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Date: Wed, 1 Aug 2012 15:23:26 -0400
From: Anthony Sabatini <tsabatini@hotmail.com>
To: "Scheffenegger, Richard" <rs@netapp.com>, tcpm@ietf.org
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Cc: "Matthew Mathis \(mattmathis@google.com\)" <mattmathis@google.com>
Subject: [tcpm] Settling Time and RFC3517bis
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From touch@isi.edu  Wed Aug  1 14:14:34 2012
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Cc: tcpm@ietf.org, "Matthew Mathis \(mattmathis@google.com\)" <mattmathis@google.com>
Subject: Re: [tcpm] Settling Time and RFC3517bis
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On 8/1/2012 12:23 PM, Anthony Sabatini wrote:
> Settling Time is the concept of how long we wait for an out of order packet.

That's already called reordering time. Settling time is a term from 
analog electronics/optics, and I don't see a good reason to import it here.

> It has been an unnamed mainstay of IPv4 from the beginning,
> masquerading under the name "Fragmentation Reassembly Timer" but at the
> TCP level its physical existence has gone unacknowledged.

There has been no masquerading, but there appears to be confusion ;-)

Fragmentation reassembly is based on the time between the arrival of the 
first fragment of a set (FWIW, I mean "first" as "first in time", not 
"the packet with an offset of 0") and the last (again, in time, not the 
fragment where MF=0), and is not affected at all by the order of the 
arrival of the fragments.

In TCP, the reordering is of segments, and is tolerated by the receive 
window.

> Settling Time is rather easy to calculate as it has a ceiling of
> .25*RTT (any number over that and you are in retransmission
> territory)

TCP should never do anything RTT-based in less than a full RTT. It has 
no business retransmitting in less than one RTT.

> and a floor of Fragmentation Reassembly Time * 1.25 (need
> to leave enough time to process reassembled packet).

The additional time is whatever time is needed to:
	- complete reassembly
	- get the resulting IP datagram to TCP for processing

This should be on the order of microseconds, vs. milliseconds for most 
RTTs; 25% of the reassembly timer is thousands of time larger.

However, again, the issue for TCP is the segment reordering time, which 
does not necessarily have anything to do with the reassembly time. E.g., 
even atomic (non-fragmented) segments can arrive out of order.

> Obviously the
> lower number is better and there are probably thousands of emails on
> how  to set the Fragmentation Reassembly Timer properly.
>
> The point here is we are dancing around issues by saying we wait for
> multiple ACKs before a SACK block is processed, you are badly
> simulating reassembly delay.  Better to act on the first appearance
> of a hole in the sequence but delay the actual call for
> retransmission by Settling Time to see if the world has changed
> before you issue the retransmission.  Also the time represented by
> multiple ACKs is extremely variable,  short segments and random times
> for WOW, long segments and continuous for file transfer.

As the use of multiple concurrent paths becomes more common, 
out-of-order segment receipt becomes the norm. Jumping the gun on 
retransmitting holes could turn a mild reordering situation into a 
congested system quickly.

I see no reason to act on holes on timescales shorter than 1-2 RTTs.

Joe


> Note for the record a DELAY TIMER is not in and of itself an event,
> it is delaying and existing event, therefore its expiration does not
> have any implications of its own, the event already  happened. >
> Tony
>
> Sent from my ASUS Pad
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From pasi.sarolahti@iki.fi  Wed Aug  1 14:21:10 2012
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: [tcpm] Draft TCPM meeting minutes
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Hi,

The draft meeting minutes are available at =
http://www.ietf.org/proceedings/84/minutes/minutes-84-tcpm . Please let =
me know if there are errors.

Many thanks to Andrew, Richard and Matt for taking notes!=20

- Pasi


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Subject: Re: [tcpm] Settling Time and RFC3517bis
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From touch@isi.edu  Thu Aug  2 11:37:52 2012
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Date: Thu, 02 Aug 2012 11:32:55 -0700
From: Joe Touch <touch@isi.edu>
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Cc: tcpm@ietf.org, "Matthew Mathis \(mattmathis@google.com\)" <mattmathis@google.com>
Subject: Re: [tcpm] Settling Time and RFC3517bis
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Hi, Tony,

On 8/2/2012 11:07 AM, Anthony Sabatini wrote:
> Joe -
>
> Thank you for your insightful comments.
>
> 1) I am used to ITU terminology. Settling time in ITU is used in
> optical, electrical and infrastructure standards. It was in that last
> context that I used it. I will change all references to "Reordering Time".

AOK.

> 2) My point about fragmentation is that "Fragmentation Reassembly
> Timer" and "Reordering Time" are different names to describe the same issue.

But they are not. Fragmentation reassembly time is the time of the 
spread-out of a set of fragments of a single IP packet -- it does not 
matter at all whether the fragments are reordered or not. There is 
neither a performance nor a time delay difference when fragments come 
out of order vs. in-order. The only impact is when fragments of 
different packets are *interleaved* - whether they are reordered or not.

Reordering time would affect only things that care about order. TCP 
doesn't care strictly about reordering of the incoming segments, but as 
you note there is a performance impact of things coming out of order at 
the TCP level.

>> 3) STANDARD TCP can not recover in less than one RTT, SACK TCP as
> written certainly can and does currently recover in less that one RTT on
> at least long links. With the changes I have been proposing nearly all
> errors can be recovered in less than one RTT.

Please explain. In less than 1 RTT the source cannot know what packets 
have arrived at the sender, so it cannot know what to retransmit. In 
less than 1 RTT, the sender has no new information, and cannot do much 
of anything.

The *receiver* can react differently, which can impact the sender later, 
but the loop cannot be closed in less than 1 RTT.

> 4) Even terrestrial Google CA to Google NY is 45 ms minimum, add
> switching equipment without MPLS and that goes to 100 ms, with MPLS who
> knows. One RTT is a huge amount of time, on a satellite or ocean link it
> is glacial. TCPM has no business discussing forward error correction
> before exploring the mechanics of making the current protocol more
> efficient, lower latency and more reliable.

TCP does not use FEC, but many links - especially satellite and oceanic 
- do. That's at the link layer. The tradeoff between ARQ-based error 
correction and FEC error correction is a known issue, and TCP is not 
exempt from it.

> 5) The 1.25 * Fragmentation Reassembly Timer was chosen as a minimum
> to include a bit more reordering time. Regardless if a packet has not
> arrived in .25 * RTT can it even remotely be considered as "out of
> order" as opposed to lost?

Reordering typically has no relationship to the RTT, except that 
reordering is always less than the max RTT.

Neither reordering nor the RTT is related at all to the fragmentation 
reassembly timer, unless a stream includes some fragments and some 
non-fragment packets. In that case, you would need to wait at least 1 * 
reassembly time to account for differences between reassembled packets 
and non-reassembled ones.

> 6) Why is the IETF spending all of its time covering for greedy
> carriers who do not install enough fiber or switching equipment? ALL of
> the changes being made in the congestion area result in a WORSE
> experience for the end user, the people we should really care about.

The IETF is focused on making sure things work in the most possible 
cases - especially in the cases of IP and TCP. Optimization is nice, but 
can *always* be sacrificed when it compromises generality.

The purpose of congestion control is to improve everyone's experience as 
a group by making some user's experience worse, by definition. The IETF 
cares about the end users as a group, but not individual greedy end 
users whose behavior impacts others.

Joe


>
> Tony
>
> Sent from my ASUS Pad
>
> Joe Touch <touch@isi.edu> wrote:
>
>>
>>
>> On 8/1/2012 12:23 PM, Anthony Sabatini wrote:
>>> Settling Time is the concept of how long we wait for an out of order packet.
>>
>> That's already called reordering time. Settling time is a term from
>> analog electronics/optics, and I don't see a good reason to import it here.
>>
>>> It has been an unnamed mainstay of IPv4 from the beginning,
>>> masquerading under the name "Fragmentation Reassembly Timer" but at the
>>> TCP level its physical existence has gone unacknowledged.
>>
>> There has been no masquerading, but there appears to be confusion ;-)
>>
>> Fragmentation reassembly is based on the time between the arrival of the
>> first fragment of a set (FWIW, I mean "first" as "first in time", not
>> "the packet with an offset of 0") and the last (again, in time, not the
>> fragment where MF=0), and is not affected at all by the order of the
>> arrival of the fragments.
>>
>> In TCP, the reordering is of segments, and is tolerated by the receive
>> window.
>>
>>> Settling Time is rather easy to calculate as it has a ceiling of
>>> .25*RTT (any number over that and you are in retransmission
>>> territory)
>>
>> TCP should never do anything RTT-based in less than a full RTT. It has
>> no business retransmitting in less than one RTT.
>>
>>> and a floor of Fragmentation Reassembly Time * 1.25 (need
>>> to leave enough time to process reassembled packet).
>>
>> The additional time is whatever time is needed to:
>> 	- complete reassembly
>> 	- get the resulting IP datagram to TCP for processing
>>
>> This should be on the order of microseconds, vs. milliseconds for most
>> RTTs; 25% of the reassembly timer is thousands of time larger.
>>
>> However, again, the issue for TCP is the segment reordering time, which
>> does not necessarily have anything to do with the reassembly time. E.g.,
>> even atomic (non-fragmented) segments can arrive out of order.
>>
>>> Obviously the
>>> lower number is better and there are probably thousands of emails on
>>> how  to set the Fragmentation Reassembly Timer properly.
>>>
>>> The point here is we are dancing around issues by saying we wait for
>>> multiple ACKs before a SACK block is processed, you are badly
>>> simulating reassembly delay.  Better to act on the first appearance
>>> of a hole in the sequence but delay the actual call for
>>> retransmission by Settling Time to see if the world has changed
>>> before you issue the retransmission.  Also the time represented by
>>> multiple ACKs is extremely variable,  short segments and random times
>>> for WOW, long segments and continuous for file transfer.
>>
>> As the use of multiple concurrent paths becomes more common,
>> out-of-order segment receipt becomes the norm. Jumping the gun on
>> retransmitting holes could turn a mild reordering situation into a
>> congested system quickly.
>>
>> I see no reason to act on holes on timescales shorter than 1-2 RTTs.
>>
>> Joe
>>
>>
>>> Note for the record a DELAY TIMER is not in and of itself an event,
>>> it is delaying and existing event, therefore its expiration does not
>>> have any implications of its own, the event already  happened. >
>>> Tony
>>>
>>> Sent from my ASUS Pad
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>


From rick.jones2@hp.com  Thu Aug  2 11:49:02 2012
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Cc: tcpm@ietf.org, "Matthew Mathis \(mattmathis@google.com\)" <mattmathis@google.com>
Subject: Re: [tcpm] Settling Time and RFC3517bis
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Joe -

Just so as I am sure I am thinking the same thing as you when you write 
"RTT" you mean that from the time a segment is first transmitted rather 
than from the time when some indication of a hole arrives at the sender yes?

rick jones

From touch@isi.edu  Thu Aug  2 11:58:48 2012
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Cc: tcpm@ietf.org, "Matthew Mathis \(mattmathis@google.com\)" <mattmathis@google.com>
Subject: Re: [tcpm] Settling Time and RFC3517bis
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On 8/2/2012 11:48 AM, Rick Jones wrote:
> Joe -
>
> Just so as I am sure I am thinking the same thing as you when you write
> "RTT" you mean that from the time a segment is first transmitted rather
> than from the time when some indication of a hole arrives at the sender
> yes?

RTT = round trip time.

It's the time delay between when a message is sent from the source to 
when any confirmation of receipt of that message could return back to 
that source.

Holes are determined by gaps of sequences of packets. They ALWAYS 
involve delays larger than 1 RTT because within 1 RTT either a packet 
arrives or it doesn't; when it doesn't, there's no indication of a gap 
until the *next* packet arrives. Packets that are reordered that cause 
that same effect are reordered because at least one packet (the earlier 
one) is delayed longer, which means the second packet arrives but the 
first didn't. The time is thus still longer than 1 RTT.

Joe

From internet-drafts@ietf.org  Thu Aug  2 12:31:07 2012
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-04.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : TCP Extensions for High Performance
	Author(s)       : David Borman
                          Bob Braden
                          Van Jacobson
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-1323bis-04.txt
	Pages           : 46
	Date            : 2012-08-02

Abstract:
   This memo presents a set of TCP extensions to improve performance
   over large bandwidth*delay product paths and to provide reliable
   operation over very high-speed paths.  It defines TCP options for
   scaled windows and timestamps, which are designed to provide
   compatible interworking with TCP's that do not implement the
   extensions.  The timestamps are used for two distinct mechanisms:
   RTTM (Round Trip Time Measurement) and PAWS (Protection Against
   Wrapped Sequences).  Selective acknowledgments are not included in
   this memo.

   This memo updates and obsoletes RFC 1323.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-1323bis

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-1323bis-04


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


From rs@netapp.com  Thu Aug  2 12:33:01 2012
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
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Hi,

While updating 1323bis for RFC2119 Phrases (see IETF84 presentation), I fou=
nd that some normative text that is directly from RFC 1323 does not strictl=
y use the normative phrasing...

Most new (diff 1323 / 1323bis) text does, though.


I would like to know the opinion of the WG as how to proceed with non-RFC 2=
119 normative phrases.

Shall they be kept unchanged (identical to 1323)?
Or shall the wording be changed, and the appropriate uppercase words includ=
ed to make these definitely normative?

So far, I updated the phrasing only in those places, that are normative, an=
d lend themselves to easy update (e.g. simply changing  the lowercase 2119 =
word to uppercase).





I also found two references to variables (TS.TStamp, TS.Stamp) that seem to=
 mean TS.Recent, but have been missed in RFC1323 so far :)=20


Please review the updated version!


Richard Scheffenegger



From michael.scharf@alcatel-lucent.com  Thu Aug  2 13:45:21 2012
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Thu, 2 Aug 2012 22:45:16 +0200
Thread-Topic: Consenus on adding a new TCPM charter milestone for accurate feedback
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Subject: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Hi all,

There has been broad support for a new TCPM charter milestone on accurate E=
CN feedback in TCP.

We therefore want to confirm that working group consensus is to add the fol=
lowing milestone to the TCPM carter:

   "Sep 2013 	Submit document on accurate ECN feedback in TCP to the IESG f=
or publication as an Experimental RFC"

The chairs believe that the consensus is not to adopt any specific document=
 as of now. However, the working group should converge on the mechanism by =
March 2013 latest.

Please feel free to comment if you have any additional thoughts, within the=
 next two weeks.

Thanks

Michael (on behalf of the TCPM co-chairs)

From ycheng@google.com  Thu Aug  2 14:13:47 2012
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From: Yuchung Cheng <ycheng@google.com>
Date: Thu, 2 Aug 2012 14:13:26 -0700
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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is accurate ECN part of "maintenance and minor extension"?



On Thu, Aug 2, 2012 at 1:45 PM, Scharf, Michael (Michael)
<michael.scharf@alcatel-lucent.com> wrote:
> Hi all,
>
> There has been broad support for a new TCPM charter milestone on accurate ECN feedback in TCP.
>
> We therefore want to confirm that working group consensus is to add the following milestone to the TCPM carter:
>
>    "Sep 2013    Submit document on accurate ECN feedback in TCP to the IESG for publication as an Experimental RFC"
>
> The chairs believe that the consensus is not to adopt any specific document as of now. However, the working group should converge on the mechanism by March 2013 latest.
>
> Please feel free to comment if you have any additional thoughts, within the next two weeks.
>
> Thanks
>
> Michael (on behalf of the TCPM co-chairs)
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From ycheng@google.com  Thu Aug  2 14:17:26 2012
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From: Yuchung Cheng <ycheng@google.com>
Date: Thu, 2 Aug 2012 14:17:04 -0700
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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On Thu, Aug 2, 2012 at 2:13 PM, Yuchung Cheng <ycheng@google.com> wrote:
> is accurate ECN part of "maintenance and minor extension"?
sorry got cut-off.

I second that to add acc-ECN to tcpm, but often times I am confused
the roles between iccrg and tcpm. To me there should be a tcp-dev to
discuss all TCP changes. Other protocols often just apply the
principles of CC in TCP.


>
>
>
> On Thu, Aug 2, 2012 at 1:45 PM, Scharf, Michael (Michael)
> <michael.scharf@alcatel-lucent.com> wrote:
>> Hi all,
>>
>> There has been broad support for a new TCPM charter milestone on accurate ECN feedback in TCP.
>>
>> We therefore want to confirm that working group consensus is to add the following milestone to the TCPM carter:
>>
>>    "Sep 2013    Submit document on accurate ECN feedback in TCP to the IESG for publication as an Experimental RFC"
>>
>> The chairs believe that the consensus is not to adopt any specific document as of now. However, the working group should converge on the mechanism by March 2013 latest.
>>
>> Please feel free to comment if you have any additional thoughts, within the next two weeks.
>>
>> Thanks
>>
>> Michael (on behalf of the TCPM co-chairs)
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm

From michael.scharf@alcatel-lucent.com  Thu Aug  2 16:10:41 2012
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Yuchung Cheng <ycheng@google.com>
Date: Fri, 3 Aug 2012 01:10:31 +0200
Thread-Topic: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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> > is accurate ECN part of "maintenance and minor extension"?

Do you argue that it is a major change?

For instance, just transporting a more accurate encoding does not imply tha=
t CC is changed at all.

> I second that to add acc-ECN to tcpm, but often times I am=20
> confused the roles between iccrg and tcpm. To me there should=20
> be a tcp-dev to discuss all TCP changes. Other protocols=20
> often just apply the principles of CC in TCP.

ICCRG is reseach... Is more research needed to find for an experimental enc=
oding that allows more than one feedback per RTT?

Anybody who thinks that this topic would be too "disruptive" for TCPM: Plea=
se speak up now.

Michael

From rs@netapp.com  Thu Aug  2 17:06:14 2012
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Yuchung Cheng <ycheng@google.com>
Thread-Topic: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Yuchung,

I believe that this work is definitely in charter for TCPM.

AccECN is about a semantic/signaling change of the (existing) wire-protocol=
 bits.

We are not proposing to make incompatible, or major changes to the wire-pro=
tocol.


There are other, standards track WGs that actually have a documented need f=
or a better congestion feedback scheme than exists today in TCP (conex). Th=
e only feedback scheme meeting these requirements is currently DCCP (Ack Ve=
ctor ECN-marked), SCTP (draft only), and RTP (draft only).



The problem you are perhaps observing is that as TCP is the de-facto standa=
rd protocol for the majority of transport needs (and we probably all agree =
that it's often not the best choice technically, but the only choice for de=
ployability), the limitations of TCP become apparent and people are still i=
nterested in addressing those within the given confines...=20
=20
Best regards,

Richard Scheffenegger


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Scharf, Michael (Michael)
> Sent: Donnerstag, 02. August 2012 16:11
> To: Yuchung Cheng
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for
> accurate feedback
>=20
> > > is accurate ECN part of "maintenance and minor extension"?
>=20
> Do you argue that it is a major change?
>=20
> For instance, just transporting a more accurate encoding does not imply
> that CC is changed at all.
>=20
> > I second that to add acc-ECN to tcpm, but often times I am confused
> > the roles between iccrg and tcpm. To me there should be a tcp-dev to
> > discuss all TCP changes. Other protocols often just apply the
> > principles of CC in TCP.
>=20
> ICCRG is reseach... Is more research needed to find for an experimental
> encoding that allows more than one feedback per RTT?
>=20
> Anybody who thinks that this topic would be too "disruptive" for TCPM:
> Please speak up now.
>=20
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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On Thu, Aug 2, 2012 at 4:10 PM, Scharf, Michael (Michael)
<michael.scharf@alcatel-lucent.com> wrote:
>> > is accurate ECN part of "maintenance and minor extension"?
>
> Do you argue that it is a major change?
>
> For instance, just transporting a more accurate encoding does not imply that CC is changed at all.
>
>> I second that to add acc-ECN to tcpm, but often times I am
>> confused the roles between iccrg and tcpm. To me there should
>> be a tcp-dev to discuss all TCP changes. Other protocols
>> often just apply the principles of CC in TCP.
>
> ICCRG is reseach... Is more research needed to find for an experimental encoding that allows more than one feedback per RTT?
>
> Anybody who thinks that this topic would be too "disruptive" for TCPM: Please speak up now.



with the proposed milestone, we'll add more signals to ECN then ?
discuss what to do about them in iccrg or other group?
what's the plan for better ECN is my question. sorry if this sounds dumb.

>
> Michael

From john@jlc.net  Thu Aug  2 18:28:32 2012
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From: John Leslie <john@jlc.net>
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Scharf, Michael (Michael) <michael.scharf@alcatel-lucent.com> wrote:
> 
> We therefore want to confirm that working group consensus is to add
> the following milestone to the TCPM carter:
> 
>    "Sep 2013 	Submit document on accurate ECN feedback in TCP to the
>     IESG for publication as an Experimental RFC"
> 
> The chairs believe that the consensus is not to adopt any specific
> document as of now. However, the working group should converge on
> the mechanism by March 2013 latest.

   I find myself worried by the name "accurate ECN feedback".

   That is a reasonable name for the draft we've seen; but I don't
think it's the right name for the _output_ of this WG.

   The issue is a bit complicated, alas: there's very good reason
to want a signal coming back to the sender which carries more
precision than the current signal (ECN-Echo in an unreliable ACK).
Quoting RFC 3168 Section 6.1:
] 
] * An ECT codepoint is set in packets transmitted by the sender to
]   indicate that ECN is supported by the transport entities for
]   these packets.
]
] * An ECN-capable router detects impending congestion and detects
]   that an ECT codepoint is set in the packet it is about to drop.
]   Instead of dropping the packet, the router chooses to set the CE
]   codepoint in the IP header and forwards the packet.
]
] * The receiver receives the packet with the CE codepoint set, and
]   sets the ECN-Echo flag in its next TCP ACK sent to the sender.
]
] * The sender receives the TCP ACK with ECN-Echo set, and reacts to
]   the congestion as if a packet had been dropped.
]
] * The sender sets the CWR flag in the TCP header of the next
]   packet sent to the receiver to acknowledge its receipt of and
]   reaction to the ECN-Echo flag.

