014 5:13 AM, Bob Briscoe wrote:
> David,
>
> There is certainly no need for an additional RTT.
>
> Let me propose a couple of concrete strawmen (concrete men?) to show
> this. I'm not claiming these are novel. I know Joe and others think that
> we've seen enough of attempts to extend the TCP options on the SYN, but
> Joe should know that when he claims something is impossible, he wakes
> sleeping dragons in people like me.

I slay dragons all the time ;-)

("Defender of the cruciform packet"* is one of my mottos)

> 1) Parallel control channel
> ___________________________
> Client A sends two SYNs back-to-back to an existing well-known port
> (e.g. 80).

You can send in whatever order you want; packets will be reordered, 
lost, and sent along alternate paths.

FWIW, do these use the same source port and ISN?
	- if they do, it'll reset the connection
	- if they don't, you're now limiting the number of
	concurrent connections to roughly half:
	http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/

> * SYN D, establishes a regular data connection, with sufficient TCP
> options to be workable but they still fit within the existing 40B option
> limit.
> * SYN C establishes another parallel connection to the same well-known
> port that looks like regular data from the outside (it could even be an
> extension to HTTP to ensure middleboxes will let it pass), but it talks
> a new app-layer 'TCP control' protocol inside.

What happens when they arrive out of order? What happens when you get D 
but not yet C? How long do you wait for C?

This is the problem with dual-stack approaches - new endpoints penalize 
legacy endpoints if there's a stall, and undermine new endpoints if they 
don't.

> If there is no support for the new app-layer protocol on port 80 the
> control channel just shuts down with a suitable HTTP error, while SYN D
> has opened a data connection with sufficient TCP options to be workable.
> If the new app-layer TCP control protocol is supported on port 80, the
> parallel control channel (C) adds unlimited additional control
> flexibility to the data channel (D) hardly any added latency.
>
> Establishing a similar control channel in the opposite direction would
> be fairly trivial.
>
> There are few, if any, middlebox problems with the above approach.
> However, there are certainly other problems, but no more insurmountable
> than all the problems that have already been discussed with taking the
> 'easy' route of EDO:
> * A secure binding would have to be added to bind channel C to a secret
> known only to the originator of channel D, otherwise it would open up
> data channels to spoof control channel attacks. This binding could be
> built on a TCP-AO option in channel D.

Yes, that's another problem.

> * Channel C would need some way to refer to the segments of channel D
> that was robust against re-segmentation.

Which means it won't work in the current Internet, because 
resegmentation is also widespread (though evil, IMO).

> * The main problem is that the two channels don't share fate;a control
> packet can be delayed relative to the point in the data stream at which
> it is attempting to exert control, possibly for a RTT if it is lost and
> has to be retransmitted. However, this is not insurmountable. The
> control protocol could include a mode to "synthesise shared fate", by
> making the data channel buffer data until an associated control segment
> had arrived. This would duplicate the latency impact of a loss or delay
> on either channel, but one can imagine mitigations that would consign
> this latency impact to corner cases.
> * It's a bit of a mess, but that comes with the territory when trying to
> fix legacy protocol problems.
> * The internal stack architecture seems to require a trombone back down
> into the kernel from user-space, but that is not insurmountable - a shim
> within the kernel on port 80 (for example) could redirect control
> channel data across to the "TCP control channel module" in the kernel,
> while passing non-control channel connections to user-space.
>
> 2) Build on LOIC
> ______________________
> Long option with invalid checksum <draft-yourtchenko-tcp-loic-00>

Won't work through current NATs, which won't recalculate the checksum 
properly.

>
> At 18:53 22/05/2014, John Leslie wrote:
>>    That's too big of a change to ask folks to believe it safe.
>
> When I read an idea, I don't take it as set in stone and just find a
> hole and dismiss it. I see it as a potential stepping stone to a
> solution and think about how it could be done better. In fact, Andrew
> Yourtchenko said that was the intention of his write-up of LOIC.
>
> I believe that an approach worth further thought would be a mixture of
> the control channel idea and the invalid checksum idea. I'm thinking of:
> * a pure control SYN (C) sent first, then a base SYN (D) sent
> back-to-back, both to the same port.

Again, please don't assume back-to-back means anything.

> * SYN C would contain something invalid to cause a legacy TCP stack or
> legacy app to discard it (and hopefully less probability that a
> middlebox would), e.g. a payload that is invalid for the application
> protocol on the port.

But so will a NAT, etc.

> * there would be additional TCP options in the payload of SYN C to be
> added to the TCP options that arrived separately on the base SYN
> * The control SYN could be bound crytographically to the base SYN (as
> already described).
> * It could use the shim-like control stack arangement described earlier.
>
> By focusing solely on extending the SYN, this would avoid the ongoing
> shared fate problems that a separate control channel suffers throughout
> the connection. There would still be shared fate problems with 2 SYNs
> (e.g. the two SYNs get re-ordered), but the protocol would have to be
> designed to be robust to that (naively, SYN D could include a new TCP
> option that told a new stack to wait a few ticks for a SYN C, but that
> would be vulnerable to meddleboxes). Not insurmountable.

AFAICT, it is.


*with a nod to Raiders 3.


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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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On Fri, May 23, 2014 at 9:32 AM, Erik Hugne <erik.hugne@ericsson.com> wrote=
:
> Maybe..
> This is a standard Linux 3.0 kernel with sack and frto enabled.
> Additional lab experiments have shown that following the path change, a s=
tream
> of lost packets occurs (almost a full window, ~400kB). The receiver sends
> dupacks on the new path to trigger fast retransmit, but then the retransm=
ission
> timer on the sender side fires and all unacked data will be resent. At th=
e RTO,
> we get the expected cwnd drop. While the receiver keeps kicking the the s=
ender
> by sending dupacks, the retransmission is reasonably fast but after a few=
 ms it
> grinds down to a stop-and-go behaviour.

It sounds like the big problems start with the RTO. In theory, if the
sender is getting enough SACKs then it ought to be able to repair all
those lost packets in one or a handful of RTTs, depending on just how
many were lost, and how much the RTT jumps upward in the switch to the
new path.

Can you retry your experiments with Linux 3.13 or newer? Recent Linux
kernels (3.13 and beyond) should do better in this regard: Yuchung
Cheng (cc-ed) has made some nice changes that take RTT measurements
from SACKs of unretransmitted data, and then keep updating the RTO
timer accordingly.

Regarding your scenario - what are the RTTs of the old and new paths?

cheers,
neal

> I failed in my efforts to find some explanation to this in the RFC's.
> What i did find was in VJ's paper from '88.
>
> -When starting or restarting after a loss, set cwnd to one packet.
> -On each ack for new data, increase cwnd by one packet
>
> If retransmitted data are not considered "new", that would explain the be=
havior
> i guess.
>
> Scripted ss -ti for the connection every 3 sec.
> In this test it wasn't even a path change, we just downed the interface o=
n a
> router in the middle for some 300ms to simulate the packetloss (This was =
done at 15:06:18)
>
> Fri May 23 15:06:01 CEST 2014
>         reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd:2215 send 15.3Mbp=
s rcv_space:14480
> Fri May 23 15:06:04 CEST 2014
>         reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd:2215 send 15.3Mbp=
s rcv_space:14480
> Fri May 23 15:06:07 CEST 2014
>         reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd:2215 send 15.3Mbp=
s rcv_space:14480
> Fri May 23 15:06:10 CEST 2014
>         reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cwnd:2215 send 15.3M=
bps rcv_space:14480
> Fri May 23 15:06:14 CEST 2014
>         reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 cwnd:2215 send 16.0=
Mbps rcv_space:14480
> Fri May 23 15:06:17 CEST 2014
>         reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cwnd:2215 send 15.3M=
bps rcv_space:14480
> Fri May 23 15:06:20 CEST 2014
>         reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwnd:3 ssthresh:1107 =
send 53.4Kbps rcv_space:14480
> Fri May 23 15:06:23 CEST 2014
>         reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwnd:3 ssthresh:1107 =
send 132.1Kbps rcv_space:14480
> Fri May 23 15:06:26 CEST 2014
>         reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cwnd:3 ssthresh:1107=
 send 165.1Kbps rcv_space:14480
> Fri May 23 15:06:29 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwnd:3 ssthresh:1107 =
send 170.8Kbps rcv_space:14480
> Fri May 23 15:06:32 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:3 ssthresh:1107 =
send 170.8Kbps rcv_space:14480
> Fri May 23 15:06:35 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:38 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:41 CEST 2014
>         reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:6 ssthresh:1107 se=
nd 340.7Kbps rcv_space:14480
> Fri May 23 15:06:44 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:47 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:50 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:53 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:56 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:59 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107 =
send 341.5Kbps rcv_space:14480
> Fri May 23 15:07:02 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 =
send 512.3Kbps rcv_space:14480
> Fri May 23 15:07:05 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 =
send 512.3Kbps rcv_space:14480
> Fri May 23 15:07:08 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 =
send 512.3Kbps rcv_space:14480
> Fri May 23 15:07:11 CEST 2014
>         reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwnd:960 ssthresh:110=
7 send 22.8Mbps rcv_space:14480
> Fri May 23 15:07:14 CEST 2014
>         reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwnd:1111 ssthresh:11=
07 send 14.9Mbps rcv_space:14480
> Fri May 23 15:07:17 CEST 2014
>         reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:1114 ssthresh:1107=
 send 14.9Mbps rcv_space:14480
> Fri May 23 15:07:20 CEST 2014
>         reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:1118 ssthresh:1107=
 send 14.9Mbps rcv_space:14480
>
> On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:
>>
>> Are you perhaps observing an effect of the ACK clock loss, when you refe=
r to a huge number of consecutive lost segments?
>>
>> Standard SACK does suffer from an issue that during loss recovery, there=
 may be a gap (SACKs don't clock out new/retransmitted packets while flight=
size > cwnd (which is now 1/2); only during the 2nd half, SACK starts to cl=
ock out packets, but at the same rate as before (which may not that bright =
an idea if a path change with differnet characteristics is now there).
>>
>> RateHalving (Linux has some kind of variant of this) / Proportional Rate=
 Reduction RFC6937 may be the thing that you are after?
>>
>>
>> ABC really only comes into play during slowstart, to make connections wh=
ich use delayed ACKs not perform much less aggressive than sessions where t=
he receiver features QuickAck (Linux non-standard).
>>
>> Best regards,
>>
>>
>> Richard Scheffenegger
>>
>>
>> > -----Original Message-----
>> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
>> > Sent: Mittwoch, 21. Mai 2014 14:55
>> > To: Mark Allman
>> > Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
>> > tcpm@ietf.org
>> > Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
>> >
>> > On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
>> > >
>> > > > TCP ABC (still in the experimental track) defines a method to
>> > > > increase the cwnd based on the bytes being acknowledged, instead o=
f
>> > > > the number of acks that arrive. I am assuming that SACK'ed (RFC201=
8)
>> > > > data should be included in this calculation, but i could find no
>> > > > mention of this in RFC3465.  Could someone confirm this?
>> > >
>> > > ABC does not include SACK information in its calculations and it
>> > > cannot include SACK information.  ABC applies to slow start---which
>> > > ends when loss occurs.  And, (essentially) SACKs only arrive during =
loss
>> > recovery.
>> > > So, there is no need for ABC to deal with SACKs.
>> > >
>> >
>> > Thanks for taking the time to explain this.
>> >
>> > The problem (this is related to Ingemars posts around "Problem with lo=
w
>> > SSthresh") is that following the path change, there is a large train o=
f
>> > segments being lost (say 150k out of a 500k window). Data at the end o=
f
>> > the window did make it through and is SACK'ed. Upon receiving these SA=
CK's
>> > the sending side enters loss recovery. It will take a very long time
>> > before recovering the 100k lost segments.
>> >
>> > An aggressive recovery algorithm would mitigate the effects of this, b=
ut i
>> > was hoping for a more elegant solution.
>> >
>> > //E
>> >
>> > _______________________________________________
>> > 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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References: <20140521073611.GA1940@eerihug-hybrid.rnd.ki.sw.ericsson.se> <20140521114628.C72DB6B339F@lawyers.icir.org> <20140521125518.GC1940@eerihug-hybrid.rnd.ki.sw.ericsson.se> <012C3117EDDB3C4781FD802A8C27DD4F4A3C3904@SACEXCMBX02-PRD.hq.netapp.com> <20140523133251.GA28880@eerihug-hybrid.rnd.ki.sw.ericsson.se>
Date: Fri, 23 May 2014 16:06:12 -0700
Message-ID: <CAO249ydqjyrK1fh8xvV6S-f3OB52ag0gwEyM0eN03AF7Yf1+Dw@mail.gmail.com>
From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: Erik Hugne <erik.hugne@ericsson.com>
Content-Type: multipart/alternative; boundary=047d7b87371e9c043904fa19464b
Archived-At: http://mailarchive.ietf.org/arch/msg/tcpm/ROOK5Onlqr8tcPmOuhYxBHQKZtQ
Cc: "ingemar.s.johansson@ericsson.com" <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>, Mark Allman <mallman@icir.org>, "martin.isaksson@ericsson.com" <martin.isaksson@ericsson.com>
Subject: [tcpm]  RFC3465:TCP ABC and SACK'ed data
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--047d7b87371e9c043904fa19464b
Content-Type: text/plain; charset=UTF-8

This is just out of my curiosity, but can we share the tcpdump file for
this?
--
Yoshi

On Friday, May 23, 2014, Erik Hugne
<erik.hugne@ericsson.com<javascript:_e(%7B%7D,'cvml','erik.hugne@ericsson.com');>>
wrote:

> Maybe..
> This is a standard Linux 3.0 kernel with sack and frto enabled.
> Additional lab experiments have shown that following the path change, a
> stream
> of lost packets occurs (almost a full window, ~400kB). The receiver sends
> dupacks on the new path to trigger fast retransmit, but then the
> retransmission
> timer on the sender side fires and all unacked data will be resent. At the
> RTO,
> we get the expected cwnd drop. While the receiver keeps kicking the the
> sender
> by sending dupacks, the retransmission is reasonably fast but after a few
> ms it
> grinds down to a stop-and-go behaviour.
>
> I failed in my efforts to find some explanation to this in the RFC's.
> What i did find was in VJ's paper from '88.
>
> -When starting or restarting after a loss, set cwnd to one packet.
> -On each ack for new data, increase cwnd by one packet
>
> If retransmitted data are not considered "new", that would explain the
> behavior
> i guess.
>
> Scripted ss -ti for the connection every 3 sec.
> In this test it wasn't even a path change, we just downed the interface on
> a
> router in the middle for some 300ms to simulate the packetloss (This was
> done at 15:06:18)
>
> Fri May 23 15:06:01 CEST 2014
>         reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd:2215 send 15.3Mbps
> rcv_space:14480
> Fri May 23 15:06:04 CEST 2014
>         reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd:2215 send 15.3Mbps
> rcv_space:14480
> Fri May 23 15:06:07 CEST 2014
>         reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd:2215 send 15.3Mbps
> rcv_space:14480
> Fri May 23 15:06:10 CEST 2014
>         reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cwnd:2215 send
> 15.3Mbps rcv_space:14480
> Fri May 23 15:06:14 CEST 2014
>         reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 cwnd:2215 send
> 16.0Mbps rcv_space:14480
> Fri May 23 15:06:17 CEST 2014
>         reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cwnd:2215 send
> 15.3Mbps rcv_space:14480
> Fri May 23 15:06:20 CEST 2014
>         reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwnd:3 ssthresh:1107
> send 53.4Kbps rcv_space:14480
> Fri May 23 15:06:23 CEST 2014
>         reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwnd:3 ssthresh:1107
> send 132.1Kbps rcv_space:14480
> Fri May 23 15:06:26 CEST 2014
>         reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cwnd:3 ssthresh:1107
> send 165.1Kbps rcv_space:14480
> Fri May 23 15:06:29 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwnd:3 ssthresh:1107
> send 170.8Kbps rcv_space:14480
> Fri May 23 15:06:32 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:3 ssthresh:1107
> send 170.8Kbps rcv_space:14480
> Fri May 23 15:06:35 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:38 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:41 CEST 2014
>         reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:6 ssthresh:1107
> send 340.7Kbps rcv_space:14480
> Fri May 23 15:06:44 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:47 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:50 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:53 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:56 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:06:59 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107
> send 341.5Kbps rcv_space:14480
> Fri May 23 15:07:02 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
> send 512.3Kbps rcv_space:14480
> Fri May 23 15:07:05 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
> send 512.3Kbps rcv_space:14480
> Fri May 23 15:07:08 CEST 2014
>         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
> send 512.3Kbps rcv_space:14480
> Fri May 23 15:07:11 CEST 2014
>         reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwnd:960 ssthresh:1107
> send 22.8Mbps rcv_space:14480
> Fri May 23 15:07:14 CEST 2014
>         reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwnd:1111
> ssthresh:1107 send 14.9Mbps rcv_space:14480
> Fri May 23 15:07:17 CEST 2014
>         reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:1114 ssthresh:1107
> send 14.9Mbps rcv_space:14480
> Fri May 23 15:07:20 CEST 2014
>         reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:1118 ssthresh:1107
> send 14.9Mbps rcv_space:14480
>
> On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:
> >
> > Are you perhaps observing an effect of the ACK clock loss, when you
> refer to a huge number of consecutive lost segments?
> >
> > Standard SACK does suffer from an issue that during loss recovery, there
> may be a gap (SACKs don't clock out new/retransmitted packets while
> flightsize > cwnd (which is now 1/2); only during the 2nd half, SACK starts
> to clock out packets, but at the same rate as before (which may not that
> bright an idea if a path change with differnet characteristics is now
> there).
> >
> > RateHalving (Linux has some kind of variant of this) / Proportional Rate
> Reduction RFC6937 may be the thing that you are after?
> >
> >
> > ABC really only comes into play during slowstart, to make connections
> which use delayed ACKs not perform much less aggressive than sessions where
> the receiver features QuickAck (Linux non-standard).
> >
> > Best regards,
> >
> >
> > Richard Scheffenegger
> >
> >
> > > -----Original Message-----
> > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
> > > Sent: Mittwoch, 21. Mai 2014 14:55
> > > To: Mark Allman
> > > Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
> > > tcpm@ietf.org
> > > Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
> > >
> > > On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
> > > >
> > > > > TCP ABC (still in the experimental track) defines a method to
> > > > > increase the cwnd based on the bytes being acknowledged, instead of
> > > > > the number of acks that arrive. I am assuming that SACK'ed
> (RFC2018)
> > > > > data should be included in this calculation, but i could find no
> > > > > mention of this in RFC3465.  Could someone confirm this?
> > > >
> > > > ABC does not include SACK information in its calculations and it
> > > > cannot include SACK information.  ABC applies to slow start---which
> > > > ends when loss occurs.  And, (essentially) SACKs only arrive during
> loss
> > > recovery.
> > > > So, there is no need for ABC to deal with SACKs.
> > > >
> > >
> > > Thanks for taking the time to explain this.
> > >
> > > The problem (this is related to Ingemars posts around "Problem with low
> > > SSthresh") is that following the path change, there is a large train of
> > > segments being lost (say 150k out of a 500k window). Data at the end of
> > > the window did make it through and is SACK'ed. Upon receiving these
> SACK's
> > > the sending side enters loss recovery. It will take a very long time
> > > before recovering the 100k lost segments.
> > >
> > > An aggressive recovery algorithm would mitigate the effects of this,
> but i
> > > was hoping for a more elegant solution.
> > >
> > > //E
> > >
> > > _______________________________________________
> > > 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
>

--047d7b87371e9c043904fa19464b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

This is just out of my curiosity, but can we share the tcpdump file for thi=
s?<div>--</div><div>Yoshi<br><br>On Friday, May 23, 2014, Erik Hugne &lt;<a=
 href=3D"javascript:_e(%7B%7D,&#39;cvml&#39;,&#39;erik.hugne@ericsson.com&#=
39;);" target=3D"_blank">erik.hugne@ericsson.com</a>&gt; wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Maybe..<br>
This is a standard Linux 3.0 kernel with sack and frto enabled.<br>
Additional lab experiments have shown that following the path change, a str=
eam<br>
of lost packets occurs (almost a full window, ~400kB). The receiver sends<b=
r>
dupacks on the new path to trigger fast retransmit, but then the retransmis=
sion<br>
timer on the sender side fires and all unacked data will be resent. At the =
RTO,<br>
we get the expected cwnd drop. While the receiver keeps kicking the the sen=
der<br>
by sending dupacks, the retransmission is reasonably fast but after a few m=
s it<br>
grinds down to a stop-and-go behaviour.<br>
<br>
I failed in my efforts to find some explanation to this in the RFC&#39;s.<b=
r>
What i did find was in VJ&#39;s paper from &#39;88.<br>
<br>
-When starting or restarting after a loss, set cwnd to one packet.<br>
-On each ack for new data, increase cwnd by one packet<br>
<br>
If retransmitted data are not considered &quot;new&quot;, that would explai=
n the behavior<br>
i guess.<br>
<br>
Scripted ss -ti for the connection every 3 sec.<br>
In this test it wasn&#39;t even a path change, we just downed the interface=
 on a<br>
router in the middle for some 300ms to simulate the packetloss (This was do=
ne at 15:06:18)<br>
<br>
Fri May 23 15:06:01 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd=
:2215 send 15.3Mbps rcv_space:14480<br>
Fri May 23 15:06:04 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd=
:2215 send 15.3Mbps rcv_space:14480<br>
Fri May 23 15:06:07 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd=
:2215 send 15.3Mbps rcv_space:14480<br>
Fri May 23 15:06:10 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cw=
nd:2215 send 15.3Mbps rcv_space:14480<br>
Fri May 23 15:06:14 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 c=
wnd:2215 send 16.0Mbps rcv_space:14480<br>
Fri May 23 15:06:17 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cw=
nd:2215 send 15.3Mbps rcv_space:14480<br>
Fri May 23 15:06:20 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwn=
d:3 ssthresh:1107 send 53.4Kbps rcv_space:14480<br>
Fri May 23 15:06:23 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwn=
d:3 ssthresh:1107 send 132.1Kbps rcv_space:14480<br>
Fri May 23 15:06:26 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cw=
nd:3 ssthresh:1107 send 165.1Kbps rcv_space:14480<br>
Fri May 23 15:06:29 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwn=
d:3 ssthresh:1107 send 170.8Kbps rcv_space:14480<br>
Fri May 23 15:06:32 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:3 ssthresh:1107 send 170.8Kbps rcv_space:14480<br>
Fri May 23 15:06:35 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:38 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:41 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:=
6 ssthresh:1107 send 340.7Kbps rcv_space:14480<br>
Fri May 23 15:06:44 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:47 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:50 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:53 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:56 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:06:59 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwn=
d:6 ssthresh:1107 send 341.5Kbps rcv_space:14480<br>
Fri May 23 15:07:02 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:9 ssthresh:1107 send 512.3Kbps rcv_space:14480<br>
Fri May 23 15:07:05 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:9 ssthresh:1107 send 512.3Kbps rcv_space:14480<br>
Fri May 23 15:07:08 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwn=
d:9 ssthresh:1107 send 512.3Kbps rcv_space:14480<br>
Fri May 23 15:07:11 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwn=
d:960 ssthresh:1107 send 22.8Mbps rcv_space:14480<br>
Fri May 23 15:07:14 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwn=
d:1111 ssthresh:1107 send 14.9Mbps rcv_space:14480<br>
Fri May 23 15:07:17 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:=
1114 ssthresh:1107 send 14.9Mbps rcv_space:14480<br>
Fri May 23 15:07:20 CEST 2014<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:=
1118 ssthresh:1107 send 14.9Mbps rcv_space:14480<br>
<br>
On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:<br>
&gt;<br>
&gt; Are you perhaps observing an effect of the ACK clock loss, when you re=
fer to a huge number of consecutive lost segments?<br>
&gt;<br>
&gt; Standard SACK does suffer from an issue that during loss recovery, the=
re may be a gap (SACKs don&#39;t clock out new/retransmitted packets while =
flightsize &gt; cwnd (which is now 1/2); only during the 2nd half, SACK sta=
rts to clock out packets, but at the same rate as before (which may not tha=
t bright an idea if a path change with differnet characteristics is now the=
re).<br>


&gt;<br>
&gt; RateHalving (Linux has some kind of variant of this) / Proportional Ra=
te Reduction RFC6937 may be the thing that you are after?<br>
&gt;<br>
&gt;<br>
&gt; ABC really only comes into play during slowstart, to make connections =
which use delayed ACKs not perform much less aggressive than sessions where=
 the receiver features QuickAck (Linux non-standard).<br>
&gt;<br>
&gt; Best regards,<br>
&gt;<br>
&gt;<br>
&gt; Richard Scheffenegger<br>
&gt;<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: tcpm [mailto:<a>tcpm-bounces@ietf.org</a>] On Behalf Of Eri=
k Hugne<br>
&gt; &gt; Sent: Mittwoch, 21. Mai 2014 14:55<br>
&gt; &gt; To: Mark Allman<br>
&gt; &gt; Cc: <a>ingemar.s.johansson@ericsson.com</a>; <a>martin.isaksson@e=
ricsson.com</a>;<br>

&gt; &gt; <a>tcpm@ietf.org</a><br>
&gt; &gt; Subject: Re: [tcpm] RFC3465:TCP ABC and SACK&#39;ed data<br>
&gt; &gt;<br>
&gt; &gt; On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; &gt; TCP ABC (still in the experimental track) defines a met=
hod to<br>
&gt; &gt; &gt; &gt; increase the cwnd based on the bytes being acknowledged=
, instead of<br>
&gt; &gt; &gt; &gt; the number of acks that arrive. I am assuming that SACK=
&#39;ed (RFC2018)<br>
&gt; &gt; &gt; &gt; data should be included in this calculation, but i coul=
d find no<br>
&gt; &gt; &gt; &gt; mention of this in RFC3465. =C2=A0Could someone confirm=
 this?<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; ABC does not include SACK information in its calculations an=
d it<br>
&gt; &gt; &gt; cannot include SACK information. =C2=A0ABC applies to slow s=
tart---which<br>
&gt; &gt; &gt; ends when loss occurs. =C2=A0And, (essentially) SACKs only a=
rrive during loss<br>
&gt; &gt; recovery.<br>
&gt; &gt; &gt; So, there is no need for ABC to deal with SACKs.<br>
&gt; &gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Thanks for taking the time to explain this.<br>
&gt; &gt;<br>
&gt; &gt; The problem (this is related to Ingemars posts around &quot;Probl=
em with low<br>
&gt; &gt; SSthresh&quot;) is that following the path change, there is a lar=
ge train of<br>
&gt; &gt; segments being lost (say 150k out of a 500k window). Data at the =
end of<br>
&gt; &gt; the window did make it through and is SACK&#39;ed. Upon receiving=
 these SACK&#39;s<br>
&gt; &gt; the sending side enters loss recovery. It will take a very long t=
ime<br>
&gt; &gt; before recovering the 100k lost segments.<br>
&gt; &gt;<br>
&gt; &gt; An aggressive recovery algorithm would mitigate the effects of th=
is, but i<br>
&gt; &gt; was hoping for a more elegant solution.<br>
&gt; &gt;<br>
&gt; &gt; //E<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; tcpm mailing list<br>
&gt; &gt; <a>tcpm@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a>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>

--047d7b87371e9c043904fa19464b--


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Date: Mon, 26 May 2014 11:12:32 +0200
From: Erik Hugne <erik.hugne@ericsson.com>
To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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Here's from a test just this morning, wgetting a 10MB file.
https://www.dropbox.com/sh/86u3nzcwukjzpzv/AAAlp_Me7wXf7PRyuWFq9Lmwa

//E

On Fri, May 23, 2014 at 04:06:12PM -0700, Yoshifumi Nishida wrote:
> This is just out of my curiosity, but can we share the tcpdump file for
> this?
> --
> Yoshi
> 
> On Friday, May 23, 2014, Erik Hugne
> <erik.hugne@ericsson.com<javascript:_e(%7B%7D,'cvml','erik.hugne@ericsson.com');>>
> wrote:
> 
> > Maybe..
> > This is a standard Linux 3.0 kernel with sack and frto enabled.
> > Additional lab experiments have shown that following the path change, a
> > stream
> > of lost packets occurs (almost a full window, ~400kB). The receiver sends
> > dupacks on the new path to trigger fast retransmit, but then the
> > retransmission
> > timer on the sender side fires and all unacked data will be resent. At the
> > RTO,
> > we get the expected cwnd drop. While the receiver keeps kicking the the
> > sender
> > by sending dupacks, the retransmission is reasonably fast but after a few
> > ms it
> > grinds down to a stop-and-go behaviour.
> >
> > I failed in my efforts to find some explanation to this in the RFC's.
> > What i did find was in VJ's paper from '88.
> >
> > -When starting or restarting after a loss, set cwnd to one packet.
> > -On each ack for new data, increase cwnd by one packet
> >
> > If retransmitted data are not considered "new", that would explain the
> > behavior
> > i guess.
> >
> > Scripted ss -ti for the connection every 3 sec.
> > In this test it wasn't even a path change, we just downed the interface on
> > a
> > router in the middle for some 300ms to simulate the packetloss (This was
> > done at 15:06:18)
> >
> > Fri May 23 15:06:01 CEST 2014
> >         reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd:2215 send 15.3Mbps
> > rcv_space:14480
> > Fri May 23 15:06:04 CEST 2014
> >         reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd:2215 send 15.3Mbps
> > rcv_space:14480
> > Fri May 23 15:06:07 CEST 2014
> >         reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd:2215 send 15.3Mbps
> > rcv_space:14480
> > Fri May 23 15:06:10 CEST 2014
> >         reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cwnd:2215 send
> > 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:14 CEST 2014
> >         reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 cwnd:2215 send
> > 16.0Mbps rcv_space:14480
> > Fri May 23 15:06:17 CEST 2014
> >         reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cwnd:2215 send
> > 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:20 CEST 2014
> >         reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwnd:3 ssthresh:1107
> > send 53.4Kbps rcv_space:14480
> > Fri May 23 15:06:23 CEST 2014
> >         reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwnd:3 ssthresh:1107
> > send 132.1Kbps rcv_space:14480
> > Fri May 23 15:06:26 CEST 2014
> >         reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cwnd:3 ssthresh:1107
> > send 165.1Kbps rcv_space:14480
> > Fri May 23 15:06:29 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwnd:3 ssthresh:1107
> > send 170.8Kbps rcv_space:14480
> > Fri May 23 15:06:32 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:3 ssthresh:1107
> > send 170.8Kbps rcv_space:14480
> > Fri May 23 15:06:35 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:38 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:41 CEST 2014
> >         reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:6 ssthresh:1107
> > send 340.7Kbps rcv_space:14480
> > Fri May 23 15:06:44 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:47 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:50 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:53 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:56 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:59 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107
> > send 341.5Kbps rcv_space:14480
> > Fri May 23 15:07:02 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
> > send 512.3Kbps rcv_space:14480
> > Fri May 23 15:07:05 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
> > send 512.3Kbps rcv_space:14480
> > Fri May 23 15:07:08 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
> > send 512.3Kbps rcv_space:14480
> > Fri May 23 15:07:11 CEST 2014
> >         reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwnd:960 ssthresh:1107
> > send 22.8Mbps rcv_space:14480
> > Fri May 23 15:07:14 CEST 2014
> >         reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwnd:1111
> > ssthresh:1107 send 14.9Mbps rcv_space:14480
> > Fri May 23 15:07:17 CEST 2014
> >         reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:1114 ssthresh:1107
> > send 14.9Mbps rcv_space:14480
> > Fri May 23 15:07:20 CEST 2014
> >         reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:1118 ssthresh:1107
> > send 14.9Mbps rcv_space:14480
> >
> > On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:
> > >
> > > Are you perhaps observing an effect of the ACK clock loss, when you
> > refer to a huge number of consecutive lost segments?
> > >
> > > Standard SACK does suffer from an issue that during loss recovery, there
> > may be a gap (SACKs don't clock out new/retransmitted packets while
> > flightsize > cwnd (which is now 1/2); only during the 2nd half, SACK starts
> > to clock out packets, but at the same rate as before (which may not that
> > bright an idea if a path change with differnet characteristics is now
> > there).
> > >
> > > RateHalving (Linux has some kind of variant of this) / Proportional Rate
> > Reduction RFC6937 may be the thing that you are after?
> > >
> > >
> > > ABC really only comes into play during slowstart, to make connections
> > which use delayed ACKs not perform much less aggressive than sessions where
> > the receiver features QuickAck (Linux non-standard).
> > >
> > > Best regards,
> > >
> > >
> > > Richard Scheffenegger
> > >
> > >
> > > > -----Original Message-----
> > > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
> > > > Sent: Mittwoch, 21. Mai 2014 14:55
> > > > To: Mark Allman
> > > > Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
> > > > tcpm@ietf.org
> > > > Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
> > > >
> > > > On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
> > > > >
> > > > > > TCP ABC (still in the experimental track) defines a method to
> > > > > > increase the cwnd based on the bytes being acknowledged, instead of
> > > > > > the number of acks that arrive. I am assuming that SACK'ed
> > (RFC2018)
> > > > > > data should be included in this calculation, but i could find no
> > > > > > mention of this in RFC3465.  Could someone confirm this?
> > > > >
> > > > > ABC does not include SACK information in its calculations and it
> > > > > cannot include SACK information.  ABC applies to slow start---which
> > > > > ends when loss occurs.  And, (essentially) SACKs only arrive during
> > loss
> > > > recovery.
> > > > > So, there is no need for ABC to deal with SACKs.
> > > > >
> > > >
> > > > Thanks for taking the time to explain this.
> > > >
> > > > The problem (this is related to Ingemars posts around "Problem with low
> > > > SSthresh") is that following the path change, there is a large train of
> > > > segments being lost (say 150k out of a 500k window). Data at the end of
> > > > the window did make it through and is SACK'ed. Upon receiving these
> > SACK's
> > > > the sending side enters loss recovery. It will take a very long time
> > > > before recovering the 100k lost segments.
> > > >
> > > > An aggressive recovery algorithm would mitigate the effects of this,
> > but i
> > > > was hoping for a more elegant solution.
> > > >
> > > > //E
> > > >
> > > > _______________________________________________
> > > > tcpm mailing list
> > > > tcpm@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/tcpm
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >


From nobody Mon May 26 02:38:09 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: Joe Touch <touch@isi.edu>
Thread-Topic: timestamp options (was Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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Date: Mon, 26 May 2014 09:37:43 +0000
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Subject: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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Hi,

On 2014-5-23, at 18:55, Joe Touch <touch@isi.edu> wrote:
> Current total for SYN options in widespread concurrent use (as already =
described in sec 6.4):
>=20
> 	2	SACK permitted
> 	10	timestamp
> 	3	window scale
> 	4	MSS
> 	------------------
> 	11 bytes

I think it was Mark Allman who questioned a while ago whether the =
benefits of the timestamp option are worth spending 10 bytes in every =
packet. (But I can't find the email now.)

> 	11	widespread basic options
> 	16	TCP-AO
> 	20	MPTCP
> 	--------------------
> 	47

If we could shave those off, we'd be at 37 even with AO. It wouldn't =
make the option space any bigger for the future, but it would get us out =
of the current situation, and would make a combination of some crypto =
and MPTCP be deployable.

Lars

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From nobody Mon May 26 03:00:44 2014
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References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <537E48CE.8040704@mti-systems.com> <537E66A7.4080907@isi.edu> <201405231003.s4NA3PAB005137@bagheera.jungle.bt.co.uk> <537F7D91.10802@isi.edu> <F4BCB99F-6133-4F3C-BD5E-3369B979EB33@netapp.com>
To: "Eggert, Lars" <lars@netapp.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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On 26 May 2014, at 11:37, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>=20
> On 2014-5-23, at 18:55, Joe Touch <touch@isi.edu> wrote:
>> Current total for SYN options in widespread concurrent use (as =
already described in sec 6.4):
>>=20
>> 	2	SACK permitted
>> 	10	timestamp
>> 	3	window scale
>> 	4	MSS
>> 	------------------
>> 	11 bytes
>=20
> I think it was Mark Allman who questioned a while ago whether the =
benefits of the timestamp option are worth spending 10 bytes in every =
packet. (But I can't find the email now.)
>=20
>> 	11	widespread basic options
>> 	16	TCP-AO
>> 	20	MPTCP
>> 	--------------------
>> 	47
>=20
> If we could shave those off, we'd be at 37 even with AO. It wouldn't =
make the option space any bigger for the future, but it would get us out =
of the current situation, and would make a combination of some crypto =
and MPTCP be deployable.

SACK and WScale are in use by the majority of sources, but (depending on =
where you're looking from), only about a third of senders ever use the =
timestamp options (lots of sources; the most recent is CCR April '14, =
Honda et al "Rekindling Network Protocol Innovation with User-Level =
Stacks", introduction), largely an effect of OS-level defaults. So an =
either-or approach for TS or TCP-AO would probably work.

(Doing this would make in-path passive measurement of RTT on =
mostly-unidirectional flows basically impossible, but in a post-BCP188 =
world we should probably be working on instrumenting endpoints for =
debugging and monitoring, anyway...)

Cheers,

Brian




From nobody Mon May 26 03:06:35 2014
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "Eggert, Lars" <lars@netapp.com>, Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Eggert, Lars
> Sent: Monday, May 26, 2014 11:38 AM
> To: Joe Touch
> Cc: tcpm@ietf.org
> Subject: [tcpm] timestamp options (was Re: New Version Notification for
> draft-touch-tcpm-tcp-edo-01.txt)
>=20
> Hi,
>=20
> On 2014-5-23, at 18:55, Joe Touch <touch@isi.edu> wrote:
> > Current total for SYN options in widespread concurrent use (as
> already described in sec 6.4):
> >
> > 	2	SACK permitted
> > 	10	timestamp
> > 	3	window scale
> > 	4	MSS
> > 	------------------
> > 	11 bytes
>=20
> I think it was Mark Allman who questioned a while ago whether the
> benefits of the timestamp option are worth spending 10 bytes in every
> packet. (But I can't find the email now.)

And we have already had a long discussion on late enablement of timestamps.=
 Given that we submitted 1323bis in the meantime, I actually think we could=
 spend some cycles on saving those 10B from SYNs (if there was agreement th=
at it is doable).=20

Michael (chair hat off)


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Date: Mon, 26 May 2014 12:32:20 +0200
From: Erik Hugne <erik.hugne@ericsson.com>
To: Neal Cardwell <ncardwell@google.com>
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Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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On Fri, May 23, 2014 at 01:37:19PM -0400, Neal Cardwell wrote:
> On Fri, May 23, 2014 at 9:32 AM, Erik Hugne <erik.hugne@ericsson.com> wrote:
> > Maybe..
> > This is a standard Linux 3.0 kernel with sack and frto enabled.
> > Additional lab experiments have shown that following the path change, a stream
> > of lost packets occurs (almost a full window, ~400kB). The receiver sends
> > dupacks on the new path to trigger fast retransmit, but then the retransmission
> > timer on the sender side fires and all unacked data will be resent. At the RTO,
> > we get the expected cwnd drop. While the receiver keeps kicking the the sender
> > by sending dupacks, the retransmission is reasonably fast but after a few ms it
> > grinds down to a stop-and-go behaviour.
> 
> It sounds like the big problems start with the RTO. In theory, if the
> sender is getting enough SACKs then it ought to be able to repair all
> those lost packets in one or a handful of RTTs, depending on just how
> many were lost, and how much the RTT jumps upward in the switch to the
> new path.
> 
> Can you retry your experiments with Linux 3.13 or newer? Recent Linux
> kernels (3.13 and beyond) should do better in this regard: Yuchung
> Cheng (cc-ed) has made some nice changes that take RTT measurements
> from SACKs of unretransmitted data, and then keep updating the RTO
> timer accordingly.
> 
> Regarding your scenario - what are the RTTs of the old and new paths?

In the synthetic experiment below, there is no path change, we just "insert"
a transient loss and the RTT are the same before and after. I'll see if i
can bump the kernel version to 3.13 and retry. 

//E




> 
> cheers,
> neal
> 
> > I failed in my efforts to find some explanation to this in the RFC's.
> > What i did find was in VJ's paper from '88.
> >
> > -When starting or restarting after a loss, set cwnd to one packet.
> > -On each ack for new data, increase cwnd by one packet
> >
> > If retransmitted data are not considered "new", that would explain the behavior
> > i guess.
> >
> > Scripted ss -ti for the connection every 3 sec.
> > In this test it wasn't even a path change, we just downed the interface on a
> > router in the middle for some 300ms to simulate the packetloss (This was done at 15:06:18)
> >
> > Fri May 23 15:06:01 CEST 2014
> >         reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:04 CEST 2014
> >         reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:07 CEST 2014
> >         reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:10 CEST 2014
> >         reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:14 CEST 2014
> >         reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 cwnd:2215 send 16.0Mbps rcv_space:14480
> > Fri May 23 15:06:17 CEST 2014
> >         reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cwnd:2215 send 15.3Mbps rcv_space:14480
> > Fri May 23 15:06:20 CEST 2014
> >         reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwnd:3 ssthresh:1107 send 53.4Kbps rcv_space:14480
> > Fri May 23 15:06:23 CEST 2014
> >         reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwnd:3 ssthresh:1107 send 132.1Kbps rcv_space:14480
> > Fri May 23 15:06:26 CEST 2014
> >         reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cwnd:3 ssthresh:1107 send 165.1Kbps rcv_space:14480
> > Fri May 23 15:06:29 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwnd:3 ssthresh:1107 send 170.8Kbps rcv_space:14480
> > Fri May 23 15:06:32 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:3 ssthresh:1107 send 170.8Kbps rcv_space:14480
> > Fri May 23 15:06:35 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:38 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:41 CEST 2014
> >         reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:6 ssthresh:1107 send 340.7Kbps rcv_space:14480
> > Fri May 23 15:06:44 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:47 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:50 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:53 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:56 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:06:59 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107 send 341.5Kbps rcv_space:14480
> > Fri May 23 15:07:02 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 send 512.3Kbps rcv_space:14480
> > Fri May 23 15:07:05 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 send 512.3Kbps rcv_space:14480
> > Fri May 23 15:07:08 CEST 2014
> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107 send 512.3Kbps rcv_space:14480
> > Fri May 23 15:07:11 CEST 2014
> >         reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwnd:960 ssthresh:1107 send 22.8Mbps rcv_space:14480
> > Fri May 23 15:07:14 CEST 2014
> >         reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwnd:1111 ssthresh:1107 send 14.9Mbps rcv_space:14480
> > Fri May 23 15:07:17 CEST 2014
> >         reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:1114 ssthresh:1107 send 14.9Mbps rcv_space:14480
> > Fri May 23 15:07:20 CEST 2014
> >         reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:1118 ssthresh:1107 send 14.9Mbps rcv_space:14480
> >
> > On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:
> >>
> >> Are you perhaps observing an effect of the ACK clock loss, when you refer to a huge number of consecutive lost segments?
> >>
> >> Standard SACK does suffer from an issue that during loss recovery, there may be a gap (SACKs don't clock out new/retransmitted packets while flightsize > cwnd (which is now 1/2); only during the 2nd half, SACK starts to clock out packets, but at the same rate as before (which may not that bright an idea if a path change with differnet characteristics is now there).
> >>
> >> RateHalving (Linux has some kind of variant of this) / Proportional Rate Reduction RFC6937 may be the thing that you are after?
> >>
> >>
> >> ABC really only comes into play during slowstart, to make connections which use delayed ACKs not perform much less aggressive than sessions where the receiver features QuickAck (Linux non-standard).
> >>
> >> Best regards,
> >>
> >>
> >> Richard Scheffenegger
> >>
> >>
> >> > -----Original Message-----
> >> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
> >> > Sent: Mittwoch, 21. Mai 2014 14:55
> >> > To: Mark Allman
> >> > Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
> >> > tcpm@ietf.org
> >> > Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
> >> >
> >> > On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
> >> > >
> >> > > > TCP ABC (still in the experimental track) defines a method to
> >> > > > increase the cwnd based on the bytes being acknowledged, instead of
> >> > > > the number of acks that arrive. I am assuming that SACK'ed (RFC2018)
> >> > > > data should be included in this calculation, but i could find no
> >> > > > mention of this in RFC3465.  Could someone confirm this?
> >> > >
> >> > > ABC does not include SACK information in its calculations and it
> >> > > cannot include SACK information.  ABC applies to slow start---which
> >> > > ends when loss occurs.  And, (essentially) SACKs only arrive during loss
> >> > recovery.
> >> > > So, there is no need for ABC to deal with SACKs.
> >> > >
> >> >
> >> > Thanks for taking the time to explain this.
> >> >
> >> > The problem (this is related to Ingemars posts around "Problem with low
> >> > SSthresh") is that following the path change, there is a large train of
> >> > segments being lost (say 150k out of a 500k window). Data at the end of
> >> > the window did make it through and is SACK'ed. Upon receiving these SACK's
> >> > the sending side enters loss recovery. It will take a very long time
> >> > before recovering the 100k lost segments.
> >> >
> >> > An aggressive recovery algorithm would mitigate the effects of this, but i
> >> > was hoping for a more elegant solution.
> >> >
> >> > //E
> >> >
> >> > _______________________________________________
> >> > 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: "Scheffenegger, Richard" <rs@netapp.com>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>, "Eggert,  Lars" <lars@netapp.com>, Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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Date: Tue, 27 May 2014 11:36:50 +0000
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References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <537E48CE.8040704@mti-systems.com> <537E66A7.4080907@isi.edu> <201405231003.s4NA3PAB005137@bagheera.jungle.bt.co.uk> <537F7D91.10802@isi.edu> <F4BCB99F-6133-4F3C-BD5E-3369B979EB33@netapp.com> <655C07320163294895BBADA28372AF5D305E00@FR712WXCHMBA15.zeu.alcatel-lucent.com>
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Subject: Re: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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Well,

(my arithmetic for the usual options comes to a total of 19 bytes, not 11).

For late TS, we would still need at least 2 Bytes during <SYN>/<SYN,ACK> to=
 allow that capability.

However, as the useful values of Window Scale only cover 0..14 [1], one app=
roach to save SYN option space would be to negotiate up to 4 binary (on/off=
) plus WS in a new single 3-byte option.

The following TCP options would lend themselves for that:

SACK permitted=20
Late TS (new)
EDO OK (new)

Some recent scans did still reveal that quite a number of paths (port 80) o=
nly negotiate MSS, but refrain from doing any other option.

	 2	SACK permitted
	10	timestamp
	 3	window scale
	 4	MSS
	------------------
	19 bytes

Could become
=09
	 4	MSS
	 3	WScale (new) + [SAckOk | LateTSOk | EDO ok]
	------------------
	 7 bytes

But

       7    compacted common options
	16	TCP-AO
	20	MPTCP
	--------------------
	43 bytes (still in need of EDO on SYN)

Regards,


Richard Scheffenegger

[1] http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-21#section-2.1

> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael
> (Michael)
> Sent: Montag, 26. Mai 2014 12:06
> To: Eggert, Lars; Joe Touch
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] timestamp options (was Re: New Version Notification
> for draft-touch-tcpm-tcp-edo-01.txt)
>=20
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Eggert, Lars
> > Sent: Monday, May 26, 2014 11:38 AM
> > To: Joe Touch
> > Cc: tcpm@ietf.org
> > Subject: [tcpm] timestamp options (was Re: New Version Notification
> > for
> > draft-touch-tcpm-tcp-edo-01.txt)
> >
> > Hi,
> >
> > On 2014-5-23, at 18:55, Joe Touch <touch@isi.edu> wrote:
> > > Current total for SYN options in widespread concurrent use (as
> > already described in sec 6.4):
> > >
> > > 	2	SACK permitted
> > > 	10	timestamp
> > > 	3	window scale
> > > 	4	MSS
> > > 	------------------
> > > 	11 bytes
> >
> > I think it was Mark Allman who questioned a while ago whether the
> > benefits of the timestamp option are worth spending 10 bytes in every
> > packet. (But I can't find the email now.)
>=20
> And we have already had a long discussion on late enablement of
> timestamps. Given that we submitted 1323bis in the meantime, I actually
> think we could spend some cycles on saving those 10B from SYNs (if there
> was agreement that it is doable).
>=20
> Michael (chair hat off)
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "Scheffenegger, Richard" <rs@netapp.com>, "Eggert, Lars" <lars@netapp.com>, Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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> For late TS, we would still need at least 2 Bytes during
> <SYN>/<SYN,ACK> to allow that capability.
>=20
> However, as the useful values of Window Scale only cover 0..14 [1], one
> approach to save SYN option space would be to negotiate up to 4 binary
> (on/off) plus WS in a new single 3-byte option.

I might be wrong, but I've thought so far that "replacing" the window scale=
 option would be pretty difficult. We already had reports of middleboxes th=
at parse this option because they try to understand rwnd (or they do not pa=
rse the option even if they should, or they have entirely broken logic to m=
ess up TCP...). I think one example are firewalls that try to detect whethe=
r segments are in-window. Another example could be PEPs that try to tune rw=
nd (if they still exist in the wild).

My understanding so far was that a solution only for timestamps would be th=
e most low-hanging fruit.

Michael


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>, "Eggert,  Lars" <lars@netapp.com>, Joe Touch <touch@isi.edu>
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Hi Michael,


Well, if it's just for Timestamps, just use an "invalid" length of 2 as a "=
Late TS permitted".=20

Maintaining the same option value and this would be rather straight forward=
.=20

Personally, I'd like to combine this with a change in semantics, if SACK is=
 negotiated on the same session as well, to allow more efficient loss recov=
ery (lost retransmission detection during a loss recovery episode). I shoul=
d have a unpublished draft sitting around here about that aspect.

The TS option handling seems to be rather generous by middleboxes, in gener=
al.=20

Richard Scheffenegger

> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael
> (Michael)
> Sent: Dienstag, 27. Mai 2014 13:49
> To: Scheffenegger, Richard; Eggert, Lars; Joe Touch
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] timestamp options (was Re: New Version Notification
> for draft-touch-tcpm-tcp-edo-01.txt)
>=20
> > For late TS, we would still need at least 2 Bytes during
> > <SYN>/<SYN,ACK> to allow that capability.
> >
> > However, as the useful values of Window Scale only cover 0..14 [1],
> > one approach to save SYN option space would be to negotiate up to 4
> > binary
> > (on/off) plus WS in a new single 3-byte option.
>=20
> I might be wrong, but I've thought so far that "replacing" the window
> scale option would be pretty difficult. We already had reports of
> middleboxes that parse this option because they try to understand rwnd (o=
r
> they do not parse the option even if they should, or they have entirely
> broken logic to mess up TCP...). I think one example are firewalls that
> try to detect whether segments are in-window. Another example could be
> PEPs that try to tune rwnd (if they still exist in the wild).
>=20
> My understanding so far was that a solution only for timestamps would be
> the most low-hanging fruit.
>=20
> Michael
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "Scheffenegger, Richard" <rs@netapp.com>, "Eggert, Lars" <lars@netapp.com>, Joe Touch <touch@isi.edu>
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> Well, if it's just for Timestamps, just use an "invalid" length of 2 as
> a "Late TS permitted".
>=20
> Maintaining the same option value and this would be rather straight
> forward.

Interesting proposal. But actually we are not really short of option number=
s. If we get e.g. 8 bytes in a significant number of SYNs (which is our rea=
l bottleneck), I think we could also spend a new option codepoint.

> Personally, I'd like to combine this with a change in semantics, if
> SACK is negotiated on the same session as well, to allow more efficient
> loss recovery (lost retransmission detection during a loss recovery
> episode). I should have a unpublished draft sitting around here about
> that aspect.

A change in semantics would also be a reason for a new codepoint, imho.

Michael


> The TS option handling seems to be rather generous by middleboxes, in
> general.
>
> Richard Scheffenegger
>=20
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf,
> Michael
> > (Michael)
> > Sent: Dienstag, 27. Mai 2014 13:49
> > To: Scheffenegger, Richard; Eggert, Lars; Joe Touch
> > Cc: tcpm@ietf.org
> > Subject: Re: [tcpm] timestamp options (was Re: New Version
> Notification
> > for draft-touch-tcpm-tcp-edo-01.txt)
> >
> > > For late TS, we would still need at least 2 Bytes during
> > > <SYN>/<SYN,ACK> to allow that capability.
> > >
> > > However, as the useful values of Window Scale only cover 0..14 [1],
> > > one approach to save SYN option space would be to negotiate up to 4
> > > binary
> > > (on/off) plus WS in a new single 3-byte option.
> >
> > I might be wrong, but I've thought so far that "replacing" the window
> > scale option would be pretty difficult. We already had reports of
> > middleboxes that parse this option because they try to understand
> rwnd (or
> > they do not parse the option even if they should, or they have
> entirely
> > broken logic to mess up TCP...). I think one example are firewalls
> that
> > try to detect whether segments are in-window. Another example could
> be
> > PEPs that try to tune rwnd (if they still exist in the wild).
> >
> > My understanding so far was that a solution only for timestamps would
> be
> > the most low-hanging fruit.
> >
> > Michael
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm


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References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <537E48CE.8040704@mti-systems.com> <537E66A7.4080907@isi.edu> <201405231003.s4NA3PAB005137@bagheera.jungle.bt.co.uk> <537F7D91.10802@isi.edu> <F4BCB99F-6133-4F3C-BD5E-3369B979EB33@netapp.com> <655C07320163294895BBADA28372AF5D305E00@FR712WXCHMBA15.zeu.alcatel-lucent.com> <012C3117EDDB3C4781FD802A8C27DD4F6F3DD808@SACEXCMBX06-PRD.hq.netapp.com> <655C07320163294895BBADA28372AF5D309C5D@FR712WXCHMBA15.zeu.alcatel-lucent.com> <012C3117EDDB3C4781FD802A8C27DD4F6F3E09CF@SACEXCMBX06-PRD.hq.netapp.com> <655C07320163294895BBADA28372AF5D309CC7@FR712WXCHMBA15.zeu.alcatel-lucent.com>
Date: Tue, 27 May 2014 11:36:48 -0700
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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--047d7b87371e848ef304fa65fac9
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Hi folks,

I've submitted a proposal for this a while ago:
http://tools.ietf.org/html/draft-nishida-tcpm-apaws-00
Although it's expired now, I will submit a new version within a couple of
weeks or so.
The late negotiation is also discussed in the draft.
As it will change the semantics of TS, it presumes to use new code point
(or use timestamp field in SYN). But, I'm happy to discuss on this point.

One tricky thing for RFC1323 is that it uses TS for two purposes:
timestamping and PAWS and one purpose of the draft is to separate them.
I appreciate any feedback on the draft.

Thanks,
--
Yoshi

On Tue, May 27, 2014 at 5:12 AM, Scharf, Michael (Michael) <
michael.scharf@alcatel-lucent.com> wrote:

> > Well, if it's just for Timestamps, just use an "invalid" length of 2 as
> > a "Late TS permitted".
> >
> > Maintaining the same option value and this would be rather straight
> > forward.
>
> Interesting proposal. But actually we are not really short of option
> numbers. If we get e.g. 8 bytes in a significant number of SYNs (which is
> our real bottleneck), I think we could also spend a new option codepoint.
>
> > Personally, I'd like to combine this with a change in semantics, if
> > SACK is negotiated on the same session as well, to allow more efficient
> > loss recovery (lost retransmission detection during a loss recovery
> > episode). I should have a unpublished draft sitting around here about
> > that aspect.
>
> A change in semantics would also be a reason for a new codepoint, imho.
>
> Michael
>
>
> > The TS option handling seems to be rather generous by middleboxes, in
> > general.
> >
> > Richard Scheffenegger
> >
> > > -----Original Message-----
> > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf,
> > Michael
> > > (Michael)
> > > Sent: Dienstag, 27. Mai 2014 13:49
> > > To: Scheffenegger, Richard; Eggert, Lars; Joe Touch
> > > Cc: tcpm@ietf.org
> > > Subject: Re: [tcpm] timestamp options (was Re: New Version
> > Notification
> > > for draft-touch-tcpm-tcp-edo-01.txt)
> > >
> > > > For late TS, we would still need at least 2 Bytes during
> > > > <SYN>/<SYN,ACK> to allow that capability.
> > > >
> > > > However, as the useful values of Window Scale only cover 0..14 [1],
> > > > one approach to save SYN option space would be to negotiate up to 4
> > > > binary
> > > > (on/off) plus WS in a new single 3-byte option.
> > >
> > > I might be wrong, but I've thought so far that "replacing" the window
> > > scale option would be pretty difficult. We already had reports of
> > > middleboxes that parse this option because they try to understand
> > rwnd (or
> > > they do not parse the option even if they should, or they have
> > entirely
> > > broken logic to mess up TCP...). I think one example are firewalls
> > that
> > > try to detect whether segments are in-window. Another example could
> > be
> > > PEPs that try to tune rwnd (if they still exist in the wild).
> > >
> > > My understanding so far was that a solution only for timestamps would
> > be
> > > the most low-hanging fruit.
> > >
> > > Michael
> > >
> > > _______________________________________________
> > > 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
>

--047d7b87371e848ef304fa65fac9
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">Hi folks,<div><br></div><div>I&#39;ve submitted a proposal=
 for this a while ago: <a href=3D"http://tools.ietf.org/html/draft-nishida-=
tcpm-apaws-00">http://tools.ietf.org/html/draft-nishida-tcpm-apaws-00</a></=
div>
<div>Although it&#39;s expired now, I will submit a new version within a co=
uple of weeks or so.</div><div>The late negotiation is also discussed in th=
e draft.=C2=A0</div><div>As it will change the semantics of TS, it presumes=
 to use new code point (or use timestamp field in SYN). But, I&#39;m happy =
to discuss on this point.</div>
<div><br></div><div>One tricky thing for RFC1323 is that it uses TS for two=
 purposes: timestamping and PAWS and one purpose of the draft is to separat=
e them.</div><div>I appreciate any feedback on the draft.</div><div><br>
</div><div>Thanks,</div><div>--</div><div>Yoshi</div><div><div class=3D"gma=
il_extra"><br><div class=3D"gmail_quote">On Tue, May 27, 2014 at 5:12 AM, S=
charf, Michael (Michael) <span dir=3D"ltr">&lt;<a href=3D"mailto:michael.sc=
harf@alcatel-lucent.com" target=3D"_blank">michael.scharf@alcatel-lucent.co=
m</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"><div class=3D"">&gt; Well, if it&#39;s just for Timestamps=
, just use an &quot;invalid&quot; length of 2 as<br>

&gt; a &quot;Late TS permitted&quot;.<br>
&gt;<br>
&gt; Maintaining the same option value and this would be rather straight<br=
>
&gt; forward.<br>
<br>
</div>Interesting proposal. But actually we are not really short of option =
numbers. If we get e.g. 8 bytes in a significant number of SYNs (which is o=
ur real bottleneck), I think we could also spend a new option codepoint.<br=
>

<div class=3D""><br>
&gt; Personally, I&#39;d like to combine this with a change in semantics, i=
f<br>
&gt; SACK is negotiated on the same session as well, to allow more efficien=
t<br>
&gt; loss recovery (lost retransmission detection during a loss recovery<br=
>
&gt; episode). I should have a unpublished draft sitting around here about<=
br>
&gt; that aspect.<br>
<br>
</div>A change in semantics would also be a reason for a new codepoint, imh=
o.<br>
<span class=3D""><font color=3D"#888888"><br>
Michael<br>
</font></span><div class=3D""><div class=3D"h5"><br>
<br>
&gt; The TS option handling seems to be rather generous by middleboxes, in<=
br>
&gt; general.<br>
&gt;<br>
&gt; Richard Scheffenegger<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: tcpm [mailto:<a href=3D"mailto:tcpm-bounces@ietf.org">tcpm-=
bounces@ietf.org</a>] On Behalf Of Scharf,<br>
&gt; Michael<br>
&gt; &gt; (Michael)<br>
&gt; &gt; Sent: Dienstag, 27. Mai 2014 13:49<br>
&gt; &gt; To: Scheffenegger, Richard; Eggert, Lars; Joe Touch<br>
&gt; &gt; Cc: <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; &gt; Subject: Re: [tcpm] timestamp options (was Re: New Version<br>
&gt; Notification<br>
&gt; &gt; for draft-touch-tcpm-tcp-edo-01.txt)<br>
&gt; &gt;<br>
&gt; &gt; &gt; For late TS, we would still need at least 2 Bytes during<br>
&gt; &gt; &gt; &lt;SYN&gt;/&lt;SYN,ACK&gt; to allow that capability.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; However, as the useful values of Window Scale only cover 0..=
14 [1],<br>
&gt; &gt; &gt; one approach to save SYN option space would be to negotiate =
up to 4<br>
&gt; &gt; &gt; binary<br>
&gt; &gt; &gt; (on/off) plus WS in a new single 3-byte option.<br>
&gt; &gt;<br>
&gt; &gt; I might be wrong, but I&#39;ve thought so far that &quot;replacin=
g&quot; the window<br>
&gt; &gt; scale option would be pretty difficult. We already had reports of=
<br>
&gt; &gt; middleboxes that parse this option because they try to understand=
<br>
&gt; rwnd (or<br>
&gt; &gt; they do not parse the option even if they should, or they have<br=
>
&gt; entirely<br>
&gt; &gt; broken logic to mess up TCP...). I think one example are firewall=
s<br>
&gt; that<br>
&gt; &gt; try to detect whether segments are in-window. Another example cou=
ld<br>
&gt; be<br>
&gt; &gt; PEPs that try to tune rwnd (if they still exist in the wild).<br>
&gt; &gt;<br>
&gt; &gt; My understanding so far was that a solution only for timestamps w=
ould<br>
&gt; be<br>
&gt; &gt; the most low-hanging fruit.<br>
&gt; &gt;<br>
&gt; &gt; Michael<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; tcpm mailing list<br>
&gt; &gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
<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>
</div></div></blockquote></div><br></div></div></div>

--047d7b87371e848ef304fa65fac9--


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Date: Tue, 27 May 2014 13:04:19 -0700
From: Joe Touch <touch@isi.edu>
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To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>, "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com>	<655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com>	<20140503122950.GM44329@verdi>	<655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com>	<201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk>	<537E3ACD.5000308@isi.edu>	<537E48CE.8040704@mti-systems.com>	<537E66A7.4080907@isi.edu>	<201405231003.s4NA3PAB005137@bagheera.jungle.bt.co.uk>	<537F7D91.10802@isi.edu>	<F4BCB99F-6133-4F3C-BD5E-3369B979EB33@netapp.com>	<655C07320163294895BBADA28372AF5D305E00@FR712WXCHMBA15.zeu.alcatel-lucent.com>	<012C3117EDDB3C4781FD802A8C27DD4F6F3DD808@SACEXCMBX06-PRD.hq.netapp.com>	<655C07320163294895BBADA28372AF5D309C5D@FR712WXCHMBA15.zeu.alcatel-lucent.com>	<012C3117EDDB3C4781FD802A8C27DD4F6F3E09CF@SACEXCMBX06-PRD.hq.netapp.com>	<655C07320163294895BBADA28372AF5D309CC7@FR712WXCHMBA15.zeu.alcatel-lucent.com> <CAO249ycpc6BzxCQv=zhJzL3BQwmvNH68ec7UGN_5MD7U1_FeUw@mail.gmail.com>
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Subject: Re: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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Hi, all,