   In fact, due to the unreliable nature of the ACK, the receiver
continues to set ECN-Echo (ECE) until it sees a CWR. From 6.1.3:
] 
] To provide robustness against the possibility of a dropped ACK packet
] carrying an ECN-Echo flag, the TCP receiver sets the ECN-Echo flag in
] a series of ACK packets sent subsequently.  The TCP receiver uses the
] CWR flag received from the TCP sender to determine when to stop
] setting the ECN-Echo flag.

   This is messy! and it is only acceptable for trying to match the
window-reduction of a packet loss.

   So there is very good reason to come up with a mechanism to provide
better reliability and precision of informing the sender about ECT
codepoints seen by the reciver.

   I entirely support such an effort, but I think "greater precision"
would be a better term than "accurate".

--
John Leslie <john@jlcnet>

From mallman@icir.org  Fri Aug  3 08:12:34 2012
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This is all pretty mis-guided.  Let me say just a few quick things.

> 3) STANDARD TCP can not recover in less than one RTT, 

Correct.  Nor would we want it to.  If you retransmit in less than one
RTT from the time a segment is originally sourced then you will always
be retransmitting blind.  I.e., you'll have no indication as to whether
that segment needs to be retransmitted or not.

>    SACK TCP as written certainly can and does currently recover in
>    less that one RTT on at least long links.  

Incorrect.  At least I am aware of no TCP SACK-based loss recovery
variant that retransmits a segment less than one RTT after it was
originally sent.  You reference 3517bis in the subject.  The algorithm
specified in RFC 3517 and 3517bis *does not* retransmit a segment in
less than one RTT.

>    With the changes I have been proposing nearly all errors can be
>    recovered in less than one RTT.

The only way to recover in less than one RTT is to retransmit blind.
I.e., retransmit segments without any indication as to whether they have
arrived at the receiver or not.  One could write such an algorithm.
And, probably show it fixes lost segments sooner.  But, the question
would immediately become "at what cost?".  I.e., how many needless
retransmits are being piled into the network to accomplish this task?
And, the answer nearly has to be proportional to the benefit you obtain.
I.e., I bet that as you speculatively retransmit more segments you will
also spurious retransmit more segments.  

And, this is needless retransmits that we know are being piled into a
congested network and so are using scarce network resources.  So, all
around a bad idea.

allman




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Anthony Sabatini spake unto us the following wisdom:
> 2) My point about fragmentation is that "Fragmentation Reassembly
> Timer" and "Reordering Time" are different names to describe the same
> issue.

They are not.  This seems to reflect a fundamental misunderstanding of
the respective roles of IP and TCP.  There are two primary differences
between IP fragmentation and TCP segmentation for the purposes of this
specific discussion, namely:

 * IP fragmentation and delivery are best-effort, while TCP segments
   are reliable (at a higher layer in the stack than the IP datagrams
   they ride in)

 * IP fragments of a single datagram are always generated nearly
   simultaneously (within the performance limits of the IP stack and
   the local transport hardware), while TCP segments may be generated
   and emitted arbitrarily far apart in time

Each of these points has an important implication.  The first means
that, if you do not receive the fragments of an IP segment quite
rapidly (and a single RTT is a reasonable upper bound, though
certainly one could envision cases where it may take longer), you are
not going to receive them at all.  The IP datagram can be discarded
blindly, and the sending machine will retransmit its contents via
whatever appropriate means if its upper-layer protocols direct so.
The TCP segment, on the other hand, must have its suspected loss
somehow transmitted back to the sending TCP so that it can be
retransmitted.

The second point means that a TCP segment can arrive at any time and
may arrive a very long time (days, months, or years, in the extreme)
after the previous segment.  It is true that, under normal conditions,
a TCP segment should not arrive long after a *succeeding* segment, but
it can happen, and as TCP is a reliable transport, this should be
accounted for.  Since TCP cannot react to loss in less than one full
RTT anyway (another misconception in your logic, more below), the
receiving TCP might as well wait a full RTT or more for progress. If
the segment is truly lost, the connection will only begin to
(substantially) suffer performance-wise if the sending 3517-compliant
TCP runs out of new data to send.  Until that time, whether the
segment is retransmitted in the meantime or not, the TCP will proceed
essentially unaltered.  (The precise timing of the retransmission will
affect segment clocking somewhat, but in the long run, in the case of
a true loss, this should be a moderate effect.)

> 3) STANDARD TCP can not recover in less than one RTT, SACK TCP as
> written certainly can and does currently recover in less that one RTT
> on at least long links.  With the changes I have been proposing nearly
> all errors can be recovered in less than one RTT.

This is entirely untrue.  It takes one RTT at minimum for a sending
TCP to receive indication from the receiver that a segment was lost.
The sending TCP must send segments following the lost segment that the
receiving TCP SACKs.  Only when those SACK blocks arrive at the
sending TCP does it know that the segment was possibly lost.

> 6) Why is the IETF spending all of its time covering for greedy
> carriers who do not install enough fiber or switching equipment?  ALL
> of the changes being made in the congestion area result in a WORSE
> experience for the end user, the people we should really care about.

I recommend that you read the following paper:

    Congestion Avoidance and Control.  Van Jacobson.  Proceedings of
    the SIGCOMM 1988 Symposium on Communications Architectures and
    Protocols.  pp. 314--329.

A version of this paper was published in the 1988 Computer
Communications Review, and a version co-authored by Michael Karels
also appeared in 1988 with additional information.

The takeaway from these papers is that, not only is congestion control
a good idea on paper, but its absence has been documented to cause
congestion collapse and complete non-responsiveness of the network to
useful traffic.  You are correct that provisioning can help this
circumstance *in some cases*, but it cannot, by definition, help in
all cases -- there can always exist a situation where the network has
more packets in flight than it can support, no matter how widely
provisioned, if the provisioning is finite.  A lesson learned from the
congestion collapses of the 80s is that this condition of over
over-subscription of valuable data does not even have to persist -- it
only has to last long enough for the retransmissions of that valuable
data to overwhelm the network.

Ethan

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Hi John,

Am I right in assuming that with "greater precision" you imply that this do=
esn't necessarily mean "exact" feedback?

(Originally we discussed using "more accurate" in the title of the draft;  =
that meant lower precision than "exact", but the "more" got lost along the =
way, leaving "accurate").


I would argue that the current ECE/CWR mechanism is perfectly reliable - up=
 to a certain precision limited by the number of bits set aside for the sig=
nal back from the receiver to the sender. What would be more reliable?

(ECE is a counter that can only increase, and remains at the maximum value =
until being reset; CWR provides the reset signal).


OTOH, if the feedback signal is encoded with enough bits, reliability can b=
e brought arbitrarily close to "perfect", even without an explicit forward =
signal (CWR).


As soon as the counter (and signal for it) cover all the packets individual=
ly in a window, the ECN feedback can be exact; The question becomes, can th=
e reset mechanism be abandoned when "enough" bits are available for the sig=
nal, or must the principal architecture (a counter increasing up to some po=
int, until it is reset) be maintained...

BTW: I found that ECN for SCTP also failed to discuss this, but there a 32 =
bit counter is used (plus a Reset signal/CWR).





To jump forward a bit, I suspect the delineation is somewhere around 5 bits=
 (2.3 bits in the codepoint draft; 3 bits in reECN; 8 bits in the TCP Optio=
n draft). Using 5 bits or less, a handshare mechanism is required; using mo=
re than 5, and a simple counter suffices to maintain exactness (until TCP s=
uffers from loss of the ACK clock etc).

The signaling is another question; Perhaps a certain loss of granularity is=
 acceptable for the currently discussed protocols (i.e. Conex; slight overe=
stimating congestion is not as bad as underestimating). But other (future) =
uses (e.g. DCTCP) would probably require the full precision - with other wo=
rds, "exact" signaling...


However, we (authors), while discussing this internally, would also be glad=
 of any opinion regarding this!


Richard Scheffenegger




> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> John Leslie
> Sent: Donnerstag, 02. August 2012 18:29
> To: Scharf, Michael (Michael)
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for
> accurate feedback
>=20
> Scharf, Michael (Michael) <michael.scharf@alcatel-lucent.com> wrote:
> >
> > We therefore want to confirm that working group consensus is to add
> > the following milestone to the TCPM carter:
> >
> >    "Sep 2013 	Submit document on accurate ECN feedback in TCP to the
> >     IESG for publication as an Experimental RFC"
> >
> > The chairs believe that the consensus is not to adopt any specific
> > document as of now. However, the working group should converge on
> > the mechanism by March 2013 latest.
>=20
>    I find myself worried by the name "accurate ECN feedback".
>=20
>    That is a reasonable name for the draft we've seen; but I don't
> think it's the right name for the _output_ of this WG.
>=20
>    The issue is a bit complicated, alas: there's very good reason
> to want a signal coming back to the sender which carries more
> precision than the current signal (ECN-Echo in an unreliable ACK).
> Quoting RFC 3168 Section 6.1:
> ]
> ] * An ECT codepoint is set in packets transmitted by the sender to
> ]   indicate that ECN is supported by the transport entities for
> ]   these packets.
> ]
> ] * An ECN-capable router detects impending congestion and detects
> ]   that an ECT codepoint is set in the packet it is about to drop.
> ]   Instead of dropping the packet, the router chooses to set the CE
> ]   codepoint in the IP header and forwards the packet.
> ]
> ] * The receiver receives the packet with the CE codepoint set, and
> ]   sets the ECN-Echo flag in its next TCP ACK sent to the sender.
> ]
> ] * The sender receives the TCP ACK with ECN-Echo set, and reacts to
> ]   the congestion as if a packet had been dropped.
> ]
> ] * The sender sets the CWR flag in the TCP header of the next
> ]   packet sent to the receiver to acknowledge its receipt of and
> ]   reaction to the ECN-Echo flag.
>=20
>    In fact, due to the unreliable nature of the ACK, the receiver
> continues to set ECN-Echo (ECE) until it sees a CWR. From 6.1.3:
> ]
> ] To provide robustness against the possibility of a dropped ACK packet
> ] carrying an ECN-Echo flag, the TCP receiver sets the ECN-Echo flag in
> ] a series of ACK packets sent subsequently.  The TCP receiver uses the
> ] CWR flag received from the TCP sender to determine when to stop
> ] setting the ECN-Echo flag.
>=20
>    This is messy! and it is only acceptable for trying to match the
> window-reduction of a packet loss.
>=20
>    So there is very good reason to come up with a mechanism to provide
> better reliability and precision of informing the sender about ECT
> codepoints seen by the reciver.
>=20
>    I entirely support such an effort, but I think "greater precision"
> would be a better term than "accurate".
>=20
> --
> John Leslie <john@jlcnet>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From john@jlc.net  Fri Aug  3 11:34:22 2012
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From: John Leslie <john@jlc.net>
To: "Scheffenegger, Richard" <rs@netapp.com>
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Scheffenegger, Richard <rs@netapp.com> wrote:
> 
> Hi John,
> 
> Am I right in assuming that with "greater precision" you imply that
> this doesn't necessarily mean "exact" feedback?

   Yes, but... This is mainly to avoid arguing what "exact" might mean.

> (Originally we discussed using "more accurate" in the title of the draft;
> that meant lower precision than "exact", but the "more" got lost along
> the way, leaving "accurate").

   (I must admit I'm at least as anxious to avoid arguing what "accurate"
might mean.)

> I would argue that the current ECE/CWR mechanism is perfectly reliable

   I might take the other side of that argument...

> - up to a certain precision limited by the number of bits set aside
> for the signal back from the receiver to the sender. What would be more
> reliable?

   "Reliable" in TCP means that everything not acknowledged will be
retransmitted. That would imply repeating each ECE until that specific
ECE is acknowledged -- and clarifying to the sender how that CWR was
understood.

   But I'm not trying to argue either way whether "reliable" would be
"better" than "unreliable".

> (ECE is a counter that can only increase, and remains at the maximumi
> value until being reset; CWR provides the reset signal).

   Umm... "remains at maximum" implies some loss of precision...

> OTOH, if the feedback signal is encoded with enough bits, reliability
> can be brought arbitrarily close to "perfect", even without an explicit
> forward signal (CWR).

   Indeed, one could argue "reliability" by unacknowledged repetition.
(I'd prefer some other term, but it certainly would serve for the
"precision" I was seeking.)

> As soon as the counter (and signal for it) cover all the packets
> individually in a window, the ECN feedback can be exact;

   Actually, there are two counters, packets and (payload) bytes.

> The question becomes, can the reset mechanism be abandoned when
> "enough" bits are available for the signal, or must the principal
> architecture (a counter increasing up to some point, until it is
> reset) be maintained...

   I don't choose to express an opinion on that yet.

> BTW: I found that ECN for SCTP also failed to discuss this, but
> there a 32 bit counter is used (plus a Reset signal/CWR).

   :^)

> To jump forward a bit, I suspect the delineation is somewhere around
> 5 bits (2.3 bits in the codepoint draft; 3 bits in reECN; 8 bits in
> the TCP Option draft). Using 5 bits or less, a handshare mechanism is
> required; using more than 5, and a simple counter suffices to maintain
> exactness (until TCP suffers from loss of the ACK clock etc).

   No comment (yet)...

> The signaling is another question; Perhaps a certain loss of
> granularity is acceptable for the currently discussed protocols
> (i.e. Conex; slight overestimating congestion is not as bad as
> underestimating). But other (future) uses (e.g. DCTCP) would probably
> require the full precision - with other words, "exact" signaling...

   "Exact" _is_ a reasonable goal...

> However, we (authors), while discussing this internally, would also
> be glad of any opinion regarding this!

   I will note that current ECN processing in TCP is awfully specialized:
I can imagine congestion-control algorithms it would serve poorly. :^(
But leaving that aside, ECN processing for other transport protocols
almost certainly deserves a different treatment.

   It would be nice if our "greater precision" effort could serve the
needs of yet-unseen future transport protocols.

--
John Leslie <john@jlc.net>

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From: "Scheffenegger, Richard" <rs@netapp.com>
To: John Leslie <john@jlc.net>
Thread-Topic: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Hi John,


Thanks for your comments so far!

> > I would argue that the current ECE/CWR mechanism is perfectly reliable
>=20
>    I might take the other side of that argument...

Could you please explain? (I should have perhaps added, RFC3168 is perfectl=
y reliable for at least, and up to, one CE mark per RTT; The sender gets th=
e information that at least one CE mark was received in the RTT under all c=
ircumstances).


>    Actually, there are two counters, packets and (payload) bytes.

Correct; but as soon as not every data segment is ACKed (and the ACK not lo=
st), the delineation (for figuring out which payload bytes got marked) beco=
mes fuzzy... Even when the sender keeps state as to the segment boundaries,=
 it's not clear which of multiple segments were marked (except in the two e=
xtreme cases), when delayed ACKs (or even ACK compression) are in play...=20

Which brings up another question that I haven't put in this forum:

Will there be a conceivable benefit when the sender knows the exact order o=
r markings?

The way the receiver ECN state-machine of DCTCP is described allows the sen=
der to reconstruct the exact order of markings (the "CE stream"), provided =
no (relevant) ACKs are lost. With the complication that the sender rarely k=
nows, when and if a relevant ACK got dropped...

This level of precision is lost, when the receiver uses some kind of summar=
y signal; But making such a signal reliable requires even more bits, and so=
me more or less complex encoding scheme...


But I am not sure if such an extreme precision will be necessary to TCP at =
least (and other transports do provide means to encode that...)


To have a proper counter (congested bytes), I think that would need to be e=
xplicit; Mirja did some experiments using implicit ACKs, but to get them wo=
rk properly, you end up with sending out more ACKs than strictly necessary =
(like what DCTCP receivers do), and single ACK loss can still result in wro=
ng signals at the sender....




Richard Scheffenegger




> -----Original Message-----
> From: John Leslie [mailto:john@jlc.net]
> Sent: Freitag, 03. August 2012 11:34
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for
> accurate feedback
>=20
> Scheffenegger, Richard <rs@netapp.com> wrote:
> >
> > Hi John,
> >
> > Am I right in assuming that with "greater precision" you imply that
> > this doesn't necessarily mean "exact" feedback?
>=20
>    Yes, but... This is mainly to avoid arguing what "exact" might mean.
>=20
> > (Originally we discussed using "more accurate" in the title of the
> > draft; that meant lower precision than "exact", but the "more" got
> > lost along the way, leaving "accurate").
>=20
>    (I must admit I'm at least as anxious to avoid arguing what "accurate"
> might mean.)
>=20
> > I would argue that the current ECE/CWR mechanism is perfectly reliable
>=20
>    I might take the other side of that argument...
>=20
> > - up to a certain precision limited by the number of bits set aside
> > for the signal back from the receiver to the sender. What would be
> > more reliable?
>=20
>    "Reliable" in TCP means that everything not acknowledged will be
> retransmitted. That would imply repeating each ECE until that specific EC=
E
> is acknowledged -- and clarifying to the sender how that CWR was
> understood.
>=20
>    But I'm not trying to argue either way whether "reliable" would be
> "better" than "unreliable".
>=20
> > (ECE is a counter that can only increase, and remains at the maximumi
> > value until being reset; CWR provides the reset signal).
>=20
>    Umm... "remains at maximum" implies some loss of precision...
>=20
> > OTOH, if the feedback signal is encoded with enough bits, reliability
> > can be brought arbitrarily close to "perfect", even without an
> > explicit forward signal (CWR).
>=20
>    Indeed, one could argue "reliability" by unacknowledged repetition.
> (I'd prefer some other term, but it certainly would serve for the
> "precision" I was seeking.)
>=20
> > As soon as the counter (and signal for it) cover all the packets
> > individually in a window, the ECN feedback can be exact;
>=20
>    Actually, there are two counters, packets and (payload) bytes.
>=20
> > The question becomes, can the reset mechanism be abandoned when
> > "enough" bits are available for the signal, or must the principal
> > architecture (a counter increasing up to some point, until it is
> > reset) be maintained...
>=20
>    I don't choose to express an opinion on that yet.
>=20
> > BTW: I found that ECN for SCTP also failed to discuss this, but there
> > a 32 bit counter is used (plus a Reset signal/CWR).
>=20
>    :^)
>=20
> > To jump forward a bit, I suspect the delineation is somewhere around
> > 5 bits (2.3 bits in the codepoint draft; 3 bits in reECN; 8 bits in
> > the TCP Option draft). Using 5 bits or less, a handshare mechanism is
> > required; using more than 5, and a simple counter suffices to maintain
> > exactness (until TCP suffers from loss of the ACK clock etc).
>=20
>    No comment (yet)...
>=20
> > The signaling is another question; Perhaps a certain loss of
> > granularity is acceptable for the currently discussed protocols (i.e.
> > Conex; slight overestimating congestion is not as bad as
> > underestimating). But other (future) uses (e.g. DCTCP) would probably
> > require the full precision - with other words, "exact" signaling...
>=20
>    "Exact" _is_ a reasonable goal...
>=20
> > However, we (authors), while discussing this internally, would also be
> > glad of any opinion regarding this!
>=20
>    I will note that current ECN processing in TCP is awfully specialized:
> I can imagine congestion-control algorithms it would serve poorly. :^( Bu=
t
> leaving that aside, ECN processing for other transport protocols almost
> certainly deserves a different treatment.
>=20
>    It would be nice if our "greater precision" effort could serve the
> needs of yet-unseen future transport protocols.
>=20
> --
> John Leslie <john@jlc.net>

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Date: Sun, 5 Aug 2012 13:02:07 -0700
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From: Matt Mathis <mattmathis@google.com>
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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I agree that "accurate" is a little bit too strong of term.  Perhaps
"counted", "proportional", "quantified", "measured", etc.

And yes, TCPM is probably the right place, since all CC standards
track docs are explicitly in scope, even as applied to other
protocols.