Late negotiation has too many problems, as have been repeatedly raised 
on this list:

	- prevents use of the timestamp to help address PAWS for
	the initial SYN

	- a new connection that either avoids TS in SYN or
	uses just "use TS" (two-byte) flag can't know when
	to engage the timestamp and what initial value to use

Joe

On 5/27/2014 11:36 AM, Yoshifumi Nishida wrote:
> Hi folks,
>
> I've submitted a proposal for this a while ago:
> http://tools.ietf.org/html/draft-nishida-tcpm-apaws-00
> Although it's expired now, I will submit a new version within a couple
> of weeks or so.
> The late negotiation is also discussed in the draft.
> As it will change the semantics of TS, it presumes to use new code point
> (or use timestamp field in SYN). But, I'm happy to discuss on this point.
>
> One tricky thing for RFC1323 is that it uses TS for two purposes:
> timestamping and PAWS and one purpose of the draft is to separate them.
> I appreciate any feedback on the draft.
>
> Thanks,
> --
> Yoshi
>
> On Tue, May 27, 2014 at 5:12 AM, Scharf, Michael (Michael)
> <michael.scharf@alcatel-lucent.com
> <mailto:michael.scharf@alcatel-lucent.com>> wrote:
>
>      > Well, if it's just for Timestamps, just use an "invalid" length
>     of 2 as
>      > a "Late TS permitted".
>      >
>      > Maintaining the same option value and this would be rather straight
>      > forward.
>
>     Interesting proposal. But actually we are not really short of option
>     numbers. If we get e.g. 8 bytes in a significant number of SYNs
>     (which is our real bottleneck), I think we could also spend a new
>     option codepoint.
>
>      > Personally, I'd like to combine this with a change in semantics, if
>      > SACK is negotiated on the same session as well, to allow more
>     efficient
>      > loss recovery (lost retransmission detection during a loss recovery
>      > episode). I should have a unpublished draft sitting around here about
>      > that aspect.
>
>     A change in semantics would also be a reason for a new codepoint, imho.
>
>     Michael
>
>
>      > The TS option handling seems to be rather generous by middleboxes, in
>      > general.
>      >
>      > Richard Scheffenegger
>      >
>      > > -----Original Message-----
>      > > From: tcpm [mailto:tcpm-bounces@ietf.org
>     <mailto:tcpm-bounces@ietf.org>] On Behalf Of Scharf,
>      > Michael
>      > > (Michael)
>      > > Sent: Dienstag, 27. Mai 2014 13:49
>      > > To: Scheffenegger, Richard; Eggert, Lars; Joe Touch
>      > > Cc: tcpm@ietf.org <mailto:tcpm@ietf.org>
>      > > Subject: Re: [tcpm] timestamp options (was Re: New Version
>      > Notification
>      > > for draft-touch-tcpm-tcp-edo-01.txt)
>      > >
>      > > > For late TS, we would still need at least 2 Bytes during
>      > > > <SYN>/<SYN,ACK> to allow that capability.
>      > > >
>      > > > However, as the useful values of Window Scale only cover
>     0..14 [1],
>      > > > one approach to save SYN option space would be to negotiate
>     up to 4
>      > > > binary
>      > > > (on/off) plus WS in a new single 3-byte option.
>      > >
>      > > I might be wrong, but I've thought so far that "replacing" the
>     window
>      > > scale option would be pretty difficult. We already had reports of
>      > > middleboxes that parse this option because they try to understand
>      > rwnd (or
>      > > they do not parse the option even if they should, or they have
>      > entirely
>      > > broken logic to mess up TCP...). I think one example are firewalls
>      > that
>      > > try to detect whether segments are in-window. Another example could
>      > be
>      > > PEPs that try to tune rwnd (if they still exist in the wild).
>      > >
>      > > My understanding so far was that a solution only for timestamps
>     would
>      > be
>      > > the most low-hanging fruit.
>      > >
>      > > Michael
>      > >
>      > > _______________________________________________
>      > > tcpm mailing list
>      > > tcpm@ietf.org <mailto:tcpm@ietf.org>
>      > > https://www.ietf.org/mailman/listinfo/tcpm
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>
>


From nobody Tue May 27 23:06:54 2014
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References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <537E48CE.8040704@mti-systems.com> <537E66A7.4080907@isi.edu> <201405231003.s4NA3PAB005137@bagheera.jungle.bt.co.uk> <537F7D91.10802@isi.edu> <F4BCB99F-6133-4F3C-BD5E-3369B979EB33@netapp.com> <655C07320163294895BBADA28372AF5D305E00@FR712WXCHMBA15.zeu.alcatel-lucent.com> <012C3117EDDB3C4781FD802A8C27DD4F6F3DD808@SACEXCMBX06-PRD.hq.netapp.com> <655C07320163294895BBADA28372AF5D309C5D@FR712WXCHMBA15.zeu.alcatel-lucent.com> <012C3117EDDB3C4781FD802A8C27DD4F6F3E09CF@SACEXCMBX06-PRD.hq.netapp.com> <655C07320163294895BBADA28372AF5D309CC7@FR712WXCHMBA15.zeu.alcatel-lucent.com> <CAO249ycpc6BzxCQv=zhJzL3BQwmvNH68ec7UGN_5MD7U1_FeUw@mail.gmail.com> <5384EFC3.50803@isi.edu>
Date: Tue, 27 May 2014 23:06:35 -0700
Message-ID: <CAO249yf6RCBtfOTmAYPt2KM7=pBD=XEDHmLTr+VyLeEOLAmwsQ@mail.gmail.com>
From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: Joe Touch <touch@isi.edu>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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--047d7b87371e5b69ff04fa6f9d11
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Hi Joe,

I'm wondering if you presume that using the existing code point for late
negotiation.
Suppose If we use new code point, do we still have problems you mentioned
below?

Thanks,
--
Yoshi

On Tue, May 27, 2014 at 1:04 PM, Joe Touch <touch@isi.edu> wrote:

> Hi, all,
>
> Late negotiation has too many problems, as have been repeatedly raised on
> this list:
>
>         - prevents use of the timestamp to help address PAWS for
>         the initial SYN
>
>         - a new connection that either avoids TS in SYN or
>         uses just "use TS" (two-byte) flag can't know when
>         to engage the timestamp and what initial value to use
>
> Joe
>
>
> On 5/27/2014 11:36 AM, Yoshifumi Nishida wrote:
>
>> Hi folks,
>>
>> I've submitted a proposal for this a while ago:
>> http://tools.ietf.org/html/draft-nishida-tcpm-apaws-00
>> Although it's expired now, I will submit a new version within a couple
>> of weeks or so.
>> The late negotiation is also discussed in the draft.
>> As it will change the semantics of TS, it presumes to use new code point
>> (or use timestamp field in SYN). But, I'm happy to discuss on this point.
>>
>> One tricky thing for RFC1323 is that it uses TS for two purposes:
>> timestamping and PAWS and one purpose of the draft is to separate them.
>> I appreciate any feedback on the draft.
>>
>> Thanks,
>> --
>> Yoshi
>>
>> On Tue, May 27, 2014 at 5:12 AM, Scharf, Michael (Michael)
>> <michael.scharf@alcatel-lucent.com
>> <mailto:michael.scharf@alcatel-lucent.com>> wrote:
>>
>>      > Well, if it's just for Timestamps, just use an "invalid" length
>>     of 2 as
>>      > a "Late TS permitted".
>>      >
>>      > Maintaining the same option value and this would be rather straight
>>      > forward.
>>
>>     Interesting proposal. But actually we are not really short of option
>>     numbers. If we get e.g. 8 bytes in a significant number of SYNs
>>     (which is our real bottleneck), I think we could also spend a new
>>     option codepoint.
>>
>>      > Personally, I'd like to combine this with a change in semantics, if
>>      > SACK is negotiated on the same session as well, to allow more
>>     efficient
>>      > loss recovery (lost retransmission detection during a loss recovery
>>      > episode). I should have a unpublished draft sitting around here
>> about
>>      > that aspect.
>>
>>     A change in semantics would also be a reason for a new codepoint,
>> imho.
>>
>>     Michael
>>
>>
>>      > The TS option handling seems to be rather generous by middleboxes,
>> in
>>      > general.
>>      >
>>      > Richard Scheffenegger
>>      >
>>      > > -----Original Message-----
>>      > > From: tcpm [mailto:tcpm-bounces@ietf.org
>>     <mailto:tcpm-bounces@ietf.org>] On Behalf Of Scharf,
>>      > Michael
>>      > > (Michael)
>>      > > Sent: Dienstag, 27. Mai 2014 13:49
>>      > > To: Scheffenegger, Richard; Eggert, Lars; Joe Touch
>>      > > Cc: tcpm@ietf.org <mailto:tcpm@ietf.org>
>>      > > Subject: Re: [tcpm] timestamp options (was Re: New Version
>>      > Notification
>>      > > for draft-touch-tcpm-tcp-edo-01.txt)
>>      > >
>>      > > > For late TS, we would still need at least 2 Bytes during
>>      > > > <SYN>/<SYN,ACK> to allow that capability.
>>      > > >
>>      > > > However, as the useful values of Window Scale only cover
>>     0..14 [1],
>>      > > > one approach to save SYN option space would be to negotiate
>>     up to 4
>>      > > > binary
>>      > > > (on/off) plus WS in a new single 3-byte option.
>>      > >
>>      > > I might be wrong, but I've thought so far that "replacing" the
>>     window
>>      > > scale option would be pretty difficult. We already had reports of
>>      > > middleboxes that parse this option because they try to understand
>>      > rwnd (or
>>      > > they do not parse the option even if they should, or they have
>>      > entirely
>>      > > broken logic to mess up TCP...). I think one example are
>> firewalls
>>      > that
>>      > > try to detect whether segments are in-window. Another example
>> could
>>      > be
>>      > > PEPs that try to tune rwnd (if they still exist in the wild).
>>      > >
>>      > > My understanding so far was that a solution only for timestamps
>>     would
>>      > be
>>      > > the most low-hanging fruit.
>>      > >
>>      > > Michael
>>      > >
>>      > > _______________________________________________
>>      > > tcpm mailing list
>>      > > tcpm@ietf.org <mailto:tcpm@ietf.org>
>>
>>      > > https://www.ietf.org/mailman/listinfo/tcpm
>>
>>     _______________________________________________
>>     tcpm mailing list
>>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/tcpm
>>
>>
>>

--047d7b87371e5b69ff04fa6f9d11
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Joe,<div><br></div><div>I&#39;m wondering if you presum=
e that using the existing code point for late negotiation.</div><div>Suppos=
e If we use new code point, do we still have problems you mentioned below?<=
/div>
<div><br></div><div>Thanks,</div><div>--</div><div>Yoshi<div class=3D"gmail=
_extra"><br><div class=3D"gmail_quote">On Tue, May 27, 2014 at 1:04 PM, Joe=
 Touch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" target=3D"_bl=
ank">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">Hi, all,<br>
<br>
Late negotiation has too many problems, as have been repeatedly raised on t=
his list:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 - prevents use of the timestamp to help address=
 PAWS for<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 the initial SYN<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 - a new connection that either avoids TS in SYN=
 or<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 uses just &quot;use TS&quot; (two-byte) flag ca=
n&#39;t know when<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 to engage the timestamp and what initial value =
to use<br>
<br>
Joe<div class=3D""><br>
<br>
On 5/27/2014 11:36 AM, Yoshifumi Nishida 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"">
Hi folks,<br>
<br>
I&#39;ve submitted a proposal for this a while ago:<br>
<a href=3D"http://tools.ietf.org/html/draft-nishida-tcpm-apaws-00" target=
=3D"_blank">http://tools.ietf.org/html/<u></u>draft-nishida-tcpm-apaws-00</=
a><br>
Although it&#39;s expired now, I will submit a new version within a couple<=
br>
of weeks or so.<br>
The late negotiation is also discussed in the draft.<br>
As it will change the semantics of TS, it presumes to use new code point<br=
>
(or use timestamp field in SYN). But, I&#39;m happy to discuss on this poin=
t.<br>
<br>
One tricky thing for RFC1323 is that it uses TS for two purposes:<br>
timestamping and PAWS and one purpose of the draft is to separate them.<br>
I appreciate any feedback on the draft.<br>
<br>
Thanks,<br>
--<br>
Yoshi<br>
<br>
On Tue, May 27, 2014 at 5:12 AM, Scharf, Michael (Michael)<br>
&lt;<a href=3D"mailto:michael.scharf@alcatel-lucent.com" target=3D"_blank">=
michael.scharf@alcatel-<u></u>lucent.com</a><br></div><div><div class=3D"h5=
">
&lt;mailto:<a href=3D"mailto:michael.scharf@alcatel-lucent.com" target=3D"_=
blank">michael.scharf@<u></u>alcatel-lucent.com</a>&gt;&gt; wrote:<br>
<br>
=C2=A0 =C2=A0 =C2=A0&gt; Well, if it&#39;s just for Timestamps, just use an=
 &quot;invalid&quot; length<br>
=C2=A0 =C2=A0 of 2 as<br>
=C2=A0 =C2=A0 =C2=A0&gt; a &quot;Late TS permitted&quot;.<br>
=C2=A0 =C2=A0 =C2=A0&gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; Maintaining the same option value and this would b=
e rather straight<br>
=C2=A0 =C2=A0 =C2=A0&gt; forward.<br>
<br>
=C2=A0 =C2=A0 Interesting proposal. But actually we are not really short of=
 option<br>
=C2=A0 =C2=A0 numbers. If we get e.g. 8 bytes in a significant number of SY=
Ns<br>
=C2=A0 =C2=A0 (which is our real bottleneck), I think we could also spend a=
 new<br>
=C2=A0 =C2=A0 option codepoint.<br>
<br>
=C2=A0 =C2=A0 =C2=A0&gt; Personally, I&#39;d like to combine this with a ch=
ange in semantics, if<br>
=C2=A0 =C2=A0 =C2=A0&gt; SACK is negotiated on the same session as well, to=
 allow more<br>
=C2=A0 =C2=A0 efficient<br>
=C2=A0 =C2=A0 =C2=A0&gt; loss recovery (lost retransmission detection durin=
g a loss recovery<br>
=C2=A0 =C2=A0 =C2=A0&gt; episode). I should have a unpublished draft sittin=
g around here about<br>
=C2=A0 =C2=A0 =C2=A0&gt; that aspect.<br>
<br>
=C2=A0 =C2=A0 A change in semantics would also be a reason for a new codepo=
int, imho.<br>
<br>
=C2=A0 =C2=A0 Michael<br>
<br>
<br>
=C2=A0 =C2=A0 =C2=A0&gt; The TS option handling seems to be rather generous=
 by middleboxes, in<br>
=C2=A0 =C2=A0 =C2=A0&gt; general.<br>
=C2=A0 =C2=A0 =C2=A0&gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; Richard Scheffenegger<br>
=C2=A0 =C2=A0 =C2=A0&gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; -----Original Message-----<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; From: tcpm [mailto:<a href=3D"mailto:tcpm-bou=
nces@ietf.org" target=3D"_blank">tcpm-bounces@ietf.org</a><br>
=C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailto:tcpm-bounces@ietf.org" target=3D=
"_blank">tcpm-bounces@ietf.org</a>&gt;<u></u>] On Behalf Of Scharf,<br>
=C2=A0 =C2=A0 =C2=A0&gt; Michael<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; (Michael)<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; Sent: Dienstag, 27. Mai 2014 13:49<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; To: Scheffenegger, Richard; Eggert, Lars; Joe=
 Touch<br></div></div><div><div class=3D"h5">
=C2=A0 =C2=A0 =C2=A0&gt; &gt; Cc: <a href=3D"mailto:tcpm@ietf.org" target=
=3D"_blank">tcpm@ietf.org</a> &lt;mailto:<a href=3D"mailto:tcpm@ietf.org" t=
arget=3D"_blank">tcpm@ietf.org</a>&gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; Subject: Re: [tcpm] timestamp options (was Re=
: New Version<br>
=C2=A0 =C2=A0 =C2=A0&gt; Notification<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; for draft-touch-tcpm-tcp-edo-01.<u></u>txt)<b=
r>
=C2=A0 =C2=A0 =C2=A0&gt; &gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt; For late TS, we would still need at leas=
t 2 Bytes during<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt; &lt;SYN&gt;/&lt;SYN,ACK&gt; to allow tha=
t capability.<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt; However, as the useful values of Window =
Scale only cover<br>
=C2=A0 =C2=A0 0..14 [1],<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt; one approach to save SYN option space wo=
uld be to negotiate<br>
=C2=A0 =C2=A0 up to 4<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt; binary<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; &gt; (on/off) plus WS in a new single 3-byte =
option.<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; I might be wrong, but I&#39;ve thought so far=
 that &quot;replacing&quot; the<br>
=C2=A0 =C2=A0 window<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; scale option would be pretty difficult. We al=
ready had reports of<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; middleboxes that parse this option because th=
ey try to understand<br>
=C2=A0 =C2=A0 =C2=A0&gt; rwnd (or<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; they do not parse the option even if they sho=
uld, or they have<br>
=C2=A0 =C2=A0 =C2=A0&gt; entirely<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; broken logic to mess up TCP...). I think one =
example are firewalls<br>
=C2=A0 =C2=A0 =C2=A0&gt; that<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; try to detect whether segments are in-window.=
 Another example could<br>
=C2=A0 =C2=A0 =C2=A0&gt; be<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; PEPs that try to tune rwnd (if they still exi=
st in the wild).<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; My understanding so far was that a solution o=
nly for timestamps<br>
=C2=A0 =C2=A0 would<br>
=C2=A0 =C2=A0 =C2=A0&gt; be<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; the most low-hanging fruit.<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; Michael<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; ______________________________<u></u>________=
_________<br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; tcpm mailing list<br></div></div>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_b=
lank">tcpm@ietf.org</a> &lt;mailto:<a href=3D"mailto:tcpm@ietf.org" target=
=3D"_blank">tcpm@ietf.org</a>&gt;<div class=3D""><br>
=C2=A0 =C2=A0 =C2=A0&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listi=
nfo/tcpm" target=3D"_blank">https://www.ietf.org/mailman/<u></u>listinfo/tc=
pm</a><br>
<br>
=C2=A0 =C2=A0 ______________________________<u></u>_________________<br>
=C2=A0 =C2=A0 tcpm mailing list<br></div>
=C2=A0 =C2=A0 <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.=
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To: Joe Touch <touch@isi.edu>, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>, "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Joe,

At 18:14 23/05/2014, Joe Touch wrote:


>On 5/23/2014 5:13 AM, Bob Briscoe wrote:
>>David,
>>
>>1) Parallel control channel
>>___________________________
>>Client A sends two SYNs back-to-back to an existing well-known port
>>(e.g. 80).
>
>You can send in whatever order you want; packets will be reordered, 
>lost, and sent along alternate paths.

Of course.

But as I suggested initially, we can standardise the protocol so that 
an upgraded host synthesises shared fate. Eg. a 2-octet TCP option on 
the D-type SYN that says "Hold for x [ms] to wait for the 
supplementary C-type SYN, where x is a lot less than the usual time 
you hold SYN connection state (e.g. x=2). If SYN C hasn't arrived by 
then, continue without it, and discard it if it arrives."

And if the C-SYN arrives first, hold it for y [ms] waiting for the 
corresponding D-SYN. Where for example y=2 as well.

>FWIW, do these use the same source port and ISN?
>         - if they do, it'll reset the connection
>         - if they don't, you're now limiting the number of
>         concurrent connections to roughly half:
>         http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/

Unless we can think of a way for the C-SYN to be discarded by a 
legacy TCP stack (but not a middlebox), I'm assuming we need to get 
the C-SYN up to the legacy app-layer before discard... So, I was 
assuming they use different source ports (otherwise, as you say, a 
legacy TCP stack could reset the D SYN connection if it arrived second)

The ISN on the C-SYN is redundant - we might be able to think of 
another use for that field.

On an upgraded host, max concurrent connections would only be 
slightly impacted, because of the v small timeout of C-SYNs.

The semantics of the option-space-extension option would be to only 
hold the C-type SYN for a timeout and only the D-type SYN creates 
full connection state (I'm deferring SYN cookie behaviour to tomorrow 
for now - this is a straw man).

Admittedly, the number of concurrent connections a /legacy/ host can 
support could reduce by up to half (if all remote TCP clients are 
sending the new option). But they have the choice of upgrading to the 
new stack to stop wasting their memory.


>>* SYN D, establishes a regular data connection, with sufficient TCP
>>options to be workable but they still fit within the existing 40B option
>>limit.
>>* SYN C establishes another parallel connection to the same well-known
>>port that looks like regular data from the outside (it could even be an
>>extension to HTTP to ensure middleboxes will let it pass), but it talks
>>a new app-layer 'TCP control' protocol inside.
>
>What happens when they arrive out of order? What happens when you 
>get D but not yet C? How long do you wait for C?

See above.
The timeouts might be standardised, or they might be declared in the 
option as a hint (for the host to ignore if it is under stress).


>This is the problem with dual-stack approaches - new endpoints 
>penalize legacy endpoints if there's a stall, and undermine new 
>endpoints if they don't.

Have I satisfied you that this can be solved sufficiently? Bearing in 
mind the gain, it's reasonable to have to accept some pain.


>>If there is no support for the new app-layer protocol on port 80 the
>>control channel just shuts down with a suitable HTTP error, while SYN D
>>has opened a data connection with sufficient TCP options to be workable.
>>If the new app-layer TCP control protocol is supported on port 80, the
>>parallel control channel (C) adds unlimited additional control
>>flexibility to the data channel (D) hardly any added latency.
>>
>>Establishing a similar control channel in the opposite direction would
>>be fairly trivial.
>>
>>There are few, if any, middlebox problems with the above approach.
>>However, there are certainly other problems, but no more insurmountable
>>than all the problems that have already been discussed with taking the
>>'easy' route of EDO:
>>* A secure binding would have to be added to bind channel C to a secret
>>known only to the originator of channel D, otherwise it would open up
>>data channels to spoof control channel attacks. This binding could be
>>built on a TCP-AO option in channel D.
>
>Yes, that's another problem.

Fairly straightforward to solve using standard techniques.


>>* Channel C would need some way to refer to the segments of channel D
>>that was robust against re-segmentation.
>
>Which means it won't work in the current Internet, because 
>resegmentation is also widespread (though evil, IMO).

Well, resegmentation isn't usually a problem on a SYN anyway.

We can't improve on the general pain caused by resegmentation. I was 
only talking about the delta pain that my strawman would suffer, 
where a resegmenting function sees data on my SYN and doesn't 
understand my strawman, so it won't patch up the damage in my 
strawman protocol that it would have patched up by altering sequence 
numbers in a regular single SYN with a payload.

I think this is a corner case.


>>* The main problem is that the two channels don't share fate;a control
>>packet can be delayed relative to the point in the data stream at which
>>it is attempting to exert control, possibly for a RTT if it is lost and
>>has to be retransmitted. However, this is not insurmountable. The
>>control protocol could include a mode to "synthesise shared fate", by
>>making the data channel buffer data until an associated control segment
>>had arrived. This would duplicate the latency impact of a loss or delay
>>on either channel, but one can imagine mitigations that would consign
>>this latency impact to corner cases.
>>* It's a bit of a mess, but that comes with the territory when trying to
>>fix legacy protocol problems.
>>* The internal stack architecture seems to require a trombone back down
>>into the kernel from user-space, but that is not insurmountable - a shim
>>within the kernel on port 80 (for example) could redirect control
>>channel data across to the "TCP control channel module" in the kernel,
>>while passing non-control channel connections to user-space.
>>
>>2) Build on LOIC
>>______________________
>>Long option with invalid checksum <draft-yourtchenko-tcp-loic-00>
>
>Won't work through current NATs, which won't recalculate the 
>checksum properly.

I'm building on the general idea of using something invalid, not 
using the checksum idea specifically.



>>At 18:53 22/05/2014, John Leslie wrote:
>>>    That's too big of a change to ask folks to believe it safe.
>>
>>When I read an idea, I don't take it as set in stone and just find a
>>hole and dismiss it. I see it as a potential stepping stone to a
>>solution and think about how it could be done better. In fact, Andrew
>>Yourtchenko said that was the intention of his write-up of LOIC.
>>
>>I believe that an approach worth further thought would be a mixture of
>>the control channel idea and the invalid checksum idea. I'm thinking of:
>>* a pure control SYN (C) sent first, then a base SYN (D) sent
>>back-to-back, both to the same port.
>
>Again, please don't assume back-to-back means anything.

See above.
Actually, even tho order isn'[t guaranteed, it is important to get 
them in the right order to optimise performance (much as TCP doesn't 
assume perfect order, but it goes smoothest with reasonable ordering).

>>* SYN C would contain something invalid to cause a legacy TCP stack or
>>legacy app to discard it (and hopefully less probability that a
>>middlebox would), e.g. a payload that is invalid for the application
>>protocol on the port.
>
>But so will a NAT, etc.

No. You can design a payload that has headers that a NAT will ignore 
but an end-host will have to process. E.g. HTTP connection control 
headers newly defined for this protocol that would be ignored by 
NATs, without any other HTTP behaviour, so a legacy host does nothing 
at the app layer.

>>* there would be additional TCP options in the payload of SYN C to be
>>added to the TCP options that arrived separately on the base SYN
>>* The control SYN could be bound crytographically to the base SYN (as
>>already described).
>>* It could use the shim-like control stack arangement described earlier.
>>
>>By focusing solely on extending the SYN, this would avoid the ongoing
>>shared fate problems that a separate control channel suffers throughout
>>the connection. There would still be shared fate problems with 2 SYNs
>>(e.g. the two SYNs get re-ordered), but the protocol would have to be
>>designed to be robust to that (naively, SYN D could include a new TCP
>>option that told a new stack to wait a few ticks for a SYN C, but that
>>would be vulnerable to meddleboxes). Not insurmountable.
>
>AFAICT, it is.

Still?


Bob



>*with a nod to Raiders 3.

________________________________________________________________
Bob Briscoe,                                                  BT 


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Joe,

I don't think this sufficiently answers the question to justify WG 
adoption. You seem to be confirming that this is an academic exercise.


Bob

At 17:55 23/05/2014, Joe Touch wrote:
>Hi, Bob,
>
>On 5/23/2014 3:03 AM, Bob Briscoe wrote:
>>Joe, and everyone else who wants to work on this,
>>
>>Just because it's easier to make a chocolate teapot than a cast-iron
>>one, doesn't imply that there is any need for chocolate teapots.
>
>You don't get a cast iron teapot just because you want one either ;-)
>
>>IOW, we will be asking the IESG to spend reviewer time on EDO, so we
>>need to give some plausible indication that someone might find it useful
>>and it's not just an academic exercise.
>
>Sometimes the answer "you can't have A, but at least here's B" is 
>more than an exercise; it educates the community. By not providing 
>either answer, we have continued to drag this issue around the block 
>for far too long -- and spent far too many cycles in this and other 
>WGs seeking solutions.
>
> > The current draft solely gives
>>SACK + MPTCP + TCP-AO as an example, but is that really something that
>>can't be done today?
>
>Current total for SYN options in widespread concurrent use (as 
>already described in sec 6.4):
>
>         2       SACK permitted
>         10      timestamp
>         3       window scale
>         4       MSS
>         ------------------
>         11 bytes
>
>The current DO field is 4 bits, with a max value of 15 = 60 bytes 
>for the total header, less 20 for the base TCP header which leaves 
>40 for options.
>
>So let's see what happens when we add:
>
>         11      widespread basic options
>         16      TCP-AO
>         20      MPTCP
>         --------------------
>         47
>
>That's more than 40.
>
>>Adding more complexity to the TCP stack (with the potential for more
>>vulnerabilities) is only worthwhile if there's an identifiable benefit,
>>otherwise few production stacks are going to implement it anyway.
>
>There are two identifiable benefits:
>
>         1) explain the ways we already know we can't extend the SYN
>         so we stop wasting time trying them repeatedly (i.e., education)
>
>         2) provide a solution for the other segments, so that can be
>         used - e.g., for large SACK responses
>
>         3) educate the community
>
>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 


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Date: Wed, 28 May 2014 10:30:55 -0700
From: Joe Touch <touch@isi.edu>
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To: Bob Briscoe <bob.briscoe@bt.com>
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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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On 5/28/2014 10:16 AM, Bob Briscoe wrote:
> Joe,
>
> I don't think this sufficiently answers the question to justify WG
> adoption. You seem to be confirming that this is an academic exercise.

You asked for a specific example - one that the MPTCP community has 
raised. It's not a proof that there's "some set" that might overload the 
space; it's based on a *specific* request.

I gave other reasons - including educating the community as to the 
issues. If you want to call that "academic", sure - in part.

But I don't see that as interfering with WG adoption - we do that sort 
of thing all the time (that's the basis of a BCP).

Joe

> At 17:55 23/05/2014, Joe Touch wrote:
>> Hi, Bob,
>>
>> On 5/23/2014 3:03 AM, Bob Briscoe wrote:
>>> Joe, and everyone else who wants to work on this,
>>>
>>> Just because it's easier to make a chocolate teapot than a cast-iron
>>> one, doesn't imply that there is any need for chocolate teapots.
>>
>> You don't get a cast iron teapot just because you want one either ;-)
>>
>>> IOW, we will be asking the IESG to spend reviewer time on EDO, so we
>>> need to give some plausible indication that someone might find it useful
>>> and it's not just an academic exercise.
>>
>> Sometimes the answer "you can't have A, but at least here's B" is more
>> than an exercise; it educates the community. By not providing either
>> answer, we have continued to drag this issue around the block for far
>> too long -- and spent far too many cycles in this and other WGs
>> seeking solutions.
>>
>> > The current draft solely gives
>>> SACK + MPTCP + TCP-AO as an example, but is that really something that
>>> can't be done today?
>>
>> Current total for SYN options in widespread concurrent use (as already
>> described in sec 6.4):
>>
>>         2       SACK permitted
>>         10      timestamp
>>         3       window scale
>>         4       MSS
>>         ------------------
>>         11 bytes
>>
>> The current DO field is 4 bits, with a max value of 15 = 60 bytes for
>> the total header, less 20 for the base TCP header which leaves 40 for
>> options.
>>
>> So let's see what happens when we add:
>>
>>         11      widespread basic options
>>         16      TCP-AO
>>         20      MPTCP
>>         --------------------
>>         47
>>
>> That's more than 40.
>>
>>> Adding more complexity to the TCP stack (with the potential for more
>>> vulnerabilities) is only worthwhile if there's an identifiable benefit,
>>> otherwise few production stacks are going to implement it anyway.
>>
>> There are two identifiable benefits:
>>
>>         1) explain the ways we already know we can't extend the SYN
>>         so we stop wasting time trying them repeatedly (i.e., education)
>>
>>         2) provide a solution for the other segments, so that can be
>>         used - e.g., for large SACK responses
>>
>>         3) educate the community
>>
>> Joe
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT


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Date: Wed, 28 May 2014 10:58:17 -0700
From: Joe Touch <touch@isi.edu>
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On 5/28/2014 10:13 AM, Bob Briscoe wrote:
> Joe,
>
> At 18:14 23/05/2014, Joe Touch wrote:
>
>
>> On 5/23/2014 5:13 AM, Bob Briscoe wrote:
>>> David,
>>>
>>> 1) Parallel control channel
>>> ___________________________
>>> Client A sends two SYNs back-to-back to an existing well-known port
>>> (e.g. 80).
>>
>> You can send in whatever order you want; packets will be reordered,
>> lost, and sent along alternate paths.
>
> Of course.
>
> But as I suggested initially, we can standardise the protocol so that an
> upgraded host synthesises shared fate.