If we separate signalling from action, some proposed actions clearly
need to mature in ICCRG for a bit.   Here is a tough question though:
does the choice of alternate actions effect the design of the
signalling?  If so, it may be premature to commit on the signalling.
(e.g. for low threshold ECN (aka DCTCP) isolated ECN marks would be
extremely unlikely and it would probably be ok to miss a few.),

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay


On Fri, Aug 3, 2012 at 1:58 PM, Scheffenegger, Richard <rs@netapp.com> wrot=
e:
>
> Hi John,
>
>
> Thanks for your comments so far!
>
>> > I would argue that the current ECE/CWR mechanism is perfectly reliable
>>
>>    I might take the other side of that argument...
>
> Could you please explain? (I should have perhaps added, RFC3168 is perfec=
tly reliable for at least, and up to, one CE mark per RTT; The sender gets =
the information that at least one CE mark was received in the RTT under all=
 circumstances).
>
>
>>    Actually, there are two counters, packets and (payload) bytes.
>
> Correct; but as soon as not every data segment is ACKed (and the ACK not =
lost), the delineation (for figuring out which payload bytes got marked) be=
comes fuzzy... Even when the sender keeps state as to the segment boundarie=
s, it's not clear which of multiple segments were marked (except in the two=
 extreme cases), when delayed ACKs (or even ACK compression) are in play...
>
> Which brings up another question that I haven't put in this forum:
>
> Will there be a conceivable benefit when the sender knows the exact order=
 or markings?
>
> The way the receiver ECN state-machine of DCTCP is described allows the s=
ender to reconstruct the exact order of markings (the "CE stream"), provide=
d no (relevant) ACKs are lost. With the complication that the sender rarely=
 knows, when and if a relevant ACK got dropped...
>
> This level of precision is lost, when the receiver uses some kind of summ=
ary signal; But making such a signal reliable requires even more bits, and =
some more or less complex encoding scheme...
>
>
> But I am not sure if such an extreme precision will be necessary to TCP a=
t least (and other transports do provide means to encode that...)
>
>
> To have a proper counter (congested bytes), I think that would need to be=
 explicit; Mirja did some experiments using implicit ACKs, but to get them =
work properly, you end up with sending out more ACKs than strictly necessar=
y (like what DCTCP receivers do), and single ACK loss can still result in w=
rong signals at the sender....
>
>
>
>
> Richard Scheffenegger
>
>
>
>
>> -----Original Message-----
>> From: John Leslie [mailto:john@jlc.net]
>> Sent: Freitag, 03. August 2012 11:34
>> To: Scheffenegger, Richard
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for
>> accurate feedback
>>
>> Scheffenegger, Richard <rs@netapp.com> wrote:
>> >
>> > Hi John,
>> >
>> > Am I right in assuming that with "greater precision" you imply that
>> > this doesn't necessarily mean "exact" feedback?
>>
>>    Yes, but... This is mainly to avoid arguing what "exact" might mean.
>>
>> > (Originally we discussed using "more accurate" in the title of the
>> > draft; that meant lower precision than "exact", but the "more" got
>> > lost along the way, leaving "accurate").
>>
>>    (I must admit I'm at least as anxious to avoid arguing what "accurate=
"
>> might mean.)
>>
>> > I would argue that the current ECE/CWR mechanism is perfectly reliable
>>
>>    I might take the other side of that argument...
>>
>> > - up to a certain precision limited by the number of bits set aside
>> > for the signal back from the receiver to the sender. What would be
>> > more reliable?
>>
>>    "Reliable" in TCP means that everything not acknowledged will be
>> retransmitted. That would imply repeating each ECE until that specific E=
CE
>> is acknowledged -- and clarifying to the sender how that CWR was
>> understood.
>>
>>    But I'm not trying to argue either way whether "reliable" would be
>> "better" than "unreliable".
>>
>> > (ECE is a counter that can only increase, and remains at the maximumi
>> > value until being reset; CWR provides the reset signal).
>>
>>    Umm... "remains at maximum" implies some loss of precision...
>>
>> > OTOH, if the feedback signal is encoded with enough bits, reliability
>> > can be brought arbitrarily close to "perfect", even without an
>> > explicit forward signal (CWR).
>>
>>    Indeed, one could argue "reliability" by unacknowledged repetition.
>> (I'd prefer some other term, but it certainly would serve for the
>> "precision" I was seeking.)
>>
>> > As soon as the counter (and signal for it) cover all the packets
>> > individually in a window, the ECN feedback can be exact;
>>
>>    Actually, there are two counters, packets and (payload) bytes.
>>
>> > The question becomes, can the reset mechanism be abandoned when
>> > "enough" bits are available for the signal, or must the principal
>> > architecture (a counter increasing up to some point, until it is
>> > reset) be maintained...
>>
>>    I don't choose to express an opinion on that yet.
>>
>> > BTW: I found that ECN for SCTP also failed to discuss this, but there
>> > a 32 bit counter is used (plus a Reset signal/CWR).
>>
>>    :^)
>>
>> > To jump forward a bit, I suspect the delineation is somewhere around
>> > 5 bits (2.3 bits in the codepoint draft; 3 bits in reECN; 8 bits in
>> > the TCP Option draft). Using 5 bits or less, a handshare mechanism is
>> > required; using more than 5, and a simple counter suffices to maintain
>> > exactness (until TCP suffers from loss of the ACK clock etc).
>>
>>    No comment (yet)...
>>
>> > The signaling is another question; Perhaps a certain loss of
>> > granularity is acceptable for the currently discussed protocols (i.e.
>> > Conex; slight overestimating congestion is not as bad as
>> > underestimating). But other (future) uses (e.g. DCTCP) would probably
>> > require the full precision - with other words, "exact" signaling...
>>
>>    "Exact" _is_ a reasonable goal...
>>
>> > However, we (authors), while discussing this internally, would also be
>> > glad of any opinion regarding this!
>>
>>    I will note that current ECN processing in TCP is awfully specialized=
:
>> I can imagine congestion-control algorithms it would serve poorly. :^( B=
ut
>> leaving that aside, ECN processing for other transport protocols almost
>> certainly deserves a different treatment.
>>
>>    It would be nice if our "greater precision" effort could serve the
>> needs of yet-unseen future transport protocols.
>>
>> --
>> John Leslie <john@jlc.net>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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On Sun, Aug 5, 2012 at 1:02 PM, Matt Mathis <mattmathis@google.com> wrote:
> I agree that "accurate" is a little bit too strong of term.  Perhaps
> "counted", "proportional", "quantified", "measured", etc.
>
> And yes, TCPM is probably the right place, since all CC standards
> track docs are explicitly in scope, even as applied to other
> protocols.
>
> If we separate signalling from action, some proposed actions clearly
> need to mature in ICCRG for a bit.   Here is a tough question though:
> does the choice of alternate actions effect the design of the
> signalling?  If so, it may be premature to commit on the signalling.
> (e.g. for low threshold ECN (aka DCTCP) isolated ECN marks would be
> extremely unlikely and it would probably be ok to miss a few.),
That's exactly my question ... thanks for presenting it in a much
better wording.

>
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
>
> On Fri, Aug 3, 2012 at 1:58 PM, Scheffenegger, Richard <rs@netapp.com> wr=
ote:
>>
>> Hi John,
>>
>>
>> Thanks for your comments so far!
>>
>>> > I would argue that the current ECE/CWR mechanism is perfectly reliabl=
e
>>>
>>>    I might take the other side of that argument...
>>
>> Could you please explain? (I should have perhaps added, RFC3168 is perfe=
ctly reliable for at least, and up to, one CE mark per RTT; The sender gets=
 the information that at least one CE mark was received in the RTT under al=
l circumstances).
>>
>>
>>>    Actually, there are two counters, packets and (payload) bytes.
>>
>> Correct; but as soon as not every data segment is ACKed (and the ACK not=
 lost), the delineation (for figuring out which payload bytes got marked) b=
ecomes fuzzy... Even when the sender keeps state as to the segment boundari=
es, it's not clear which of multiple segments were marked (except in the tw=
o extreme cases), when delayed ACKs (or even ACK compression) are in play..=
.
>>
>> Which brings up another question that I haven't put in this forum:
>>
>> Will there be a conceivable benefit when the sender knows the exact orde=
r or markings?
>>
>> The way the receiver ECN state-machine of DCTCP is described allows the =
sender to reconstruct the exact order of markings (the "CE stream"), provid=
ed no (relevant) ACKs are lost. With the complication that the sender rarel=
y knows, when and if a relevant ACK got dropped...
>>
>> This level of precision is lost, when the receiver uses some kind of sum=
mary signal; But making such a signal reliable requires even more bits, and=
 some more or less complex encoding scheme...
>>
>>
>> But I am not sure if such an extreme precision will be necessary to TCP =
at least (and other transports do provide means to encode that...)
>>
>>
>> To have a proper counter (congested bytes), I think that would need to b=
e explicit; Mirja did some experiments using implicit ACKs, but to get them=
 work properly, you end up with sending out more ACKs than strictly necessa=
ry (like what DCTCP receivers do), and single ACK loss can still result in =
wrong signals at the sender....
>>
>>
>>
>>
>> Richard Scheffenegger
>>
>>
>>
>>
>>> -----Original Message-----
>>> From: John Leslie [mailto:john@jlc.net]
>>> Sent: Freitag, 03. August 2012 11:34
>>> To: Scheffenegger, Richard
>>> Cc: tcpm@ietf.org
>>> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for
>>> accurate feedback
>>>
>>> Scheffenegger, Richard <rs@netapp.com> wrote:
>>> >
>>> > Hi John,
>>> >
>>> > Am I right in assuming that with "greater precision" you imply that
>>> > this doesn't necessarily mean "exact" feedback?
>>>
>>>    Yes, but... This is mainly to avoid arguing what "exact" might mean.
>>>
>>> > (Originally we discussed using "more accurate" in the title of the
>>> > draft; that meant lower precision than "exact", but the "more" got
>>> > lost along the way, leaving "accurate").
>>>
>>>    (I must admit I'm at least as anxious to avoid arguing what "accurat=
e"
>>> might mean.)
>>>
>>> > I would argue that the current ECE/CWR mechanism is perfectly reliabl=
e
>>>
>>>    I might take the other side of that argument...
>>>
>>> > - up to a certain precision limited by the number of bits set aside
>>> > for the signal back from the receiver to the sender. What would be
>>> > more reliable?
>>>
>>>    "Reliable" in TCP means that everything not acknowledged will be
>>> retransmitted. That would imply repeating each ECE until that specific =
ECE
>>> is acknowledged -- and clarifying to the sender how that CWR was
>>> understood.
>>>
>>>    But I'm not trying to argue either way whether "reliable" would be
>>> "better" than "unreliable".
>>>
>>> > (ECE is a counter that can only increase, and remains at the maximumi
>>> > value until being reset; CWR provides the reset signal).
>>>
>>>    Umm... "remains at maximum" implies some loss of precision...
>>>
>>> > OTOH, if the feedback signal is encoded with enough bits, reliability
>>> > can be brought arbitrarily close to "perfect", even without an
>>> > explicit forward signal (CWR).
>>>
>>>    Indeed, one could argue "reliability" by unacknowledged repetition.
>>> (I'd prefer some other term, but it certainly would serve for the
>>> "precision" I was seeking.)
>>>
>>> > As soon as the counter (and signal for it) cover all the packets
>>> > individually in a window, the ECN feedback can be exact;
>>>
>>>    Actually, there are two counters, packets and (payload) bytes.
>>>
>>> > The question becomes, can the reset mechanism be abandoned when
>>> > "enough" bits are available for the signal, or must the principal
>>> > architecture (a counter increasing up to some point, until it is
>>> > reset) be maintained...
>>>
>>>    I don't choose to express an opinion on that yet.
>>>
>>> > BTW: I found that ECN for SCTP also failed to discuss this, but there
>>> > a 32 bit counter is used (plus a Reset signal/CWR).
>>>
>>>    :^)
>>>
>>> > To jump forward a bit, I suspect the delineation is somewhere around
>>> > 5 bits (2.3 bits in the codepoint draft; 3 bits in reECN; 8 bits in
>>> > the TCP Option draft). Using 5 bits or less, a handshare mechanism is
>>> > required; using more than 5, and a simple counter suffices to maintai=
n
>>> > exactness (until TCP suffers from loss of the ACK clock etc).
>>>
>>>    No comment (yet)...
>>>
>>> > The signaling is another question; Perhaps a certain loss of
>>> > granularity is acceptable for the currently discussed protocols (i.e.
>>> > Conex; slight overestimating congestion is not as bad as
>>> > underestimating). But other (future) uses (e.g. DCTCP) would probably
>>> > require the full precision - with other words, "exact" signaling...
>>>
>>>    "Exact" _is_ a reasonable goal...
>>>
>>> > However, we (authors), while discussing this internally, would also b=
e
>>> > glad of any opinion regarding this!
>>>
>>>    I will note that current ECN processing in TCP is awfully specialize=
d:
>>> I can imagine congestion-control algorithms it would serve poorly. :^( =
But
>>> leaving that aside, ECN processing for other transport protocols almost
>>> certainly deserves a different treatment.
>>>
>>>    It would be nice if our "greater precision" effort could serve the
>>> needs of yet-unseen future transport protocols.
>>>
>>> --
>>> John Leslie <john@jlc.net>
>> _______________________________________________
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From: Yuchung Cheng <ycheng@google.com>
Date: Sun, 5 Aug 2012 15:32:08 -0700
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Cc: tcpm@ietf.org, draft-ietf-tcpm-fastopen@tools.ietf.org
Subject: Re: [tcpm] Some ideas building on draft-ietf-tcpm-fastopen-01
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Hi Bob,

Re-inst. our discussions during the meeting.

On Thu, Jul 26, 2012 at 10:19 PM,  <bob.briscoe@bt.com> wrote:
> Yuchung, Jerry, Sivasankar, Arvind,
>
> I liked this enough to want to work on improving it. I have a couple of
> proposals that occurred to me, which I've finally made time to write up.
>
> (I'll send editorial nits from reviewing the draft later)
>
> ==Proposal #1/ ==
> Fast Open Cookie Request MUST (SHOULD?) be on a FIN
>
> I believe it would be better for the reasons below if:
> * the client put a fast open cookie request in its FIN, not its SYN (whether
> or not the server has sent a FIN first).
> * Then the server can send the fastopen cookie on the FIN-ACK.
> * If the client times out waiting for a FIN-ACK, it just re-sends as usual.
> * If the server's FIN-ACK never arrives (perhaps because a middlebox has
> eaten it), the client just doesn't have a cookie for the next session and
> uses a 3WHS as normal.
>
> Note, only the Request would have to be on a FIN. The Fast Open Cookie
> itself would obviously still be on the SYN of subsequent connections. Then,
> when a subsequent connection finishes, it can refresh the cookie on its FIN,
> ready for another connection later.
>
> This solves a couple of potential problems with a fastopen request-response
> on the initial SYN exchange:
> a) A FIN-cookie sidesteps the risk of unpredictable delay of the very first
> SYN due to some middleboxes dropping new TCP options (any timeouts due to
> non-delivery don't hold up the session, which has already finished)
> b) Transports GETting typical Web pages could (ab)use the fastopen cookie on
> the SYN exchange by opening multiple TCP connections during the initial
> connection without each one suffering the delay of a 3WHS. This creates a
> perverse incentive to use multiple parallel connections, because you can get
> the same latency benefit as HTTP pipelining [Yoshifumi posed this problem
> some time ago].  If clients can only request a fastopen cookie on a FIN,
> they don't have this perverse incentive to not bother with pipelining.
>
> I have no evidence, but it seems intuitive that if the TCP option got
> through on the first FIN, it might be more likely to get through on the next
> SYN, altho I admit options on SYNs are looked at more closely by
> middleboxes.

Like Jerry's response, I also like your ideas of cookie in FIN to
mitigate middlebox drops.
But I don't think doing so will practically diminish parallel
connections, because
most other time it works better: this change only affects the (first) connection
to get cookies. I personally don't mind adding a MAY for cookie
exchange in FIN stage.


>
>
> ==Proposal #2/ ==
> Fastopen cookies (optionally) authenticate both the client's network address
> and the its knowledge of the TCP timestamp option on the segment carrying
> the server's cookie.
>
> I would like to propose that if the server TCP-timestamps the segment
> carrying its cookie, the client MUST use a timestamp option on the segment
> that starts the subsequent fast open, and in its timestamp echo field it
> must echo the timestamp sent with the earlier cookie. Then:
> i) the server can use the timestamp as part of the material for generating
> and validating the cookie and
> ii) the server can set an expiry_duration, and check whether the echoed
> timestamp is older than time(now) - expiry_duration.
>
> Reasoning:
> The draft currently says:
> "   3. The cookie expires after a certain amount of time. The reason is
>        detailed in the "Security Consideration" section. This can be
>        done by either periodically changing the server key used to
>        generate cookies or including a timestamp in the cookie.
> "
>
> Let's take each in turn:
> * Periodically changing the server key: Let's imagine the key changes every
> 30 min. Then all cookies issued in the last 30 mins stop being useful, even
> those just issued a second ago, or a minute ago, or 15 minutes ago...
> * Including a timestamp in the cookie. This is only useful if the timestamp
> is in the clear within the cookie, otherwise the server cannot use it to key
> the cookie. But a timestamp in the clear takes valuable bits away from the
> cookie, weakening its strength. This is unnecessary if there's already space
> for a timestamp option on the segment (which is very common).
>
> Security Considerations mentions only two ways a client can collect valid
> cookies from other hosts:
> i) compromising other hosts, or
> ii) (legally) taking over the lease of an IP address from a DHCP server.
> But there is a much more problematic third way:
> iii) sitting behind the same NAT as a neighbour (esp. a carrier-grade NAT
> serving thousands of clients), where the NAT uses a common IP address for
> many clients.
>
> As it stands, the cookie is only dependent on the client's network address.
> Binding a TCP timestamp option into the cookie as well adds a lot more
> entropy.
>
> With only the IP address for entropy, a client behind a NAT (or an attacker
> in control of a compromised host behind a NAT) can get its own (valid)
> cookie and use it in a brute-force spoof of numerous neighbouring clients -
> to either SYN-flood the server or launch a reflection attack on its
> neighbours by fast-opening loads of new TCP connections requesting large
> data items on behalf of its neighbours, bypassing the safety check of the
> 3WHS.
>
> In contrast, if a cookie depends on both the client's network address and
> server's timestamp, a cookie requested by a client behind a NAT will only be
> the same as a neighbours cookie served by the same server within the same
> clock tick.
>
> This doesn't stop attackers getting valid cookies from compromised hosts,
> but it does effectively prevent these more trivial attacks on neighbours.
I also like TS idea but I see two problems in proposal #2:
a) scale: servers need to remember the timestamps. NOTE: in our implementation,
   servers do not need to remember the cookie.

b) a change in TS RFC: SYN ECR value is not 0. Are Mr. firewalls ok with that?


>
>
> ==Proposal #3/ ==
> Special cookie values
>
> (A minor point compared to the two above)
>
> 4.1.2 Server Cookie handling
>
> "  Also note the server may want to use special cookie values, e.g.,
>    "0", for specific scenarios. For example, the server wants to notify
>    the client the support of TFO, but chooses not to return a valid
>    cookie for security or performance reasons upon receiving a TFO
>    request.
> "
>
> I suggest the server just returns the cookie option with length 2 and no
> cookie data, to show it supports the option, but chooses not to give a
Acked. I prefer so actually.

> cookie. Then the server doesn't have to run an extra 'if' test every time it
> generates a cookie, just to check the cookie doesn't collide with a special
> value.
>
> This would require the value of the cookie option request kind to be
> different from the cookie option kind. Otherwise a response meaning "cookie
> supported but empty" would be confused with a cookie request for the
> half-connection in the opposite direction.
>
>
> == Outstanding concern ==
>
> Similar to a concern of Joe Touch's and Richard Scheffeneger's: dividing up
> TCP implementations into ones with idempotent semantics and ones without.
> This confuses the simple applicability of TCP. We need to work on that - but
> manyana.
Great point. We MUST be more specific about that point in the next revision.

All and all thanks for reviewing our draft :)

>
>
>
> Bob
>
>
>
> [PS. Apologies if you already got this once - I've been 'experimenting' with
> my email]
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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From: Yuchung Cheng <ycheng@google.com>
Date: Sun, 5 Aug 2012 17:12:28 -0700
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Cc: Eric Dumazet <edumazet@google.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-experimental-options-01.txt
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On Wed, May 30, 2012 at 3:45 PM, Joe Touch <touch@isi.edu> wrote:
> Hi, all,
>
> This version incorporates feedback from the list. In specific:
>
> - minimum magic number length is not specified
>
> There are minor reorganization and revisions as well.
>
> Joe
Two questions:

1) how to share the magic value with other developers (for other OSes)?

Since I don't know, I am doing this on the list:
We implemented this draft in the Fast Open for Linux. The magic is 0xF989
by this simple formula (suggested by Eric Dumazet):
time_t now = time(NULL);
printf("0x%04x\n", 0xFFFF & (now ^ (now >> 16)));

   Maybe a simple text file under ietf.org work?

2) While the draft does not address the transition plan, we developers have
   to. TFO plan is that once IANA grants the precious number, Linux will
   support both values for a while, until most clients use the new value.
   It's hassle but well ... but other TCP features may have different plans.

   Should this kind of transition plan be part of any new TCP draft that
   needs new option?

Thanks.

>
>
> On 5/30/2012 3:40 PM, internet-drafts@ietf.org wrote:
>>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories. This draft is a work item of the TCP Maintenance and Minor
>> Extensions Working Group of the IETF.
>>
>>         Title           : Shared Use of Experimental TCP Options
>>         Author(s)       : Joe Touch
>>         Filename        : draft-ietf-tcpm-experimental-options-01.txt
>>         Pages           : 8
>>         Date            : 2012-05-30
>>
>>     This document describes how TCP option codepoints can support
>>     concurrent experiments using a magic number field. This mechanism
>>     avoids the need for a coordinated registry, and is backward-
>>     compatible with currently known uses.
>>
>>
>> A URL for this Internet-Draft is:
>>
>> http://www.ietf.org/internet-drafts/draft-ietf-tcpm-experimental-options-01.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> This Internet-Draft can be retrieved at:
>>
>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpm-experimental-options-01.txt
>>
>> The IETF datatracker page for this Internet-Draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-experimental-options/
>>
>> _______________________________________________
>> 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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Cc: Eric Dumazet <edumazet@google.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-experimental-options-01.txt
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On Aug 5, 2012, at 5:12 PM, Yuchung Cheng wrote:

> On Wed, May 30, 2012 at 3:45 PM, Joe Touch <touch@isi.edu> wrote:
>> Hi, all,
>>=20
>> This version incorporates feedback from the list. In specific:
>>=20
>> - minimum magic number length is not specified
>>=20
>> There are minor reorganization and revisions as well.
>>=20
>> Joe
> Two questions:
>=20
> 1) how to share the magic value with other developers (for other =
OSes)?
>=20
> Since I don't know, I am doing this on the list:
> We implemented this draft in the Fast Open for Linux. The magic is =
0xF989
> by this simple formula (suggested by Eric Dumazet):
> time_t now =3D time(NULL);
> printf("0x%04x\n", 0xFFFF & (now ^ (now >> 16)));
>=20
>   Maybe a simple text file under ietf.org work?