Fate is dependent on path.

> Eg. a 2-octet TCP option on the
> D-type SYN that says "Hold for x [ms] to wait for the supplementary
> C-type SYN, where x is a lot less than the usual time you hold SYN
> connection state (e.g. x=2). If SYN C hasn't arrived by then, continue
> without it, and discard it if it arrives."
 >
> And if the C-SYN arrives first, hold it for y [ms] waiting for the
> corresponding D-SYN. Where for example y=2 as well.

(C=legacy; D=extended - from your earlier email, for context)

Legacy clients now experience connection delays when contacting upgraded 
servers - why is that acceptable?

FWIW, many endpoints don't hold SYN state at all if they implement SYN 
cookies, so 2ms would end up overloading the server with state much 
larger than it currently keeps.

>> FWIW, do these use the same source port and ISN?
>>         - if they do, it'll reset the connection
>>         - if they don't, you're now limiting the number of
>>         concurrent connections to roughly half:
>>         http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/
>
> Unless we can think of a way for the C-SYN to be discarded by a legacy
> TCP stack (but not a middlebox), I'm assuming we need to get the C-SYN
> up to the legacy app-layer before discard... So, I was assuming they use
> different source ports (otherwise, as you say, a legacy TCP stack could
> reset the D SYN connection if it arrived second)

If they use different source ports, you could be limiting the connection 
rate. It's not enough to just "discard" late D-SYNs; they could be for 
new connections (how do you know the two are paired?), which means you 
need to RST them and keep state for 2MSL.

> The ISN on the C-SYN is redundant - we might be able to think of another
> use for that field.

But it's not redundant if you're talking to a legacy server, so you 
can't do anything with that field.

> On an upgraded host, max concurrent connections would only be slightly
> impacted, because of the v small timeout of C-SYNs.

See above; you need to know how to pair C-SYNs and D-SYNs. Consider that 
legacy hosts and upgraded hosts might be behind the same NAT, at which 
point how do you know the two SYNs are even from the same host?

> The semantics of the option-space-extension option would be to only hold
> the C-type SYN for a timeout and only the D-type SYN creates full
> connection state (I'm deferring SYN cookie behaviour to tomorrow for now
> - this is a straw man).

I don't think you can defer a widely-deployed feature like SYN cookies, 
and that seems to kill this.

> Admittedly, the number of concurrent connections a /legacy/ host can
> support could reduce by up to half (if all remote TCP clients are
> sending the new option). But they have the choice of upgrading to the
> new stack to stop wasting their memory.

Both legacy clients and legacy servers are impacted. We already know 
that servers are overloaded with connections, so halving that number 
seems like a non-starter. As does adding x ms delay for *each* legacy 
client.

>>> * SYN D, establishes a regular data connection, with sufficient TCP
>>> options to be workable but they still fit within the existing 40B option
>>> limit.
>>> * SYN C establishes another parallel connection to the same well-known
>>> port that looks like regular data from the outside (it could even be an
>>> extension to HTTP to ensure middleboxes will let it pass), but it talks
>>> a new app-layer 'TCP control' protocol inside.
>>
>> What happens when they arrive out of order? What happens when you get
>> D but not yet C? How long do you wait for C?
>
> See above.
> The timeouts might be standardised, or they might be declared in the
> option as a hint (for the host to ignore if it is under stress).
 >
  >> This is the problem with dual-stack approaches - new endpoints
>> penalize legacy endpoints if there's a stall, and undermine new
>> endpoints if they don't.
>
> Have I satisfied you that this can be solved sufficiently? Bearing in
> mind the gain, it's reasonable to have to accept some pain.

The pain is the problem. The smaller the timeout, the smaller the pain 
to legacy clients but the lower probability the extension will be useful 
to upgraded clients. The method cuts the rate of connections for legacy 
servers in half - which is *huge*. And it adds to the state overhead of 
the upgraded clients that need to do double the work for *every* connection.

Ultimately, it's roughly equivalent to "try both and shut down the one 
you don't want" -- the dual-stack approach we already know about.

>>> If there is no support for the new app-layer protocol on port 80 the
>>> control channel just shuts down with a suitable HTTP error, while SYN D
>>> has opened a data connection with sufficient TCP options to be workable.
>>> If the new app-layer TCP control protocol is supported on port 80, the
>>> parallel control channel (C) adds unlimited additional control
>>> flexibility to the data channel (D) hardly any added latency.
>>>
>>> Establishing a similar control channel in the opposite direction would
>>> be fairly trivial.
>>>
>>> There are few, if any, middlebox problems with the above approach.
>>> However, there are certainly other problems, but no more insurmountable
>>> than all the problems that have already been discussed with taking the
>>> 'easy' route of EDO:
>>> * A secure binding would have to be added to bind channel C to a secret
>>> known only to the originator of channel D, otherwise it would open up
>>> data channels to spoof control channel attacks. This binding could be
>>> built on a TCP-AO option in channel D.
>>
>> Yes, that's another problem.
>
> Fairly straightforward to solve using standard techniques.

I think that warrants more than a claim. You need to explain how you 
avoid false postives through NATs.

>>> * Channel C would need some way to refer to the segments of channel D
>>> that was robust against re-segmentation.
>>
>> Which means it won't work in the current Internet, because
>> resegmentation is also widespread (though evil, IMO).
>
> Well, resegmentation isn't usually a problem on a SYN anyway.
>
> We can't improve on the general pain caused by resegmentation. I was
> only talking about the delta pain that my strawman would suffer, where a
> resegmenting function sees data on my SYN and doesn't understand my
> strawman, so it won't patch up the damage in my strawman protocol that
> it would have patched up by altering sequence numbers in a regular
> single SYN with a payload.

You're right, in the sense that SYN data would typically be discarded 
anyway at the endpoint if it implemented SYN cookies.

> I think this is a corner case.

It is, but you raised it in a general sense; were you thinking of it in 
terms of just SYNs or other segments?

>>> * The main problem is that the two channels don't share fate;a control
>>> packet can be delayed relative to the point in the data stream at which
>>> it is attempting to exert control, possibly for a RTT if it is lost and
>>> has to be retransmitted. However, this is not insurmountable. The
>>> control protocol could include a mode to "synthesise shared fate", by
>>> making the data channel buffer data until an associated control segment
>>> had arrived. This would duplicate the latency impact of a loss or delay
>>> on either channel, but one can imagine mitigations that would consign
>>> this latency impact to corner cases.
>>> * It's a bit of a mess, but that comes with the territory when trying to
>>> fix legacy protocol problems.
>>> * The internal stack architecture seems to require a trombone back down
>>> into the kernel from user-space, but that is not insurmountable - a shim
>>> within the kernel on port 80 (for example) could redirect control
>>> channel data across to the "TCP control channel module" in the kernel,
>>> while passing non-control channel connections to user-space.
>>>
>>> 2) Build on LOIC
>>> ______________________
>>> Long option with invalid checksum <draft-yourtchenko-tcp-loic-00>
>>
>> Won't work through current NATs, which won't recalculate the checksum
>> properly.
>
> I'm building on the general idea of using something invalid, not using
> the checksum idea specifically.

Anything considered invalid by an endpoint might be checked by an 
overzealous midbox.

>>> At 18:53 22/05/2014, John Leslie wrote:
>>>>    That's too big of a change to ask folks to believe it safe.
>>>
>>> When I read an idea, I don't take it as set in stone and just find a
>>> hole and dismiss it. I see it as a potential stepping stone to a
>>> solution and think about how it could be done better. In fact, Andrew
>>> Yourtchenko said that was the intention of his write-up of LOIC.
>>>
>>> I believe that an approach worth further thought would be a mixture of
>>> the control channel idea and the invalid checksum idea. I'm thinking of:
>>> * a pure control SYN (C) sent first, then a base SYN (D) sent
>>> back-to-back, both to the same port.
>>
>> Again, please don't assume back-to-back means anything.
>
> See above.
> Actually, even tho order isn'[t guaranteed, it is important to get them
> in the right order to optimise performance (much as TCP doesn't assume
> perfect order, but it goes smoothest with reasonable ordering).

Right, but I want to emphasize that things emitted back-to-back could 
end up being delivered days later (e.g., when a link goes down, some 
routers just keep the queues there). The impact in those cases needs to 
be considered.

>>> * SYN C would contain something invalid to cause a legacy TCP stack or
>>> legacy app to discard it (and hopefully less probability that a
>>> middlebox would), e.g. a payload that is invalid for the application
>>> protocol on the port.
>>
>> But so will a NAT, etc.
>
> No. You can design a payload that has headers that a NAT will ignore but
> an end-host will have to process.

You can try, but if an end-host looks at it you can be sure that a 
midbox will do the same - if not today, then soon.

 > E.g. HTTP connection control headers
> newly defined for this protocol that would be ignored by NATs, without
> any other HTTP behaviour, so a legacy host does nothing at the app layer.

Those are rewritten by "helpful" midboxes all the time, FWIW.

>>> * there would be additional TCP options in the payload of SYN C to be
>>> added to the TCP options that arrived separately on the base SYN
>>> * The control SYN could be bound crytographically to the base SYN (as
>>> already described).
>>> * It could use the shim-like control stack arangement described earlier.
>>>
>>> By focusing solely on extending the SYN, this would avoid the ongoing
>>> shared fate problems that a separate control channel suffers throughout
>>> the connection. There would still be shared fate problems with 2 SYNs
>>> (e.g. the two SYNs get re-ordered), but the protocol would have to be
>>> designed to be robust to that (naively, SYN D could include a new TCP
>>> option that told a new stack to wait a few ticks for a SYN C, but that
>>> would be vulnerable to meddleboxes). Not insurmountable.
>>
>> AFAICT, it is.
>
> Still?

If it means halving the performance of legacy servers and delaying 
legacy clients, yes.

Joe


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From: Yuchung Cheng <ycheng@google.com>
Date: Wed, 28 May 2014 20:31:01 -0700
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To: Erik Hugne <erik.hugne@ericsson.com>
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Cc: "ingemar.s.johansson@ericsson.com" <ingemar.s.johansson@ericsson.com>, "martin.isaksson@ericsson.com" <martin.isaksson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>, Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
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On Mon, May 26, 2014 at 2:12 AM, Erik Hugne <erik.hugne@ericsson.com> wrote:
> Here's from a test just this morning, wgetting a 10MB file.
> https://www.dropbox.com/sh/86u3nzcwukjzpzv/AAAlp_Me7wXf7PRyuWFq9Lmwa

The problem is neither ABC nor SACK. It's bad interaction of
DSACK/undo, F-RTO, and rate-halving in the old kernel. These features
have gone through major changes since 3.0 (F-RTO re-implementation,
PRR, dozens fixes of TCP undo) so I'd recommend trying 3.12+ kernel as
well.

00:00:00.000026 IP s > c: Flags [.], seq 83359754:83368442, ack 145,
win 122, options [nop,nop,TS val 1294920005 ecr 1943986835], length
8688

// Timeout & retransmit the head
00:00:01.806873 IP s > c: Flags [.], seq 80269362:80270810, ack 145,
win 122, options [nop,nop,TS val 1294920482 ecr 1943986836], length
1448

// a DSACK indicating the timeout is spurious, however cwnd is not
reverted and F-RTO is activated to send a new data packet
00:00:00.305013 IP c > s: Flags [.], ack 80270810, win 6132, options
[nop,nop,TS val 1943987388 ecr 1294920482,nop,nop,sack 1
{80269362:80270810}], length 0
00:00:00.000070 IP s > c: Flags [.], seq 83368442:83372786, ack 145,
win 122, options [nop,nop,TS val 1294920558 ecr 1943987388], length
4344

// On receiving the SACK for the F-RTO probe, we got "stuck" in the
stop and go, possibly a bug in rate-halving/loss-recovery/f-rto?
00:00:00.200690 IP c > s: Flags [.], ack 80270810, win 6132, options
[nop,nop,TS val 1943987438 ecr 1294920482,nop,nop,sack 1
{83368442:83372786}], length 0
00:00:00.000053 IP s > c: Flags [.], seq 80270810:80272258, ack 145,
win 122, options [nop,nop,TS val 1294920608 ecr 1943987438], length
1448
00:00:00.000009 IP s > c: Flags [.], seq 80272258:80273706, ack 145,
win 122, options [nop,nop,TS val 1294920608 ecr 1943987438], length
1448
00:00:00.000003 IP s > c: Flags [.], seq 80273706:80275154, ack 145,
win 122, options [nop,nop,TS val 1294920608 ecr 1943987438], length
1448
00:00:00.200629 IP c > s: Flags [.], ack 80275154, win 6126, options
[nop,nop,TS val 1943987488 ecr 1294920608,nop,nop,sack 1
{83368442:83372786}], length 0
00:00:00.000033 IP s > c: Flags [.], seq 80275154:80276602, ack 145,
win 122, options [nop,nop,TS val 1294920658 ecr 1943987488], length
1448
00:00:00.000007 IP s > c: Flags [.], seq 80276602:80278050, ack 145,
win 122, options [nop,nop,TS val 1294920658 ecr 1943987488], length
1448
00:00:00.000004 IP s > c: Flags [.], seq 80278050:80279498, ack 145,
win 122, options [nop,nop,TS val 1294920658 ecr 1943987488], length
1448

>
> //E
>
> On Fri, May 23, 2014 at 04:06:12PM -0700, Yoshifumi Nishida wrote:
>> This is just out of my curiosity, but can we share the tcpdump file for
>> this?
>> --
>> Yoshi
>>
>> On Friday, May 23, 2014, Erik Hugne
>> <erik.hugne@ericsson.com<javascript:_e(%7B%7D,'cvml','erik.hugne@ericsson.com');>>
>> wrote:
>>
>> > Maybe..
>> > This is a standard Linux 3.0 kernel with sack and frto enabled.
>> > Additional lab experiments have shown that following the path change, a
>> > stream
>> > of lost packets occurs (almost a full window, ~400kB). The receiver sends
>> > dupacks on the new path to trigger fast retransmit, but then the
>> > retransmission
>> > timer on the sender side fires and all unacked data will be resent. At the
>> > RTO,
>> > we get the expected cwnd drop. While the receiver keeps kicking the the
>> > sender
>> > by sending dupacks, the retransmission is reasonably fast but after a few
>> > ms it
>> > grinds down to a stop-and-go behaviour.
>> >
>> > I failed in my efforts to find some explanation to this in the RFC's.
>> > What i did find was in VJ's paper from '88.
>> >
>> > -When starting or restarting after a loss, set cwnd to one packet.
>> > -On each ack for new data, increase cwnd by one packet
>> >
>> > If retransmitted data are not considered "new", that would explain the
>> > behavior
>> > i guess.
>> >
>> > Scripted ss -ti for the connection every 3 sec.
>> > In this test it wasn't even a path change, we just downed the interface on
>> > a
>> > router in the middle for some 300ms to simulate the packetloss (This was
>> > done at 15:06:18)
>> >
>> > Fri May 23 15:06:01 CEST 2014
>> >         reno wscale:9,7 rto:1892 rtt:1678/7 ato:40 cwnd:2215 send 15.3Mbps
>> > rcv_space:14480
>> > Fri May 23 15:06:04 CEST 2014
>> >         reno wscale:9,7 rto:1888 rtt:1678/8 ato:40 cwnd:2215 send 15.3Mbps
>> > rcv_space:14480
>> > Fri May 23 15:06:07 CEST 2014
>> >         reno wscale:9,7 rto:1888 rtt:1676/9 ato:40 cwnd:2215 send 15.3Mbps
>> > rcv_space:14480
>> > Fri May 23 15:06:10 CEST 2014
>> >         reno wscale:9,7 rto:1884 rtt:1675.5/5 ato:40 cwnd:2215 send
>> > 15.3Mbps rcv_space:14480
>> > Fri May 23 15:06:14 CEST 2014
>> >         reno wscale:9,7 rto:1816 rtt:1607.5/22 ato:40 cwnd:2215 send
>> > 16.0Mbps rcv_space:14480
>> > Fri May 23 15:06:17 CEST 2014
>> >         reno wscale:9,7 rto:1892 rtt:1681.5/7 ato:40 cwnd:2215 send
>> > 15.3Mbps rcv_space:14480
>> > Fri May 23 15:06:20 CEST 2014
>> >         reno wscale:9,7 rto:1548 rtt:651/225 ato:40 cwnd:3 ssthresh:1107
>> > send 53.4Kbps rcv_space:14480
>> > Fri May 23 15:06:23 CEST 2014
>> >         reno wscale:9,7 rto:1224 rtt:263/110 ato:40 cwnd:3 ssthresh:1107
>> > send 132.1Kbps rcv_space:14480
>> > Fri May 23 15:06:26 CEST 2014
>> >         reno wscale:9,7 rto:1172 rtt:210.5/18 ato:40 cwnd:3 ssthresh:1107
>> > send 165.1Kbps rcv_space:14480
>> > Fri May 23 15:06:29 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/6 ato:40 cwnd:3 ssthresh:1107
>> > send 170.8Kbps rcv_space:14480
>> > Fri May 23 15:06:32 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:3 ssthresh:1107
>> > send 170.8Kbps rcv_space:14480
>> > Fri May 23 15:06:35 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:38 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:41 CEST 2014
>> >         reno wscale:9,7 rto:1168 rtt:204/4 ato:40 cwnd:6 ssthresh:1107
>> > send 340.7Kbps rcv_space:14480
>> > Fri May 23 15:06:44 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:47 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:50 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:53 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:56 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:06:59 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/4 ato:40 cwnd:6 ssthresh:1107
>> > send 341.5Kbps rcv_space:14480
>> > Fri May 23 15:07:02 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
>> > send 512.3Kbps rcv_space:14480
>> > Fri May 23 15:07:05 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
>> > send 512.3Kbps rcv_space:14480
>> > Fri May 23 15:07:08 CEST 2014
>> >         reno wscale:9,7 rto:1164 rtt:203.5/3 ato:40 cwnd:9 ssthresh:1107
>> > send 512.3Kbps rcv_space:14480
>> > Fri May 23 15:07:11 CEST 2014
>> >         reno wscale:9,7 rto:1260 rtt:487.5/8 ato:40 cwnd:960 ssthresh:1107
>> > send 22.8Mbps rcv_space:14480
>> > Fri May 23 15:07:14 CEST 2014
>> >         reno wscale:9,7 rto:1244 rtt:863.5/6 ato:40 cwnd:1111
>> > ssthresh:1107 send 14.9Mbps rcv_space:14480
>> > Fri May 23 15:07:17 CEST 2014
>> >         reno wscale:9,7 rto:1148 rtt:864/6 ato:40 cwnd:1114 ssthresh:1107
>> > send 14.9Mbps rcv_space:14480
>> > Fri May 23 15:07:20 CEST 2014
>> >         reno wscale:9,7 rto:1096 rtt:868/8 ato:40 cwnd:1118 ssthresh:1107
>> > send 14.9Mbps rcv_space:14480
>> >
>> > On Thu, May 22, 2014 at 06:47:56AM +0000, Scheffenegger, Richard wrote:
>> > >
>> > > Are you perhaps observing an effect of the ACK clock loss, when you
>> > refer to a huge number of consecutive lost segments?
>> > >
>> > > Standard SACK does suffer from an issue that during loss recovery, there
>> > may be a gap (SACKs don't clock out new/retransmitted packets while
>> > flightsize > cwnd (which is now 1/2); only during the 2nd half, SACK starts
>> > to clock out packets, but at the same rate as before (which may not that
>> > bright an idea if a path change with differnet characteristics is now
>> > there).
>> > >
>> > > RateHalving (Linux has some kind of variant of this) / Proportional Rate
>> > Reduction RFC6937 may be the thing that you are after?
>> > >
>> > >
>> > > ABC really only comes into play during slowstart, to make connections
>> > which use delayed ACKs not perform much less aggressive than sessions where
>> > the receiver features QuickAck (Linux non-standard).
>> > >
>> > > Best regards,
>> > >
>> > >
>> > > Richard Scheffenegger
>> > >
>> > >
>> > > > -----Original Message-----
>> > > > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Erik Hugne
>> > > > Sent: Mittwoch, 21. Mai 2014 14:55
>> > > > To: Mark Allman
>> > > > Cc: ingemar.s.johansson@ericsson.com; martin.isaksson@ericsson.com;
>> > > > tcpm@ietf.org
>> > > > Subject: Re: [tcpm] RFC3465:TCP ABC and SACK'ed data
>> > > >
>> > > > On Wed, May 21, 2014 at 07:46:28AM -0400, Mark Allman wrote:
>> > > > >
>> > > > > > TCP ABC (still in the experimental track) defines a method to
>> > > > > > increase the cwnd based on the bytes being acknowledged, instead of
>> > > > > > the number of acks that arrive. I am assuming that SACK'ed
>> > (RFC2018)
>> > > > > > data should be included in this calculation, but i could find no
>> > > > > > mention of this in RFC3465.  Could someone confirm this?
>> > > > >
>> > > > > ABC does not include SACK information in its calculations and it
>> > > > > cannot include SACK information.  ABC applies to slow start---which
>> > > > > ends when loss occurs.  And, (essentially) SACKs only arrive during
>> > loss
>> > > > recovery.
>> > > > > So, there is no need for ABC to deal with SACKs.
>> > > > >
>> > > >
>> > > > Thanks for taking the time to explain this.
>> > > >
>> > > > The problem (this is related to Ingemars posts around "Problem with low
>> > > > SSthresh") is that following the path change, there is a large train of
>> > > > segments being lost (say 150k out of a 500k window). Data at the end of
>> > > > the window did make it through and is SACK'ed. Upon receiving these
>> > SACK's
>> > > > the sending side enters loss recovery. It will take a very long time
>> > > > before recovering the 100k lost segments.
>> > > >
>> > > > An aggressive recovery algorithm would mitigate the effects of this,
>> > but i
>> > > > was hoping for a more elegant solution.
>> > > >
>> > > > //E
>> > > >
>> > > > _______________________________________________
>> > > > 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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Date: Fri, 30 May 2014 10:55:10 +0100
To: Joe Touch <touch@isi.edu>
From: Bob Briscoe <bob.briscoe@bt.com>
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References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <537E48CE.8040704@mti-systems.com> <537E66A7.4080907@isi.edu> <201405231003.s4NA3PAB005137@bagheera.jungle.bt.co.uk> <537F7D91.10802@isi.edu> <201405281716.s4SHG29Y014642@bagheera.jungle.bt.co.uk> <53861D4F.60709@isi.edu>
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Joe,

I was asking for an example of something useful that /can/ be done with EDO.

I'm sure you can come up with one. However, you have shown 
conclusively that the example in the draft (SACK + MPTCP + TCP-AO), 
when added to the widespread basic options, /cannot/ be done with 
EDO, because it needs 7 more bytes of options than a SYN allows.


Bob

At 18:30 28/05/2014, Joe Touch wrote:


>On 5/28/2014 10:16 AM, Bob Briscoe wrote:
>>Joe,
>>
>>I don't think this sufficiently answers the question to justify WG
>>adoption. You seem to be confirming that this is an academic exercise.
>
>You asked for a specific example - one that the MPTCP community has 
>raised. It's not a proof that there's "some set" that might overload 
>the space; it's based on a *specific* request.
>
>I gave other reasons - including educating the community as to the 
>issues. If you want to call that "academic", sure - in part.
>
>But I don't see that as interfering with WG adoption - we do that 
>sort of thing all the time (that's the basis of a BCP).
>
>Joe
>
>>At 17:55 23/05/2014, Joe Touch wrote:
>>>Hi, Bob,
>>>
>>>On 5/23/2014 3:03 AM, Bob Briscoe wrote:
>>>>Joe, and everyone else who wants to work on this,
>>>>
>>>>Just because it's easier to make a chocolate teapot than a cast-iron
>>>>one, doesn't imply that there is any need for chocolate teapots.
>>>
>>>You don't get a cast iron teapot just because you want one either ;-)
>>>
>>>>IOW, we will be asking the IESG to spend reviewer time on EDO, so we
>>>>need to give some plausible indication that someone might find it useful
>>>>and it's not just an academic exercise.
>>>
>>>Sometimes the answer "you can't have A, but at least here's B" is more
>>>than an exercise; it educates the community. By not providing either
>>>answer, we have continued to drag this issue around the block for far
>>>too long -- and spent far too many cycles in this and other WGs
>>>seeking solutions.
>>>
>>> > The current draft solely gives
>>>>SACK + MPTCP + TCP-AO as an example, but is that really something that
>>>>can't be done today?
>>>
>>>Current total for SYN options in widespread concurrent use (as already
>>>described in sec 6.4):
>>>
>>>         2       SACK permitted
>>>         10      timestamp
>>>         3       window scale
>>>         4       MSS
>>>         ------------------
>>>         11 bytes
>>>
>>>The current DO field is 4 bits, with a max value of 15 = 60 bytes for
>>>the total header, less 20 for the base TCP header which leaves 40 for
>>>options.
>>>
>>>So let's see what happens when we add:
>>>
>>>         11      widespread basic options
>>>         16      TCP-AO
>>>         20      MPTCP
>>>         --------------------
>>>         47
>>>
>>>That's more than 40.
>>>
>>>>Adding more complexity to the TCP stack (with the potential for more
>>>>vulnerabilities) is only worthwhile if there's an identifiable benefit,
>>>>otherwise few production stacks are going to implement it anyway.
>>>
>>>There are two identifiable benefits:
>>>
>>>         1) explain the ways we already know we can't extend the SYN
>>>         so we stop wasting time trying them repeatedly (i.e., education)
>>>
>>>         2) provide a solution for the other segments, so that can be
>>>         used - e.g., for large SACK responses
>>>
>>>         3) educate the community
>>>
>>>Joe
>>
>>________________________________________________________________
>>Bob Briscoe,                                                  BT
>
>________________________________________________________________
>Bob Briscoe,                                                  BT 


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Bob Briscoe <bob.briscoe@bt.com> wrote:
> 
> I was asking for an example of something useful that /can/ be done with EDO.
> 
> I'm sure you can come up with one. However, you have shown 
> conclusively that the example in the draft (SACK + MPTCP + TCP-AO), 
> when added to the widespread basic options, /cannot/ be done with 
> EDO, because it needs 7 more bytes of options than a SYN allows.

   The claim that MPTCP needs 20 bytes in the SYN is getting on my
nerves...

   As I read RFC 6824, it seems pretty clear that the initial SYN calls
for 12 bytes, not 20. It's the SYN-ACK that uses 20. Somebody please
correct me if I'm reading this wrong.

--
John Leslie <john@jlc.net>


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On 30/05/14 - 13:50:44, John Leslie wrote:
> Bob Briscoe <bob.briscoe@bt.com> wrote:
> > 
> > I was asking for an example of something useful that /can/ be done with EDO.
> > 
> > I'm sure you can come up with one. However, you have shown 
> > conclusively that the example in the draft (SACK + MPTCP + TCP-AO), 
> > when added to the widespread basic options, /cannot/ be done with 
> > EDO, because it needs 7 more bytes of options than a SYN allows.
> 
>    The claim that MPTCP needs 20 bytes in the SYN is getting on my
> nerves...
> 
>    As I read RFC 6824, it seems pretty clear that the initial SYN calls
> for 12 bytes, not 20. It's the SYN-ACK that uses 20. Somebody please
> correct me if I'm reading this wrong.

Indeed, it uses 12 bytes on both the SYN and the SYN/ACK.

Only the third ACK carries 20 bytes.


Cheers,
Christoph


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Date: Fri, 30 May 2014 17:42:37 +0100
To: Joe Touch <touch@isi.edu>
From: Bob Briscoe <bob.briscoe@bt.com>
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Joe,

At 18:58 28/05/2014, Joe Touch wrote:


>On 5/28/2014 10:13 AM, Bob Briscoe wrote:
>>Joe,
>>At 18:14 23/05/2014, Joe Touch wrote:
>>>On 5/23/2014 5:13 AM, Bob Briscoe wrote:
>>>>David,
>>>>
>>>>1) Parallel control channel
>>>>___________________________
>>>>Client A sends two SYNs back-to-back to an existing well-known port
>>>>(e.g. 80).
>>>
>>>You can send in whatever order you want; packets will be reordered,
>>>lost, and sent along alternate paths.
>>
>>Of course.
>>
>>But as I suggested initially, we can standardise the protocol so that an
>>upgraded host synthesises shared fate.
>
>Fate is dependent on path.
>
>>Eg. a 2-octet TCP option on the
>>D-type SYN that says "Hold for x [ms] to wait for the supplementary
>>C-type SYN, where x is a lot less than the usual time you hold SYN
>>connection state (e.g. x=2). If SYN C hasn't arrived by then, continue
>>without it, and discard it if it arrives."
> >
>>And if the C-SYN arrives first, hold it for y [ms] waiting for the
>>corresponding D-SYN. Where for example y=2 as well.
>
>(C=legacy; D=extended - from your earlier email, for context)

THis is not a good way to describe them, because neither SYN is 
legacy, but even if you did think this way, it would be the other way round
(mnemonic: D = primary *D*ata connection, C = additional *C*ontrol options).

>Legacy clients now experience connection delays when contacting 
>upgraded servers - why is that acceptable?

Nope. A legacy client experiences no extra delay. As I said, both 
non-legacy SYNs carry a flag TCP option that says that the server 
should wait for the other if it hasn't seen it yet. A legacy client 
wouldn't add this option, so there would never be any delay.

Only upgraded clients see any extra delay, and then only if the C-SYN is lost.
In the 'regular' case where the two SYNs both arrive, even upgraded 
clients see no delay. There's only additional delay if the C-SYN is 
lost AND both client and server are upgraded. If the D-SYN is lost, 
there would have been a delay anyway.


>FWIW, many endpoints don't hold SYN state at all if they implement 
>SYN cookies, so 2ms would end up overloading the server with state 
>much larger than it currently keeps.

Let's deal with whether SYN cookies can be supported alongside this 
dual-SYN scheme first...

Because both client and server have to support using a pair of SYNs, 
the server doesn't have to exactly mimic the SYN cookie mechanism, it 
just has to achieve the same outcome as a SYN cookie. So the 
generalised question is: how does a server accept C-SYNs and D-SYNs, 
but also reflect all its connection state back off the client, 
without holding any state itself, while at the same time responding 
to any negotiation requests in the TCP options on both SYNs, so as 
not to delay negotiations.

==SYN Cookie v2==
The high level answer would surely be for the server to use an EDO 
TCP option on the SYN-ACK, and put all the connection state in a 
suitably large TCP option, for the client to reflect back. On the 
SYN-ACK, the server would also include any responses to TCP options 
(on either SYN) that expected responses.

Back to your original question, paraphrasing: "Doesn't holding state 
for 2ms risk more overload?"
The answer comes in two parts:

1) The server reduces y (the time it holds C-SYNs) the more it is 
under pressure. Then if necessary it reduces x (the time it holds 
D-SYNs), until both are zero.

2) This seems to allow a flooding attack to make a server remove the 
SYN extension facility, but we can prevent that... For either SYN, a 
server doesn't just discard it when it reaches the timeout. For SYN-D 
it returns a SYN cookie, and for SYN-C it returns a SYN Cookie v2 
(described eariler). For a genuine client this doesn't mean that an 
attack prevents the use of dual SYNs, it merely defers binding the 
two SYNs together until the next round. This only affects any options 
that depend on being combined with information in the other SYN, 
which should be a corner case.

(...I'm making this up as I go along, but it's still holding water 
pretty well isn't it.)

Perhaps this points to an alternative solution, where SYN cookie v2 
is always used, and the binding between the two SYNs only happens on 
the second round.

>>>FWIW, do these use the same source port and ISN?
>>>         - if they do, it'll reset the connection
>>>         - if they don't, you're now limiting the number of
>>>         concurrent connections to roughly half:
>>>         http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/
>>
>>Unless we can think of a way for the C-SYN to be discarded by a legacy
>>TCP stack (but not a middlebox), I'm assuming we need to get the C-SYN
>>up to the legacy app-layer before discard... So, I was assuming they use
>>different source ports (otherwise, as you say, a legacy TCP stack could
>>reset the D SYN connection if it arrived second)
>
>If they use different source ports, you could be limiting the 
>connection rate. It's not enough to just "discard" late D-SYNs; they 
>could be for new connections (how do you know the two are paired?), 
>which means you need to RST them and keep state for 2MSL.

The upgraded listener knows which D-SYN that a C-SYN is paired with as follows:
1) On the D-SYN, the TCP option declaring that it is a D-SYN would 
have to carry a parameter duplicating the src port in the main TCP 
header of the D-SYN (so it would survive NATing, unlike the src port itself).
2) On the C-SYN, the TCP option declaring that it is a C-SYN makes 
the server read the payload of the C-SYN, which contains the original 
value of the source port of the D-SYN.
3) The server can then match the src port inside the C-SYN to the 
original src port of the D-SYN.

Once the server has both SYNs, it can check they are 
cryptographically paired, using other material that would have to be 
included in the C-SYN payload (tbd).

Assuming you really meant discarding late C-SYNs,...

Two cases, both don't need a 2MSL wait:

1) A legacy server doesn't have to RST the connection, because the 
app-layer payload is a valid app-layer header. Normally, of course, 
the server will hold back the payload on a SYN so it won't get to the 
app-layer unless the client responds to the SYN-ACK with an ACK. But 
that's OK, 'cos we don't need the paylkoad to get to the app-layer, 
we just need it to look like it would be valid if it did. The 
upgraded client could complete the 3WHS with a regular ACK, or it 
could respond with a RST. When a TCP server in SYN_SENT state 
receives a RST it returns to LISTEN state, and I believe it 
immediately discards connection state, because it never synchronised 
the connection.

2) If an upgraded host gets the C-SYN late, it knows it's a C-SYN, so 
when it tries to match it to a D-SYN it finds it already gave up 
waiting and went ahead with a connection based on the D-SYN alone. So 
it just silently discards the C-SYN with no need to keep to the TCP 
timeout rules (because it understands that a C-SYN doesn't open a 
connection itself).

>>The ISN on the C-SYN is redundant - we might be able to think of another
>>use for that field.
>
>But it's not redundant if you're talking to a legacy server, so you 
>can't do anything with that field.

The ISN can be any number, and legacy TCPs will accept it. I just 
meant the field could be used for a number with some other useful 
meaning, because it never has to be used to synch any TCP sequence space.

>>On an upgraded host, max concurrent connections would only be slightly
>>impacted, because of the v small timeout of C-SYNs.
>
>See above; you need to know how to pair C-SYNs and D-SYNs. Consider 
>that legacy hosts and upgraded hosts might be behind the same NAT, 
>at which point how do you know the two SYNs are even from the same host?

The crypto binding info in the C-SYN payload (tbd) would distinguish 
two hosts that a NAT had given the same src IP address. If the NAT 
gives different IP addresses to two connections from the same host, 
that would be a problem. We could get round that using a similar 
trick to that we already used to protect the src port value, but 
perhaps we don't need to, 'cos I think NATs like this are fairly rare (?).


>>The semantics of the option-space-extension option would be to only hold
>>the C-type SYN for a timeout and only the D-type SYN creates full
>>connection state (I'm deferring SYN cookie behaviour to tomorrow for now
>>- this is a straw man).
>
>I don't think you can defer a widely-deployed feature like SYN 
>cookies, and that seems to kill this.

Solved above, I think.

>>Admittedly, the number of concurrent connections a /legacy/ host can
>>support could reduce by up to half (if all remote TCP clients are
>>sending the new option). But they have the choice of upgrading to the
>>new stack to stop wasting their memory.
>
>Both legacy clients and legacy servers are impacted. We already know 
>that servers are overloaded with connections, so halving that number 
>seems like a non-starter. As does adding x ms delay for *each* legacy client.

Legacy clients are not impacted at all (as explained above).
Legacy servers impact upgraded clients with delay ~2ms only if a 
C-SYN is lost. Zero otherwise.

Legacy servers have to hold more connection state, but half is the 
worst case for the last legacy server on the planet after all clients 
have upgraded.

If fraction f of clients have upgraded, a legacy server's connection 
state for n connections increases to n(1-f) + n(2f) = n(1+f).
Ie. Legacy server connection state increases by fraction f, and at 
any time it can solve this by upgrading to coupled SYNs. Usually 
servers upgrade earlier than clients anyway.

(Assumption: connection durations used by upgraded and legacy clients 
follow a similar distribution.)


>>>>* SYN D, establishes a regular data connection, with sufficient TCP
>>>>options to be workable but they still fit within the existing 40B option
>>>>limit.
>>>>* SYN C establishes another parallel connection to the same well-known
>>>>port that looks like regular data from the outside (it could even be an
>>>>extension to HTTP to ensure middleboxes will let it pass), but it talks
>>>>a new app-layer 'TCP control' protocol inside.
>>>
>>>What happens when they arrive out of order? What happens when you get
>>>D but not yet C? How long do you wait for C?
>>
>>See above.
>>The timeouts might be standardised, or they might be declared in the
>>option as a hint (for the host to ignore if it is under stress).
> >
>  >> This is the problem with dual-stack approaches - new endpoints
>>>penalize legacy endpoints if there's a stall, and undermine new
>>>endpoints if they don't.
>>
>>Have I satisfied you that this can be solved sufficiently? Bearing in
>>mind the gain, it's reasonable to have to accept some pain.
>
>The pain is the problem. The smaller the timeout, the smaller the 
>pain to legacy clients but the lower probability the extension will 
>be useful to upgraded clients.

Solved using the SYN cookie 2 idea above.

>The method cuts the rate of connections for legacy servers in half - 
>which is *huge*.

Nope. It's cut by 1/(1+f) where f is the fraction of upgraded 
clients. And the legacy server can itself upgrade to avoid this completely.

>And it adds to the state overhead of the upgraded clients that need 
>to do double the work for *every* connection.

Accepted. However, state overhead is rarely a concern on clients. 
Anyway, they don't need to use more TCP options on every connection - 
only those connections that need them.

>Ultimately, it's roughly equivalent to "try both and shut down the 
>one you don't want" -- the dual-stack approach we already know about.

That's a bit of a wild dismissal of a scheme that so far seems to 
have pretty useful properties.

>>>>If there is no support for the new app-layer protocol on port 80 the
>>>>control channel just shuts down with a suitable HTTP error, while SYN D
>>>>has opened a data connection with sufficient TCP options to be workable.
>>>>If the new app-layer TCP control protocol is supported on port 80, the
>>>>parallel control channel (C) adds unlimited additional control
>>>>flexibility to the data channel (D) hardly any added latency.
>>>>
>>>>Establishing a similar control channel in the opposite direction would
>>>>be fairly trivial.
>>>>
>>>>There are few, if any, middlebox problems with the above approach.
>>>>However, there are certainly other problems, but no more insurmountable
>>>>than all the problems that have already been discussed with taking the
>>>>'easy' route of EDO:
>>>>* A secure binding would have to be added to bind channel C to a secret
>>>>known only to the originator of channel D, otherwise it would open up
>>>>data channels to spoof control channel attacks. This binding could be
>>>>built on a TCP-AO option in channel D.
>>>
>>>Yes, that's another problem.
>>
>>Fairly straightforward to solve using standard techniques.
>
>I think that warrants more than a claim. You need to explain how you 
>avoid false postives through NATs.

Certainly. But I'm not going to do that now. I have no need for 
bigger TCP options in my day job (or my night job), so I would hope 
someone else would pick this up and specify it fully if they need it. 
I was just irked by your impossibility claim.

I won't believe this dual SYN idea is useful myself until it's fully 
specified, implemented and tested. Bitter experience of trying to 
extend major protocols like IP has taught me that one little niggle 
can ruin years of work, even if it looked promising until the last moment.

However, I still don't give up, or discourage others by burning time 
writing RFCs that claim it's impossible.

>>>>* Channel C would need some way to refer to the segments of channel D
>>>>that was robust against re-segmentation.
>>>
>>>Which means it won't work in the current Internet, because
>>>resegmentation is also widespread (though evil, IMO).
>>
>>Well, resegmentation isn't usually a problem on a SYN anyway.
>>
>>We can't improve on the general pain caused by resegmentation. I was
>>only talking about the delta pain that my strawman would suffer, where a
>>resegmenting function sees data on my SYN and doesn't understand my
>>strawman, so it won't patch up the damage in my strawman protocol that
>>it would have patched up by altering sequence numbers in a regular
>>single SYN with a payload.
>
>You're right, in the sense that SYN data would typically be 
>discarded anyway at the endpoint if it implemented SYN cookies.
>
>>I think this is a corner case.
>
>It is, but you raised it in a general sense; were you thinking of it 
>in terms of just SYNs or other segments?

It was a general point that I figured might affect any scheme that 
enables TCP options that might want to refer to byte offsets, when 
those bytes are in a different packet. For the scheme that has 
developed over the course of this email thread, I don't think it's a 
concern now.

>>>>* The main problem is that the two channels don't share fate;a control
>>>>packet can be delayed relative to the point in the data stream at which
>>>>it is attempting to exert control, possibly for a RTT if it is lost and
>>>>has to be retransmitted. However, this is not insurmountable. The
>>>>control protocol could include a mode to "synthesise shared fate", by
>>>>making the data channel buffer data until an associated control segment
>>>>had arrived. This would duplicate the latency impact of a loss or delay
>>>>on either channel, but one can imagine mitigations that would consign
>>>>this latency impact to corner cases.
>>>>* It's a bit of a mess, but that comes with the territory when trying to
>>>>fix legacy protocol problems.
>>>>* The internal stack architecture seems to require a trombone back down
>>>>into the kernel from user-space, but that is not insurmountable - a shim
>>>>within the kernel on port 80 (for example) could redirect control
>>>>channel data across to the "TCP control channel module" in the kernel,
>>>>while passing non-control channel connections to user-space.
>>>>
>>>>2) Build on LOIC
>>>>______________________
>>>>Long option with invalid checksum <draft-yourtchenko-tcp-loic-00>
>>>
>>>Won't work through current NATs, which won't recalculate the checksum
>>>properly.
>>
>>I'm building on the general idea of using something invalid, not using
>>the checksum idea specifically.
>
>Anything considered invalid by an endpoint might be checked by an 
>overzealous midbox.

True. But in this case, the client can put something perfectly valid 
in the app-layer payload, because it can even RST the connection to 
stop the payload getting to the app.

Theoretically, we could even encode the extended TCP options in HTML 
as the argument to an HTTP PUT, so that if an HTTP server did ever 
parse them, it might even display a prettily rendered list of 
extended TCP options :)

>>>>At 18:53 22/05/2014, John Leslie wrote:
>>>>>    That's too big of a change to ask folks to believe it safe.
>>>>
>>>>When I read an idea, I don't take it as set in stone and just find a
>>>>hole and dismiss it. I see it as a potential stepping stone to a
>>>>solution and think about how it could be done better. In fact, Andrew
>>>>Yourtchenko said that was the intention of his write-up of LOIC.
>>>>
>>>>I believe that an approach worth further thought would be a mixture of
>>>>the control channel idea and the invalid checksum idea. I'm thinking of:
>>>>* a pure control SYN (C) sent first, then a base SYN (D) sent
>>>>back-to-back, both to the same port.
>>>
>>>Again, please don't assume back-to-back means anything.
>>
>>See above.
>>Actually, even tho order isn'[t guaranteed, it is important to get them
>>in the right order to optimise performance (much as TCP doesn't assume
>>perfect order, but it goes smoothest with reasonable ordering).
>
>Right, but I want to emphasize that things emitted back-to-back 
>could end up being delivered days later (e.g., when a link goes 
>down, some routers just keep the queues there). The impact in those 
>cases needs to be considered.

Considered, certainly.

>>>>* SYN C would contain something invalid to cause a legacy TCP stack or
>>>>legacy app to discard it (and hopefully less probability that a
>>>>middlebox would), e.g. a payload that is invalid for the application
>>>>protocol on the port.
>>>
>>>But so will a NAT, etc.
>>
>>No. You can design a payload that has headers that a NAT will ignore but
>>an end-host will have to process.
>
>You can try, but if an end-host looks at it you can be sure that a 
>midbox will do the same - if not today, then soon.

Agree in general. But see above re HTML & PUT.

> > E.g. HTTP connection control headers
>>newly defined for this protocol that would be ignored by NATs, without
>>any other HTTP behaviour, so a legacy host does nothing at the app layer.
>
>Those are rewritten by "helpful" midboxes all the time, FWIW.

Again, see HTML-PUT commment.

>>>>* there would be additional TCP options in the payload of SYN C to be
>>>>added to the TCP options that arrived separately on the base SYN
>>>>* The control SYN could be bound crytographically to the base SYN (as
>>>>already described).
>>>>* It could use the shim-like control stack arangement described earlier.
>>>>
>>>>By focusing solely on extending the SYN, this would avoid the ongoing
>>>>shared fate problems that a separate control channel suffers throughout
>>>>the connection. There would still be shared fate problems with 2 SYNs
>>>>(e.g. the two SYNs get re-ordered), but the protocol would have to be
>>>>designed to be robust to that (naively, SYN D could include a new TCP
>>>>option that told a new stack to wait a few ticks for a SYN C, but that
>>>>would be vulnerable to meddleboxes). Not insurmountable.
>>>
>>>AFAICT, it is.
>>
>>Still?
>
>If it means halving the performance of legacy servers and delaying 
>legacy clients, yes.

Not so. See above.

Regards


Bob


>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 


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John,

We needed to lose 7B and we've lost 8B.
So (SACK + MPTCP + TCP-AO + widespread basic options, incl 
TCPtimestamp) do fit in a SYN, therefore the example in the draft is valid.

However, as everyone has pointed out, jubilation may be short-lived. 
For instance, even now, people are working on a lower latency MPTCP 
that will need more option space.

Thx.


Bob


At 15:14 30/05/2014, Christoph Paasch wrote:
>On 30/05/14 - 13:50:44, John Leslie wrote:
> > Bob Briscoe <bob.briscoe@bt.com> wrote:
> > >
> > > I was asking for an example of something useful that /can/ be 
> done with EDO.
> > >
> > > I'm sure you can come up with one. However, you have shown
> > > conclusively that the example in the draft (SACK + MPTCP + TCP-AO),
> > > when added to the widespread basic options, /cannot/ be done with
> > > EDO, because it needs 7 more bytes of options than a SYN allows.
> >
> >    The claim that MPTCP needs 20 bytes in the SYN is getting on my
> > nerves...
> >
> >    As I read RFC 6824, it seems pretty clear that the initial SYN calls
> > for 12 bytes, not 20. It's the SYN-ACK that uses 20. Somebody please
> > correct me if I'm reading this wrong.
>
>Indeed, it uses 12 bytes on both the SYN and the SYN/ACK.
>
>Only the third ACK carries 20 bytes.
>
>
>Cheers,
>Christoph

________________________________________________________________
Bob Briscoe,                                                  BT 


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I'll update the doc accordingly.

I had also been thinking of SACK not being as limited in its use of 
space, nor MPTCP.

Joe

On 5/30/2014 10:13 AM, Bob Briscoe wrote:
> John,
>
> We needed to lose 7B and we've lost 8B.
> So (SACK + MPTCP + TCP-AO + widespread basic options, incl TCPtimestamp)
> do fit in a SYN, therefore the example in the draft is valid.
>
> However, as everyone has pointed out, jubilation may be short-lived. For
> instance, even now, people are working on a lower latency MPTCP that
> will need more option space.
>
> Thx.
>
>
> Bob
>
>
> At 15:14 30/05/2014, Christoph Paasch wrote:
>> On 30/05/14 - 13:50:44, John Leslie wrote:
>> > Bob Briscoe <bob.briscoe@bt.com> wrote:
>> > >
>> > > I was asking for an example of something useful that /can/ be done
>> with EDO.
>> > >
>> > > I'm sure you can come up with one. However, you have shown
>> > > conclusively that the example in the draft (SACK + MPTCP + TCP-AO),
>> > > when added to the widespread basic options, /cannot/ be done with
>> > > EDO, because it needs 7 more bytes of options than a SYN allows.
>> >
>> >    The claim that MPTCP needs 20 bytes in the SYN is getting on my
>> > nerves...
>> >
>> >    As I read RFC 6824, it seems pretty clear that the initial SYN calls
>> > for 12 bytes, not 20. It's the SYN-ACK that uses 20. Somebody please
>> > correct me if I'm reading this wrong.
>>
>> Indeed, it uses 12 bytes on both the SYN and the SYN/ACK.
>>
>> Only the third ACK carries 20 bytes.
>>
>>
>> Cheers,
>> Christoph
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT


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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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Bob,
>
> Returning to the question of adoption, we have to address the question
> of why previous attempts to do this have failed. I don't believe it is
> as simple as that they tried to include options on SYNs, so all we have
> to do is avoid the SYN problem.
...
>
> 4) Finally, the EDO draft cites <draft-ananth-tcpm-tcpoptext-00> as if
> it is just another solution. It's not. It's actually a very useful
> survey of all the previous attempts to solve this problem, including a
> useful enumeration of the problems that have to be surmounted.
>
> The arguments on this thread show that we don't agree on the problem
> space. So, I suggest we adopt Anatha's draft, and as we develop it, we
> agree on the problem we are trying to solve first. Boring, but
> apparently necessary.


I was offline this week and am now catching up the email thread. I agree 
that starting from this draft would be an excellent idea to document in 
an information draft the existing solutions and their advantages and 
drawbacks. With this documentation, we could then build another solution 
that works efficiently.


Olivier



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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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David, Joe, Bob,

>> What is impossible is to expand the option space in the initial SYN
>> packet and maintain 100% backwards compatibility with the following
>> restrictions:
>>     1) No additional packets are sent, just the initial SYN
>>     2) There is no a priori knowledge as to whether or not the remote
>> host understands expanded option space.
>
> I would pose the conditions as follows:
>
>      1) is robust to packet loss, reordering, and alternate paths
>          any assumption of "back to back" packets isn't
>
>      2) does not negatively impact legacy hosts
>          e.g., dual-stack/multiple SYN approaches fail here when
>          a legacy system is trying to connect to an updated one
>          (which has to wait to see if the updated SYN will show)
>
>      3) succeeds through basic NAT/NAPTs
>          anything that uses a nonstandard checksum will fail
>          to support the extended space
>
>      4) addresses new connections to hosts not previously contacted
>          it's trivial to retain state across sequences of
>          connections, as per RFC 2140.


Let me make two rough proposals that could be worth being explored or at 
least documented.

1. The TCP option space is limited, but not the IPv6 destination options 
in the IPv6 header. One possibility to extend the TCP option space, 
notably (but not only) in the SYN would be to place part of the TCP 
option inside an IPv6 destination option. Given that the IPv6 
destination option is only limited by the IPv6 packet size, we are safe 
for the long term and the solution seems to be clean.

  This approach has two limitations :

  - It is a violation of the layering principle since the receiving host 
needs to provide the content of the IPv6 destination option to the TCP 
stack.

  - It only works with IPv6 and does not work with IPv4. In the short 
term, IPv4 support is mandatory, but in the long term IPv6 support is 
clearly more important.


2. TCP assumes that the client proposes the TCP extensions and the 
server selects the ones that it supports. In practice, measurements show 
that the TCP stack running on servers is usually upgraded after the 
stack running on clients. It might be interesting to allow the client to 
insert in the SYN a special option that indicates that it would accept 
new options proposed in the SYN+ACK (or perhaps later). These options 
proposed in the SYN+ACK would be confirmed in the third ACK.


If one of these solutions does not seems completely crazy, I can write a 
more detailed proposal to see how it would work in practice


Olivier



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Hi, Bob,

Let's get back to the core, in a simpler fashion, so other can follow it.

I stand by my "there's no way to extend the space in the initial SYN", 
but you've convinced me there *might* be a way to provide extended space 
that can occur during the first phase of the TWHS. I think the dual-SYN 
approach still isn't viable, but I've outlined an alternative below 
that's similar but doesn't have the same baggage, IMO.

Again, I'm still concerned by what midboxes might do to this...

What do others think??

Joe

For quick review, here's what I understand:

		dso = dual-syn option
			dso-D = data
			dso-C = control
		conn_id = identifier to link the two SYNs together
		extra_opt = options that didn't fit in legacy SYN
		fit_opt = options that do fit in the legacy SYN

	new endpoint sends
		port A SYN + dso-D + conn_id + fit_opt
		port B SYN + dso-C + conn_id + extra_opt

			legacy endpoint sends back two connections:
				port A SYN-ACK + fit_opt
				port B SYN-ACK + ??
				(it's interpretation of extra_opt)
			new endpoint responds:
				port A ACK (established)
				port B RST

			Notes about legacy servers:
				- they do twice the work on SYNs
				- they might keep twice the state
				(if not using cookies)
				- they might clean state if the RST
				is received, but that state might
				persist indefinitely (until the next
				connection, depending on timeouts, etc.)

			-----

			new endpoint sends back one connection:
				port A SYN=ACK + dso-D + ....
			
			Notes:
				- can stall when dso-D SYN arrives
				before dso-C SYN, up to some limit
				- twice the work on SYNs (or more)

Here's what I was assuming, though admittedly it's not documented (yet):

	- no significant impact on TCP connection rate for
	legacy servers

	- no significant impact on TCP connection rate for
	legacy clients

	- impact dominated by processing the extended option space
	for extended clients

	- impact dominated by processing the extended option space
	for extended servers

	- compatible with typical TCP processing optimizations,
	notably SYN cookies
		you did provide a potential way forward for these

	- capable of successfully traversing typical NATs

Your approach has the following properties:

	- halves the server connection rate for updated servers
	from legacy clients when this option is in use

	- lowers (to some extent, if not halves) the client
	connection rate of updated clients to all servers
	when this option is in use

	- halves (roughly) the server rate for all servers
	when this option is in use

It also:

	- doubles the number of SYNs in the network

	- susceptible to lack of fate-sharing problems, e.g.,
	if the two SYNs experience different firewall configurations

	- reduces the space available for fit_opt due to the need
	for the conn_id even in the fall-back D-SYN, which means
	less option space in the SYNs for fall-back connections

	the conn_id which may need to be very large because it
	needs to be unique per source port and source IP address
	because that information is lost during NAT translation

	- requires the ISNs to be related (see RFC6528 - if there's
	a rule to generate it, there will be code to validate that
	rule, and eventually a BCP to encourage that validation -
	typically from the same RFC author)

I agree that you have proposed potentially viable ways to deal with the 
SYN cookie, and that RST state is not an issue.

However, there are too many problems with this, IMO, to call it viable.

>> Ultimately, it's roughly equivalent to "try both and shut down the one
>> you don't want" -- the dual-stack approach we already know about.
>
> That's a bit of a wild dismissal of a scheme that so far seems to have
> pretty useful properties.

I said it was roughly equivalent - if you call that "wild dismissal", so 
be it.

Here's another trick that might clean up the above a little:

	aso - after SYN option

	FBP - front bumper packet (best I could do on names today)
		a packet with a sequence number BEFORE the ISN of
		the SYN

	new endpoint sends:

		SYN + aso + fix_opt
		FBP + aso + extra_opt + seqno-outsidewindow

			legacy endpoint sends back one connections:
				SYN-ACK + fix_opt

				if seg arrives before SYN, legacy
				will send a RST - which the new endpoint
				can ignore

				if seg arrives after SYN, it'll be
				discarded as out of window

			----

			new endpoint sends back one connection:

				SYN-ACK + options + ....

			a) if FBP arrives before SYN, it will generate
			a RST - which the new endpoint will ignore

			b) if FPB arrives with the SYN, they can be
			processed together

				the SYN-ACK can include responses to
				the extra_opts in addition to the
				fix_opts, and says "FBP received"

			c) if FPB arrives after the SYN:

				SYN-ACK proceeds, but sends
				back "wait for option response".

				at this point, the source re-sends FBP
				until an ACK is received that indicates
				"FBP received", or times-out as with
				any connection that doesn't finish TWHS

			I'm still thinking as to whether the ACK number
			might indicate whether FBP has been received,
			e.g., send back ACK = ISN -1 until FBP,
			at which point it sends back ISN. The connection
			proceeds when ISN is ACK'd.

This is cleaner as follows:

	- no need for conn_id coordination

	- no need for conn_id to consume option space for fall-back

	- avoids double-load for legacy servers

	- no problem with fate-sharing

	- traverses a NAT just fine

Upgraded servers still need to wait for the 'seg', but they could get 
that retransmitted if necessary.

It causes additional processing for legacy endpoints, but not 
necessarily in a way that decreases the rate because it's not trying to 
start a second connection; the RSTs sent back are lightweight, and the 
additional ACK.

----


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Hi, Oliver,

On 5/30/2014 10:53 AM, Olivier Bonaventure wrote:
...
> Let me make two rough proposals that could be worth being explored or at
> least documented.
>
> 1. The TCP option space is limited, but not the IPv6 destination options
> in the IPv6 header.

IP option space would not be protected by TCP-AO, and likely not by 
whatever the TCPUSEC WG creates (e.g.,my proposal is here:
http://tools.ietf.org/html/draft-touch-tcp-ao-encrypt-01)

Additionally, DPI, NAT processing, etc. are all impacted by IPv6 
options, so they're generally discouraged AFAIR.

Joe


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To: Bob Briscoe <bob.briscoe@bt.com>
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On 5/30/2014 2:55 AM, Bob Briscoe wrote:
> Joe,
>
> I was asking for an example of something useful that /can/ be done with
> EDO.

The obvious one is larger amounts of SACK feedback and space for MPTCP 
information.

Joe


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Hi, all,

Some additional information below about "another trick" I proposed, 
inspired by Bob's dual-SYN mechanism, courtesy a few long discussions 
today with Ted Faber.

I'll be glad to offer this as a potential solution in the doc.

Joe

On 5/30/2014 1:34 PM, Joe Touch wrote:
...
> Here's another trick that might clean up the above a little:

FWIW, I had explained it below as being based on sending out-of-window 
data; Ted pointed out that I had been assuming that the FBP ACK bit 
wasn't set - which means the sequence number might be more usefully 
matched to that of the SYN.

See below...

>      aso - after SYN option

		length = 2 (just a flag)
		length = 3 (or 4) indicating the length of the
			FBP expected

>      FBP - front bumper packet (best I could do on names today)
>          a packet
		ISN = same as the associated SYN
		ACK = cleared (i.e., a data packet NOT part of
		synchronized connection - see why that's useful below)

>      new endpoint sends:
>
>          SYN + aso + fix_opt
>          FBP + aso + extra_opt

		extra_opt in the data field
		total length < min MTU (576 for IPv4, 1280 for IPv6)
		again ACK bit is zero

>              legacy endpoint sends back one connections:
>                  SYN-ACK + fix_opt
>
>                  if seg arrives before SYN,

			it is silently dropped, because
			the ACK bit is clear (this is
			explicit in RFC793)

>                  if seg arrives after SYN,

			it is silently dropped, because
			the ACK bit is clear (this is also
			explicit in RFC793)

>              ----
>
>              new endpoint sends back one connection:
>
>                  SYN-ACK + options + ....
>
>              a) if FBP arrives before SYN,

			it can be silently dropped, but
			it's probably useful for new endpoints
			to hold onto these (without action)
			for a while; they can be silently
			discarded if there are too many
			(which will just result in a
			retransmission and an extra RTT)

>              b) if FPB arrives with the SYN, they can be
>              processed together
>
>                  the SYN-ACK can include responses to
>                  the extra_opts in addition to the
>                  fix_opts, and says "FBP received"
>
>              c) if FPB arrives after the SYN:
>
>                  SYN-ACK proceeds, but sends
>                  back "wait for option response".
>
>                  at this point, the source re-sends FBP
>                  until an ACK is received that indicates
>                  "FBP received", or times-out as with
>                  any connection that doesn't finish TWHS
>
>              I'm still thinking as to whether the ACK number
>              might indicate whether FBP has been received,

There are a few ways to handle this, but IMO the best is to have:

		SYN-ACK aso with length=2 means "waiting for FBP"

		SYN-ACK aso with length=3 (or 4) could mean
		"got the FBP", or might even indicate how
		many bytes the FBP extension contained

Note - the SYN-ACK and all subsequent segments can assume that aso == 
edo, i.e., they can use the same EDO as spec'd in version 01 of this draft.

> This is cleaner as follows:
>
>      - no need for conn_id coordination
>
>      - no need for conn_id to consume option space for fall-back
>
>      - avoids double-load for legacy servers
>
>      - no problem with fate-sharing
>
>      - traverses a NAT just fine
>
> Upgraded servers still need to wait for the 'seg', but they could get
> that retransmitted if necessary.

And there's a small amount of additional processing to discard the FBP 
at legacy endpoints, but silently discarding one packet per connection 
doesn't seem like a huge effort to me.

Note - there's no special RST processing, based on Ted's observation 
about "data without ACK" being considered "data sent in the 
unsynchronzed state" -- something legacy TCP explicitly silently discards.

-----


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On 5/27/2014 11:06 PM, Yoshifumi Nishida wrote:
> Hi Joe,
>
> I'm wondering if you presume that using the existing code point for late
> negotiation.

I had.

> Suppose If we use new code point, do we still have problems you
> mentioned below?

It depends:

- I don't think it matters vs. using the existing codepoint with a 
different length (e.g., as a flag in the SYN); they are equivalent in 
being uniquely identifiable.

- I don't like the idea of setting new TCP flags or changing the meaning 
of flags in the middle of a connection. That undermines the notion of a 
connection IMO.

To do so, the source would have to keep track of the last byte of the 
segment in which the change occurs, and wait for a confirmation in the 
ACK of that byte; ditto for the other end.

They'd need to resend things if lost *with the same flags* to indicate 
the change.

I.e., you'd need to implement a new TWHS per option that changes state. 
That's a mess. And I don't even want to think about what happens when 
you have multiple options that change state at the same time.

IMO, it's not worth the trouble.