The magic numbers should be noted in the document that defines the =
option. We can ask IANA to manage the list of such known numbers, but =
two key issues need to be ensured:

1) the numbers in such a list are voluntarily reported, and IANA doesn't =
enforce ownership
	i.e., it's just a list of known uses, not a registry

2) numbers in the list need to comply with the recommendations of this =
doc, i.e., at least 32-bits long


> 2) While the draft does not address the transition plan, we developers =
have
>   to. TFO plan is that once IANA grants the precious number, Linux =
will
>   support both values for a while, until most clients use the new =
value.

Correct. We'll make that more clear in an update of this doc.

Support for the magic-number variant can be dropped either when it's =
desired (to reduce code) or to deliberately transition to production =
code (vs. experimental, which may differ).

>   It's hassle but well ... but other TCP features may have different =
plans.
>=20
>   Should this kind of transition plan be part of any new TCP draft =
that
>   needs new option?

Yes. I'll add that recommendation to this doc as well.

Note that the assigned option SHOULD NOT include the magic number, =
though - to conserve SYN space.

Joe

>=20
> Thanks.
>=20
>>=20
>>=20
>> On 5/30/2012 3:40 PM, internet-drafts@ietf.org wrote:
>>>=20
>>>=20
>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>> directories. This draft is a work item of the TCP Maintenance and =
Minor
>>> Extensions Working Group of the IETF.
>>>=20
>>>        Title           : Shared Use of Experimental TCP Options
>>>        Author(s)       : Joe Touch
>>>        Filename        : draft-ietf-tcpm-experimental-options-01.txt
>>>        Pages           : 8
>>>        Date            : 2012-05-30
>>>=20
>>>    This document describes how TCP option codepoints can support
>>>    concurrent experiments using a magic number field. This mechanism
>>>    avoids the need for a coordinated registry, and is backward-
>>>    compatible with currently known uses.
>>>=20
>>>=20
>>> A URL for this Internet-Draft is:
>>>=20
>>> =
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-experimental-options-0=
1.txt
>>>=20
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>=20
>>> This Internet-Draft can be retrieved at:
>>>=20
>>> =
ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpm-experimental-options-01=
.txt
>>>=20
>>> The IETF datatracker page for this Internet-Draft is:
>>> =
https://datatracker.ietf.org/doc/draft-ietf-tcpm-experimental-options/
>>>=20
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


From rs@netapp.com  Tue Aug  7 09:40:38 2012
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Matt Mathis <mattmathis@google.com>
Thread-Topic: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Hi Matt,

> does the choice of alternate actions effect the design of the signalling?
> If so, it may be premature to commit on the signalling.
> (e.g. for low threshold ECN (aka DCTCP) isolated ECN marks would be
> extremely unlikely and it would probably be ok to miss a few.),

I believe that any signaling would still have to cater for legacy style, lo=
w-density ECN marking (RED) for some time.=20

Further I suspect that the way how the sender handles segments (segment-ori=
ented or bytestream-oriented TCP) can also have an indirect implication as =
to the quality of a feedback signal, if only a few bits can be used for it.

I asked this question a couple days ago, if the exact ordering in which CE =
marks are received on the receiver need to be conveyed back to the sender. =
That would allow the sender to infer exactly which packet was marked. One a=
rgument in favor of this which was brought forward was that this could allo=
w the sender to know how many "congestion bytes" were marked. However, that=
 particular aspect can also be addressed with a byte-counting feedback sign=
al, and the full CE "bit stream" is not required then.

The last question I still wonder is, would there be any benefit to the netw=
ork to know when the sender did react to a control signal...

When using Conex, CWR really is redundant (but it does mark the exact segme=
nt when TCP CC reacted; conex signals don't necessarily have that, but the =
draft is still in flux there).

Today, CWR is also (primarily) used as a "reset" signal to the receiver, as=
 much as a signal that the sender TCP CC reacted.


If we could find enough bits (as mentioned my experiments point to slightly=
 over 5 bits as being sufficient), a very simple, straight forward signalin=
g scheme should cater even unknown future needs (except for the exact CE bi=
tstream case mentioned above).

Best regards,

Richard Scheffenegger


> -----Original Message-----
> From: Matt Mathis [mailto:mattmathis@google.com]
> Sent: Sonntag, 05. August 2012 13:02
> To: Scheffenegger, Richard
> Cc: John Leslie; tcpm@ietf.org
> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for
> accurate feedback
>=20
> I agree that "accurate" is a little bit too strong of term.  Perhaps
> "counted", "proportional", "quantified", "measured", etc.
>=20
> And yes, TCPM is probably the right place, since all CC standards track
> docs are explicitly in scope, even as applied to other protocols.
>=20
> If we separate signalling from action, some proposed actions clearly
> need to mature in ICCRG for a bit.   Here is a tough question though:
> does the choice of alternate actions effect the design of the signalling?
> If so, it may be premature to commit on the signalling.
> (e.g. for low threshold ECN (aka DCTCP) isolated ECN marks would be
> extremely unlikely and it would probably be ok to miss a few.),
>=20
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>=20
>=20
> On Fri, Aug 3, 2012 at 1:58 PM, Scheffenegger, Richard <rs@netapp.com>
> wrote:
> >
> > Hi John,
> >
> >
> > Thanks for your comments so far!
> >
> >> > I would argue that the current ECE/CWR mechanism is perfectly
> >> > reliable
> >>
> >>    I might take the other side of that argument...
> >
> > Could you please explain? (I should have perhaps added, RFC3168 is
> perfectly reliable for at least, and up to, one CE mark per RTT; The
> sender gets the information that at least one CE mark was received in the
> RTT under all circumstances).
> >
> >
> >>    Actually, there are two counters, packets and (payload) bytes.
> >
> > Correct; but as soon as not every data segment is ACKed (and the ACK no=
t
> lost), the delineation (for figuring out which payload bytes got marked)
> becomes fuzzy... Even when the sender keeps state as to the segment
> boundaries, it's not clear which of multiple segments were marked (except
> in the two extreme cases), when delayed ACKs (or even ACK compression) ar=
e
> in play...
> >
> > Which brings up another question that I haven't put in this forum:
> >
> > Will there be a conceivable benefit when the sender knows the exact
> order or markings?
> >
> > The way the receiver ECN state-machine of DCTCP is described allows the
> sender to reconstruct the exact order of markings (the "CE stream"),
> provided no (relevant) ACKs are lost. With the complication that the
> sender rarely knows, when and if a relevant ACK got dropped...
> >
> > This level of precision is lost, when the receiver uses some kind of
> summary signal; But making such a signal reliable requires even more bits=
,
> and some more or less complex encoding scheme...
> >
> >
> > But I am not sure if such an extreme precision will be necessary to
> > TCP at least (and other transports do provide means to encode that...)
> >
> >
> > To have a proper counter (congested bytes), I think that would need to
> be explicit; Mirja did some experiments using implicit ACKs, but to get
> them work properly, you end up with sending out more ACKs than strictly
> necessary (like what DCTCP receivers do), and single ACK loss can still
> result in wrong signals at the sender....
> >
> >
> >
> >
> > Richard Scheffenegger
> >
> >
> >
> >
> >> -----Original Message-----
> >> From: John Leslie [mailto:john@jlc.net]
> >> Sent: Freitag, 03. August 2012 11:34
> >> To: Scheffenegger, Richard
> >> Cc: tcpm@ietf.org
> >> Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone
> >> for accurate feedback
> >>
> >> Scheffenegger, Richard <rs@netapp.com> wrote:
> >> >
> >> > Hi John,
> >> >
> >> > Am I right in assuming that with "greater precision" you imply that
> >> > this doesn't necessarily mean "exact" feedback?
> >>
> >>    Yes, but... This is mainly to avoid arguing what "exact" might mean=
.
> >>
> >> > (Originally we discussed using "more accurate" in the title of the
> >> > draft; that meant lower precision than "exact", but the "more" got
> >> > lost along the way, leaving "accurate").
> >>
> >>    (I must admit I'm at least as anxious to avoid arguing what
> "accurate"
> >> might mean.)
> >>
> >> > I would argue that the current ECE/CWR mechanism is perfectly
> >> > reliable
> >>
> >>    I might take the other side of that argument...
> >>
> >> > - up to a certain precision limited by the number of bits set aside
> >> > for the signal back from the receiver to the sender. What would be
> >> > more reliable?
> >>
> >>    "Reliable" in TCP means that everything not acknowledged will be
> >> retransmitted. That would imply repeating each ECE until that
> >> specific ECE is acknowledged -- and clarifying to the sender how that
> >> CWR was understood.
> >>
> >>    But I'm not trying to argue either way whether "reliable" would be
> >> "better" than "unreliable".
> >>
> >> > (ECE is a counter that can only increase, and remains at the
> >> > maximumi value until being reset; CWR provides the reset signal).
> >>
> >>    Umm... "remains at maximum" implies some loss of precision...
> >>
> >> > OTOH, if the feedback signal is encoded with enough bits,
> >> > reliability can be brought arbitrarily close to "perfect", even
> >> > without an explicit forward signal (CWR).
> >>
> >>    Indeed, one could argue "reliability" by unacknowledged repetition.
> >> (I'd prefer some other term, but it certainly would serve for the
> >> "precision" I was seeking.)
> >>
> >> > As soon as the counter (and signal for it) cover all the packets
> >> > individually in a window, the ECN feedback can be exact;
> >>
> >>    Actually, there are two counters, packets and (payload) bytes.
> >>
> >> > The question becomes, can the reset mechanism be abandoned when
> >> > "enough" bits are available for the signal, or must the principal
> >> > architecture (a counter increasing up to some point, until it is
> >> > reset) be maintained...
> >>
> >>    I don't choose to express an opinion on that yet.
> >>
> >> > BTW: I found that ECN for SCTP also failed to discuss this, but
> >> > there a 32 bit counter is used (plus a Reset signal/CWR).
> >>
> >>    :^)
> >>
> >> > To jump forward a bit, I suspect the delineation is somewhere
> >> > around
> >> > 5 bits (2.3 bits in the codepoint draft; 3 bits in reECN; 8 bits in
> >> > the TCP Option draft). Using 5 bits or less, a handshare mechanism
> >> > is required; using more than 5, and a simple counter suffices to
> >> > maintain exactness (until TCP suffers from loss of the ACK clock
> etc).
> >>
> >>    No comment (yet)...
> >>
> >> > The signaling is another question; Perhaps a certain loss of
> >> > granularity is acceptable for the currently discussed protocols (i.e=
.
> >> > Conex; slight overestimating congestion is not as bad as
> >> > underestimating). But other (future) uses (e.g. DCTCP) would
> >> > probably require the full precision - with other words, "exact"
> signaling...
> >>
> >>    "Exact" _is_ a reasonable goal...
> >>
> >> > However, we (authors), while discussing this internally, would also
> >> > be glad of any opinion regarding this!
> >>
> >>    I will note that current ECN processing in TCP is awfully
> specialized:
> >> I can imagine congestion-control algorithms it would serve poorly.
> >> :^( But leaving that aside, ECN processing for other transport
> >> protocols almost certainly deserves a different treatment.
> >>
> >>    It would be nice if our "greater precision" effort could serve the
> >> needs of yet-unseen future transport protocols.
> >>
> >> --
> >> John Leslie <john@jlc.net>
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm

From pasi.sarolahti@iki.fi  Wed Aug  8 02:32:53 2012
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: [tcpm] WGLC for Proportional Rate Reduction
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Hi,

As discussed in the meeting last week, we will now start working group =
last call for "Proportional Rate Reduction for TCP" =
(draft-ietf-tcpm-proportional-rate-reduction-02), to be published as =
Experimental RFC. The WGLC runs until Friday, August 24th. Please read =
the draft and send comments by then. If you have read the draft and =
think it is ok, notifying just that is also helpful (even without =
comments).

The draft is available at =
http://tools.ietf.org/html/draft-ietf-tcpm-proportional-rate-reduction-02=20=


- Pasi


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From: Murari Sridharan <muraris@microsoft.com>
To: Matt Mathis <mattmathis@google.com>, rtp-congestion <RTP-congestion@alvestrand.no>, "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: [R-C] Fwd: video and TCP interaction
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Subject: Re: [tcpm] [R-C] Fwd: video and TCP interaction
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It is interesting that this paper references RFC 5681 when it finds that th=
e OS after a period of inactivity goes back to initial (restart) cwnd of 10=
 packets :-)

Thanks,
Murari

-----Original Message-----
From: rtp-congestion-bounces@alvestrand.no [mailto:rtp-congestion-bounces@a=
lvestrand.no] On Behalf Of Matt Mathis
Sent: Thursday, August 9, 2012 9:56 AM
To: rtp-congestion
Cc: Nandita Dukkipati
Subject: [R-C] Fwd: video and TCP interaction

This seem particularly relevant here.

Confused, Timid, and Unstable:
Picking a Video Streaming Rate is Hard

http://www.stanford.edu/~huangty/videoRateAdapt.pdf

>From IMC 2012

Thanks,
--MM--
_______________________________________________
Rtp-congestion mailing list
Rtp-congestion@alvestrand.no
http://www.alvestrand.no/mailman/listinfo/rtp-congestion






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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Murari Sridharan <muraris@microsoft.com>, Matt Mathis <mattmathis@google.com>, rtp-congestion <RTP-congestion@alvestrand.no>, "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: [R-C] Fwd: video and TCP interaction
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Subject: Re: [tcpm] [R-C] Fwd: video and TCP interaction
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SSR is a killer in http streaming apps and there are certain ways to avoid =
that long inactivity that will trigger SSR. There are quite a number of ser=
vers out there running a particular OS that do not implement SSR at all.

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of M=
urari Sridharan
> Sent: Thursday, August 09, 2012 6:05 PM
> To: Matt Mathis; rtp-congestion; tcpm (tcpm@ietf.org)
> Subject: Re: [tcpm] [R-C] Fwd: video and TCP interaction
>=20
> It is interesting that this paper references RFC 5681 when it finds that =
the OS after a period of inactivity goes back to initial
> (restart) cwnd of 10 packets :-)
>=20
> Thanks,
> Murari
>=20
> -----Original Message-----
> From: rtp-congestion-bounces@alvestrand.no [mailto:rtp-congestion-bounces=
@alvestrand.no] On Behalf Of Matt Mathis
> Sent: Thursday, August 9, 2012 9:56 AM
> To: rtp-congestion
> Cc: Nandita Dukkipati
> Subject: [R-C] Fwd: video and TCP interaction
>=20
> This seem particularly relevant here.
>=20
> Confused, Timid, and Unstable:
> Picking a Video Streaming Rate is Hard
>=20
> http://www.stanford.edu/~huangty/videoRateAdapt.pdf
>=20
> From IMC 2012
>=20
> Thanks,
> --MM--
> _______________________________________________
> Rtp-congestion mailing list
> Rtp-congestion@alvestrand.no
> http://www.alvestrand.no/mailman/listinfo/rtp-congestion
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From rick.jones2@hp.com  Thu Aug  9 15:37:00 2012
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Matt Mathis <mattmathis@google.com>, rtp-congestion <RTP-congestion@alvestrand.no>
Subject: Re: [tcpm] [R-C] Fwd: video and TCP interaction
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On 08/09/2012 03:05 PM, Murari Sridharan wrote:
> It is interesting that this paper references RFC 5681 when it finds
> that the OS after a period of inactivity goes back to initial
> (restart) cwnd of 10 packets :-)

I personally am not much of a fan of slow-start after idle (or persist I 
suppose) but this:

>> The competing wget flow has already filled the buffer during the
>> OFF period, and so the video flow sees very high packet loss.

caught my eye and immediately made me think of the likes of codel and 
wonder how that might have changed behaviour of the overall system.  It 
also made me wonder if the three services might have tested their 
heuristics in an environment with lots and lots of buffering at the 
bottleneck...

rick jones

> Thanks,
> Murari
>
> -----Original Message-----
> From: rtp-congestion-bounces@alvestrand.no [mailto:rtp-congestion-bounces@alvestrand.no] On Behalf Of Matt Mathis
> Sent: Thursday, August 9, 2012 9:56 AM
> To: rtp-congestion
> Cc: Nandita Dukkipati
> Subject: [R-C] Fwd: video and TCP interaction
>
> This seem particularly relevant here.
>
> Confused, Timid, and Unstable:
> Picking a Video Streaming Rate is Hard
>
> http://www.stanford.edu/~huangty/videoRateAdapt.pdf
>
>>From IMC 2012
>
> Thanks,
> --MM--
> _______________________________________________
> Rtp-congestion mailing list
> Rtp-congestion@alvestrand.no
> http://www.alvestrand.no/mailman/listinfo/rtp-congestion
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From mzanaty@cisco.com  Thu Aug  9 19:17:31 2012
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Murari Sridharan <muraris@microsoft.com>, Matt Mathis <mattmathis@google.com>, rtp-congestion <RTP-congestion@alvestrand.no>, "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: [R-C] Fwd: video and TCP interaction
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Subject: Re: [tcpm] [R-C] Fwd: video and TCP interaction
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More interesting than IW10 on CDNs, "Service C" (which must be Vudu) stream=
s at 9M! The RMCAT evaluation criteria should include testing with competin=
g DASH flows (Vudu for fireworks bigger than those in Vancouver).

Mo

-----Original Message-----
From: rtp-congestion-bounces@alvestrand.no [mailto:rtp-congestion-bounces@a=
lvestrand.no] On Behalf Of Murari Sridharan
Sent: Thursday, August 09, 2012 6:05 PM
To: Matt Mathis; rtp-congestion; tcpm (tcpm@ietf.org)
Cc: Nandita Dukkipati
Subject: Re: [R-C] Fwd: video and TCP interaction

It is interesting that this paper references RFC 5681 when it finds that th=
e OS after a period of inactivity goes back to initial (restart) cwnd of 10=
 packets :-)

Thanks,
Murari

-----Original Message-----
From: rtp-congestion-bounces@alvestrand.no [mailto:rtp-congestion-bounces@a=
lvestrand.no] On Behalf Of Matt Mathis
Sent: Thursday, August 9, 2012 9:56 AM
To: rtp-congestion
Cc: Nandita Dukkipati
Subject: [R-C] Fwd: video and TCP interaction

This seem particularly relevant here.

Confused, Timid, and Unstable:
Picking a Video Streaming Rate is Hard

http://www.stanford.edu/~huangty/videoRateAdapt.pdf

>From IMC 2012

Thanks,
--MM--
_______________________________________________
Rtp-congestion mailing list
Rtp-congestion@alvestrand.no
http://www.alvestrand.no/mailman/listinfo/rtp-congestion





_______________________________________________
Rtp-congestion mailing list
Rtp-congestion@alvestrand.no
http://www.alvestrand.no/mailman/listinfo/rtp-congestion

From nishida@sfc.wide.ad.jp  Fri Aug 10 11:40:25 2012
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Subject: [tcpm] next steps for draft-ietf-tcpm-initcwnd
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Hello folks,

Since the WGLC duration for this draft has been expired and there seems to
be no continuous discussions, we are thinking how to proceed this.

Our primary decision is we would like to suggest authors to update the
draft based on the discussions during the WGLC.
There has been discussion on restructuring the draft. While this appears to
be a valid suggestion, we conclude it's not strong enough to impose extra
efforts on the authors at this time.
(However, if authors want to do, I won't stop it. I will respect author's
efforts. :) )

So, authors, please revise the draft based on the comments you received.
If there's no strong objection on the revised version, we'll send it to
IESG.

We really appreciate everyone who has been taking your time for reviewing
and discussing the draft.

Thanks,
--
Yoshifumi
one of tcpm co-chairs

--f46d0442810415411f04c6edad89
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Hello folks,<br><br>Since the WGLC duration for this draft has been expired=
 and there seems to be no continuous discussions, we are thinking how to pr=
oceed this. <br><br>Our primary decision is we would like to suggest author=
s to update the draft based on the discussions during the WGLC.<br>

There has been discussion on restructuring the draft. While this appears to=
 be a valid suggestion, we conclude it&#39;s not strong enough to impose ex=
tra efforts on the authors at this time. <br>(However, if authors want to d=
o, I won&#39;t stop it. I will respect author&#39;s efforts. :) )<br>

<br>So, authors, please revise the draft based on the comments you received=
.<br>If there&#39;s no strong objection on the revised version, we&#39;ll s=
end it to IESG.<br><br>We really appreciate everyone who has been taking yo=
ur time for reviewing and discussing the draft. <br>

<br>Thanks,<br>--<br>Yoshifumi<br>one of tcpm co-chairs<br>

--f46d0442810415411f04c6edad89--

From tsabatini@hotmail.com  Sat Aug 11 14:24:00 2012
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Cc: tcpm <tcpm@ietf.org>, Matt Mathis <mattmathis@google.com>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] Reordering Time and RFC3517bis
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All -

After reviewing things a bit more I realize no one has given serious though=
t to Fragmentation Reassembly Timer=2C Reordering Time and the pseudo equiv=
alent of three acks that is used to substitute for a more rigorous Reorderi=
ng Time.