Joe


>
> Thanks,
> --
> Yoshi
>
> On Tue, May 27, 2014 at 1:04 PM, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
>
>     Hi, all,
>
>     Late negotiation has too many problems, as have been repeatedly
>     raised on this list:
>
>              - prevents use of the timestamp to help address PAWS for
>              the initial SYN
>
>              - a new connection that either avoids TS in SYN or
>              uses just "use TS" (two-byte) flag can't know when
>              to engage the timestamp and what initial value to use
>
>     Joe
>
>
>     On 5/27/2014 11:36 AM, Yoshifumi Nishida wrote:
>
>         Hi folks,
>
>         I've submitted a proposal for this a while ago:
>         http://tools.ietf.org/html/__draft-nishida-tcpm-apaws-00
>         <http://tools.ietf.org/html/draft-nishida-tcpm-apaws-00>
>         Although it's expired now, I will submit a new version within a
>         couple
>         of weeks or so.
>         The late negotiation is also discussed in the draft.
>         As it will change the semantics of TS, it presumes to use new
>         code point
>         (or use timestamp field in SYN). But, I'm happy to discuss on
>         this point.
>
>         One tricky thing for RFC1323 is that it uses TS for two purposes:
>         timestamping and PAWS and one purpose of the draft is to
>         separate them.
>         I appreciate any feedback on the draft.
>
>         Thanks,
>         --
>         Yoshi
>
>         On Tue, May 27, 2014 at 5:12 AM, Scharf, Michael (Michael)
>         <michael.scharf@alcatel-__lucent.com
>         <mailto:michael.scharf@alcatel-lucent.com>
>         <mailto:michael.scharf@__alcatel-lucent.com
>         <mailto:michael.scharf@alcatel-lucent.com>>> wrote:
>
>               > Well, if it's just for Timestamps, just use an "invalid"
>         length
>              of 2 as
>               > a "Late TS permitted".
>               >
>               > Maintaining the same option value and this would be
>         rather straight
>               > forward.
>
>              Interesting proposal. But actually we are not really short
>         of option
>              numbers. If we get e.g. 8 bytes in a significant number of SYNs
>              (which is our real bottleneck), I think we could also spend
>         a new
>              option codepoint.
>
>               > Personally, I'd like to combine this with a change in
>         semantics, if
>               > SACK is negotiated on the same session as well, to allow
>         more
>              efficient
>               > loss recovery (lost retransmission detection during a
>         loss recovery
>               > episode). I should have a unpublished draft sitting
>         around here about
>               > that aspect.
>
>              A change in semantics would also be a reason for a new
>         codepoint, imho.
>
>              Michael
>
>
>               > The TS option handling seems to be rather generous by
>         middleboxes, in
>               > general.
>               >
>               > Richard Scheffenegger
>               >
>               > > -----Original Message-----
>               > > From: tcpm [mailto:tcpm-bounces@ietf.org
>         <mailto:tcpm-bounces@ietf.org>
>              <mailto:tcpm-bounces@ietf.org
>         <mailto:tcpm-bounces@ietf.org>>__] On Behalf Of Scharf,
>               > Michael
>               > > (Michael)
>               > > Sent: Dienstag, 27. Mai 2014 13:49
>               > > To: Scheffenegger, Richard; Eggert, Lars; Joe Touch
>               > > Cc: tcpm@ietf.org <mailto:tcpm@ietf.org>
>         <mailto:tcpm@ietf.org <mailto:tcpm@ietf.org>>
>               > > Subject: Re: [tcpm] timestamp options (was Re: New Version
>               > Notification
>               > > for draft-touch-tcpm-tcp-edo-01.__txt)
>               > >
>               > > > For late TS, we would still need at least 2 Bytes during
>               > > > <SYN>/<SYN,ACK> to allow that capability.
>               > > >
>               > > > However, as the useful values of Window Scale only cover
>              0..14 [1],
>               > > > one approach to save SYN option space would be to
>         negotiate
>              up to 4
>               > > > binary
>               > > > (on/off) plus WS in a new single 3-byte option.
>               > >
>               > > I might be wrong, but I've thought so far that
>         "replacing" the
>              window
>               > > scale option would be pretty difficult. We already had
>         reports of
>               > > middleboxes that parse this option because they try to
>         understand
>               > rwnd (or
>               > > they do not parse the option even if they should, or
>         they have
>               > entirely
>               > > broken logic to mess up TCP...). I think one example
>         are firewalls
>               > that
>               > > try to detect whether segments are in-window. Another
>         example could
>               > be
>               > > PEPs that try to tune rwnd (if they still exist in the
>         wild).
>               > >
>               > > My understanding so far was that a solution only for
>         timestamps
>              would
>               > be
>               > > the most low-hanging fruit.
>               > >
>               > > Michael
>               > >
>               > > _________________________________________________
>               > > 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>
>
>              _________________________________________________
>              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>
>
>
>


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Date: Fri, 30 May 2014 18:07:29 -0700
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Subject: Re: [tcpm] timestamp options (was Re: New Version Notification for draft-touch-tcpm-tcp-edo-01.txt)
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On 5/28/2014 1:17 AM, Scheffenegger, Richard wrote:
> Hi Joe,
>
>> Late negotiation has too many problems, as have been repeatedly raised on
>> this list:
>>
>> 	- prevents use of the timestamp to help address PAWS for
>> 	the initial SYN
>
> This one is genuine. However, given the prevalence of non-TS TCP
> sessions and no wide-spread havoc caused by NOT having PAWS on the SYN,
> it appears to me that the impact is manageable. (Of course it would be
> nicer to have PAWS on SYN, but as long as SYN option space is limited...)

I don't know the stats, but I'd prefer if initial connections were 
PAWS-protected.

>> 	- a new connection that either avoids TS in SYN or
>> 	uses just "use TS" (two-byte) flag can't know when
>> 	to engage the timestamp and what initial value to use
>
> Well, as this would be a new mechanism, there should be some broad
> guidance given; I would assume that the latest onset - if late TS is
> negotiated for - would be when either direction has sent at most 2^31 -
> (2^16 << Window.scale) bytes; and a receiver that sees the full TS must
> then start to reflect TS as usual, independent of its own relative
> offset into the session. And never relinquish from sending TS even when
> some packets - ie. retransmissions, or due to middleboxes on a changed
> path - not contain a TS from the other side.

I had thought of a similar mechanism, e.g., where there would be an 
extended sequence number (ESN), and the upper bits would be assumed zero 
until actually needed.

The advantage to that would be:
	ESN flag would occur in all segments
		length = 2 until needed
		length can increase as additional bytes/words are needed

i.e., the meaning and use of the flag would never change; it would be 
synchronized at the start in both directions. It would just be 
compressed until actually needed. If you have a recent connection that 
already wrapped, you might start with ESN of length = 4 rather than 
length = 2 (0 in both directions), so you can sync non-zero ESNs.

The only benefit would be that this is a lot shorter than timestamps - 
i.e., it'd be 4 rather than 10 bytes.

(is this "Extended Sequence Number" option worth explaining in detail, 
e.g., in a separate draft?)

---

Doing real TS's later in the connection requires a new sort of 
synchronization in the middle of the connection - which, as I noted in 
my other post, seems to require a TWHS mechanism, and I think that's too 
cumbersome.

> As for the semantic change, I propose that when the session has SACK
> enabled as well as late TS, then the TS should reflect the TS of the
> segment triggering the ACK (we can discuss what should happen during
> in-sequence delayed ACK processing though; the relevant aspect for me is
> during loss - SACK recovery - to not mask the timestamps of
> retransmitted packets as per 1323bis; For legacy RTTM processing, the
> presence of a SACK option is indicative enough to NOT do normal RTTM
> processing - although a sender could choose to do so particularly for
> the loss recovery episodes. Having the (unique) Timestamp/SACK block
> combinations during loss recovery would allow more efficient, faster and
> also packet conserving recovery of lost retransmissions - unlike the
> current method of last resort, RTO.

Perhaps, but I don't see this paragraph as being related to late TS.

Joe


From nobody Fri May 30 18:46:23 2014
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To: Joe Touch <touch@isi.edu>
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Joe Touch <touch@isi.edu> wrote:
> 
> Some additional information below about "another trick" I proposed, 
> inspired by Bob's dual-SYN mechanism, courtesy a few long discussions 
> today with Ted Faber.

   This _is- as interesting idea...

> I'll be glad to offer this as a potential solution in the doc.

   I'd _much_ rather this be detailed in a separate document. The edo
proposal is reasonably easy to understand, while this contains multiple
layers of complexity.

   I _don't_ claim to understand it all -- if we try to publish this
in an RFC, a _lot_ of work will be needed to clarify everything...

> On 5/30/2014 1:34 PM, Joe Touch wrote:
> ...
>> Here's another trick that might clean up the above a little:
> 
> FWIW, I had explained it below as being based on sending out-of-window 
> data; Ted pointed out that I had been assuming that the FBP ACK bit 
> wasn't set - which means the sequence number might be more usefully 
> matched to that of the SYN.

   I see you now give the "second" packet the same sequence number.
I feel a need to describe the FBP more exactly. I'm guessing it's the
same port pair (but I very much dislike guessing!). Obviously it's
the same IP-address pair (but can we really depend upon middleboxes
to translate both packets the same?). I won't venture a guess as to
whether it has the SYN bit set (but thanks for saying the ACK bit
is clear). Et cetera...

> See below...
> 
>>     aso - after SYN option

   This is an option within the Options field of the first SYN, right?

> 		length = 2 (just a flag)
> 		length = 3 (or 4) indicating the length of the
> 			FBP expected

   What does length==2 tell us about the FBP?

>>     FBP - front bumper packet (best I could do on names today)
>>         a packet
> 		ISN = same as the associated SYN
> 		ACK = cleared (i.e., a data packet NOT part of
> 		synchronized connection - see why that's useful below)

   Does this have SYN set?

   How does the sender of the FBP learn what the receiver of the FBP
took it to mean? (Assuming, of course, it was received at all...)

>>     new endpoint sends:

   What's a "new endpoint"?

>>         SYN + aso + fix_opt
>>         FBP + aso + extra_opt
> 
> 		extra_opt in the data field
> 		total length < min MTU (576 for IPv4, 1280 for IPv6)
> 		again ACK bit is zero

   I get very nervous about using the data field before the handshake
is complete.

>>             legacy endpoint sends back one connections:
>>                 SYN-ACK + fix_opt

   Presumably the legacy endpoint ignores the FBP... but what happens
if only the FBP arrives (the initial SYN being lost or mangled by a
middlebox)?

>>                 if seg arrives before SYN,

   Am I supposed to assume "seg" here means FBP?

> 			it is silently dropped, because
> 			the ACK bit is clear (this is
> 			explicit in RFC793)
> 
> >                 if seg arrives after SYN,
> 
> 			it is silently dropped, because
> 			the ACK bit is clear (this is also
> 			explicit in RFC793)
>>             ----
>>
>>             new endpoint sends back one connection:
>>
>>                 SYN-ACK + options + ....
>>
>>             a) if FBP arrives before SYN,
> 
> 			it can be silently dropped, but
> 			it's probably useful for new endpoints
> 			to hold onto these (without action)
> 			for a while; they can be silently
> 			discarded if there are too many
> 			(which will just result in a
> 			retransmission and an extra RTT)

   Your hands are waving...

>>             b) if FPB arrives with the SYN, they can be
>>             processed together
>>
>>                 the SYN-ACK can include responses to
>>                 the extra_opts in addition to the
>>                 fix_opts, and says "FBP received"

   How would it say "FBP received"?

>>             c) if FPB arrives after the SYN:
>>
>>                 SYN-ACK proceeds, but sends
>>                 back "wait for option response".

   How would it send that?

>>                 at this point, the source re-sends FBP
>>                 until an ACK is received that indicates
>>                 "FBP received", or times-out as with
>>                 any connection that doesn't finish TWHS

   This sounds like the timeout wouldn't be rare -- a middlebox might
pretty automatically drop the FBP, depending...

>>             I'm still thinking as to whether the ACK number
>>             might indicate whether FBP has been received,

   That sounds strange...

> There are a few ways to handle this, but IMO the best is to have:
> 
> 		SYN-ACK aso with length=2 means "waiting for FBP"
> 
> 		SYN-ACK aso with length=3 (or 4) could mean
> 		"got the FBP", or might even indicate how
> 		many bytes the FBP extension contained

   That's limiting...

> Note - the SYN-ACK and all subsequent segments can assume that aso == 
> edo, i.e., they can use the same EDO as spec'd in version 01 of this draft.

   Please! No!

   (Option numbers aren't that scarce.)

--
John Leslie <john@jlc.net>


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On 5/30/2014 6:46 PM, John Leslie wrote:
> Joe Touch <touch@isi.edu> wrote:
>>
>> Some additional information below about "another trick" I proposed,
>> inspired by Bob's dual-SYN mechanism, courtesy a few long discussions
>> today with Ted Faber.
>
>     This _is- as interesting idea...
>
>> I'll be glad to offer this as a potential solution in the doc.
>
>     I'd _much_ rather this be detailed in a separate document. The edo
> proposal is reasonably easy to understand, while this contains multiple
> layers of complexity.
>
>     I _don't_ claim to understand it all -- if we try to publish this
> in an RFC, a _lot_ of work will be needed to clarify everything...

Sure...

>> On 5/30/2014 1:34 PM, Joe Touch wrote:
>> ...
>>> Here's another trick that might clean up the above a little:
>>
>> FWIW, I had explained it below as being based on sending out-of-window
>> data; Ted pointed out that I had been assuming that the FBP ACK bit
>> wasn't set - which means the sequence number might be more usefully
>> matched to that of the SYN.
>
>     I see you now give the "second" packet the same sequence number.

Yes.

> I feel a need to describe the FBP more exactly.

Of course.

> I'm guessing it's the
> same port pair (but I very much dislike guessing!).

Oh, yes - that's the way it ensures fate-sharing with the SYN.

> Obviously it's
> the same IP-address pair

Yes.

> (but can we really depend upon middleboxes
> to translate both packets the same?).

If the FBP comes after the SYN, yes. If not, the FBP will be dropped, 
and retransmitted when the SYN-ACK indicates the FBP hasn't been 
received (for upgraded servers).

 > I won't venture a guess as to
> whether it has the SYN bit set (but thanks for saying the ACK bit
> is clear). Et cetera...

It can't have the SYN bit - that would be two SYNs to the same place, 
and if they have different options that'd be confusing for legacy 
servers. If they had the same options, there'd be no increase in space - 
it's not possible to use the data field of a SYN for options because 
legacy hosts will interpret that as user data.

>> See below...
>>
>>>      aso - after SYN option
>
>     This is an option within the Options field of the first SYN, right?
>
>> 		length = 2 (just a flag)
>> 		length = 3 (or 4) indicating the length of the
>> 			FBP expected
>
>     What does length==2 tell us about the FBP?

Here's more specifically what I was thinking:

	Length = 2 (1 byte), Kind = fpb-option-codepoint (1 byte)

or

	Length = 3, Kind = fbp-option-codepoint, fbp-length (1 byte)

Length == 4 just means a longer fbp-length field, or maybe it means:

	Length, Kind, Flags, FBP-length
		(you'll see below where the flags might come in)

Length == 2 just means there's no fbp-length field. That could mean that 
the length of the FBP should be determined from the FBP packet itself 
(using fbp-length operates as a double-check that the FBP packet 
matches); alternately, it could mean that there's no FBP - i.e., that 
this is just the EDO that extends non-SYN packets.

Alternately, we could use different option codepoints for FBP and EDO, 
in which case the FBP option just means "FBP packet should be expected".

>>>      FBP - front bumper packet (best I could do on names today)
>>>          a packet
>> 		ISN = same as the associated SYN
>> 		ACK = cleared (i.e., a data packet NOT part of
>> 		synchronized connection - see why that's useful below)
>
>     Does this have SYN set?

No. that would be a second SYN, which causes load problems for legacy 
servers.

>     How does the sender of the FBP learn what the receiver of the FBP
> took it to mean? (Assuming, of course, it was received at all...)

The ACK that comes back, either in the SYN-ACK (if the FBP has been 
received) or a second ACK after that will have the response to the FBP 
options.

>>>      new endpoint sends:
>
>     What's a "new endpoint"?

An upgraded one, i.e., one that supports this new option...

>>>          SYN + aso + fix_opt
>>>          FBP + aso + extra_opt
>>
>> 		extra_opt in the data field
>> 		total length < min MTU (576 for IPv4, 1280 for IPv6)
>> 		again ACK bit is zero
>
>     I get very nervous about using the data field before the handshake
> is complete.

It's safe - as per RFC793. This packet is outside the connection because 
the ACK isn't set. That's effectively saying "this packet is for the 
unsynchronized state". RFC793 says all such packets get silently dropped 
(we're talking about a packet with ACK clear, and NO other control flags 
set - no SYN, no RST, no FIN).

>>>              legacy endpoint sends back one connections:
>>>                  SYN-ACK + fix_opt
>
>     Presumably the legacy endpoint ignores the FBP... but what happens
> if only the FBP arrives (the initial SYN being lost or mangled by a
> middlebox)?

They ignore it at all times, except if received in CLOSED at which point 
they send a RST (just like receiving any other packet in that state).

If the SYN is lost, the same thing happens as any time the SYN is lost - 
the client resends the SYN after a timeout, and after some number of 
failed timeouts it gives up.

Upgraded endpoints might keep FBPs for a short time in the hope of 
getting a matching SYN, but if they don't or discard it before it comes, 
the client will resend it when the SYN-ACK indicates its missing.

>>>                  if seg arrives before SYN,
>
>     Am I supposed to assume "seg" here means FBP?

Sorry - typed too fast. Yes, here and throughout the rest...

>> 			it is silently dropped, because
>> 			the ACK bit is clear (this is
>> 			explicit in RFC793)
>>
>>>                  if seg arrives after SYN,
>>
>> 			it is silently dropped, because
>> 			the ACK bit is clear (this is also
>> 			explicit in RFC793)
>>>              ----
>>>
>>>              new endpoint sends back one connection:
>>>
>>>                  SYN-ACK + options + ....
>>>
>>>              a) if FBP arrives before SYN,
>>
>> 			it can be silently dropped, but
>> 			it's probably useful for new endpoints
>> 			to hold onto these (without action)
>> 			for a while; they can be silently
>> 			discarded if there are too many
>> 			(which will just result in a
>> 			retransmission and an extra RTT)
>
>     Your hands are waving...

Not that hard. Keeping a cache of recently received FBPs is a simple 
thing to do, and the mechanism is robust to that cache either not 
existing or being small.

>>>              b) if FPB arrives with the SYN, they can be
>>>              processed together
>>>
>>>                  the SYN-ACK can include responses to
>>>                  the extra_opts in addition to the
>>>                  fix_opts, and says "FBP received"
>
>     How would it say "FBP received"?

I thought of a few ways - they all end up being equivalent in spirit to 
a flag in an option field in the SYN-ACK, so let's say it's that for now:

	set FBP-received flag in the aso option

>>>              c) if FPB arrives after the SYN:
>>>
>>>                  SYN-ACK proceeds, but sends
>>>                  back "wait for option response".
>
>     How would it send that?

The FBP-received flag in the SYN-ACK aso option isn't set.

>>>                  at this point, the source re-sends FBP
>>>                  until an ACK is received that indicates
>>>                  "FBP received", or times-out as with
>>>                  any connection that doesn't finish TWHS
>
>     This sounds like the timeout wouldn't be rare -- a middlebox might
> pretty automatically drop the FBP, depending...

If the FBP comes before the SYN, yes. If not, it shouldn't AFAICT.

>>>              I'm still thinking as to whether the ACK number
>>>              might indicate whether FBP has been received,
>
>     That sounds strange...

Yeah - nevermind that.

>> There are a few ways to handle this, but IMO the best is to have:
>>
>> 		SYN-ACK aso with length=2 means "waiting for FBP"
>>
>> 		SYN-ACK aso with length=3 (or 4) could mean
>> 		"got the FBP", or might even indicate how
>> 		many bytes the FBP extension contained
>
>     That's limiting...

I don't see why (FBP can indicate 32-bit words, just like the current 
Data Offset field, which means 255 = 1020 bytes of option space in the 
FBP -- note that this is all contained in the data portion of the FBP 
segment).

>> Note - the SYN-ACK and all subsequent segments can assume that aso ==
>> edo, i.e., they can use the same EDO as spec'd in version 01 of this draft.
>
>     Please! No!
>
>     (Option numbers aren't that scarce.)

See above. We could say:
	
	EDO length = 0 in a SYN means no FBP
	EDO length = N in a SYN means FBP with length N

But let's not get hung on that detail either, IMO.

Joe


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Joe,
>>
>> I was asking for an example of something useful that /can/ be done with
>> EDO.
>
> The obvious one is larger amounts of SACK feedback and space for MPTCP
> information.

It would be intersting to look at packet traces to see how often the 
SACK option is not large enough to fit into x bytes. This would give an 
indication of its impact. Has someone already seen such a study in the 
literature ?


Olivier


-- 
INL, ICTEAM, UCLouvain, Belgium, http://inl.info.ucl.ac.be


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Joe,
>
> On 5/30/2014 10:53 AM, Olivier Bonaventure wrote:
> ...
>> Let me make two rough proposals that could be worth being explored or at
>> least documented.
>>
>> 1. The TCP option space is limited, but not the IPv6 destination options
>> in the IPv6 header.
>
> IP option space would not be protected by TCP-AO, and likely not by
> whatever the TCPUSEC WG creates (e.g.,my proposal is here:
> http://tools.ietf.org/html/draft-touch-tcp-ao-encrypt-01)

If the destination option is used to transport TCP options, then it 
would define how AO would need to take into account the TCP option 
inserted by the sender in the IPv6 option.


> Additionally, DPI, NAT processing, etc. are all impacted by IPv6
> options, so they're generally discouraged AFAIR.


IPv6 options are much cleaner than IPv4 options. IPv4 options are 
discouraged because they impact the forwarding on routers. For IPv6 
options, this is easier since the option type indicates that the option 
should only be processed by the destination. For NATs or DPI, supporting 
TCP options as an IPv6 destination option or as any other proposal would 
be the same problem


Olivier


-- 
INL, ICTEAM, UCLouvain, Belgium, http://inl.info.ucl.ac.be


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To: Joe Touch <touch@isi.edu>
From: Bob Briscoe <bob.briscoe@bt.com>
In-Reply-To: <5388EB6F.4010405@isi.edu>
References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <1AD79820-22C1-4500-84D1-1383F264D68C@weston.borman.com> <201405231213.s4NCDa5P005525@bagheera.jungle.bt.co.uk> <537F8202.4020907@isi.edu> <201405281715.s4SHFMm0014634@bagheera.jungle.bt.co.uk> <538623B9.2060209@isi.edu> <201405301642.s4UGgcvY030471@bagheera.jungle.bt.co.uk> <5388EB6F.4010405@isi.edu>
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Subject: [tcpm] More TCP option space on SYNs
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Joe,

Thx for consolidating this thread. I've given it a new subject line.

1) You've silently made an important alteration to the proposed 
protocol. You've put the extra-options directly in the TCP option 
space of the C-SYN, not within the payload. This creates problems:
a) it limits additional options to another 40B, beyond which we will 
need a third SYN, then a fourth.
b) it perpetuates the deployment problem that every newly defined TCP 
option will have: when deployed in year y, there will be a new crop 
of middleboxes that only forward options defined up to year y-1.

I had deliberately squirrelled the options away in the app layer to:
a) provide expansion space for options on a SYN, limited only by the 
max segment size
b) reduce the chances that middleboxes will alter the extra options, 
given there is a higher bar to altering the payload.
c) allow for future structured ways to make extra options invisible 
and/or immutable by middleboxes.

Yes, your altered proposal is cleaner. However, don't imagine I 
didn't think of this. I did and I deliberately didn't do it this way. 
We have a choice:
         clean and vulnerable vs. messy but robust.

I'm not wedded to using port 80 and http headers, but this is perhaps 
the most pragmatic approach. It will be really unorthodox to define 
such a protocol I know. We would have to say something like

         "The dst port of the C-SYN MUST be 80, and the payload MUST start
         with the constant magic_token, where
         magic_token = 'PUT / HTTP/1.1<CRLF>Connection : DSO<CRLF><CRLF>'
         "

I'm sorry if even thinking about this makes you feel dirty :|

Other suggestions for inner protocols are welcome, including 
tunnelled protocols, as long as middleboxes widely forward them, 
given their dst port.


2) The main problem with your notation is it doesn't say /where/ the 
info is placed.
I've added notation as follows:
TCP(base header [TCP options [APP(header[payload])]])

And for the record I've made the if-else logic clearer.

Where I've made more than clarifying edits inline, I've described 
them and tagged them with [BB].

At 21:34 30/05/2014, Joe Touch wrote:
>Hi, Bob,
>
>Let's get back to the core, in a simpler fashion, so other can follow it.
>
>I stand by my "there's no way to extend the space in the initial 
>SYN", but you've convinced me there *might* be a way to provide 
>extended space that can occur during the first phase of the TWHS. I 
>think the dual-SYN approach still isn't viable, but I've outlined an 
>alternative below that's similar but doesn't have the same baggage, IMO.
>
>Again, I'm still concerned by what midboxes might do to this...
>
>What do others think??
>
>Joe
>
>For quick review, here's what I understand:
>
>                 dso = dual-syn option
>                         dso-D = data
>                         dso-C = control
>                 conn_id = identifier to link the two SYNs together
>                 extra_opt = options that didn't fit in legacy SYN
>                 fit_opt = options that do fit in the legacy SYN
         new client endpoint sends
                 TCP(port A SYN [dso-D(conn_id) + fit_opt] )
                 TCP(port B SYN [dso-C [APP(headers [conn_id + 
extra_opt] ) ] ] )

[BB]: i/APP(headers...)/

                         if (legacy server endpoint) { sends back two 
connections:
                                 TCP(port A SYN-ACK [fit_opt] )
                                 TCP(port B SYN-ACK [??] )
>                                 (it's interpretation of extra_opt)
                                         new client endpoint responds:
                                         TCP(port A ACK) (established)
                                         TCP(port B RST)

>                         Notes about legacy servers:
>                                 - they do twice the work on SYNs
>                                 - they might keep twice the state
>                                 (if not using cookies)
>                                 - they might clean state if the RST
>                                 is received, but that state might
>                                 persist indefinitely (until the next
>                                 connection, depending on timeouts, etc.)
>
>                         -----
                         } elif (new server endpoint) { sends back 
one connection:
                                 TCP(port A SYN-ACK [edo + fit_opt + 
extra_opt] )

[BB]: s/dso-d/edo/
                                         new client endpoint responds:
                                         TCP(port A ACK) (established)

>
>                         Notes:
>                                 - can stall when dso-D SYN arrives
>                                 before dso-C SYN, up to some limit
>                                 - twice the work on SYNs (or more)
                         }

>Here's what I was assuming, though admittedly it's not documented (yet):
>
>         - no significant impact on TCP connection rate for
>         legacy servers
>
>         - no significant impact on TCP connection rate for
>         legacy clients
>
>         - impact dominated by processing the extended option space
>         for extended clients
>
>         - impact dominated by processing the extended option space
>         for extended servers
>
>         - compatible with typical TCP processing optimizations,
>         notably SYN cookies
>                 you did provide a potential way forward for these
>
>         - capable of successfully traversing typical NATs
>
>Your approach has the following properties:

The 3 bullets below are not useful ways to describe performance 
impact. They selectively describe whichever gives the most 
pessimistic picture out of:
a) either the instantaneous performance change at the moment of connection
b) or the worst-case long-run performance impact

They don't describe the average long-run performance impact, which is 
important for sizing machines.

Worse, the instantaneous performance impact is only significant when 
a machine's SYN processing time is large relative to the e2e delay, 
which would be a highly unusual scenario on public networks (even in 
scenarios such as intra-data-centre, it's hard to reduce e2e delay to 
approach SYN processing time, but you could for intra-machine connections).


>         - halves the server connection rate for updated servers
>         from legacy clients when this option is in use

Eh? The long-run server connection rate will be fractionally 
decreased due to updated clients using extra options (which is your 
third case below), but the instantaneous server connection rate seen 
by a legacy client is unchanged, because it only sends one SYN.


>         - lowers (to some extent, if not halves) the client
>         connection rate of updated clients to all servers
>         when this option is in use
>
>         - halves (roughly) the server rate for all servers
>         when this option is in use

Nope. All long-run server rates are reduced by 1/(1+e), where e is 
the fraction of connections using extra options.


>It also:
>
>         - doubles the number of SYNs in the network

Nope. The number of SYNs in the network is inflated by e where e is 
the fraction of connections using extra options.


>         - susceptible to lack of fate-sharing problems, e.g.,
>         if the two SYNs experience different firewall configurations

Nope. It's fairer to say it's potentially susceptible to second-order 
fate-sharing problems like your firewall example (the first-order 
fate sharing problems have been addressed).


>         - reduces the space available for fit_opt due to the need
>         for the conn_id even in the fall-back D-SYN, which means
>         less option space in the SYNs for fall-back connections

Yup.


>         the conn_id which may need to be very large because it
>         needs to be unique per source port and source IP address
>         because that information is lost during NAT translation

Given many NATs will typically make the src IPs of both SYNs the 
same, I suggest a larger conn_id should be a fall-back option for the 
client, not a default.

Even if the src IPs of both SYNs are different once they reach the 
server, the high end bits will invariably be the same. So the max 
size of the contents of the DSO TCP option can be 6B, and the server 
can take the rest of conn_id from the higher bits of the src IP addr 
of each SYN. This is a variant of the idea in <draft-wing-nat-reveal-option>.

In fact, the server doesn't even need a small conn_id for clients 
that know they are not behind a NAT and that want more option space 
in the D-SYN - then the server could use the src port & src IP for the conn_id.

To summarise, these options could be distinguished by the length 
field of the dual-syn option.
Length = 2B                             => conn_id = src netaddr + src port
Length = 6B = 2B +4B conn_id_short      => conn_id = src netaddr + 
conn_id_short
Length = 8B = 2B +6B conn_id_long       => conn_id = hsb(src netaddr) 
+ conn_id_long

Given there have been numerous other attempts to reveal a connection 
ID that is preserved through middleboxes [RFC6967], rather than 
defining a dual-syn option that carries a conn_id, we might want to 
design a TCP connection ID option with a flag to say whether it is 
also part of a dual SYN pair or not.

Where
* hsb(src netaddr) is the netaddr with the lowest 16 bits truncated
* src netaddr is the network address (IPv4, IPv6, or any other 
network protocol)

To reduce latency, a host could use the default short_conn_id for all 
connections at first, then:
- if it finds that DSO persistently doesn't work it falls back to the 
long_conn_id for all connections
- it occasionally tests the short and zero options to see if it can 
use shorter DSO options.


>         - requires the ISNs to be related (see RFC6528 - if there's
>         a rule to generate it, there will be code to validate that
>         rule, and eventually a BCP to encourage that validation -
>         typically from the same RFC author)

Eh? The ISNs can and should be independent. To be robust against 
middleboxes that rewrite sequence numbers, we must not required ISNs 
to be related.


>I agree that you have proposed potentially viable ways to deal with 
>the SYN cookie, and that RST state is not an issue.

A feature that I think it's fair to add:
         - Good chance of passing through app-layer middleboxes that forward
         unrecognised TCP options unchanged, but not those that discard them.


>However, there are too many problems with this, IMO, to call it viable.

Once your over-pessimistic analyses of the performance impact are 
corrected, and my ideas to reduce the size of the conn_id are taken 
into account, it's a different story.

But it's up to the WG to decide whether this is worth taking further. 
Not just you or I.

>Here's another trick that might clean up the above a little:

<snip - I'll respond separately to your later updates on this ASO 
idea, with ACK=0>

Cheers


Bob


________________________________________________________________
Bob Briscoe,                                                  BT  


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Date: Sat, 31 May 2014 12:20:54 -0700
From: Joe Touch <touch@isi.edu>
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To: Bob Briscoe <bob.briscoe@bt.com>
References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <1AD79820-22C1-4500-84D1-1383F264D68C@weston.borman.com> <201405231213.s4NCDa5P005525@bagheera.jungle.bt.co.uk> <537F8202.4020907@isi.edu> <201405281715.s4SHFMm0014634@bagheera.jungle.bt.co.uk> <538623B9.2060209@isi.edu> <201405301642.s4UGgcvY030471@bagheera.jungle.bt.co.uk> <5388EB6F.4010405@isi.edu> <201405311819.s4VIJt2V003823@bagheera.jungle.bt.co.uk>
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Subject: Re: [tcpm] More TCP option space on SYNs
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Hi, Bob,

On 5/31/2014 11:19 AM, Bob Briscoe wrote:
> Joe,
>
> Thx for consolidating this thread. I've given it a new subject line.
>
> 1) You've silently made an important alteration to the proposed
> protocol. You've put the extra-options directly in the TCP option space
> of the C-SYN, not within the payload.

I hadn't assumed that; I was describing only which SYN carried which 
options, not where they were. As you do, I was assuming the C-SYN 
options were in the data space.

To be clear, in my variant - using data outside the sync'd connection 
(with ACK clear), the option space is similarly in the data portion of 
the 'extra' segment.

> Yes, your altered proposal is cleaner. However, don't imagine I didn't
> think of this. I did and I deliberately didn't do it this way.We have a
> choice:
>          clean and vulnerable vs. messy but robust.
>
> I'm not wedded to using port 80 and http headers, but this is perhaps
> the most pragmatic approach.

That's fraught with a whole slew of other problems, notably the 
middleboxes that currently try to translate, proxy, or otherwise modify 
anything they see as port 80.

Further, that would limit the number of pending connections.

> It will be really unorthodox to define such
> a protocol I know. We would have to say something like
>
>          "The dst port of the C-SYN MUST be 80, and the payload MUST start
>          with the constant magic_token, where
>          magic_token = 'PUT / HTTP/1.1<CRLF>Connection : DSO<CRLF><CRLF>'
>          "
>
> I'm sorry if even thinking about this makes you feel dirty :|
>
> Other suggestions for inner protocols are welcome, including tunnelled
> protocols, as long as middleboxes widely forward them, given their dst
> port.

I don't see what this gets us other than potential interactions. There's 
no reason to use HTTP to format TCP options, and it would necessitate 
new encodings and new processing for existing options that might end up 
in the extended space.

> 2) The main problem with your notation is it doesn't say /where/ the
> info is placed.

EDO says the options are at the head of the data segment, just with a 
longer pointer - the same way as current options.

For extending the SYN using a separate segment - whether a C-SYN or 
unsync'd data segment - the data similarly comes at the head of the 
data, just that there's no user data permitted after it.

> I've added notation as follows:
> TCP(base header [TCP options [APP(header[payload])]])
>
> And for the record I've made the if-else logic clearer.
>
> Where I've made more than clarifying edits inline, I've described them
> and tagged them with [BB].
>
> At 21:34 30/05/2014, Joe Touch wrote:
>> Hi, Bob,
>>
>> Let's get back to the core, in a simpler fashion, so other can follow it.
>>
>> I stand by my "there's no way to extend the space in the initial SYN",
>> but you've convinced me there *might* be a way to provide extended
>> space that can occur during the first phase of the TWHS. I think the
>> dual-SYN approach still isn't viable, but I've outlined an alternative
>> below that's similar but doesn't have the same baggage, IMO.
>>
>> Again, I'm still concerned by what midboxes might do to this...
>>
>> What do others think??
>>
>> Joe
>>
>> For quick review, here's what I understand:
>>
>>                 dso = dual-syn option
>>                         dso-D = data
>>                         dso-C = control
>>                 conn_id = identifier to link the two SYNs together
>>                 extra_opt = options that didn't fit in legacy SYN
>>                 fit_opt = options that do fit in the legacy SYN
>          new client endpoint sends
>                  TCP(port A SYN [dso-D(conn_id) + fit_opt] )
>                  TCP(port B SYN [dso-C [APP(headers [conn_id +
> extra_opt] ) ] ] )

IMO, the conn_ID in the C-SYN should similarly be in the regular option 
space, but I don't think it matters.

> [BB]: i/APP(headers...)/
>
>                          if (legacy server endpoint) { sends back two
> connections:
>                                  TCP(port A SYN-ACK [fit_opt] )
>                                  TCP(port B SYN-ACK [??] )
>>                                 (it's interpretation of extra_opt)

I'm assuming EDO after the SYN, which means the SYN-ACK has as much 
space as it wants for options, as would every other segment.

>                                          new client endpoint responds:
>                                          TCP(port A ACK) (established)
>                                          TCP(port B RST)
>
>>                         Notes about legacy servers:
>>                                 - they do twice the work on SYNs
>>                                 - they might keep twice the state
>>                                 (if not using cookies)
>>                                 - they might clean state if the RST
>>                                 is received, but that state might
>>                                 persist indefinitely (until the next
>>                                 connection, depending on timeouts, etc.)
>>
>>                         -----
>                          } elif (new server endpoint) { sends back one
> connection:
>                                  TCP(port A SYN-ACK [edo + fit_opt +
> extra_opt] )
>
> [BB]: s/dso-d/edo/

Sure. At this point we're basically in EDO-land.

>                                          new client endpoint responds:
>                                          TCP(port A ACK) (established)
>
>>
>>                         Notes:
>>                                 - can stall when dso-D SYN arrives
>>                                 before dso-C SYN, up to some limit
>>                                 - twice the work on SYNs (or more)
>                          }
>
>> Here's what I was assuming, though admittedly it's not documented (yet):
>>
>>         - no significant impact on TCP connection rate for
>>         legacy servers
>>
>>         - no significant impact on TCP connection rate for
>>         legacy clients
>>
>>         - impact dominated by processing the extended option space
>>         for extended clients
>>
>>         - impact dominated by processing the extended option space
>>         for extended servers
>>
>>         - compatible with typical TCP processing optimizations,
>>         notably SYN cookies
>>                 you did provide a potential way forward for these
>>
>>         - capable of successfully traversing typical NATs
>>
>> Your approach has the following properties:
>
> The 3 bullets below are not useful ways to describe performance impact.
> They selectively describe whichever gives the most pessimistic picture
> out of:
> a) either the instantaneous performance change at the moment of connection
> b) or the worst-case long-run performance impact
>
> They don't describe the average long-run performance impact, which is
> important for sizing machines.
>
> Worse, the instantaneous performance impact is only significant when a
> machine's SYN processing time is large relative to the e2e delay, which
> would be a highly unusual scenario on public networks (even in scenarios
> such as intra-data-centre, it's hard to reduce e2e delay to approach SYN
> processing time, but you could for intra-machine connections).

SYN processing is a known load - and potential overload - on machines. 
That's why it's reasonable to focus on it.

>>         - halves the server connection rate for updated servers
>>         from legacy clients when this option is in use
>
> Eh? The long-run server connection rate will be fractionally decreased
> due to updated clients using extra options (which is your third case
> below), but the instantaneous server connection rate seen by a legacy
> client is unchanged, because it only sends one SYN.

Agreed. I was doing too many of these at once...

>>         - lowers (to some extent, if not halves) the client
>>         connection rate of updated clients to all servers
>>         when this option is in use
>>
>>         - halves (roughly) the server rate for all servers
>>         when this option is in use
>
> Nope. All long-run server rates are reduced by 1/(1+e), where e is the
> fraction of connections using extra options.

the rate for connections when this option is in use is halved - I didn't 
see that the sentence above could be parsed the other way. the overall 
rate depends on the fraction of connections using the option.

>> It also:
>>
>>         - doubles the number of SYNs in the network
>
> Nope. The number of SYNs in the network is inflated by e where e is the
> fraction of connections using extra options.

You keep bringing this up - I don't like a solution that should be used 
only a little. If it works, we should assume it works everywhere all the 
time.

>>         - susceptible to lack of fate-sharing problems, e.g.,
>>         if the two SYNs experience different firewall configurations
>
> Nope. It's fairer to say it's potentially susceptible to second-order
> fate-sharing problems like your firewall example (the first-order fate
> sharing problems have been addressed).

I have no idea why you think firewalls are a second-order fate-sharing, 
or what that even means. Fate sharing means fate is shared. When fate is 
not shared by the SYNs, this method has problems. There's no mechanism 
to recover the missing D-SYN or C-SYN.

>>         - reduces the space available for fit_opt due to the need
>>         for the conn_id even in the fall-back D-SYN, which means
>>         less option space in the SYNs for fall-back connections
>
> Yup.
>
>
>>         the conn_id which may need to be very large because it
>>         needs to be unique per source port and source IP address
>>         because that information is lost during NAT translation
>
> Given many NATs will typically make the src IPs of both SYNs the same, I
> suggest a larger conn_id should be a fall-back option for the client,
> not a default.

Multiple NATs are quite common - carrier grade NATs in the net, and user 
NATs at the home WIFI point.

Any multiple-NAT, or any path through even a single carrier-grade NAT 
will end up potentially un-doing or over-doing source IP overlap, 
creating false positives and false negatives that break this mechanism.

That's really the crux - fate sharing and the need to bind the two SYNs 
- that make this untenable, IMO.

> Even if the src IPs of both SYNs are different once they reach the
> server, the high end bits will invariably be the same.

You can't know this - the two SYNs could have gone through different 
numbers of NATs or different NATs entirely, so the bits could change 
arbitrarily.

> So the max size
> of the contents of the DSO TCP option can be 6B, and the server can take
> the rest of conn_id from the higher bits of the src IP addr of each SYN.
> This is a variant of the idea in <draft-wing-nat-reveal-option>.

FWIW, you're now in the land where we would have to start explaining the 
privacy and tracking implications of those bits. I'd really like to 
avoid that.

> In fact, the server doesn't even need a small conn_id for clients that
> know they are not behind a NAT and that want more option space in the
> D-SYN - then the server could use the src port & src IP for the conn_id.
>
> To summarise, these options could be distinguished by the length field
> of the dual-syn option.
> Length = 2B                             => conn_id = src netaddr + src port
> Length = 6B = 2B +4B conn_id_short      => conn_id = src netaddr +
> conn_id_short
> Length = 8B = 2B +6B conn_id_long       => conn_id = hsb(src netaddr) +
> conn_id_long
>
> Given there have been numerous other attempts to reveal a connection ID
> that is preserved through middleboxes [RFC6967], rather than defining a
> dual-syn option that carries a conn_id, we might want to design a TCP
> connection ID option with a flag to say whether it is also part of a
> dual SYN pair or not.
>
> Where
> * hsb(src netaddr) is the netaddr with the lowest 16 bits truncated
> * src netaddr is the network address (IPv4, IPv6, or any other network
> protocol)
>
> To reduce latency, a host could use the default short_conn_id for all
> connections at first, then:
> - if it finds that DSO persistently doesn't work it falls back to the
> long_conn_id for all connections
> - it occasionally tests the short and zero options to see if it can use
> shorter DSO options.
>
>
>>         - requires the ISNs to be related (see RFC6528 - if there's
>>         a rule to generate it, there will be code to validate that
>>         rule, and eventually a BCP to encourage that validation -
>>         typically from the same RFC author)
>
> Eh? The ISNs can and should be independent. To be robust against
> middleboxes that rewrite sequence numbers, we must not required ISNs to
> be related.
>
>
>> I agree that you have proposed potentially viable ways to deal with
>> the SYN cookie, and that RST state is not an issue.
>
> A feature that I think it's fair to add:
>          - Good chance of passing through app-layer middleboxes that
> forward
>          unrecognised TCP options unchanged, but not those that discard
> them.
>
>
>> However, there are too many problems with this, IMO, to call it viable.
>
> Once your over-pessimistic analyses of the performance impact are
> corrected, and my ideas to reduce the size of the conn_id are taken into
> account, it's a different story.
>
> But it's up to the WG to decide whether this is worth taking further.
> Not just you or I.

Agreed.

>> Here's another trick that might clean up the above a little:
>
> <snip - I'll respond separately to your later updates on this ASO idea,
> with ACK=0>
>
> Cheers
>
>
> Bob
>
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT


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Subject: Re: [tcpm] New Version Notification for draft-touch-tcpm-tcp-edo-01.txt
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On May 31, 2014, at 5:55 AM, Olivier Bonaventure =
<Olivier.Bonaventure@uclouvain.be> wrote:

> Joe,
>>=20
>> On 5/30/2014 10:53 AM, Olivier Bonaventure wrote:
>> ...
>>> Let me make two rough proposals that could be worth being explored =
or at
>>> least documented.
>>>=20
>>> 1. The TCP option space is limited, but not the IPv6 destination =
options
>>> in the IPv6 header.
>>=20
>> IP option space would not be protected by TCP-AO, and likely not by
>> whatever the TCPUSEC WG creates (e.g.,my proposal is here:
>> http://tools.ietf.org/html/draft-touch-tcp-ao-encrypt-01)
>=20
> If the destination option is used to transport TCP options,
> then it would define how AO would need to take into account the TCP =
option inserted by the sender in the IPv6 option.

TCP doesn't necessarily get the entire IP header. It's not practical to =
claim that TCP-AO can protect something it might never see - and =
certainly might not have control over once it's inserted in the IPv6 =
header.


>> Additionally, DPI, NAT processing, etc. are all impacted by IPv6
>> options, so they're generally discouraged AFAIR.
>=20
>=20
> IPv6 options are much cleaner than IPv4 options. IPv4 options are =
discouraged because they impact the forwarding on routers. For IPv6 =
options, this is easier since the option type indicates that the option =
should only be processed by the destination. For NATs or DPI, supporting =
TCP options as an IPv6 destination option or as any other proposal would =
be the same problem

IPv6 options push the location of TCP port numbers to places that need =
to be computed, rather than being known in advance - and wired into =
hardware.

Anything that looks at ports might toss something with *any* IP options =
- v4 or v6 - to slow-path.

Joe=


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On May 31, 2014, at 5:57 AM, Olivier Bonaventure =
<Olivier.Bonaventure@uclouvain.be> wrote:

> Joe,
>>>=20
>>> I was asking for an example of something useful that /can/ be done =
with
>>> EDO.
>>=20
>> The obvious one is larger amounts of SACK feedback and space for =
MPTCP
>> information.
>=20
> It would be intersting to look at packet traces to see how often the =
SACK option is not large enough to fit into x bytes.

You'll never see that in a trace - what you'll see is SACK options that =
fit, of course.

> This would give an indication of its impact. Has someone already seen =
such a study in the literature ?

I don't know, though there are armies that mine traces for all sorts of =
stuff, so I wouldn't be surprised.

Joe=


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To: Joe Touch <touch@isi.edu>
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Subject: [tcpm] SYN extension using ACK=0 data packets
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Joe,

Hope it's ok to have changed the subject line, with tcpm still in cc.

I'm afraid I'm not as excited about ACK=0 as you are. It's certainly 
cleaner than anything we've come up with so far.

However, I see the goal as finding a way to send a supplement to a 
SYN that is invalid in some way to legacy TCP servers, but likely to 
appear valid to many/most middleboxes. I suspect many middleboxes and 
firewalls will discard the ACK=0 segment.

When assessing the DSO scheme, you were adamant that anything invalid 
to an endpoint would always eventually become invalid to middleboxes. 
In the excitement of finding a nice clean way of doing ASO, that 
critique seems to have suddenly become unimportant.

Whatever, there may be a place for both solutions:
a) ASO (ACK=0) for paths where a NAT might be the worst middlebox on the path
b) DSO for paths with stateful firewalls, TCP normalisers etc.

The cellular world is certainly more like (b). Whether there are many 
parts of the public Internet left like (a) is unclear. (b) seems to 
have become the norm for most paths.

More inline...

At 01:45 31/05/2014, Joe Touch wrote:
>Hi, all,
>
>Some additional information below about "another trick" I proposed, 
>inspired by Bob's dual-SYN mechanism, courtesy a few long 
>discussions today with Ted Faber.
>
>I'll be glad to offer this as a potential solution in the doc.
>
>Joe
>
>On 5/30/2014 1:34 PM, Joe Touch wrote:
>...
>>Here's another trick that might clean up the above a little:
>
>FWIW, I had explained it below as being based on sending 
>out-of-window data; Ted pointed out that I had been assuming that 
>the FBP ACK bit wasn't set - which means the sequence number might 
>be more usefully matched to that of the SYN.
>
>See below...
>
>>      aso - after SYN option
>
>                 length = 2 (just a flag)
>                 length = 3 (or 4) indicating the length of the
>                         FBP expected
>
>>      FBP - front bumper packet (best I could do on names today)
>>          a packet
>                 ISN = same as the associated SYN
>                 ACK = cleared (i.e., a data packet NOT part of
>                 synchronized connection - see why that's useful below)

For the avoidance of doubt,
I assume these two packets have the same src port.
and I assume the FBP has SYN=0.

Coincidentally, a couple of weeks ago, when I was looking for places 
to find more bits in the TCP header, I noted that during an 
established connection ACK=0 is never used, so I looked into setting 
ACK=0 and overloading the ack_number field.


>>      new endpoint sends:
>>
>>          SYN + aso + fix_opt
>>          FBP + aso + extra_opt
>
>                 extra_opt in the data field
>                 total length < min MTU (576 for IPv4, 1280 for IPv6)
>                 again ACK bit is zero
>
>>              legacy endpoint sends back one connections:
>>                  SYN-ACK + fix_opt
>>
>>                  if seg arrives before SYN,
>
>                         it is silently dropped, because
>                         the ACK bit is clear (this is
>                         explicit in RFC793)
>
>>                  if seg arrives after SYN,
>
>                         it is silently dropped, because
>                         the ACK bit is clear (this is also
>                         explicit in RFC793)

Two important questions:
1) are most/all legacy TCP implementations faithful to the spec?
2) if a TCP endpoint is meant to drop SYN=0, ACK=0, then many 
middleboxes surely will.

Both these will need experimental testing.

Nit: I wouldn't exactly say RFC793 is clear. You have to follow 2 
pages of quite imprecise descriptive logic, starting from p66. But, 
yes it is eventually fairly unambiguous. Paraphrasing, it says:

SEGMENT ARRIVES
         ...
         if state = SYN-SENT
                 1. Check ACK bit
                         if ACK = 1
                                 do stuff
                 2. Check RST bit
                         if RST = 1
                                 if ACK OK enter CLOSED state
                                 elif no ACK, drop
                 3. Check security & precedence
                         Do stuff
                 4. Check SYN bit
                         Should only get here if ACK OK, or no ACK and no RST
                         if SYN = 1
                                 do stuff
                 5. if SYN = 0 and RST = 0
                         drop


>>              ----
>>
>>              new endpoint sends back one connection:
>>
>>                  SYN-ACK + options + ....
>>
>>              a) if FBP arrives before SYN,
>
>                         it can be silently dropped, but
>                         it's probably useful for new endpoints
>                         to hold onto these (without action)
>                         for a while; they can be silently
>                         discarded if there are too many
>                         (which will just result in a
>                         retransmission and an extra RTT)
>
>>              b) if FPB arrives with the SYN, they can be
>>              processed together

By 'arrives with' I assume you mean 'within some time duration'.

>>                  the SYN-ACK can include responses to
>>                  the extra_opts in addition to the
>>                  fix_opts, and says "FBP received"
>>
>>
>>              c) if FPB arrives after the SYN:
>>
>>                  SYN-ACK proceeds, but sends
>>                  back "wait for option response".
>>
>>                  at this point, the source re-sends FBP
>>                  until an ACK is received that indicates
>>                  "FBP received", or times-out as with
>>                  any connection that doesn't finish TWHS

There's a dilemma whether the server:
* prioritises latency and goes ahead without the extra options,
* or prioritise completeness and blocks until the extra options arrive.

I would take the view that if the FBP is late there's a chance it got 
snarled up in a middlebox, and it will never get through no matter 
how many times it's retransmitted.

Rather than make the choice between latency and completeness at 
protocol design time, we need a flag in either scheme (on the DSO or 
ASO option) for the client to say which it wants. Then:
* the client can choose latency if it knows the extra_opts are not critical.
* otherwise it can choose completeness, and if it doesn't work after 
a retransmit or two, the client won't block for ever; it can work out 
the compromise set of options that fit in a single SYN but still 'work'.

>>              I'm still thinking as to whether the ACK number
>>              might indicate whether FBP has been received,
>
>There are a few ways to handle this, but IMO the best is to have:
>
>                 SYN-ACK aso with length=2 means "waiting for FBP"
>
>                 SYN-ACK aso with length=3 (or 4) could mean
>                 "got the FBP", or might even indicate how
>                 many bytes the FBP extension contained
>
>Note - the SYN-ACK and all subsequent segments can assume that aso 
>== edo, i.e., they can use the same EDO as spec'd in version 01 of this draft.
>
>>This is cleaner as follows:
>>
>>      - no need for conn_id coordination
>>
>>      - no need for conn_id to consume option space for fall-back
>>
>>      - avoids double-load for legacy servers

With a typical legacy server that reflects SYN cookies, less activity 
is doubled:
* my scheme: sends out two SYN cookies
* your scheme: sends out one SYN cookie and drops one packet, 
probably after a long chain of logic, because the FBP is an unusual packet.

>>      - no problem with fate-sharing

To be as consistently pessimistic as you were with my scheme, I think 
you mean that you have created machinery to synthesise fate sharing 
between a connection and an invalid segment, and it may not work in 
cases that you may not have thought of.

>>      - traverses a NAT just fine

As above, surely TCP normalising middleboxes and incoming stateful 
firewalls at the server end are likely to discard the FBP.

>>Upgraded servers still need to wait for the 'seg', but they could get
>>that retransmitted if necessary.

See discussion of latency vs completeness dilemma above.


>And there's a small amount of additional processing to discard the 
>FBP at legacy endpoints, but silently discarding one packet per 
>connection doesn't seem like a huge effort to me.

Despite this being an exceptional drop (see earlier), this is still a 
reasonably fair statement.


>Note - there's no special RST processing, based on Ted's observation 
>about "data without ACK" being considered "data sent in the 
>unsynchronzed state" -- something legacy TCP explicitly silently discards.

...at least in theory.


Cheers


Bob




________________________________________________________________
Bob Briscoe,                                                  BT 


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Subject: Re: [tcpm] SYN extension using ACK=0 data packets
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Hi, Bob,

On May 31, 2014, at 2:13 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:

> Joe,
>=20
> Hope it's ok to have changed the subject line, with tcpm still in cc.
>=20
> I'm afraid I'm not as excited about ACK=3D0 as you are. It's certainly =
cleaner than anything we've come up with so far.
>=20
> However, I see the goal as finding a way to send a supplement to a SYN =
that is invalid in some way to legacy TCP servers, but likely to appear =
valid to many/most middleboxes. I suspect many middleboxes and firewalls =
will discard the ACK=3D0 segment.
>=20
> When assessing the DSO scheme, you were adamant that anything invalid =
to an endpoint would always eventually become invalid to middleboxes. In =
the excitement of finding a nice clean way of doing ASO, that critique =
seems to have suddenly become unimportant.

I worry about checksums in particular - which get recalculated, so using =
a different checksum means we're sure to fail through a middlebox that =
doesn't recalculate it properly (if it's stored in a new place), or that =
won't validate it (if it's in the current checksum field).

I can't say that middleboxes don't check for ACK=3D0 packets, but it =
seems a lot of work to do unless they're perceived to be an issue. I =
doubt they look at the ACK at all unless the SYN or FIN is set.

> Whatever, there may be a place for both solutions:
> a) ASO (ACK=3D0) for paths where a NAT might be the worst middlebox on =
the path
> b) DSO for paths with stateful firewalls, TCP normalisers etc.

DSO is better through stateful firewalls only when the FBP packet =
precedes the SYN. Keep in mind that the FBP can also be sent multiple =
times after the SYN anyway, with small (1ms) delays; only the first will =
end up being used, and loss of all of them would still be recoverable.

> The cellular world is certainly more like (b).

Cellular uses GGN (carrier-grade NATs), which could make DSO SYN pairing =
more difficult.

> Whether there are many parts of the public Internet left like (a) is =
unclear. (b) seems to have become the norm for most paths.

If we're counting numbers, there are a *lot* more hosts behind NATs. The =
ones behind CGNs probably swamp all others combined.

> More inline...
>=20
> At 01:45 31/05/2014, Joe Touch wrote:
>> Hi, all,
>>=20
>> Some additional information below about "another trick" I proposed, =
inspired by Bob's dual-SYN mechanism, courtesy a few long discussions =
today with Ted Faber.
>>=20
>> I'll be glad to offer this as a potential solution in the doc.
>>=20
>> Joe
>>=20
>> On 5/30/2014 1:34 PM, Joe Touch wrote:
>> ...
>>> Here's another trick that might clean up the above a little:
>>=20
>> FWIW, I had explained it below as being based on sending =
out-of-window data; Ted pointed out that I had been assuming that the =
FBP ACK bit wasn't set - which means the sequence number might be more =
usefully matched to that of the SYN.
>>=20
>> See below...
>>=20
>>>     aso - after SYN option
>>=20
>>                length =3D 2 (just a flag)
>>                length =3D 3 (or 4) indicating the length of the
>>                        FBP expected
>>=20
>>>     FBP - front bumper packet (best I could do on names today)
>>>         a packet
>>                ISN =3D same as the associated SYN
>>                ACK =3D cleared (i.e., a data packet NOT part of
>>                synchronized connection - see why that's useful below)
>=20
> For the avoidance of doubt,
> I assume these two packets have the same src port.

Same IP addresses, same ports, *because* they're associated with the =
same connection.

> and I assume the FBP has SYN=3D0.

Yes. All control bits are 0, including ACK.

> Coincidentally, a couple of weeks ago, when I was looking for places =
to find more bits in the TCP header, I noted that during an established =
connection ACK=3D0 is never used, so I looked into setting ACK=3D0 and =
overloading the ack_number field.
>=20
>=20
>>>     new endpoint sends:
>>>=20
>>>         SYN + aso + fix_opt
>>>         FBP + aso + extra_opt
>>=20
>>                extra_opt in the data field
>>                total length < min MTU (576 for IPv4, 1280 for IPv6)
>>                again ACK bit is zero
>>=20
>>>             legacy endpoint sends back one connections:
>>>                 SYN-ACK + fix_opt
>>>=20
>>>                 if seg arrives before SYN,
>>=20
>>                        it is silently dropped, because
>>                        the ACK bit is clear (this is
>>                        explicit in RFC793)
>>=20
>>>                 if seg arrives after SYN,
>>=20
>>                        it is silently dropped, because
>>                        the ACK bit is clear (this is also
>>                        explicit in RFC793)
>=20
> Two important questions:
> 1) are most/all legacy TCP implementations faithful to the spec?

That's something I'll take a look at.=20

> 2) if a TCP endpoint is meant to drop SYN=3D0, ACK=3D0, then many =
middleboxes surely will.

Middleboxes drop things whose checksum fails, or when a packet comes in =
for a connection that hasn't been established. They don't go out of =
their way to do a lot of other work AFAICT.

> Both these will need experimental testing.

Certainly.

> Nit: I wouldn't exactly say RFC793 is clear. You have to follow 2 =
pages of quite imprecise descriptive logic, starting from p66. But, yes =
it is eventually fairly unambiguous. Paraphrasing, it says:
>=20
> SEGMENT ARRIVES
>        ...
>        if state =3D SYN-SENT
>                1. Check ACK bit
>                        if ACK =3D 1
>                                do stuff
>                2. Check RST bit
>                        if RST =3D 1
>                                if ACK OK enter CLOSED state
>                                elif no ACK, drop
>                3. Check security & precedence
>                        Do stuff
>                4. Check SYN bit
>                        Should only get here if ACK OK, or no ACK and =
no RST
>                        if SYN =3D 1
>                                do stuff
>                5. if SYN =3D 0 and RST =3D 0
>                        drop

In CLOSED, it says to send a RST (like any other packet for a =
non-connection).

In LISTEN, it says:

        Any other control or text-bearing segment (not containing SYN)
        must have an ACK and thus would be discarded by the ACK
        processing.  An incoming RST segment could not be valid, since
        it could not have been sent in response to anything sent by this
        incarnation of the connection.  So you are unlikely to get here,
(where 'here' is a segment with ACK=3D0 and no other control bits set)
        but if you do, drop the segment, and return.

In SYN-SENT, it says:

      fifth, if neither of the SYN or RST bits is set then drop the
      segment and return.

In all other states, it says:

      if the ACK bit is off drop the segment and return

If that's not direct and explicit, I don't know what is.

>>>             ----
>>>=20
>>>             new endpoint sends back one connection:
>>>=20
>>>                 SYN-ACK + options + ....
>>>=20
>>>             a) if FBP arrives before SYN,
>>=20
>>                        it can be silently dropped, but
>>                        it's probably useful for new endpoints
>>                        to hold onto these (without action)
>>                        for a while; they can be silently
>>                        discarded if there are too many
>>                        (which will just result in a
>>                        retransmission and an extra RTT)
>>=20
>>>             b) if FPB arrives with the SYN, they can be
>>>             processed together
>=20
> By 'arrives with' I assume you mean 'within some time duration'.

Yes - I was thinking of the case where the FBP is either cached (as per =
(a)), or is in the segment queue at the time the SYN is being processed.

>>>                 the SYN-ACK can include responses to
>>>                 the extra_opts in addition to the
>>>                 fix_opts, and says "FBP received"
>>>=20
>>>=20
>>>             c) if FPB arrives after the SYN:
>>>=20
>>>                 SYN-ACK proceeds, but sends
>>>                 back "wait for option response".
>>>=20
>>>                 at this point, the source re-sends FBP
>>>                 until an ACK is received that indicates
>>>                 "FBP received", or times-out as with
>>>                 any connection that doesn't finish TWHS
>=20
> There's a dilemma whether the server:
> * prioritises latency and goes ahead without the extra options,
> * or prioritise completeness and blocks until the extra options =
arrive.

There's no dilemma - if the SYN says "FBP is coming", then it means that =
an upgraded server MUST wait for the FBP.

I.e., the FBP is paired with the SYN-ACK, and the handshake doesn't =
complete at the client - the client should NOT send the final ACK of the =
TWHS - until the SYN-ACK is received *and* confirmation that the FBP has =
been received. That can happen together inside the SYN-ACK if known, or =
the SYN-ACK would say "FBP missing" and the client would retransmit the =
FBP *until* the server confirms it (it would send an option =
confirmation, not an ACK).

> I would take the view that if the FBP is late there's a chance it got =
snarled up in a middlebox, and it will never get through no matter how =
many times it's retransmitted.

The client can timeout, and can retry without the extended SYN space in =
that case.

> Rather than make the choice between latency and completeness at =
protocol design time, we need a flag in either scheme (on the DSO or ASO =
option) for the client to say which it wants. Then:
> * the client can choose latency if it knows the extra_opts are not =
critical.
> * otherwise it can choose completeness, and if it doesn't work after a =
retransmit or two, the client won't block for ever; it can work out the =
compromise set of options that fit in a single SYN but still 'work'.

A flag that says "do what a legacy server would do if you have to wait X =
ms" seems fine to me too.

>>>             I'm still thinking as to whether the ACK number
>>>             might indicate whether FBP has been received,
>>=20
>> There are a few ways to handle this, but IMO the best is to have:
>>=20
>>                SYN-ACK aso with length=3D2 means "waiting for FBP"
>>=20
>>                SYN-ACK aso with length=3D3 (or 4) could mean
>>                "got the FBP", or might even indicate how
>>                many bytes the FBP extension contained
>>=20
>> Note - the SYN-ACK and all subsequent segments can assume that aso =3D=3D=
 edo, i.e., they can use the same EDO as spec'd in version 01 of this =
draft.
>>=20
>>> This is cleaner as follows:
>>>=20
>>>     - no need for conn_id coordination
>>>=20
>>>     - no need for conn_id to consume option space for fall-back
>>>=20
>>>     - avoids double-load for legacy servers
>=20
> With a typical legacy server that reflects SYN cookies, less activity =
is doubled:
> * my scheme: sends out two SYN cookies
> * your scheme: sends out one SYN cookie and drops one packet, probably =
after a long chain of logic, because the FBP is an unusual packet.

The activity involved in parsing a packet is small compared to that of =
setting up a SYN cookie or creating TCB state (where SYN cookies aren't =
used).

>>>     - no problem with fate-sharing
>=20
> To be as consistently pessimistic as you were with my scheme, I think =
you mean that you have created machinery to synthesise fate sharing =
between a connection and an invalid segment, and it may not work in =
cases that you may not have thought of.

Your scheme uses two different SYNs on different ports - that's a recipe =
for fates NOT being shared.

Mine uses two segments on the same addresses with the same ports - if =
they're not fate-shared, then neither will the rest of the TCP =
connection be, and TCP won't work (if that fate matters, e.g., through =
different NATs).

>>>     - traverses a NAT just fine
>=20
> As above, surely TCP normalising middleboxes and incoming stateful =
firewalls at the server end are likely to discard the FBP.