First let us examine fragmentation reassembly timer.  Since=2C as was point=
ed out=2C all fragments are generated at the same time if we do not conside=
r reordering of the fragments then the timer need be no greater than the tr=
ansmission delay inherent in the fragment length as it relates to link spee=
d plus an amount for processing delay=2C relatively speaking a very small a=
mount of time.  Even if we include a reroute through a satellite link that =
adds only 500 ms to the timer=2C there is NO scenario under which reorderin=
g will take over two seconds so with an abundance of caution you could set =
fragment reassembly time to four seconds and be done with it.  Yet EVERY IP=
 stack sets this value to above fifteen seconds (Linux is 15)=2C some at th=
irty seconds (Microsoft is 30) and the rest at SIXTY SECONDS!! (Solaris=2C =
BSD).  These numbers are unsupportable and make fragmentation a good attack=
 vector because of it (interesting note=2C unlike the operating system peop=
le=2C Cisco who has to deal with this in the real world sets the value to 5=
 seconds).

Now let us look at 3517bis.  No where is there a mention of Reordering time=
 yet by definition it has to be considered.  Indeed if we did not consider =
reordering time then we could simply queue the retransmission of a missing =
segment as soon as a SACK block arrives that implies the segment is missing=
.  As it is if we saved the block and then waited an amount of time equal t=
o the reordering time=2C if we did not receive an ACK or SACK covering the =
range we could and should queue it for retransmission WHETHER WE RECEIVED A=
NY ACKs OR SACKs IN THE INTERIM.

The problem is three ACKs is a very very bad way to simulate reordering tim=
e since on a busy link it will trigger too early casing duplicate transmiss=
ions and if the link goes idle it will fire way to slowly or not at all=2C =
even though it could correctly have been used for retransmission with only =
one occurrence.

Also for you to think about - a Reordering time of greater than .25*RTT deg=
rades the throughput of the link in direct proportion to its value.

.Tony

Anthony Sabatini
200 West 20th Street
Apt. 1216
New York=2C NY 10011

Phone: (212) 867-7179
Mobile: (917) 224-8388

=20


> Date: Fri=2C 3 Aug 2012 19:37:32 -0400
> From: tsabatini@hotmail.com
> To: mallman@icir.org
> CC: tcpm@ietf.org=3B mattmathis@google.com=3B touch@isi.edu
> Subject: Re: [tcpm] Settling Time and RFC3517bis
>=20
> All -
>=20
> Because of the wording of the original comment it was not clear whether t=
he RTT referred to the generated SACK information from the receiver or the =
original transmission from the sender.  By definition=2C with respect to th=
e sender=2C it is not possible to properly retransmit the message in less t=
han one RTT plus Reordering time plus processing time.
>=20
> I am sorry for this misunderstanding.
>=20
> Tony
>=20
> Sent from my ASUS Pad
>=20
> Mark Allman <mallman@icir.org> wrote:
>=20
> >
> >This is all pretty mis-guided.  Let me say just a few quick things.
> >
> >> 3) STANDARD TCP can not recover in less than one RTT=2C=20
> >
> >Correct.  Nor would we want it to.  If you retransmit in less than one
> >RTT from the time a segment is originally sourced then you will always
> >be retransmitting blind.  I.e.=2C you'll have no indication as to whethe=
r
> >that segment needs to be retransmitted or not.
> >
> >>    SACK TCP as written certainly can and does currently recover in
> >>    less that one RTT on at least long links. =20
> >
> >Incorrect.  At least I am aware of no TCP SACK-based loss recovery
> >variant that retransmits a segment less than one RTT after it was
> >originally sent.  You reference 3517bis in the subject.  The algorithm
> >specified in RFC 3517 and 3517bis *does not* retransmit a segment in
> >less than one RTT.
> >
> >>    With the changes I have been proposing nearly all errors can be
> >>    recovered in less than one RTT.
> >
> >The only way to recover in less than one RTT is to retransmit blind.
> >I.e.=2C retransmit segments without any indication as to whether they ha=
ve
> >arrived at the receiver or not.  One could write such an algorithm.
> >And=2C probably show it fixes lost segments sooner.  But=2C the question
> >would immediately become "at what cost?".  I.e.=2C how many needless
> >retransmits are being piled into the network to accomplish this task?
> >And=2C the answer nearly has to be proportional to the benefit you obtai=
n.
> >I.e.=2C I bet that as you speculatively retransmit more segments you wil=
l
> >also spurious retransmit more segments. =20
> >
> >And=2C this is needless retransmits that we know are being piled into a
> >congested network and so are using scarce network resources.  So=2C all
> >around a bad idea.
> >
> >allman
> >
> >
> >
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
 		 	   		  =

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<html>
<head>
<style><!--
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body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
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<body class=3D'hmmessage'><div dir=3D'ltr'>All -<br><br>After reviewing thi=
ngs a bit more I realize no one has given serious thought to Fragmentation =
Reassembly Timer=2C Reordering Time and the pseudo equivalent of three acks=
 that is used to substitute for a more rigorous Reordering Time.<br><br>Fir=
st let us examine fragmentation reassembly timer.&nbsp=3B Since=2C as was p=
ointed out=2C all fragments are generated at the same time if we do not con=
sider reordering of the fragments then the timer need be no greater than th=
e transmission delay inherent in the fragment length as it relates to link =
speed plus an amount for processing delay=2C relatively speaking a very sma=
ll amount of time.&nbsp=3B Even if we include a reroute through a satellite=
 link that adds only 500 ms to the timer=2C there is NO scenario under whic=
h reordering will take over two seconds so with an abundance of caution you=
 could set fragment reassembly time to four seconds and be done with it.&nb=
sp=3B Yet EVERY IP stack sets this value to above fifteen seconds (Linux is=
 15)=2C some at thirty seconds (Microsoft is 30) and the rest at SIXTY SECO=
NDS!! (Solaris=2C BSD).&nbsp=3B These numbers are unsupportable and make fr=
agmentation a good attack vector because of it (interesting note=2C unlike =
the operating system people=2C Cisco who has to deal with this in the real =
world sets the value to 5 seconds).<br><br>Now let us look at 3517bis.&nbsp=
=3B No where is there a mention of Reordering time yet by definition it has=
 to be considered.&nbsp=3B Indeed if we did not consider reordering time th=
en we could simply queue the retransmission of a missing segment as soon as=
 a SACK block arrives that implies the segment is missing.&nbsp=3B As it is=
 if we saved the block and then waited an amount of time equal to the reord=
ering time=2C if we did not receive an ACK or SACK covering the range we co=
uld and should queue it for retransmission WHETHER WE RECEIVED ANY ACKs OR =
SACKs IN THE INTERIM.<br><br>The problem is three ACKs is a very very bad w=
ay to simulate reordering time since on a busy link it will trigger too ear=
ly casing duplicate transmissions and if the link goes idle it will fire wa=
y to slowly or not at all=2C even though it could correctly have been used =
for retransmission with only one occurrence.<br><br>Also for you to think a=
bout - a Reordering time of greater than .25*RTT degrades the throughput of=
 the link in direct proportion to its value.<br><br>.Tony<br><br>Anthony Sa=
batini<br>200&nbsp=3BWest 20th Street<br>Apt. 1216<br>New York=2C NY 10011<=
br><br>Phone: (212) 867-7179<br>Mobile: (917) 224-8388<br><br>&nbsp=3B<br><=
br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B Date: Fri=2C 3 Aug=
 2012 19:37:32 -0400<br>&gt=3B From: tsabatini@hotmail.com<br>&gt=3B To: ma=
llman@icir.org<br>&gt=3B CC: tcpm@ietf.org=3B mattmathis@google.com=3B touc=
h@isi.edu<br>&gt=3B Subject: Re: [tcpm] Settling Time and RFC3517bis<br>&gt=
=3B <br>&gt=3B All -<br>&gt=3B <br>&gt=3B Because of the wording of the ori=
ginal comment it was not clear whether the RTT referred to the generated SA=
CK information from the receiver or the original transmission from the send=
er.  By definition=2C with respect to the sender=2C it is not possible to p=
roperly retransmit the message in less than one RTT plus Reordering time pl=
us processing time.<br>&gt=3B <br>&gt=3B I am sorry for this misunderstandi=
ng.<br>&gt=3B <br>&gt=3B Tony<br>&gt=3B <br>&gt=3B Sent from my ASUS Pad<br=
>&gt=3B <br>&gt=3B Mark Allman &lt=3Bmallman@icir.org&gt=3B wrote:<br>&gt=
=3B <br>&gt=3B &gt=3B<br>&gt=3B &gt=3BThis is all pretty mis-guided.  Let m=
e say just a few quick things.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B 3) S=
TANDARD TCP can not recover in less than one RTT=2C <br>&gt=3B &gt=3B<br>&g=
t=3B &gt=3BCorrect.  Nor would we want it to.  If you retransmit in less th=
an one<br>&gt=3B &gt=3BRTT from the time a segment is originally sourced th=
en you will always<br>&gt=3B &gt=3Bbe retransmitting blind.  I.e.=2C you'll=
 have no indication as to whether<br>&gt=3B &gt=3Bthat segment needs to be =
retransmitted or not.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B    SACK TCP a=
s written certainly can and does currently recover in<br>&gt=3B &gt=3B&gt=
=3B    less that one RTT on at least long links.  <br>&gt=3B &gt=3B<br>&gt=
=3B &gt=3BIncorrect.  At least I am aware of no TCP SACK-based loss recover=
y<br>&gt=3B &gt=3Bvariant that retransmits a segment less than one RTT afte=
r it was<br>&gt=3B &gt=3Boriginally sent.  You reference 3517bis in the sub=
ject.  The algorithm<br>&gt=3B &gt=3Bspecified in RFC 3517 and 3517bis *doe=
s not* retransmit a segment in<br>&gt=3B &gt=3Bless than one RTT.<br>&gt=3B=
 &gt=3B<br>&gt=3B &gt=3B&gt=3B    With the changes I have been proposing ne=
arly all errors can be<br>&gt=3B &gt=3B&gt=3B    recovered in less than one=
 RTT.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3BThe only way to recover in less than=
 one RTT is to retransmit blind.<br>&gt=3B &gt=3BI.e.=2C retransmit segment=
s without any indication as to whether they have<br>&gt=3B &gt=3Barrived at=
 the receiver or not.  One could write such an algorithm.<br>&gt=3B &gt=3BA=
nd=2C probably show it fixes lost segments sooner.  But=2C the question<br>=
&gt=3B &gt=3Bwould immediately become "at what cost?".  I.e.=2C how many ne=
edless<br>&gt=3B &gt=3Bretransmits are being piled into the network to acco=
mplish this task?<br>&gt=3B &gt=3BAnd=2C the answer nearly has to be propor=
tional to the benefit you obtain.<br>&gt=3B &gt=3BI.e.=2C I bet that as you=
 speculatively retransmit more segments you will<br>&gt=3B &gt=3Balso spuri=
ous retransmit more segments.  <br>&gt=3B &gt=3B<br>&gt=3B &gt=3BAnd=2C thi=
s is needless retransmits that we know are being piled into a<br>&gt=3B &gt=
=3Bcongested network and so are using scarce network resources.  So=2C all<=
br>&gt=3B &gt=3Baround a bad idea.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3Ballman<=
br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B ______________=
_________________________________<br>&gt=3B tcpm mailing list<br>&gt=3B tcp=
m@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo/tcpm<br></div> 	=
	 	   		  </div></body>
</html>=

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Date: Sat, 11 Aug 2012 18:16:40 -0500
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Subject: Re: [tcpm] Reordering Time and RFC3517bis
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On Aug 11, 2012, at 4:23 PM, Anthony Sabatini <tsabatini@hotmail.com> wrote:=

> First let us examine fragmentation reassembly timer.  Since, as was pointe=
d out, all fragments are generated at the same time

Fragmentation of fragments can also occur, which is not synchronized.=20

> if we do not consider reordering of the fragments

We do.=20

> then the timer need be no greater than the transmission delay inherent in t=
he fragment length as it relates to link speed plus an amount for processing=
 delay, relatively speaking a very small amount of time.

See multipath routing.=20

...
> These numbers are unsupportable and make fragmentation a good attack vecto=
r because of it (interesting note, unlike the operating system people, Cisco=
 who has to deal with this in the real world sets the value to 5 seconds).

Implementations regularly trade compliance - and robustness sometimes - for c=
ost.=20

The resulting drops aren't well tracked, so we shouldn't assume shore reasse=
mbly timers are ok.=20

Joe=

From eblanton@cs.ohiou.edu  Sat Aug 11 19:02:01 2012
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Subject: Re: [tcpm] Reordering Time and RFC3517bis
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Anthony Sabatini spake unto us the following wisdom:
> After reviewing things a bit more I realize no one has given serious
> thought to Fragmentation Reassembly Timer, Reordering Time and the
> pseudo equivalent of three acks that is used to substitute for a more
> rigorous Reordering Time.

MANY people have thought about this.  This means only that you are not
familiar with the literature.

    http://scholar.google.com/scholar?q=ip+fragmentation

    http://scholar.google.com/scholar?q=tcp+reordering

The second link seems to have many more useful papers on the first
page, but certainly there is a wealth of information on both.

[snip]

> Even if we include a reroute through a satellite link that adds only
> 500 ms to the timer, there is NO scenario under which reordering will
> take over two seconds so with an abundance of caution you could set
> fragment reassembly time to four seconds and be done with it.

Queuing delay.  The rest of this paragraph is invalid, before and
after this erroneous statement.

[snip]

> Now let us look at 3517bis.  No where is there a mention of Reordering
> time yet by definition it has to be considered.  Indeed if we did not
> consider reordering time then we could simply queue the retransmission
> of a missing segment as soon as a SACK block arrives that implies the
> segment is missing.  As it is if we saved the block and then waited an
> amount of time equal to the reordering time, if we did not receive an
> ACK or SACK covering the range we could and should queue it for
> retransmission WHETHER WE RECEIVED ANY ACKs OR SACKs IN THE INTERIM.

Reordering was *carefully* considered by the community, as well as
specifically by the authors of RFC 3517 and 3517bis, myself included.
I have published several papers and RFCs specifically on TCP
reordering-related issues.  You will find them in the above Google
Scholar link.  Your overly simplistic analysis fails to take into
consideration the basic nature of packet switched networks with best
effort forwarding and forwarding queues.

> The problem is three ACKs is a very very bad way to simulate
> reordering time since on a busy link it will trigger too early casing
> duplicate transmissions and if the link goes idle it will fire way to
> slowly or not at all, even though it could correctly have been used
> for retransmission with only one occurrence.

This is also a problem that has been extensively considered in the
literature.  In fact, it is the precise reason we define the 3517
algorithms in terms of DupThresh, rather than the number 3.  There are
adaptive schemes for tuning DupThresh, as well as networks that simply
use a number larger than three.  Both 5681 and 3517 permit this,
provided that the resulting algorithm is no more aggressive than the
algorithms described in 5681.  See, for example, RFC 4653.

Three dup ACKs is certainly a compromise, but it is not "a very very
bad way to simulate reordering time" -- in fact, it is almost always
correct, and there are studies to show this.  There are certain
network pathologies on which it is reliably wrong, and there are
situations where it may degrade performance by a small constant
factor.  These are a trade-off for the heaps of situations where it
just Does the Right Thing without complicated timers and predictions.
I'm not actually aware of any algorithm that performs markedly better
in a wide range of network situations, but I would be shocked to see
one that is anywhere near as simple.

> Also for you to think about - a Reordering time of greater than
> .25*RTT degrades the throughput of the link in direct proportion to
> its value.

Unecessary retransmissions have a MUCH larger impact than this, as
each spurious retransmission cuts cwnd by 1/2.

You need to carefully revisit your premises, this thread indicates
that you do not understand the interactions of packet switched
networks and reliable transport protocols.  I would be happy to help
you understand these issues, but I will not reply to any more spurious
assertions that the well-studied approaches we are currently taking
are wrong unless they are accompanied by careful analysis and a clear
understanding of the subject matter.

Ethan

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Subject: Re: [tcpm] Reordering Time and RFC3517bis
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On Sat, Aug 11, 2012 at 2:23 PM, Anthony Sabatini <tsabatini@hotmail.com> wrote:
> All -
>
> After reviewing things a bit more I realize no one has given serious thought
> to Fragmentation Reassembly Timer, Reordering Time and the pseudo equivalent
> of three acks that is used to substitute for a more rigorous Reordering
> Time.
>
> First let us examine fragmentation reassembly timer.  Since, as was pointed
> out, all fragments are generated at the same time if we do not consider
> reordering of the fragments then the timer need be no greater than the
> transmission delay inherent in the fragment length as it relates to link
> speed plus an amount for processing delay, relatively speaking a very small
> amount of time.  Even if we include a reroute through a satellite link that
> adds only 500 ms to the timer, there is NO scenario under which reordering
> will take over two seconds so with an abundance of caution you could set
> fragment reassembly time to four seconds and be done with it.  Yet EVERY IP
> stack sets this value to above fifteen seconds (Linux is 15), some at thirty
> seconds (Microsoft is 30) and the rest at SIXTY SECONDS!! (Solaris, BSD).
> These numbers are unsupportable and make fragmentation a good attack vector
> because of it (interesting note, unlike the operating system people, Cisco
> who has to deal with this in the real world sets the value to 5 seconds).
>
> Now let us look at 3517bis.  No where is there a mention of Reordering time
> yet by definition it has to be considered.  Indeed if we did not consider
> reordering time then we could simply queue the retransmission of a missing
> segment as soon as a SACK block arrives that implies the segment is missing.
> As it is if we saved the block and then waited an amount of time equal to
> the reordering time, if we did not receive an ACK or SACK covering the range
> we could and should queue it for retransmission WHETHER WE RECEIVED ANY ACKs
> OR SACKs IN THE INTERIM.
Hi Anthony,

RFC 5827 suggests a lower dupthresh and similar reordering time logic in the
appendix. I implemented RFC 5827 with features suggested in Appendix A.1 and A.3
 in Linux 3.5. Coincidentally the imlementation waits .25 RTT to
trigger ER as well.

>
> The problem is three ACKs is a very very bad way to simulate reordering time
> since on a busy link it will trigger too early casing duplicate
> transmissions and if the link goes idle it will fire way to slowly or not at
> all, even though it could correctly have been used for retransmission with
> only one occurrence.
>
> Also for you to think about - a Reordering time of greater than .25*RTT
> degrades the throughput of the link in direct proportion to its value.
>
> .Tony
>
> Anthony Sabatini
> 200 West 20th Street
> Apt. 1216
> New York, NY 10011
>
> Phone: (212) 867-7179
> Mobile: (917) 224-8388
>
>
>
>
>> Date: Fri, 3 Aug 2012 19:37:32 -0400
>> From: tsabatini@hotmail.com
>> To: mallman@icir.org
>> CC: tcpm@ietf.org; mattmathis@google.com; touch@isi.edu
>> Subject: Re: [tcpm] Settling Time and RFC3517bis
>>
>> All -
>>
>> Because of the wording of the original comment it was not clear whether
>> the RTT referred to the generated SACK information from the receiver or the
>> original transmission from the sender. By definition, with respect to the
>> sender, it is not possible to properly retransmit the message in less than
>> one RTT plus Reordering time plus processing time.
>>
>> I am sorry for this misunderstanding.
>>
>> Tony
>>
>> Sent from my ASUS Pad
>>
>> Mark Allman <mallman@icir.org> wrote:
>>
>> >
>> >This is all pretty mis-guided. Let me say just a few quick things.
>> >
>> >> 3) STANDARD TCP can not recover in less than one RTT,
>> >
>> >Correct. Nor would we want it to. If you retransmit in less than one
>> >RTT from the time a segment is originally sourced then you will always
>> >be retransmitting blind. I.e., you'll have no indication as to whether
>> >that segment needs to be retransmitted or not.
>> >
>> >> SACK TCP as written certainly can and does currently recover in
>> >> less that one RTT on at least long links.
>> >
>> >Incorrect. At least I am aware of no TCP SACK-based loss recovery
>> >variant that retransmits a segment less than one RTT after it was
>> >originally sent. You reference 3517bis in the subject. The algorithm
>> >specified in RFC 3517 and 3517bis *does not* retransmit a segment in
>> >less than one RTT.
>> >
>> >> With the changes I have been proposing nearly all errors can be
>> >> recovered in less than one RTT.
>> >
>> >The only way to recover in less than one RTT is to retransmit blind.
>> >I.e., retransmit segments without any indication as to whether they have
>> >arrived at the receiver or not. One could write such an algorithm.
>> >And, probably show it fixes lost segments sooner. But, the question
>> >would immediately become "at what cost?". I.e., how many needless
>> >retransmits are being piled into the network to accomplish this task?
>> >And, the answer nearly has to be proportional to the benefit you obtain.
>> >I.e., I bet that as you speculatively retransmit more segments you will
>> >also spurious retransmit more segments.
>> >
>> >And, this is needless retransmits that we know are being piled into a
>> >congested network and so are using scarce network resources. So, all
>> >around a bad idea.
>> >
>> >allman
>> >
>> >
>> >
>> _______________________________________________
>> 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 mirja.kuehlewind@ikr.uni-stuttgart.de  Mon Aug 13 13:39:13 2012
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From: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
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To: Jerry Chu <hkchu@google.com>
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Hi Jerry,

I still believe restructuring the document would help to make it better=20
readable but I'm okay to keep the same structure than RFC 3390. Please chec=
k=20
if all paragraph are actually in the right section or more subsection are=20
needed to improve readability. Someone who wants to implement the change=20
might not read the whole document and might very easy overread any=20
SHOULDs/MUSTs in any later section than section 3. Maybe it is an option to=
=20
add a reference to the later SHOULDs/MUSTs  in section 2?

Regarding the comments below. You several times named a reference to=20
slides/papers/mails instead of actually addressing my comment in the draft.=
 I=20
know where to find the information as I've been following the discussion. B=
ut=20
someone who later reads the document, might not have followed mail discussi=
on=20
or ieft slides. Thus you really should add the references in the document o=
r=20
even better, explain the outcome better in the document such that the=20
references are not need anymore.

Mirja


On Wednesday 01 August 2012 07:32:31 Jerry Chu wrote:
> On Wed, Jul 25, 2012 at 2:58 PM, Jerry Chu <hkchu@google.com> wrote:
> > On Wed, Jul 25, 2012 at 11:12 AM, Mirja Kuehlewind
> >
> > <mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:
> >> Hi Jerry,
> >>
> >> just one quick comment:
> >>> Moreover, the TOC provides a very clear index of the contents so a
> >>> savvy reader can
> >>> quickly pick the sections of interesting. E.g., if you only want to
> >>> focus on the change
> >>> you'll know to read section 1 and 2 only. That's it.
> >>
> >> And that's not really true (anymore). There are several SHOULD and MUST
> >> in the other sections as well. Thus you have to read all the doc to
> >> correctly implement the changes. I know that the intent was to have
> >> everything in section 1 and 2 that is of interest for implementers but
> >> due to the many changes that have been done in previous versions, it's
> >> not the case anymore.
> >
> > Actually the only other specification related languages is section 9,
> > and it's not
> > essential as it simply reiterates what is described in rfc6298.
> >
> > Yes other SHOULD and MUST can be found in the newly added section 12
> > "Usage and Deployment Recommendations", but that section is mainly about
> > what one must do in any future large scale experiment, closely monitori=
ng
> > a number of key metrics... and not really about protocol specification.
> >
> > So to summarize, this draft is not just about protocol specification,
> > it also addresses
> > many other questions like why, and how to go about doing more
> > experiments safely.
> >
> >> That why I said, all the content is there, would be nice to
> >> clean-up/restructure a little and then it's done.
> >>
> >> It's good to have a similar structure than RFC3390 (didn't realize that
> >> you tried to have exactly the same sections here) but if the content
> >> would fit
> >
> > It was spelled out up front in 1.  Introduction
> >
> >    "This document proposes to raise the upper bound on TCP's initial
> >    window (IW) to 10 segments or roughly 15KB. It is patterned after and
> >    borrows heavily from RFC 3390 [RFC3390] and earlier work in this
> >    area."
> >
> >> better with a different structure, I'd prefer to have a good readable
> >> doc over having a doc that looks similar to RFC3390.
> >
> > Let's stick to the current format but I will respond to your one other
> > criticism on
> > section 12 first. The section is quite specific about the metrics to
> > monitor: "A key metric to monitor is the rate of packet losses, ECN
> > marking, or segment retransmissions during the initial burst." But as
> > others have pointed out it can be
> > quite difficult to try to get more specific than that, i.e., how much
> > performance
> > deteriorates before one must back off. As such all we can do for now is
> > to have enough warning signs for now but leave some details for future
> > investigation.
> >
> > Jerry
> >
> >> Mirja
> >>
> >>> >Hence I'd like to
> >>> > propose quite a little restructuring before moving the document
> >>> > forward as it is hard to easily catch the main points at the moment.
> >>> > More concrete, I'd move at least section 4, 10 and 11, maybe also 5,
> >>> > 6 and 7 (or at least partly) to the appendix. The first two paragra=
ph
> >>> > of section 7 might actually belong in
> >>>
> >>> Yes it's doable but I think the current flow is fine if you realize
> >>> the main purpose of
> >>> the document is to justify the change, not the change itself. Also TOC
> >>> will make it
> >>> much easier to navigate through.
> >>>
> >>> > the security consideration section. And section 15 (conclusion) is
> >>> > not needed at all. More detailed comments below.
> >>> >
> >>> > Mirja
> >>> >
> >>> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>> > Section 1 Introduction:
> >>> > ----------------------------
> >>> > - s/roughly 15KB/maxium 14600B/
> >>>
> >>> Ok.
> >>>
> >>> > - 2. paragraph: Don't like this two sentences at all. I hope the
> >>> > reason to increase IW is not just the fact there the queues are lar=
ge
> >>> > enough to store 10 packets. You basically saying "Let's send more
> >>> > packets at once to blow the queues!". I'd say the reason is that fl=
ow
> >>> > sizes have increased and therefore IW10 would improve latency a lot.
> >>> > Only because queues are likely to be large than 10 packets today, it
> >>> > is possible at all to send more than 10 packet at once. Btw. even if
> >>> > the queue would be small, pacing could help to still allow an IW of
> >>> > 10. Moreover, the fact that there might be several concurrent flows,
> >>> > does not mean that the buffer has to be larger (only if those flow
> >>> > start at the same time with IW10). And even if there is a large
> >>> > buffer, if there is a long-lived TCP connection, this flow will fill
> >>> > up the queue and there might be by chance still no space in the que=
ue
> >>> > for 10 additional packets.
> >>>
> >>> Guess you read the paragraph differently from what the authors had in
> >>> mind. What we have in mind was  "Ten segments are likely to fit into
> >>> queue space" because access link drains much faster these days (due to
> >>> b/w growth)...
> >>>
> >>> Will try to make it more clear.
> >>>
> >>> Will continue later,
> >>>
> >>> Jerry
> >>>
> >>> > - 3. paragraph: Don't agree. You might not know that there is a low
> >>> > speed link somewhere on the path. There are cases where the endpoint
> >>> > can protect its path by setting the receiver window but that's not
> >>> > the general case.
>
> This is exactly the point - it's not the general case but may be the
> vast majority.
> I.e., 95% of severely b/w limited hosts may be just 1 hop away or even
> directly attached to a slow link (e.g., dailup modem) hence can be
> mitigated much easily. No one suggested a solution here to solve the
> general problem of
> detecting bottleneck
> b/w. That's the job TCP's CC algorithm.
>
> >>> > - 5. paragraph: "We recommend that all TCP implementations have a
> >>> > settable TCP IW parameter..."
> >>> >   -> I'd like to see this not only in the introduction but also in
> >>> > the main section (2.) that is describing the changes and maybe even
> >>> > as a SHOULD
> >>> >
> >>> >
> >>> > Section 2 TCP Modifications
> >>> > -------------
> >>> > - Maybe have a more specific title like "Setting the IW" and some
> >>> > subsections
>
> Plagiarized from RFC3390 :)
>
> >>> > - equation 1: say somewhere that numbers are in KByte
>
> Not really.
>
> >>> > - 4. paragraph ("Note that...") does not belong in this section
>
> Why not? (It's more or less a disclaim because no one has done any study
> on non-Ethernet MTU.
>
> >>> > - maybe new subsection for paragraph 5 "Resetting the IW after SYN
> >>> > loss"; what are "multiple SYN or SYN/ACK retransmissions" -> maybe
> >>> > say a number
>
> Ok. Changed to "more than one".
>
> >>> > - last paragraph: maybe s/implementations SHOULD fall
> >>> > back/implementations MUST fall back/ ?
> >>> >
> >>> >
> >>> > Section 3 Implementation Issues
> >>> > -------------
> >>> > - Maybe move the 3 paragraph into section 2 such that the setting of
> >>> > the receive window is a MUST or SHOULD of the changes to TCP
>
> Non-essential. Remember even IW10 is just an upperbound.
>
> >>> > - 4. paragraph: Please give some explanation (or remove)
>
> This was brought up and discussed on the list long ago.
>
> >>> > Section 4 Background
> >>> > ------------
> >>> > - Move most parts in the appendix
> >>> >
> >>> > - Move 2. paragraph into Introduction
> >>> >
> >>> > - Maybe improve structure with subsections e.g. paragraph 5 and 6
> >>> > as "Recommendation for HTTP Traffic" or something
>
> See my other response. I prefer not to change the structure at this time.
>
> >>> > - 7. paragraph: "pacing will not be necessary"
> >>> >   -> I can't remember having seen any results on this. Please expla=
in
> >>> > further or remove!
>
> The statement was really "We suspect" pacing will not be necessary because
> our test results do not show any severe congestion from IW10...
>
> >>> > Section 5 Advantages of a Larger Initial Window
> >>> > -----------
> >>> > - section 5.1 Reducing Latency
> >>> >    You should also mention that for larger transmission with several
> >>> > 100s of RTT a save of 4 RTT does not really help. Thus transmissions
> >>> > which will finish within a small number of RTT will benefit the mos=
t.
>
> Isn't it obvious already?
>
> >>> > - section 5.2
> >>> >   You make some quite general statements here and then u only refer
> >>> > to google data. Maybe you can find some other references as well.
>
> It's not just google data (see all the refs.)
>
> >>> > Section 6
> >>> > ----------
> >>> > - paragraph 3 and 4 belong in an evaluation section/appendix (maybe
> >>> > own subsection on "Influence of simultaneous opened connections")
> >>> >
> >>> > Section 7
> >>> > --------
> >>> > - Move 1. paragraph to security consideration
> >>> >
> >>> > - Move rest in appendix or even parts in  the introduction
> >>> >
> >>> > Section 9
> >>> > ----------
> >>> > - move to main section which  is introducing the TCP changes as own
> >>> > subsection as a MUST is in here
>
> Non-essential because it simply restates what is said in rfc6298. It was
> added to address some concern (or confusion) that was brought up in the
> past that the initial burst of 10 pkts won't bid well for RTO hence causi=
ng
> spurious rexmits.
>
> >>> > - can you explain better why this is need or what would happen
> >>> > otherwise?
> >>> >
> >>> > Section 10
> >>> > ----------
> >>> > - Move to appendix, have more subsections and better titles for the
> >>> > subsections e.g. "Improvements in Latency", "Impact on the
> >>> > Retransmission Rate" and "Multiple TCP Connections"
> >>> >
> >>> > - The numbers for the latency improvement are only on (google) web
> >>> > search. Of course there are quite large improvements possible but t=
he
> >>> > Internet contains also other traffic. Not sure if your number are
> >>> > really that significant...?
>
> We just stated what we have.
>
> >>> > Section 11
> >>> > ---------
> >>> > - move to appendix
> >>> >
> >>> > - It's nice that you list all the studies. Could you please quickly
> >>> > summarize their results/findings, too? Because I guess that the
> >>> > intersting part to know.
>
> If you are interested please go read them yourself.
>
> >>> > Section 12
> >>> > --------
> >>> > - Its good to have this section but it's quite unspecific. E.g. you
> >>> > say "An increased initial window MUST NOT be turned on by default on
> >>> > systems without such monitoring capabilities." But what exactly
> >>> > should be monitored and how frequently and so on. Also "any
> >>> > significant deterioration" doesn't say anything to me. Is it possib=
le
> >>> > to have a concrete approach/algorithm here what to monitor when and
> >>> > what to do in which cases? And than have this even as part of the
> >>> > main section with TCP changes as this section has also some SHOULD
> >>> > and MUST and might be overread otherwise.
> >>> >
> >>> > Section 14
> >>> > ---------
> >>> > - "highly unlikely to lead to a persistent state of network
> >>> > congestion" -> Even a "highly unlikely" is concerning me in this
> >>> > case! Also there is no reasoning for this. But I guess you can even
> >>> > say, that this change cannot cause persistent network congestion as
> >>> > congestion control is still used....
>
> See 1st paragraph of section 7.
>
> >>> > Section 15 Conclusion
> >>> > --------
> >>> > - Is not needed here at all, as no new information
>
> Most rfcs have a conclusion section for those who have time only for
> the 1st and last sections.
>
> >>> > Appendix
> >>> > ---------
> >>> > - "One notable exception is YouTube because we don't think
> >>> >      the large initial window will have much material impact, either
> >>> >      positive or negative, on bulk data services."
> >>> >   -> Why is this document than not recommending to NOT use IW10 for
> >>> > bulk data services?
>
> Why should it recommend NOT? How would the TCP stack know apriori bulk
> or not? So it will require input from the apps... Is it really worth
> the complexity?
>
> >>> > - "[CW10] contains some result from a testbed study on how short
> >>> > flows with a larger initial window might affect the throughput
> >>> > performance of other co-existing, long lived, bulk data transfers."
> >>> > -> Please also describe what the results are!
>
> Read the slides please.
>
> >>> > - paragraph on "Why 10 segment?"
> >>> >   -> This is something I would prefer to have in the body of the
> >>> > document not in the appendix
> >>> >
> >>> > - "Some simulation studies [JNDK10, JNDK10-1] have been conducted to
> >>> >      study the effect of a larger initial window on wireless links
> >>> > from 2G to 4G networks (EGDE/HSPA/LTE). The overall result seems
> >>> > mixed in both raw performance and the fairness index."
> >>> >   -> Should it be recommend to not use IW10 in wireless scenarios (=
if
> >>> > known)?
>
> Studies are still on-going. I don't remember there has been a conclusive
> result.
>
> Jerry
>
> >> --
> >> -------------------------------------------------------------------
> >> Dipl.-Ing. Mirja K=FChlewind
> >> Institute of Communication Networks and Computer Engineering (IKR)
> >> University of Stuttgart, Germany
> >> Pfaffenwaldring 47, D-70569 Stuttgart
> >>
> >> tel: +49(0)711/685-67973
> >> email: mirja.kuehlewind@ikr.uni-stuttgart.de
> >> web: www.ikr.uni-stuttgart.de
> >> -------------------------------------------------------------------



=2D-=20
=2D------------------------------------------------------------------
Dipl.-Ing. Mirja K=FChlewind
Institute of Communication Networks and Computer Engineering (IKR)
University of Stuttgart, Germany
Pfaffenwaldring 47, D-70569 Stuttgart

tel: +49(0)711/685-67973
email: mirja.kuehlewind@ikr.uni-stuttgart.de
web: www.ikr.uni-stuttgart.de
=2D------------------------------------------------------------------

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Subject: Re: [tcpm] Reordering Time and RFC3517bis
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Yuchung -

Thank you for your reference to RFC5827 appendix=2C it speaks to the issue =
at hand.

My point has been if you are cleaning up RFC3517 then reference should be m=
ade to this in the "bis" draft.  The truth has always been no matter how ma=
ny ACKs you have seen=2C if a SACK hole has not been filled in Reordering T=
ime=2C then the packet can be safely assumed lost.  Obviously this makes yo=
u more vulnerable to lost ACKs but the only way to truly fix that would be =
to adopt my token strategy.  This is why I proposed sending another ACK at =
.25*RTT if the link goes idle.  This fixes both the Last ACK lost problem a=
nd kills a duplicate retransmit if that ACK filled a hole (obviously we wou=
ld adjust the two times to not collide=2C probably setting the safety ACK t=
o .20*RTT and the delay before considering a hole valid to .30*RTT for exam=
ple).

Also as a personal comment 3517bis has a "first read" problem.  You know wh=
at you mean but a casual reader does not.  You need to be careful to spell =
things out and not use words and expressions common to the working group bu=
t which may be misunderstood by people lacking that context.

Now with respect to those who say I know nothing and have read nothing - le=
t us get back to the whole theoretical underpinning of the "three ACK" rule=
=2C Vern Paxson's 1999 paper "End-to-End Internet Packet Dynamics"=2C which=
 discusses in great detail that at the time the rule is a great substitute =
for a fixed delay of between 20 ms and 85 ms.  He noted rater pointedly tha=
t those numbers were selected based on current link speeds of 56 kbps and 1=
=2C544 kbps (T1).  The reason for avoiding short fixed delays in TCP were o=
utlined in RFC791=2C the fact that timers of that era were based on 60 cycl=
e power line frequency or 33 ms minimum resolution (two ticks since that co=
uld be anywhere from 17 to 33 ms).  Obviously all of these assumptions are =
invalid today when average high speed internet is 4 mbps and we are progres=
sing toward 10 mbps and microsecond timing services are available and wide =
spread.  As pointed out to me rather forcefully queueing delays are a major=
 issue=2C which was rather unneccessary as I have been dealing with the iss=
ue for the last 40 years=2C so "three ACKs" is no longer a very useful meas=
ure in the real world.  I would=2C however=2C point out that there is no ro=
uter in common use that has enough memory to buffer more than 5 seconds wor=
th of high speed traffic=2C nor would it want to as the information then de=
livered would be highly irrelevant.=20