I don't think so, but I agree we'll have to try. This is a completely =
different situation, IMO, than expecting boxes that *already* are known =
to validate TCP checksums to do otherwise. In this case, we don't *know* =
the solution won't work from the start.

>>> Upgraded servers still need to wait for the 'seg', but they could =
get
>>> that retransmitted if necessary.
>=20
> See discussion of latency vs completeness dilemma above.

Latency vs. completeness is true for all variants that use multiple =
packets - that's one reason I don't like that aspect of any of these =
approaches.

>> And there's a small amount of additional processing to discard the =
FBP at legacy endpoints, but silently discarding one packet per =
connection doesn't seem like a huge effort to me.
>=20
> Despite this being an exceptional drop (see earlier), this is still a =
reasonably fair statement.

And I'm glad to concede that we don't know the cost on the code cache, =
etc. of exercising a dusty, dark path of processing.=20

>> Note - there's no special RST processing, based on Ted's observation =
about "data without ACK" being considered "data sent in the =
unsynchronzed state" -- something legacy TCP explicitly silently =
discards.
>=20
> ...at least in theory.

And in requirements. Yes, I'll see if I can find out what BSD and Linux =
do - though I have more hope of BSD being correct than Linux.

Joe=


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To: Joe Touch <touch@isi.edu>
From: Bob Briscoe <bob.briscoe@bt.com>
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Subject: Re: [tcpm] More TCP option space on SYNs
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Joe,

At 20:20 31/05/2014, Joe Touch wrote:
>Hi, Bob,
>
>On 5/31/2014 11:19 AM, Bob Briscoe wrote:
>>Joe,
>>
>>Thx for consolidating this thread. I've given it a new subject line.
>>
>>1) You've silently made an important alteration to the proposed
>>protocol. You've put the extra-options directly in the TCP option space
>>of the C-SYN, not within the payload.
>
>I hadn't assumed that; I was describing only which SYN carried which 
>options, not where they were. As you do, I was assuming the C-SYN 
>options were in the data space.

OK.


>To be clear, in my variant - using data outside the sync'd 
>connection (with ACK clear), the option space is similarly in the 
>data portion of the 'extra' segment.

Yup, that was understood already.


>>Yes, your altered proposal is cleaner. However, don't imagine I didn't
>>think of this. I did and I deliberately didn't do it this way.We have a
>>choice:
>>          clean and vulnerable vs. messy but robust.
>>
>>I'm not wedded to using port 80 and http headers, but this is perhaps
>>the most pragmatic approach.
>
>That's fraught with a whole slew of other problems, notably the 
>middleboxes that currently try to translate, proxy, or otherwise 
>modify anything they see as port 80.
>
>Further, that would limit the number of pending connections.
>
>>It will be really unorthodox to define such
>>a protocol I know. We would have to say something like
>>
>>          "The dst port of the C-SYN MUST be 80, and the payload MUST start
>>          with the constant magic_token, where
>>          magic_token = 'PUT / HTTP/1.1<CRLF>Connection : DSO<CRLF><CRLF>'
>>          "
>>
>>I'm sorry if even thinking about this makes you feel dirty :|
>>
>>Other suggestions for inner protocols are welcome, including tunnelled
>>protocols, as long as middleboxes widely forward them, given their dst
>>port.
>
>I don't see what this gets us other than potential interactions. 
>There's no reason to use HTTP to format TCP options, and it would 
>necessitate new encodings and new processing for existing options 
>that might end up in the extended space.

To be clear, I would put the general requirement as :

"       Pick a port number, then make sure the app-layer headers at 
the start of the TCP payload appear valid for that port number, so 
that middleboxes will forward it, then include the extra TCP options 
as the body of this faked up app-layer protocol.
"

Then the dilemma becomes: pick a number, any number as long as it's 
80. The alternative is often to be blocked. Particularly if you 
connect from inside a company that has bought one of those myopic Web 
gateways, or you connect to a cellular network, or... you just 
connect to the Internet via a box from a vendor beginning with a 
letter in the alphabet,... or a digit.


>>2) The main problem with your notation is it doesn't say /where/ the
>>info is placed.
>
>EDO says the options are at the head of the data segment, just with 
>a longer pointer - the same way as current options.
>
>For extending the SYN using a separate segment - whether a C-SYN or 
>unsync'd data segment - the data similarly comes at the head of the 
>data, just that there's no user data permitted after it.

I understood all this. I just meant that the notation you introduced 
in your email didn't have the facility to specify at which layer 
fields were placed.


>>I've added notation as follows:
>>TCP(base header [TCP options [APP(header[payload])]])
>>
>>And for the record I've made the if-else logic clearer.
>>
>>Where I've made more than clarifying edits inline, I've described them
>>and tagged them with [BB].
>>
>>At 21:34 30/05/2014, Joe Touch wrote:
>>>Hi, Bob,
>>>
>>>Let's get back to the core, in a simpler fashion, so other can follow it.
>>>
>>>I stand by my "there's no way to extend the space in the initial SYN",
>>>but you've convinced me there *might* be a way to provide extended
>>>space that can occur during the first phase of the TWHS. I think the
>>>dual-SYN approach still isn't viable, but I've outlined an alternative
>>>below that's similar but doesn't have the same baggage, IMO.
>>>
>>>Again, I'm still concerned by what midboxes might do to this...
>>>
>>>What do others think??
>>>
>>>Joe
>>>
>>>For quick review, here's what I understand:
>>>
>>>                 dso = dual-syn option
>>>                         dso-D = data
>>>                         dso-C = control
>>>                 conn_id = identifier to link the two SYNs together
>>>                 extra_opt = options that didn't fit in legacy SYN
>>>                 fit_opt = options that do fit in the legacy SYN
>>          new client endpoint sends
>>                  TCP(port A SYN [dso-D(conn_id) + fit_opt] )
>>                  TCP(port B SYN [dso-C [APP(headers [conn_id +
>>extra_opt] ) ] ] )
>
>IMO, the conn_ID in the C-SYN should similarly be in the regular 
>option space, but I don't think it matters.
>
>>[BB]: i/APP(headers...)/
>>
>>                          if (legacy server endpoint) { sends back two
>>connections:
>>                                  TCP(port A SYN-ACK [fit_opt] )
>>                                  TCP(port B SYN-ACK [??] )
>>>                                 (it's interpretation of extra_opt)
>
>I'm assuming EDO after the SYN, which means the SYN-ACK has as much 
>space as it wants for options, as would every other segment.
>
>>                                          new client endpoint responds:
>>                                          TCP(port A ACK) (established)
>>                                          TCP(port B RST)
>>
>>>                         Notes about legacy servers:
>>>                                 - they do twice the work on SYNs
>>>                                 - they might keep twice the state
>>>                                 (if not using cookies)
>>>                                 - they might clean state if the RST
>>>                                 is received, but that state might
>>>                                 persist indefinitely (until the next
>>>                                 connection, depending on timeouts, etc.)
>>>
>>>                         -----
>>                          } elif (new server endpoint) { sends back one
>>connection:
>>                                  TCP(port A SYN-ACK [edo + fit_opt +
>>extra_opt] )
>>
>>[BB]: s/dso-d/edo/
>
>Sure. At this point we're basically in EDO-land.
>
>>                                          new client endpoint responds:
>>                                          TCP(port A ACK) (established)
>>
>>>
>>>                         Notes:
>>>                                 - can stall when dso-D SYN arrives
>>>                                 before dso-C SYN, up to some limit
>>>                                 - twice the work on SYNs (or more)
>>                          }
>>
>>>Here's what I was assuming, though admittedly it's not documented (yet):
>>>
>>>         - no significant impact on TCP connection rate for
>>>         legacy servers
>>>
>>>         - no significant impact on TCP connection rate for
>>>         legacy clients
>>>
>>>         - impact dominated by processing the extended option space
>>>         for extended clients
>>>
>>>         - impact dominated by processing the extended option space
>>>         for extended servers
>>>
>>>         - compatible with typical TCP processing optimizations,
>>>         notably SYN cookies
>>>                 you did provide a potential way forward for these
>>>
>>>         - capable of successfully traversing typical NATs
>>>
>>>Your approach has the following properties:
>>
>>The 3 bullets below are not useful ways to describe performance impact.
>>They selectively describe whichever gives the most pessimistic picture
>>out of:
>>a) either the instantaneous performance change at the moment of connection
>>b) or the worst-case long-run performance impact
>>
>>They don't describe the average long-run performance impact, which is
>>important for sizing machines.
>>
>>Worse, the instantaneous performance impact is only significant when a
>>machine's SYN processing time is large relative to the e2e delay, which
>>would be a highly unusual scenario on public networks (even in scenarios
>>such as intra-data-centre, it's hard to reduce e2e delay to approach SYN
>>processing time, but you could for intra-machine connections).
>
>SYN processing is a known load - and potential overload - on 
>machines. That's why it's reasonable to focus on it.

But we should primarily think in terms of SYN cookie processing for 
legacy hosts, and the equivalent for upgraded hosts. We should design 
thia as the norm, not the exception. Possibly the /only/ way to 
implement this new form of SYN processing.

Then, for the dual-SYN scheme, the upgraded host doesn't have a 
normal SYN cookie processing load with a C-SYN, so we can no longer 
rely on known-load formulae.


>>>         - halves the server connection rate for updated servers
>>>         from legacy clients when this option is in use
>>
>>Eh? The long-run server connection rate will be fractionally decreased
>>due to updated clients using extra options (which is your third case
>>below), but the instantaneous server connection rate seen by a legacy
>>client is unchanged, because it only sends one SYN.
>
>Agreed. I was doing too many of these at once...
>
>>>         - lowers (to some extent, if not halves) the client
>>>         connection rate of updated clients to all servers
>>>         when this option is in use
>>>
>>>         - halves (roughly) the server rate for all servers
>>>         when this option is in use
>>
>>Nope. All long-run server rates are reduced by 1/(1+e), where e is the
>>fraction of connections using extra options.
>
>the rate for connections when this option is in use is halved - I 
>didn't see that the sentence above could be parsed the other way. 
>the overall rate depends on the fraction of connections using the option.
>
>>>It also:
>>>
>>>         - doubles the number of SYNs in the network
>>
>>Nope. The number of SYNs in the network is inflated by e where e is the
>>fraction of connections using extra options.
>
>You keep bringing this up - I don't like a solution that should be 
>used only a little. If it works, we should assume it works 
>everywhere all the time.

It doesn't make sense to predict that all connections will always use 
MSS, timestamp, SACK, window-scale, MPTCP, TCP-AO, new option X, new 
option y, etc etc.

For a start, less than 1% of connections contain enough packets to 
make MPTCP worth starting.

The word option means optional.

>>>         - susceptible to lack of fate-sharing problems, e.g.,
>>>         if the two SYNs experience different firewall configurations
>>
>>Nope. It's fairer to say it's potentially susceptible to second-order
>>fate-sharing problems like your firewall example (the first-order fate
>>sharing problems have been addressed).
>
>I have no idea why you think firewalls are a second-order 
>fate-sharing, or what that even means. Fate sharing means fate is 
>shared. When fate is not shared by the SYNs, this method has 
>problems. There's no mechanism to recover the missing D-SYN or C-SYN.

What I mean by first-order fate-sharing problems is those obvious 
cases (reordering, drop) that we have addressed by resynthesising 
fate sharing.

There is certainly a possibility that two SYNs between the same 
endpoints will pass through different firewalls, but it is small in 
the Internet as a whole. There is a further possibility that these 
two firewalls will be configured differently, but it is small. There 
is a further possibility that the differences in configuration will 
be significant enough to distinguish the behaviour wrt the two SYNs, 
but it is small. When three small probabilities are multiplied 
together, you get a really small probability. This is what I call a 
second-order fate-sharing problem.

You must surely see that it is inconsistent to say:
* "this method has problems" about the DSO idea, based on such a corner case
* "no problem with fate-sharing" about the ASO idea, when all you 
mean is that the FBP will follow the same path as the SYN, and they 
will therefore likely pass through the same devices.

In the ASO proposal, splitting the options on a SYN over two packets, 
one of which is considered invalid by TCP, will cause the FBP to be 
discarded by numerous TCP normalisers and stateful firewalls. That is 
a fate sharing problem of the first order. Likely to be on a much 
larger scale than the 'second-order' firewall problem above.


>>>         - reduces the space available for fit_opt due to the need
>>>         for the conn_id even in the fall-back D-SYN, which means
>>>         less option space in the SYNs for fall-back connections
>>
>>Yup.
>>
>>
>>>         the conn_id which may need to be very large because it
>>>         needs to be unique per source port and source IP address
>>>         because that information is lost during NAT translation
>>
>>Given many NATs will typically make the src IPs of both SYNs the same, I
>>suggest a larger conn_id should be a fall-back option for the client,
>>not a default.
>
>Multiple NATs are quite common - carrier grade NATs in the net, and 
>user NATs at the home WIFI point.
>
>Any multiple-NAT, or any path through even a single carrier-grade 
>NAT will end up potentially un-doing or over-doing source IP 
>overlap, creating false positives and false negatives that break 
>this mechanism.
>
>That's really the crux - fate sharing and the need to bind the two 
>SYNs - that make this untenable, IMO.
>
>>Even if the src IPs of both SYNs are different once they reach the
>>server, the high end bits will invariably be the same.
>
>You can't know this - the two SYNs could have gone through different 
>numbers of NATs or different NATs entirely, so the bits could change 
>arbitrarily.

That's what 'invariably' means. I'm not claiming this works 100%. I 
am saying I expect it to work sufficiently to be worth trying it out 
to see if I'm right. If it doesn't work, on some paths, the fall-back 
has been described - the client has to rely on the TCP options it can 
fit in a single SYN.


>>So the max size
>>of the contents of the DSO TCP option can be 6B, and the server can take
>>the rest of conn_id from the higher bits of the src IP addr of each SYN.
>>This is a variant of the idea in <draft-wing-nat-reveal-option>.
>
>FWIW, you're now in the land where we would have to start explaining 
>the privacy and tracking implications of those bits. I'd really like 
>to avoid that.

Nope. I think you're reading this the wrong way round. I'm not 
revealing private IDs in TCP option fields. I'm reducing the size of 
the (random) conn_id in cases where the /public/ src port or the 
upper bits of the /public/ src IP have sufficient entropy to be used 
to construct the rest of the conn_id.

I was acknowledging Dan Wing 'cos he gave us the idea. But it's a 
/variant/ that doesn't reveal anything private.

I'm going offline, possibly until Tue now.

Regards


Bob


>>In fact, the server doesn't even need a small conn_id for clients that
>>know they are not behind a NAT and that want more option space in the
>>D-SYN - then the server could use the src port & src IP for the conn_id.
>>
>>To summarise, these options could be distinguished by the length field
>>of the dual-syn option.
>>Length = 2B                             => conn_id = src netaddr + src port
>>Length = 6B = 2B +4B conn_id_short      => conn_id = src netaddr +
>>conn_id_short
>>Length = 8B = 2B +6B conn_id_long       => conn_id = hsb(src netaddr) +
>>conn_id_long
>>
>>Given there have been numerous other attempts to reveal a connection ID
>>that is preserved through middleboxes [RFC6967], rather than defining a
>>dual-syn option that carries a conn_id, we might want to design a TCP
>>connection ID option with a flag to say whether it is also part of a
>>dual SYN pair or not.
>>
>>Where
>>* hsb(src netaddr) is the netaddr with the lowest 16 bits truncated
>>* src netaddr is the network address (IPv4, IPv6, or any other network
>>protocol)
>>
>>To reduce latency, a host could use the default short_conn_id for all
>>connections at first, then:
>>- if it finds that DSO persistently doesn't work it falls back to the
>>long_conn_id for all connections
>>- it occasionally tests the short and zero options to see if it can use
>>shorter DSO options.
>>
>>
>>>         - requires the ISNs to be related (see RFC6528 - if there's
>>>         a rule to generate it, there will be code to validate that
>>>         rule, and eventually a BCP to encourage that validation -
>>>         typically from the same RFC author)
>>
>>Eh? The ISNs can and should be independent. To be robust against
>>middleboxes that rewrite sequence numbers, we must not required ISNs to
>>be related.
>>
>>
>>>I agree that you have proposed potentially viable ways to deal with
>>>the SYN cookie, and that RST state is not an issue.
>>
>>A feature that I think it's fair to add:
>>          - Good chance of passing through app-layer middleboxes that
>>forward
>>          unrecognised TCP options unchanged, but not those that discard
>>them.
>>
>>
>>>However, there are too many problems with this, IMO, to call it viable.
>>
>>Once your over-pessimistic analyses of the performance impact are
>>corrected, and my ideas to reduce the size of the conn_id are taken into
>>account, it's a different story.
>>
>>But it's up to the WG to decide whether this is worth taking further.
>>Not just you or I.
>
>Agreed.
>
>>>Here's another trick that might clean up the above a little:
>>
>><snip - I'll respond separately to your later updates on this ASO idea,
>>with ACK=0>
>>
>>Cheers
>>
>>
>>Bob
>>
>>
>>________________________________________________________________
>>Bob Briscoe,                                                  BT
>
>________________________________________________________________
>Bob Briscoe,                                                  BT 


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Date: Sat, 31 May 2014 23:45:34 +0100
To: Joe Touch <touch@isi.edu>
From: Bob Briscoe <bob.briscoe@bt.com>
In-Reply-To: <6C4E6E63-F3F4-4364-9459-794957DC8799@isi.edu>
References: <20140425221257.12559.43206.idtracker@ietfa.amsl.com> <2586_1398464386_535ADF82_2586_915_1_535ADF56.9050106@isi.edu> <CF8D8E25-E435-4199-8FD6-3F7066447292@iki.fi> <5363AF84.8090701@mti-systems.com> <5363B397.8090009@isi.edu> <CAO249yeyr5q21-=e6p5azwULOh1_jUsniZ6YPcDYd69av8MMYw@mail.gmail.com> <DCC98F94-EA74-4AAA-94AE-E399A405AF13@isi.edu> <655C07320163294895BBADA28372AF5D2CFE36@FR712WXCHMBA15.zeu.alcatel-lucent.com> <20140503122950.GM44329@verdi> <655C07320163294895BBADA28372AF5D2D009E@FR712WXCHMBA15.zeu.alcatel-lucent.com> <201405221710.s4MHAY4S002037@bagheera.jungle.bt.co.uk> <537E3ACD.5000308@isi.edu> <1AD79820-22C1-4500-84D1-1383F264D68C@weston.borman.com> <201405231213.s4NCDa5P005525@bagheera.jungle.bt.co.uk> <537F8202.4020907@isi.edu> <201405281715.s4SHFMm0014634@bagheera.jungle.bt.co.uk> <538623B9.2060209@isi.edu> <201405301642.s4UGgcvY030471@bagheera.jungle.bt.co.uk> <5388EB6F.4010405@isi.edu> <5389263C.8010202@isi.edu> <201405312113.s4VLDEbG004301@baghee! ra.jungle.bt.co.uk> <6C4E6E63-F3F4-4364-9459-794957DC8799@isi.edu>
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Subject: Re: [tcpm] SYN extension using ACK=0 data packets
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Joe,

Nothing further to add on this thread.
I do think you're seeing the ASO scheme thru rose-coloured spectacles.

Cheers


Bob

At 23:01 31/05/2014, Joe Touch wrote:
>Hi, Bob,
>
>On May 31, 2014, at 2:13 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:
>
> > Joe,
> >
> > Hope it's ok to have changed the subject line, with tcpm still in cc.
> >
> > I'm afraid I'm not as excited about ACK=0 as you are. It's 
> certainly cleaner than anything we've come up with so far.
> >
> > However, I see the goal as finding a way to send a supplement to 
> a SYN that is invalid in some way to legacy TCP servers, but likely 
> to appear valid to many/most middleboxes. I suspect many 
> middleboxes and firewalls will discard the ACK=0 segment.
> >
> > When assessing the DSO scheme, you were adamant that anything 
> invalid to an endpoint would always eventually become invalid to 
> middleboxes. In the excitement of finding a nice clean way of doing 
> ASO, that critique seems to have suddenly become unimportant.
>
>I worry about checksums in particular - which get recalculated, so 
>using a different checksum means we're sure to fail through a 
>middlebox that doesn't recalculate it properly (if it's stored in a 
>new place), or that won't validate it (if it's in the current checksum field).
>
>I can't say that middleboxes don't check for ACK=0 packets, but it 
>seems a lot of work to do unless they're perceived to be an issue. I 
>doubt they look at the ACK at all unless the SYN or FIN is set.
>
> > Whatever, there may be a place for both solutions:
> > a) ASO (ACK=0) for paths where a NAT might be the worst middlebox 
> on the path
> > b) DSO for paths with stateful firewalls, TCP normalisers etc.
>
>DSO is better through stateful firewalls only when the FBP packet 
>precedes the SYN. Keep in mind that the FBP can also be sent 
>multiple times after the SYN anyway, with small (1ms) delays; only 
>the first will end up being used, and loss of all of them would 
>still be recoverable.
>
> > The cellular world is certainly more like (b).
>
>Cellular uses GGN (carrier-grade NATs), which could make DSO SYN 
>pairing more difficult.
>
> > Whether there are many parts of the public Internet left like (a) 
> is unclear. (b) seems to have become the norm for most paths.
>
>If we're counting numbers, there are a *lot* more hosts behind NATs. 
>The ones behind CGNs probably swamp all others combined.
>
> > More inline...
> >
> > At 01:45 31/05/2014, Joe Touch wrote:
> >> Hi, all,
> >>
> >> Some additional information below about "another trick" I 
> proposed, inspired by Bob's dual-SYN mechanism, courtesy a few long 
> discussions today with Ted Faber.
> >>
> >> I'll be glad to offer this as a potential solution in the doc.
> >>
> >> Joe
> >>
> >> On 5/30/2014 1:34 PM, Joe Touch wrote:
> >> ...
> >>> Here's another trick that might clean up the above a little:
> >>
> >> FWIW, I had explained it below as being based on sending 
> out-of-window data; Ted pointed out that I had been assuming that 
> the FBP ACK bit wasn't set - which means the sequence number might 
> be more usefully matched to that of the SYN.
> >>
> >> See below...
> >>
> >>>     aso - after SYN option
> >>
> >>                length = 2 (just a flag)
> >>                length = 3 (or 4) indicating the length of the
> >>                        FBP expected
> >>
> >>>     FBP - front bumper packet (best I could do on names today)
> >>>         a packet
> >>                ISN = same as the associated SYN
> >>                ACK = cleared (i.e., a data packet NOT part of
> >>                synchronized connection - see why that's useful below)
> >
> > For the avoidance of doubt,
> > I assume these two packets have the same src port.
>
>Same IP addresses, same ports, *because* they're associated with the 
>same connection.
>
> > and I assume the FBP has SYN=0.
>
>Yes. All control bits are 0, including ACK.
>
> > Coincidentally, a couple of weeks ago, when I was looking for 
> places to find more bits in the TCP header, I noted that during an 
> established connection ACK=0 is never used, so I looked into 
> setting ACK=0 and overloading the ack_number field.
> >
> >
> >>>     new endpoint sends:
> >>>
> >>>         SYN + aso + fix_opt
> >>>         FBP + aso + extra_opt
> >>
> >>                extra_opt in the data field
> >>                total length < min MTU (576 for IPv4, 1280 for IPv6)
> >>                again ACK bit is zero
> >>
> >>>             legacy endpoint sends back one connections:
> >>>                 SYN-ACK + fix_opt
> >>>
> >>>                 if seg arrives before SYN,
> >>
> >>                        it is silently dropped, because
> >>                        the ACK bit is clear (this is
> >>                        explicit in RFC793)
> >>
> >>>                 if seg arrives after SYN,
> >>
> >>                        it is silently dropped, because
> >>                        the ACK bit is clear (this is also
> >>                        explicit in RFC793)
> >
> > Two important questions:
> > 1) are most/all legacy TCP implementations faithful to the spec?
>
>That's something I'll take a look at.
>
> > 2) if a TCP endpoint is meant to drop SYN=0, ACK=0, then many 
> middleboxes surely will.
>
>Middleboxes drop things whose checksum fails, or when a packet comes 
>in for a connection that hasn't been established. They don't go out 
>of their way to do a lot of other work AFAICT.
>
> > Both these will need experimental testing.
>
>Certainly.
>
> > Nit: I wouldn't exactly say RFC793 is clear. You have to follow 2 
> pages of quite imprecise descriptive logic, starting from p66. But, 
> yes it is eventually fairly unambiguous. Paraphrasing, it says:
> >
> > SEGMENT ARRIVES
> >        ...
> >        if state = SYN-SENT
> >                1. Check ACK bit
> >                        if ACK = 1
> >                                do stuff
> >                2. Check RST bit
> >                        if RST = 1
> >                                if ACK OK enter CLOSED state
> >                                elif no ACK, drop
> >                3. Check security & precedence
> >                        Do stuff
> >                4. Check SYN bit
> >                        Should only get here if ACK OK, or no ACK and no RST
> >                        if SYN = 1
> >                                do stuff
> >                5. if SYN = 0 and RST = 0
> >                        drop
>
>In CLOSED, it says to send a RST (like any other packet for a non-connection).
>
>In LISTEN, it says:
>
>         Any other control or text-bearing segment (not containing SYN)
>         must have an ACK and thus would be discarded by the ACK
>         processing.  An incoming RST segment could not be valid, since
>         it could not have been sent in response to anything sent by this
>         incarnation of the connection.  So you are unlikely to get here,
>(where 'here' is a segment with ACK=0 and no other control bits set)
>         but if you do, drop the segment, and return.
>
>In SYN-SENT, it says:
>
>       fifth, if neither of the SYN or RST bits is set then drop the
>       segment and return.
>
>In all other states, it says:
>
>       if the ACK bit is off drop the segment and return
>
>If that's not direct and explicit, I don't know what is.
>
> >>>             ----
> >>>
> >>>             new endpoint sends back one connection:
> >>>
> >>>                 SYN-ACK + options + ....
> >>>
> >>>             a) if FBP arrives before SYN,
> >>
> >>                        it can be silently dropped, but
> >>                        it's probably useful for new endpoints
> >>                        to hold onto these (without action)
> >>                        for a while; they can be silently
> >>                        discarded if there are too many
> >>                        (which will just result in a
> >>                        retransmission and an extra RTT)
> >>
> >>>             b) if FPB arrives with the SYN, they can be
> >>>             processed together
> >
> > By 'arrives with' I assume you mean 'within some time duration'.
>
>Yes - I was thinking of the case where the FBP is either cached (as 
>per (a)), or is in the segment queue at the time the SYN is being processed.
>
> >>>                 the SYN-ACK can include responses to
> >>>                 the extra_opts in addition to the
> >>>                 fix_opts, and says "FBP received"
> >>>
> >>>
> >>>             c) if FPB arrives after the SYN:
> >>>
> >>>                 SYN-ACK proceeds, but sends
> >>>                 back "wait for option response".
> >>>
> >>>                 at this point, the source re-sends FBP
> >>>                 until an ACK is received that indicates
> >>>                 "FBP received", or times-out as with
> >>>                 any connection that doesn't finish TWHS
> >
> > There's a dilemma whether the server:
> > * prioritises latency and goes ahead without the extra options,
> > * or prioritise completeness and blocks until the extra options arrive.
>
>There's no dilemma - if the SYN says "FBP is coming", then it means 
>that an upgraded server MUST wait for the FBP.
>
>I.e., the FBP is paired with the SYN-ACK, and the handshake doesn't 
>complete at the client - the client should NOT send the final ACK of 
>the TWHS - until the SYN-ACK is received *and* confirmation that the 
>FBP has been received. That can happen together inside the SYN-ACK 
>if known, or the SYN-ACK would say "FBP missing" and the client 
>would retransmit the FBP *until* the server confirms it (it would 
>send an option confirmation, not an ACK).
>
> > I would take the view that if the FBP is late there's a chance it 
> got snarled up in a middlebox, and it will never get through no 
> matter how many times it's retransmitted.
>
>The client can timeout, and can retry without the extended SYN space 
>in that case.
>
> > Rather than make the choice between latency and completeness at 
> protocol design time, we need a flag in either scheme (on the DSO 
> or ASO option) for the client to say which it wants. Then:
> > * the client can choose latency if it knows the extra_opts are 
> not critical.
> > * otherwise it can choose completeness, and if it doesn't work 
> after a retransmit or two, the client won't block for ever; it can 
> work out the compromise set of options that fit in a single SYN but 
> still 'work'.
>
>A flag that says "do what a legacy server would do if you have to 
>wait X ms" seems fine to me too.
>
> >>>             I'm still thinking as to whether the ACK number
> >>>             might indicate whether FBP has been received,
> >>
> >> There are a few ways to handle this, but IMO the best is to have:
> >>
> >>                SYN-ACK aso with length=2 means "waiting for FBP"
> >>
> >>                SYN-ACK aso with length=3 (or 4) could mean
> >>                "got the FBP", or might even indicate how
> >>                many bytes the FBP extension contained
> >>
> >> Note - the SYN-ACK and all subsequent segments can assume that 
> aso == edo, i.e., they can use the same EDO as spec'd in version 01 
> of this draft.
> >>
> >>> This is cleaner as follows:
> >>>
> >>>     - no need for conn_id coordination
> >>>
> >>>     - no need for conn_id to consume option space for fall-back
> >>>
> >>>     - avoids double-load for legacy servers
> >
> > With a typical legacy server that reflects SYN cookies, less 
> activity is doubled:
> > * my scheme: sends out two SYN cookies
> > * your scheme: sends out one SYN cookie and drops one packet, 
> probably after a long chain of logic, because the FBP is an unusual packet.
>
>The activity involved in parsing a packet is small compared to that 
>of setting up a SYN cookie or creating TCB state (where SYN cookies 
>aren't used).
>
> >>>     - no problem with fate-sharing
> >
> > To be as consistently pessimistic as you were with my scheme, I 
> think you mean that you have created machinery to synthesise fate 
> sharing between a connection and an invalid segment, and it may not 
> work in cases that you may not have thought of.
>
>Your scheme uses two different SYNs on different ports - that's a 
>recipe for fates NOT being shared.
>
>Mine uses two segments on the same addresses with the same ports - 
>if they're not fate-shared, then neither will the rest of the TCP 
>connection be, and TCP won't work (if that fate matters, e.g., 
>through different NATs).
>
> >>>     - traverses a NAT just fine
> >
> > As above, surely TCP normalising middleboxes and incoming 
> stateful firewalls at the server end are likely to discard the FBP.
>
>I don't think so, but I agree we'll have to try. This is a 
>completely different situation, IMO, than expecting boxes that 
>*already* are known to validate TCP checksums to do otherwise. In 
>this case, we don't *know* the solution won't work from the start.
>
> >>> Upgraded servers still need to wait for the 'seg', but they could get
> >>> that retransmitted if necessary.
> >
> > See discussion of latency vs completeness dilemma above.
>
>Latency vs. completeness is true for all variants that use multiple 
>packets - that's one reason I don't like that aspect of any of these 
>approaches.
>
> >> And there's a small amount of additional processing to discard 
> the FBP at legacy endpoints, but silently discarding one packet per 
> connection doesn't seem like a huge effort to me.
> >
> > Despite this being an exceptional drop (see earlier), this is 
> still a reasonably fair statement.
>
>And I'm glad to concede that we don't know the cost on the code 
>cache, etc. of exercising a dusty, dark path of processing.
>
> >> Note - there's no special RST processing, based on Ted's 
> observation about "data without ACK" being considered "data sent in 
> the unsynchronzed state" -- something legacy TCP explicitly silently discards.
> >
> > ...at least in theory.
>
>And in requirements. Yes, I'll see if I can find out what BSD and 
>Linux do - though I have more hope of BSD being correct than Linux.
>
>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 