Tony

Anthony Sabatini
200 West 20th Street
Apt. 1216
New York=2C NY 10011

Phone: (212) 867-7179
Mobile: (917) 224-8388

=20


> From: ycheng@google.com
> Date: Sun=2C 12 Aug 2012 10:33:22 -0700
> Subject: Re: [tcpm] Reordering Time and RFC3517bis
> To: tsabatini@hotmail.com
> CC: tcpm@ietf.org
>=20
> On Sat=2C Aug 11=2C 2012 at 2:23 PM=2C Anthony Sabatini <tsabatini@hotmai=
l.com> wrote:
> > All -
> >
> > After reviewing things a bit more I realize no one has given serious th=
ought
> > to Fragmentation Reassembly Timer=2C Reordering Time and the pseudo equ=
ivalent
> > of three acks that is used to substitute for a more rigorous Reordering
> > Time.
> >
> > First let us examine fragmentation reassembly timer.  Since=2C as was p=
ointed
> > out=2C all fragments are generated at the same time if we do not consid=
er
> > reordering of the fragments then the timer need be no greater than the
> > transmission delay inherent in the fragment length as it relates to lin=
k
> > speed plus an amount for processing delay=2C relatively speaking a very=
 small
> > amount of time.  Even if we include a reroute through a satellite link =
that
> > adds only 500 ms to the timer=2C there is NO scenario under which reord=
ering
> > will take over two seconds so with an abundance of caution you could se=
t
> > fragment reassembly time to four seconds and be done with it.  Yet EVER=
Y IP
> > stack sets this value to above fifteen seconds (Linux is 15)=2C some at=
 thirty
> > seconds (Microsoft is 30) and the rest at SIXTY SECONDS!! (Solaris=2C B=
SD).
> > These numbers are unsupportable and make fragmentation a good attack ve=
ctor
> > because of it (interesting note=2C unlike the operating system people=
=2C Cisco
> > who has to deal with this in the real world sets the value to 5 seconds=
).
> >
> > Now let us look at 3517bis.  No where is there a mention of Reordering =
time
> > yet by definition it has to be considered.  Indeed if we did not consid=
er
> > reordering time then we could simply queue the retransmission of a miss=
ing
> > segment as soon as a SACK block arrives that implies the segment is mis=
sing.
> > As it is if we saved the block and then waited an amount of time equal =
to
> > the reordering time=2C if we did not receive an ACK or SACK covering th=
e range
> > we could and should queue it for retransmission WHETHER WE RECEIVED ANY=
 ACKs
> > OR SACKs IN THE INTERIM.
> Hi Anthony=2C
>=20
> RFC 5827 suggests a lower dupthresh and similar reordering time logic in =
the
> appendix. I implemented RFC 5827 with features suggested in Appendix A.1 =
and A.3
>  in Linux 3.5. Coincidentally the imlementation waits .25 RTT to
> trigger ER as well.
>=20
> >
> > The problem is three ACKs is a very very bad way to simulate reordering=
 time
> > since on a busy link it will trigger too early casing duplicate
> > transmissions and if the link goes idle it will fire way to slowly or n=
ot at
> > all=2C even though it could correctly have been used for retransmission=
 with
> > only one occurrence.
> >
> > Also for you to think about - a Reordering time of greater than .25*RTT
> > degrades the throughput of the link in direct proportion to its value.
> >
> > .Tony
> >
> > Anthony Sabatini
> > 200 West 20th Street
> > Apt. 1216
> > New York=2C NY 10011
> >
> > Phone: (212) 867-7179
> > Mobile: (917) 224-8388
> >
> >
> >
> >
> >> Date: Fri=2C 3 Aug 2012 19:37:32 -0400
> >> From: tsabatini@hotmail.com
> >> To: mallman@icir.org
> >> CC: tcpm@ietf.org=3B mattmathis@google.com=3B touch@isi.edu
> >> Subject: Re: [tcpm] Settling Time and RFC3517bis
> >>
> >> All -
> >>
> >> Because of the wording of the original comment it was not clear whethe=
r
> >> the RTT referred to the generated SACK information from the receiver o=
r the
> >> original transmission from the sender. By definition=2C with respect t=
o the
> >> sender=2C it is not possible to properly retransmit the message in les=
s than
> >> one RTT plus Reordering time plus processing time.
> >>
> >> I am sorry for this misunderstanding.
> >>
> >> Tony
> >>
> >> Sent from my ASUS Pad
> >>
> >> Mark Allman <mallman@icir.org> wrote:
> >>
> >> >
> >> >This is all pretty mis-guided. Let me say just a few quick things.
> >> >
> >> >> 3) STANDARD TCP can not recover in less than one RTT=2C
> >> >
> >> >Correct. Nor would we want it to. If you retransmit in less than one
> >> >RTT from the time a segment is originally sourced then you will alway=
s
> >> >be retransmitting blind. I.e.=2C you'll have no indication as to whet=
her
> >> >that segment needs to be retransmitted or not.
> >> >
> >> >> SACK TCP as written certainly can and does currently recover in
> >> >> less that one RTT on at least long links.
> >> >
> >> >Incorrect. At least I am aware of no TCP SACK-based loss recovery
> >> >variant that retransmits a segment less than one RTT after it was
> >> >originally sent. You reference 3517bis in the subject. The algorithm
> >> >specified in RFC 3517 and 3517bis *does not* retransmit a segment in
> >> >less than one RTT.
> >> >
> >> >> With the changes I have been proposing nearly all errors can be
> >> >> recovered in less than one RTT.
> >> >
> >> >The only way to recover in less than one RTT is to retransmit blind.
> >> >I.e.=2C retransmit segments without any indication as to whether they=
 have
> >> >arrived at the receiver or not. One could write such an algorithm.
> >> >And=2C probably show it fixes lost segments sooner. But=2C the questi=
on
> >> >would immediately become "at what cost?". I.e.=2C how many needless
> >> >retransmits are being piled into the network to accomplish this task?
> >> >And=2C the answer nearly has to be proportional to the benefit you ob=
tain.
> >> >I.e.=2C I bet that as you speculatively retransmit more segments you =
will
> >> >also spurious retransmit more segments.
> >> >
> >> >And=2C this is needless retransmits that we know are being piled into=
 a
> >> >congested network and so are using scarce network resources. So=2C al=
l
> >> >around a bad idea.
> >> >
> >> >allman
> >> >
> >> >
> >> >
> >> _______________________________________________
> >> 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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<body class=3D'hmmessage'><div dir=3D'ltr'>Yuchung -<br><br>Thank you for y=
our reference to RFC5827 appendix=2C it speaks to the issue at hand.<br><br=
>My point has been if you are cleaning up RFC3517 then reference should be =
made to this in the "bis" draft.&nbsp=3B The truth has always been no matte=
r how many ACKs you have seen=2C if a SACK hole has not been filled in Reor=
dering Time=2C then the packet can be safely assumed lost.&nbsp=3B Obviousl=
y this makes you more vulnerable to lost ACKs but the only way to truly fix=
 that would be to adopt my token strategy.&nbsp=3B This is why I proposed s=
ending another ACK at .25*RTT if the link goes idle.&nbsp=3B This fixes bot=
h the Last ACK lost problem and kills a duplicate retransmit if that ACK fi=
lled a hole (obviously we would adjust the two times to not collide=2C prob=
ably setting the safety ACK to .20*RTT and the delay before considering a h=
ole valid to .30*RTT for example).<br><br>Also as a personal comment 3517bi=
s has a "first read" problem.&nbsp=3B You know what you mean but a casual r=
eader does not.&nbsp=3B You need to be careful to spell things out and not =
use words and expressions common to the working group but which may be misu=
nderstood by people lacking that context.<br><br>Now with respect to those =
who say I know nothing and have read nothing - let us get back to the whole=
 theoretical underpinning of the "three ACK" rule=2C Vern Paxson's 1999 pap=
er "End-to-End Internet Packet Dynamics"=2C which discusses in great detail=
 that at the time the rule is a great substitute for a fixed delay of betwe=
en 20 ms and 85 ms.&nbsp=3B He noted rater pointedly that those numbers wer=
e selected based on current link speeds of 56 kbps and 1=2C544 kbps (T1).&n=
bsp=3B The reason for avoiding short fixed delays in TCP were outlined in R=
FC791=2C the fact that timers of that era were based on 60 cycle power line=
 frequency or 33 ms minimum resolution (two ticks since that could be anywh=
ere from 17 to 33 ms).&nbsp=3B Obviously all of these assumptions are inval=
id today when average high speed internet is 4 mbps and we are progressing =
toward 10 mbps and microsecond timing services are available and wide sprea=
d.&nbsp=3B As pointed out to me rather forcefully queueing delays are a maj=
or issue=2C which was rather unneccessary as I have been dealing with the i=
ssue for the last 40 years=2C so "three ACKs" is no longer a very useful me=
asure in the real world.&nbsp=3B I would=2C however=2C point out that there=
 is no router in common use that has enough memory to buffer more than 5 se=
conds worth of high speed traffic=2C nor would it want to as the informatio=
n then delivered would be highly irrelevant. <br><br>Tony<br><br>Anthony Sa=
batini<br>200&nbsp=3BWest 20th Street<br>Apt. 1216<br>New York=2C NY 10011<=
br><br>Phone: (212) 867-7179<br>Mobile: (917) 224-8388<br><br>&nbsp=3B<br><=
br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B From: ycheng@googl=
e.com<br>&gt=3B Date: Sun=2C 12 Aug 2012 10:33:22 -0700<br>&gt=3B Subject: =
Re: [tcpm] Reordering Time and RFC3517bis<br>&gt=3B To: tsabatini@hotmail.c=
om<br>&gt=3B CC: tcpm@ietf.org<br>&gt=3B <br>&gt=3B On Sat=2C Aug 11=2C 201=
2 at 2:23 PM=2C Anthony Sabatini &lt=3Btsabatini@hotmail.com&gt=3B wrote:<b=
r>&gt=3B &gt=3B All -<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B After reviewing thi=
ngs a bit more I realize no one has given serious thought<br>&gt=3B &gt=3B =
to Fragmentation Reassembly Timer=2C Reordering Time and the pseudo equival=
ent<br>&gt=3B &gt=3B of three acks that is used to substitute for a more ri=
gorous Reordering<br>&gt=3B &gt=3B Time.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B =
First let us examine fragmentation reassembly timer.  Since=2C as was point=
ed<br>&gt=3B &gt=3B out=2C all fragments are generated at the same time if =
we do not consider<br>&gt=3B &gt=3B reordering of the fragments then the ti=
mer need be no greater than the<br>&gt=3B &gt=3B transmission delay inheren=
t in the fragment length as it relates to link<br>&gt=3B &gt=3B speed plus =
an amount for processing delay=2C relatively speaking a very small<br>&gt=
=3B &gt=3B amount of time.  Even if we include a reroute through a satellit=
e link that<br>&gt=3B &gt=3B adds only 500 ms to the timer=2C there is NO s=
cenario under which reordering<br>&gt=3B &gt=3B will take over two seconds =
so with an abundance of caution you could set<br>&gt=3B &gt=3B fragment rea=
ssembly time to four seconds and be done with it.  Yet EVERY IP<br>&gt=3B &=
gt=3B stack sets this value to above fifteen seconds (Linux is 15)=2C some =
at thirty<br>&gt=3B &gt=3B seconds (Microsoft is 30) and the rest at SIXTY =
SECONDS!! (Solaris=2C BSD).<br>&gt=3B &gt=3B These numbers are unsupportabl=
e and make fragmentation a good attack vector<br>&gt=3B &gt=3B because of i=
t (interesting note=2C unlike the operating system people=2C Cisco<br>&gt=
=3B &gt=3B who has to deal with this in the real world sets the value to 5 =
seconds).<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B Now let us look at 3517bis.  No=
 where is there a mention of Reordering time<br>&gt=3B &gt=3B yet by defini=
tion it has to be considered.  Indeed if we did not consider<br>&gt=3B &gt=
=3B reordering time then we could simply queue the retransmission of a miss=
ing<br>&gt=3B &gt=3B segment as soon as a SACK block arrives that implies t=
he segment is missing.<br>&gt=3B &gt=3B As it is if we saved the block and =
then waited an amount of time equal to<br>&gt=3B &gt=3B the reordering time=
=2C if we did not receive an ACK or SACK covering the range<br>&gt=3B &gt=
=3B we could and should queue it for retransmission WHETHER WE RECEIVED ANY=
 ACKs<br>&gt=3B &gt=3B OR SACKs IN THE INTERIM.<br>&gt=3B Hi Anthony=2C<br>=
&gt=3B <br>&gt=3B RFC 5827 suggests a lower dupthresh and similar reorderin=
g time logic in the<br>&gt=3B appendix. I implemented RFC 5827 with feature=
s suggested in Appendix A.1 and A.3<br>&gt=3B  in Linux 3.5. Coincidentally=
 the imlementation waits .25 RTT to<br>&gt=3B trigger ER as well.<br>&gt=3B=
 <br>&gt=3B &gt=3B<br>&gt=3B &gt=3B The problem is three ACKs is a very ver=
y bad way to simulate reordering time<br>&gt=3B &gt=3B since on a busy link=
 it will trigger too early casing duplicate<br>&gt=3B &gt=3B transmissions =
and if the link goes idle it will fire way to slowly or not at<br>&gt=3B &g=
t=3B all=2C even though it could correctly have been used for retransmissio=
n with<br>&gt=3B &gt=3B only one occurrence.<br>&gt=3B &gt=3B<br>&gt=3B &gt=
=3B Also for you to think about - a Reordering time of greater than .25*RTT=
<br>&gt=3B &gt=3B degrades the throughput of the link in direct proportion =
to its value.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B .Tony<br>&gt=3B &gt=3B<br>&=
gt=3B &gt=3B Anthony Sabatini<br>&gt=3B &gt=3B 200 West 20th Street<br>&gt=
=3B &gt=3B Apt. 1216<br>&gt=3B &gt=3B New York=2C NY 10011<br>&gt=3B &gt=3B=
<br>&gt=3B &gt=3B Phone: (212) 867-7179<br>&gt=3B &gt=3B Mobile: (917) 224-=
8388<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br=
>&gt=3B &gt=3B&gt=3B Date: Fri=2C 3 Aug 2012 19:37:32 -0400<br>&gt=3B &gt=
=3B&gt=3B From: tsabatini@hotmail.com<br>&gt=3B &gt=3B&gt=3B To: mallman@ic=
ir.org<br>&gt=3B &gt=3B&gt=3B CC: tcpm@ietf.org=3B mattmathis@google.com=3B=
 touch@isi.edu<br>&gt=3B &gt=3B&gt=3B Subject: Re: [tcpm] Settling Time and=
 RFC3517bis<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B All -<br>&gt=3B &=
gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B Because of the wording of the original c=
omment it was not clear whether<br>&gt=3B &gt=3B&gt=3B the RTT referred to =
the generated SACK information from the receiver or the<br>&gt=3B &gt=3B&gt=
=3B original transmission from the sender. By definition=2C with respect to=
 the<br>&gt=3B &gt=3B&gt=3B sender=2C it is not possible to properly retran=
smit the message in less than<br>&gt=3B &gt=3B&gt=3B one RTT plus Reorderin=
g time plus processing time.<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B =
I am sorry for this misunderstanding.<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=
=3B&gt=3B Tony<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B Sent from my A=
SUS Pad<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B Mark Allman &lt=3Bmal=
lman@icir.org&gt=3B wrote:<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B &g=
t=3B<br>&gt=3B &gt=3B&gt=3B &gt=3BThis is all pretty mis-guided. Let me say=
 just a few quick things.<br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=
=3B &gt=3B&gt=3B 3) STANDARD TCP can not recover in less than one RTT=2C<br=
>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B &gt=3BCorrect. Nor would=
 we want it to. If you retransmit in less than one<br>&gt=3B &gt=3B&gt=3B &=
gt=3BRTT from the time a segment is originally sourced then you will always=
<br>&gt=3B &gt=3B&gt=3B &gt=3Bbe retransmitting blind. I.e.=2C you'll have =
no indication as to whether<br>&gt=3B &gt=3B&gt=3B &gt=3Bthat segment needs=
 to be retransmitted or not.<br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B=
&gt=3B &gt=3B&gt=3B SACK TCP as written certainly can and does currently re=
cover in<br>&gt=3B &gt=3B&gt=3B &gt=3B&gt=3B less that one RTT on at least =
long links.<br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B &gt=3BInco=
rrect. At least I am aware of no TCP SACK-based loss recovery<br>&gt=3B &gt=
=3B&gt=3B &gt=3Bvariant that retransmits a segment less than one RTT after =
it was<br>&gt=3B &gt=3B&gt=3B &gt=3Boriginally sent. You reference 3517bis =
in the subject. The algorithm<br>&gt=3B &gt=3B&gt=3B &gt=3Bspecified in RFC=
 3517 and 3517bis *does not* retransmit a segment in<br>&gt=3B &gt=3B&gt=3B=
 &gt=3Bless than one RTT.<br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=
=3B &gt=3B&gt=3B With the changes I have been proposing nearly all errors c=
an be<br>&gt=3B &gt=3B&gt=3B &gt=3B&gt=3B recovered in less than one RTT.<b=
r>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B &gt=3BThe only way to r=
ecover in less than one RTT is to retransmit blind.<br>&gt=3B &gt=3B&gt=3B =
&gt=3BI.e.=2C retransmit segments without any indication as to whether they=
 have<br>&gt=3B &gt=3B&gt=3B &gt=3Barrived at the receiver or not. One coul=
d write such an algorithm.<br>&gt=3B &gt=3B&gt=3B &gt=3BAnd=2C probably sho=
w it fixes lost segments sooner. But=2C the question<br>&gt=3B &gt=3B&gt=3B=
 &gt=3Bwould immediately become "at what cost?". I.e.=2C how many needless<=
br>&gt=3B &gt=3B&gt=3B &gt=3Bretransmits are being piled into the network t=
o accomplish this task?<br>&gt=3B &gt=3B&gt=3B &gt=3BAnd=2C the answer near=
ly has to be proportional to the benefit you obtain.<br>&gt=3B &gt=3B&gt=3B=
 &gt=3BI.e.=2C I bet that as you speculatively retransmit more segments you=
 will<br>&gt=3B &gt=3B&gt=3B &gt=3Balso spurious retransmit more segments.<=
br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B &gt=3BAnd=2C this is n=
eedless retransmits that we know are being piled into a<br>&gt=3B &gt=3B&gt=
=3B &gt=3Bcongested network and so are using scarce network resources. So=
=2C all<br>&gt=3B &gt=3B&gt=3B &gt=3Baround a bad idea.<br>&gt=3B &gt=3B&gt=
=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B &gt=3Ballman<br>&gt=3B &gt=3B&gt=3B &gt=
=3B<br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &gt=3B&gt=3B &gt=3B<br>&gt=3B &=
gt=3B&gt=3B _______________________________________________<br>&gt=3B &gt=
=3B&gt=3B tcpm mailing list<br>&gt=3B &gt=3B&gt=3B tcpm@ietf.org<br>&gt=3B =
&gt=3B&gt=3B https://www.ietf.org/mailman/listinfo/tcpm<br>&gt=3B &gt=3B<br=
>&gt=3B &gt=3B _______________________________________________<br>&gt=3B &g=
t=3B tcpm mailing list<br>&gt=3B &gt=3B tcpm@ietf.org<br>&gt=3B &gt=3B http=
s://www.ietf.org/mailman/listinfo/tcpm<br>&gt=3B &gt=3B<br></div> 		 	   		=
  </div></body>
</html>=

--_8f90a6fa-7129-4673-bc2f-a70b5bb2f78b_--

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Bottom line ...

> so "three ACKs" is no longer a very useful measure in the real world.

Show us.  In other words, make your case empirically.  I have looked at
(ahem) a few traces in my life and read (ahem) a few papers on the
subject and it seems to me the empirical evidence is that three dupacks
is actually pretty useful.  It provides for reasonably timely
retransmission and it is generally able to disambiguate loss from
reordering.  If you have some empirical observation to the
contrary---rather than hand waves---I think we'd all like to see it.

There are known issues with three dupacks in that there are cases when
you don't or can't get three dupacks back.  We have dealt with that in
3517bis, in RFC 3042, in RFC 5827 and the current rtorestart is also
aiming to help that situation.  So, we know it isn't perfect.

So, write down the problem you see, this real world evidence of the
problem and your proposed solution.

allman

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----------ma46947-1
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> As discussed in the meeting last week, we will now start working group
> last call for "Proportional Rate Reduction for TCP"
> (draft-ietf-tcpm-proportional-rate-reduction-02), to be published as
> Experimental RFC. The WGLC runs until Friday, August 24th. Please read
> the draft and send comments by then. If you have read the draft and
> think it is ok, notifying just that is also helpful (even without
> comments). 

I have reviewed the draft.  I have some comments.

  - It seems strange that the intro does not offer a sketch and critique
    of 3517 given that 5681, rate halving and PRR are all sketched and
    the natural question of "well, how does 3517 related?" comes to
    mind.

  - And, this is even more problematic when you start making comparisons
    to 3517 in the 5th paragraph of the intro.

  - In general, I think you could intuitively frame these algorithms
    better in non-normative words.  There is clear similarities and
    differences in the strategies that even if there are muddled (e.g.,
    how many segments get sent in the RTT following recovery, or whether
    those segments are spread out over the RTT or bunched up together,
    or how fast segments are sent when they are sent (as distinct from
    how many are sent)).

  - It'd be more useful to cite Van's SIGCOMM paper than to cite Van.

  - It struck me as odd that you noted 3517 as non-conservative.  I
    think that is wrong.  But, after reading the entire document I
    believe I understand what you mean.  

    And, in fairness I must have been reading the intro too quickly
    because you in fact define it there, but I missed the subtle shit
    you have introduced.  In particular, you say:

      [...] packet conservation principle: segments delivered to the
      receiver are used to as the clock to trigger sending the same
      number of segments back into the network.

    However, the SIGCOMM 88 paper is what is in my head:

      By `conservation of packets' we mean that for a connection `in
      equilibrium', i.e., running stably with a full window of data in
      transit, the packet flow is what a physicist would call
      `conservative': A new packet isn't put into the network until an
      old packet leaves.

    Being "delivered to the receiver" is only one way a packet can leave
    the network.  RFC 3517 is packet conserving assuming (a) one admits
    there is another way packets can leave the network and (b) its
    judgment of losses is reasonable.  This latter *can* be unreasonable
    if the path has a high degree of reordering.  In this case the
    estimate produced by IsLost() can be wrong and RFC 3517 will in fact
    add packets to the network before previous versions have left the
    network.  

    That said, while I know of empirical measurements that suggest one
    can find such paths, I know of none that shows this sort of
    reordering to be common.  So, I think the assumption holds.  And, I
    think RFC 3517 is packet conserving.

    I think at the very least you should be very clear in calling out
    your definition of "packet conservation" in the intro---and note
    that it is not the only interpretation.  The document is a lot
    easier to understand if one has this bit in mind.  Otherwise, it
    comes off as confused.

  - In definitions you should remind us of the snd.una definition since
    you use it in many of your definitions.

  - In 3.1 you should cite Limited Transmit.

  - There is also a 3517bis which should pop out any day now as RFC
    6675.  You should at least note this.

    And, because it is a little different than 3517 in your examples
    because loss recovery can now be triggered on the first dupack if
    there is enough SACKed data in the incoming ACK to indicate that
    three segments have left the network (either lost or delivered).
    So, e.g., in your second example the ACK of 15 would trigger the 7R
    burst.

    I don't think you need to work things in terms of RFC 6675 because
    your measurements are in terms of RFC 3517.  But, at least a note to
    help folks follow the relationship seems worthwhile.

  - On page 8 you note that your *measurements* suggest something.  But,
    later on it seems clear that the measurements show something.  I
    suggest cleaning this up.

  - I find this business about pipe being a lousy estimate on page 10 to
    be pretty dubious.

    I understand your point that errors in the estimate are spread
    across the RTT.  That is even nice.

    But, there is no evidence (provided; and I am aware of none) that
    pipe is in general a bad estimate.  And, even if there is some
    reordering issues we have defined a way to get around that (RFC
    4653) (which probably boils down to something like rate halving in
    one of the two variants).  So, at least some mention of this would
    seem germane and I'd suggest just toning down the whole thing a bit
    given that there is no reason to think in most cases pipe is busted.

    (Or, point to evidence that says pipe is busted.  But, if it is then
    we should also be thinking of adjusting the dupack threshold.)

  - In the second paragraph of section 5 you mention 'production use'.
    I get it after reading a couple of paragraphs ahead.  You might move
    the notion that you ran all this in Google production servers to the
    front of the section.

  - You note that for 3517 that in 32% of the cases pipe is less than
    ssthresh when entering loss recovery.  I assume this means the
    ssthresh that is set upon entering, right?  You might make this more
    clear.

So, in general I think this document is fine.  It needs cleaned up,
IMO.  But, the mechanism is nice.

I do wonder why it is going experimental.  That seems too cautious to
me.  It seems these things are no more aggressive than 3517 and that is
standards track and so why should this spin at experimental?  What is
the experiment here?  What are we concerned about?  I say let it go as
standards track.

allman




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From wwwrun@rfc-editor.org  Fri Aug 24 16:33:11 2012
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Subject: [tcpm] RFC 6675 on A Conservative Loss Recovery Algorithm Based on Selective Acknowledgment (SACK) for TCP
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6675

        Title:      A Conservative Loss Recovery Algorithm 
                    Based on Selective Acknowledgment (SACK) for 
                    TCP 
        Author:     E. Blanton, M. Allman,
                    L. Wang, I. Jarvinen,
                    M. Kojo, Y. Nishida
        Status:     Standards Track
        Stream:     IETF
        Date:       August 2012
        Mailbox:    elb@psg.com, 
                    mallman@icir.org, 
                    liliw@juniper.net,
                    ilpo.jarvinen@helsinki.fi, 
                    kojo@cs.helsinki.fi,
                    nishida@wide.ad.jp
        Pages:      15
        Characters: 34484
        Obsoletes:  RFC3517

        I-D Tag:    draft-ietf-tcpm-3517bis-02.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6675.txt

This document presents a conservative loss recovery algorithm for TCP
that is based on the use of the selective acknowledgment (SACK) TCP 
option.  The algorithm presented in this document conforms to the spirit 
of the current congestion control specification (RFC 5681), but allows TCP 
senders to recover more effectively when multiple segments are lost from a 
single flight of data.  This document obsoletes RFC 3517 and describes 
changes from it.  [STANDARDS-TRACK]

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

This is now a Proposed Standard Protocol.

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

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Hi Mark,

On Fri, Aug 24, 2012 at 10:18 AM, Mark Allman <mallman@icir.org> wrote:

>
> I do wonder why it is going experimental.  That seems too cautious to
> me.  It seems these things are no more aggressive than 3517 and that is
> standards track and so why should this spin at experimental?  What is
> the experiment here?  What are we concerned about?  I say let it go as
> standards track.
>
>
I agree with the logic appears to be conservative enough, but I'm not very
sure if it's a enough reason to publish a tcp's CC logic as a standard.
I'm wondering how to justify this.

Thanks,
--
Yoshifumi

--f46d0408913109f03e04c81a142b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Mark,<br><br><div class=3D"gmail_quote">On Fri, Aug 24, 2012 at 10:18 AM=
, Mark Allman <span dir=3D"ltr">&lt;<a href=3D"mailto:mallman@icir.org" tar=
get=3D"_blank">mallman@icir.org</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">

<br>
I do wonder why it is going experimental. =A0That seems too cautious to<br>
me. =A0It seems these things are no more aggressive than 3517 and that is<b=
r>
standards track and so why should this spin at experimental? =A0What is<br>
the experiment here? =A0What are we concerned about? =A0I say let it go as<=
br>
standards track.<br>
<span></span><br></blockquote><div><br>I agree with the logic appears to be=
 conservative enough, but I&#39;m not very sure if it&#39;s a enough reason=
 to publish a tcp&#39;s CC logic as a standard.<br>I&#39;m wondering how to=
 justify this.<br>
<br>
Thanks,<br>--<br>Yoshifumi<br><br></div></div>

--f46d0408913109f03e04c81a142b--

From eblanton@cs.ohiou.edu  Sat Aug 25 15:26:31 2012
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From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Yoshifumi Nishida spake unto us the following wisdom:
> > I do wonder why it is going experimental.  That seems too cautious to
> > me.  It seems these things are no more aggressive than 3517 and that is
> > standards track and so why should this spin at experimental?  What is
> > the experiment here?  What are we concerned about?  I say let it go as
> > standards track.
>
> I agree with the logic appears to be conservative enough, but I'm not very
> sure if it's a enough reason to publish a tcp's CC logic as a standard.
> I'm wondering how to justify this.

As I read 5681, any scheme which is at least as conservative as 5681
is implicitly allowable.  I have no problem, in general, with allowing
any behavior at least as conservative as a standards track algorithm
to go standards track; this suggests that any penalty paid for using
the algorithm will be paid by the host using the algorithm, which
seems OK to me.

As Mark asks, what is the experiment?  If WG consensus is that an
algorithm is OK out in the wild, standards track is the right target.

(Disclaimer: I haven't yet read the draft.)

Ethan

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From pasi.sarolahti@iki.fi  Sun Aug 26 13:02:36 2012
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Hi Mark,

Thanks for the detailed review.

On Aug 24, 2012, at 8:18 PM, Mark Allman wrote:

> I do wonder why it is going experimental.  That seems too cautious to
> me.  It seems these things are no more aggressive than 3517 and that =
is
> standards track and so why should this spin at experimental?  What is
> the experiment here?  What are we concerned about?  I say let it go as
> standards track.

Fair questions. I checked the minutes from IETF-81, where this was =
adopted specifically as experimental according to the minutes. I wasn't =
there at the time, so I don't know if the PS vs. Exp issue was discussed =
at all, but no one seems to have questioned the experimental status =
until now.

I wonder about the process issues related to changing the status at this =
point. People who have read the draft earlier may have done so with the =
mindset of evaluating an experimental document. If we wanted to change =
the intended status now, we might need a separate consensus call to =
verify that the rough consensus supports it.

- Pasi


From michawe@ifi.uio.no  Sun Aug 26 13:22:33 2012
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I just had to share this, it's cool:
http://engineering.stanford.edu/news/stanford-biologist-computer-scientist-discover-anternet

Cheers
Michael


From michael.scharf@alcatel-lucent.com  Sun Aug 26 13:32:10 2012
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> > I do wonder why it is going experimental.  That seems too=20
> cautious to=20
> > me.  It seems these things are no more aggressive than 3517=20
> and that=20
> > is standards track and so why should this spin at=20
> experimental?  What=20
> > is the experiment here?  What are we concerned about?  I=20
> say let it go=20
> > as standards track.
>=20
> Fair questions. I checked the minutes from IETF-81, where=20
> this was adopted specifically as experimental according to=20
> the minutes. I wasn't there at the time, so I don't know if=20
> the PS vs. Exp issue was discussed at all, but no one seems=20
> to have questioned the experimental status until now.

I don't recall any discussion on a status other than experimental.
=20
> I wonder about the process issues related to changing the=20
> status at this point. People who have read the draft earlier=20
> may have done so with the mindset of evaluating an=20
> experimental document. If we wanted to change the intended=20
> status now, we might need a separate consensus call to verify=20
> that the rough consensus supports it.

Yes, I think that this would require more feedback from the community.

Michael=

From michawe@ifi.uio.no  Sun Aug 26 13:49:58 2012
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Subject: Re: [tcpm] WGLC for Proportional Rate Reduction
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On Aug 26, 2012, at 10:31 PM, Scharf, Michael (Michael) wrote:

>>> I do wonder why it is going experimental.  That seems too
>> cautious to
>>> me.  It seems these things are no more aggressive than 3517
>> and that
>>> is standards track and so why should this spin at
>> experimental?  What
>>> is the experiment here?  What are we concerned about?  I
>> say let it go
>>> as standards track.
>>
>> Fair questions. I checked the minutes from IETF-81, where
>> this was adopted specifically as experimental according to
>> the minutes. I wasn't there at the time, so I don't know if
>> the PS vs. Exp issue was discussed at all, but no one seems
>> to have questioned the experimental status until now.
>
> I don't recall any discussion on a status other than experimental.
>
>> I wonder about the process issues related to changing the
>> status at this point. People who have read the draft earlier
>> may have done so with the mindset of evaluating an
>> experimental document. If we wanted to change the intended
>> status now, we might need a separate consensus call to verify
>> that the rough consensus supports it.
>
> Yes, I think that this would require more feedback from the community.

I read it, with experimental in mind, but I'd support seeing this on  
standards track. I agree with Mark's points.

Cheers,
Michael


From mallman@icir.org  Mon Aug 27 07:42:06 2012
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> > I do wonder why it is going experimental.  That seems too cautious
> > to me.  It seems these things are no more aggressive than 3517 and
> > that is standards track and so why should this spin at experimental?
> > What is the experiment here?  What are we concerned about?  I say
> > let it go as standards track.
>
> I agree with the logic appears to be conservative enough, but I'm not
> very sure if it's a enough reason to publish a tcp's CC logic as a
> standard.  I'm wondering how to justify this.

It sounds to me like this path to experimental was chosen by default.
I.e., without thinking about status very hard we can always start at
experimental.  So, lets think about it for a few minutes.

I can argue both ways on technical grounds.

On the one hand, PRR does not set the amount of transmission rate
reduction, it merely sets the sending pattern based on an externally
provided reduction.  So, if the algorithm is fed a reduction of 50% (a
la current standard TCP) then it will send at most 50% of the previous
flight size in the RTT following loss detection.  This is absolutely
consistent with the spirit of TCP congestion control (RFCs 2914, 5681,
3517, 6675 (== 3517bis)).  This reading of things says that even if we
call it 'experimental' it is consistent with the "or more conservative"
notions allowed by the various RFCs and so PRR is implicitly allowed as
standards track regardless of what label you put on it.  So, we might as
well label it accurately.

On the other hand, one could argue that the *pattern* of sending is
different under PRR and hence this makes it different enough as to not
be covered under the argument above.  PRR is at once more conservative
and more aggressive than current standards track schemes (a point which
could be better made in the PRR document, IMHO).  Consider the first
example in the draft whereby all schemes send the same number of
packets.  PRR is more conservative in the rate it sends those packets
into the network.  The rate is roughly half that before loss detection.
However, PRR is also more aggressive in that it starts this sending
right away whereas 3517(bis) leaves the network idle for a bit (1/2 RTT)
before sending.  So, we could view PRR as more aggressive in that it
sends sooner.

So, I think one can argue either side here.  My own hit is that the
first order is the reduction factor (half for PRR just as for earlier
standard CC schemes) and the second order is the pattern.  The pattern
might be important in some cases and PRR might be a better pattern than
3517 or rate halving.  But, ultimately for safety I think the magnitude
of the MD is the important thing.  So, of the two arguments above, I buy
the first---i.e., that PRR is consistent with the standards and so using
it could not be called non-standard and so we should just label it
accurately as standards track.

allman




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> I wonder about the process issues related to changing the status at
> this point.

The process is that y'all say someone has suggested the status should be
standards track and you want to know what the WG thinks of that.  This
document has not left the WG.  I have not seen any declaration that it
passed WGLC.  It is within this WG's purview to consider this question.

allman




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From wes@mti-systems.com  Mon Aug 27 10:47:46 2012
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On 8/27/2012 10:43 AM, Mark Allman wrote:
> 
>> I wonder about the process issues related to changing the status at
>> this point.
> 
> The process is that y'all say someone has suggested the status should be
> standards track and you want to know what the WG thinks of that.  This
> document has not left the WG.  I have not seen any declaration that it
> passed WGLC.  It is within this WG's purview to consider this question.


As AD, that's correct :).

Personally, I wouldn't have any real problem with PS based on what's
known at this point.


-- 
Wes Eddy
MTI Systems

From pasi.sarolahti@iki.fi  Mon Aug 27 11:27:57 2012
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On Aug 27, 2012, at 5:43 PM, Mark Allman wrote:

>> I wonder about the process issues related to changing the status at
>> this point.
>=20
> The process is that y'all say someone has suggested the status should =
be
> standards track and you want to know what the WG thinks of that.  This
> document has not left the WG.  I have not seen any declaration that it
> passed WGLC.  It is within this WG's purview to consider this =
question.

Yep. That said, I hope more people can voice their opinions on this.

One question, though: the document gives two variants of the algorithm. =
If we go for standards track, wouldn't it be better to clearly recommend =
one of them? The current advise in conclusions:

   "At this time we see no reason not to test and deploy PRR-SSRB on a
   large scale.  Implementers worried about any potential impact of
   raising the window during recovery may want to optionally support
   PRR-CRB (which is actually simpler to implement) for comparison
   studies."

sounds like there is something to be tested still.

- Pasi


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Sorry for the long radio silence here.

The only reason to bother with standards track is if there is a need
to compel reluctant manufacturers to implement and deploy it.
Customers can then write in their requests for quotes specifications
such as "must support SACK (RFC 2018), and then suddenly the lack of
SACK support becomes a high priority issue for some large manufacture
who shall remain nameless.

Back in the old days (before 1990) RFC meant "request for comments"
and there was no such concept as "standards track". We wrote down our
ideas, and if they were valuable, they became part of the installed
base.   For PRR this has already come to pass for the installed base
that is most important to me.

If management at Apple or Microsoft think PRR isn't worth the effort,
then making PRR a standard will permit their customers to influence
their priorities.   Given the extent to which PRR is an across the
board win, I really don't think this is required.  (However, I could
change my mind if a TCP engineer at one of the above companies
indicate that they are getting pushback from their management.)

Finally there are some details in PRR that are effected by Laminar.  I
would rather not go to the effort of fixing the current draft to make
it a proper normative PS, but not totally correct.  (Hint: Yoshifumi
Nishida's "off by one" comment reflects the difference between pipe
and total_pipe).

So here is what I was planning to do:  Push out the current document
as experimental, and then after Laminar is further along, generate a
normative respecification using Laminar state variables (probably as
one section of a larger doc, but that remains TBD).

Does that make sense?

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay


On Mon, Aug 27, 2012 at 7:43 AM, Mark Allman <mallman@icir.org> wrote:
>
>> I wonder about the process issues related to changing the status at
>> this point.
>
> The process is that y'all say someone has suggested the status should be
> standards track and you want to know what the WG thinks of that.  This
> document has not left the WG.  I have not seen any declaration that it
> passed WGLC.  It is within this WG's purview to consider this question.
>
> allman
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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From: Mark Allman <mallman@icir.org>
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> >> I wonder about the process issues related to changing the status at
> >> this point.
> > 
> > The process is that y'all say someone has suggested the status should be
> > standards track and you want to know what the WG thinks of that.  This
> > document has not left the WG.  I have not seen any declaration that it
> > passed WGLC.  It is within this WG's purview to consider this question.
> 
> Yep. That said, I hope more people can voice their opinions on this.

Agreed.

> One question, though: the document gives two variants of the
> algorithm. If we go for standards track, wouldn't it be better to
> clearly recommend one of them? The current advise in conclusions:
> 
>    "At this time we see no reason not to test and deploy PRR-SSRB on a
>    large scale.  Implementers worried about any potential impact of
>    raising the window during recovery may want to optionally support
>    PRR-CRB (which is actually simpler to implement) for comparison
>    studies."
> 
> sounds like there is something to be tested still.

I don't think the question should be: Is there anything more to learn?
If that was the question we'd spin everything at experimental forever.

Both variants are at most as conservative as current standards.  There
is a fine tradition in any number of the TCP docs of giving implementers
choices.  I view this document as doing just that.  And, in this case,
both choices are conservative in my view.

allman




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From michael.scharf@alcatel-lucent.com  Tue Aug 28 00:27:17 2012
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Tue, 28 Aug 2012 09:27:06 +0200
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So, let's discuss wordsmithing. What about simply saying "more detailed"?

 "Sep 2013 	Submit document on more detailed ECN feedback in TCP to the IES=
G for publication as an Experimental RFC"

Comments are welcome.

Michael



> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
> Behalf Of Matt Mathis
> Sent: Sunday, August 05, 2012 10:02 PM
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Consenus on adding a new TCPM charter=20
> milestone for accurate feedback
>=20
> I agree that "accurate" is a little bit too strong of term. =20
> Perhaps "counted", "proportional", "quantified", "measured", etc.
>=20
> And yes, TCPM is probably the right place, since all CC=20
> standards track docs are explicitly in scope, even as applied=20
> to other protocols.
>=20
> If we separate signalling from action, some proposed actions clearly
> need to mature in ICCRG for a bit.   Here is a tough question though:
> does the choice of alternate actions effect the design of the=20
> signalling?  If so, it may be premature to commit on the signalling.
> (e.g. for low threshold ECN (aka DCTCP) isolated ECN marks=20
> would be extremely unlikely and it would probably be ok to=20
> miss a few.),
>=20
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>=20
>=20
> On Fri, Aug 3, 2012 at 1:58 PM, Scheffenegger, Richard=20
> <rs@netapp.com> wrote:
> >
> > Hi John,
> >
> >
> > Thanks for your comments so far!
> >
> >> > I would argue that the current ECE/CWR mechanism is perfectly=20
> >> > reliable
> >>
> >>    I might take the other side of that argument...
> >
> > Could you please explain? (I should have perhaps added,=20
> RFC3168 is perfectly reliable for at least, and up to, one CE=20
> mark per RTT; The sender gets the information that at least=20
> one CE mark was received in the RTT under all circumstances).
> >
> >
> >>    Actually, there are two counters, packets and (payload) bytes.
> >
> > Correct; but as soon as not every data segment is ACKed=20
> (and the ACK not lost), the delineation (for figuring out=20
> which payload bytes got marked) becomes fuzzy... Even when=20
> the sender keeps state as to the segment boundaries, it's not=20
> clear which of multiple segments were marked (except in the=20
> two extreme cases), when delayed ACKs (or even ACK=20
> compression) are in play...
> >
> > Which brings up another question that I haven't put in this forum:
> >
> > Will there be a conceivable benefit when the sender knows=20
> the exact order or markings?
> >
> > The way the receiver ECN state-machine of DCTCP is=20
> described allows the sender to reconstruct the exact order of=20
> markings (the "CE stream"), provided no (relevant) ACKs are=20
> lost. With the complication that the sender rarely knows,=20
> when and if a relevant ACK got dropped...
> >
> > This level of precision is lost, when the receiver uses=20
> some kind of summary signal; But making such a signal=20
> reliable requires even more bits, and some more or less=20
> complex encoding scheme...
> >
> >
> > But I am not sure if such an extreme precision will be necessary to=20
> > TCP at least (and other transports do provide means to=20
> encode that...)
> >
> >
> > To have a proper counter (congested bytes), I think that=20
> would need to be explicit; Mirja did some experiments using=20
> implicit ACKs, but to get them work properly, you end up with=20
> sending out more ACKs than strictly necessary (like what=20
> DCTCP receivers do), and single ACK loss can still result in=20
> wrong signals at the sender....
> >
> >
> >
> >
> > Richard Scheffenegger
> >
> >
> >
> >
> >> -----Original Message-----
> >> From: John Leslie [mailto:john@jlc.net]
> >> Sent: Freitag, 03. August 2012 11:34
> >> To: Scheffenegger, Richard
> >> Cc: tcpm@ietf.org
> >> Subject: Re: [tcpm] Consenus on adding a new TCPM charter=20
> milestone=20
> >> for accurate feedback
> >>
> >> Scheffenegger, Richard <rs@netapp.com> wrote:
> >> >
> >> > Hi John,
> >> >
> >> > Am I right in assuming that with "greater precision" you=20
> imply that=20
> >> > this doesn't necessarily mean "exact" feedback?
> >>
> >>    Yes, but... This is mainly to avoid arguing what=20
> "exact" might mean.
> >>
> >> > (Originally we discussed using "more accurate" in the=20
> title of the=20
> >> > draft; that meant lower precision than "exact", but the=20
> "more" got=20
> >> > lost along the way, leaving "accurate").
> >>
> >>    (I must admit I'm at least as anxious to avoid arguing=20
> what "accurate"
> >> might mean.)
> >>
> >> > I would argue that the current ECE/CWR mechanism is perfectly=20
> >> > reliable
> >>
> >>    I might take the other side of that argument...
> >>
> >> > - up to a certain precision limited by the number of=20
> bits set aside=20
> >> > for the signal back from the receiver to the sender.=20
> What would be=20
> >> > more reliable?
> >>
> >>    "Reliable" in TCP means that everything not=20
> acknowledged will be=20
> >> retransmitted. That would imply repeating each ECE until that=20
> >> specific ECE is acknowledged -- and clarifying to the=20
> sender how that=20
> >> CWR was understood.
> >>
> >>    But I'm not trying to argue either way whether=20
> "reliable" would be=20
> >> "better" than "unreliable".
> >>
> >> > (ECE is a counter that can only increase, and remains at the=20
> >> > maximumi value until being reset; CWR provides the reset signal).
> >>
> >>    Umm... "remains at maximum" implies some loss of precision...
> >>
> >> > OTOH, if the feedback signal is encoded with enough bits,=20
> >> > reliability can be brought arbitrarily close to "perfect", even=20
> >> > without an explicit forward signal (CWR).
> >>
> >>    Indeed, one could argue "reliability" by unacknowledged=20
> repetition.
> >> (I'd prefer some other term, but it certainly would serve for the=20
> >> "precision" I was seeking.)
> >>
> >> > As soon as the counter (and signal for it) cover all the packets=20
> >> > individually in a window, the ECN feedback can be exact;
> >>
> >>    Actually, there are two counters, packets and (payload) bytes.
> >>
> >> > The question becomes, can the reset mechanism be abandoned when=20
> >> > "enough" bits are available for the signal, or must the=20
> principal=20
> >> > architecture (a counter increasing up to some point, until it is
> >> > reset) be maintained...
> >>
> >>    I don't choose to express an opinion on that yet.
> >>
> >> > BTW: I found that ECN for SCTP also failed to discuss this, but=20
> >> > there a 32 bit counter is used (plus a Reset signal/CWR).
> >>
> >>    :^)
> >>
> >> > To jump forward a bit, I suspect the delineation is somewhere=20
> >> > around
> >> > 5 bits (2.3 bits in the codepoint draft; 3 bits in=20
> reECN; 8 bits in=20
> >> > the TCP Option draft). Using 5 bits or less, a handshare=20
> mechanism=20
> >> > is required; using more than 5, and a simple counter suffices to=20
> >> > maintain exactness (until TCP suffers from loss of the=20
> ACK clock etc).
> >>
> >>    No comment (yet)...
> >>
> >> > The signaling is another question; Perhaps a certain loss of=20
> >> > granularity is acceptable for the currently discussed=20
> protocols (i.e.
> >> > Conex; slight overestimating congestion is not as bad as=20
> >> > underestimating). But other (future) uses (e.g. DCTCP) would=20
> >> > probably require the full precision - with other words,=20
> "exact" signaling...
> >>
> >>    "Exact" _is_ a reasonable goal...
> >>
> >> > However, we (authors), while discussing this internally,=20
> would also=20
> >> > be glad of any opinion regarding this!
> >>
> >>    I will note that current ECN processing in TCP is=20
> awfully specialized:
> >> I can imagine congestion-control algorithms it would serve poorly.=20
> >> :^( But leaving that aside, ECN processing for other transport=20
> >> protocols almost certainly deserves a different treatment.
> >>
> >>    It would be nice if our "greater precision" effort=20
> could serve the=20
> >> needs of yet-unseen future transport protocols.
> >>
> >> --
> >> John Leslie <john@jlc.net>
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> =

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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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I like it!

Michael


On 28. aug. 2012, at 09:27, Scharf, Michael (Michael) wrote:

> So, let's discuss wordsmithing. What about simply saying "more =
detailed"?
>=20
> "Sep 2013 	Submit document on more detailed ECN feedback in TCP to =
the IESG for publication as an Experimental RFC"
>=20
> Comments are welcome.
>=20
> Michael
>=20
>=20
>=20
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
>> Behalf Of Matt Mathis
>> Sent: Sunday, August 05, 2012 10:02 PM
>> To: Scheffenegger, Richard
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] Consenus on adding a new TCPM charter=20
>> milestone for accurate feedback
>>=20
>> I agree that "accurate" is a little bit too strong of term. =20
>> Perhaps "counted", "proportional", "quantified", "measured", etc.
>>=20
>> And yes, TCPM is probably the right place, since all CC=20
>> standards track docs are explicitly in scope, even as applied=20
>> to other protocols.
>>=20
>> If we separate signalling from action, some proposed actions clearly
>> need to mature in ICCRG for a bit.   Here is a tough question though:
>> does the choice of alternate actions effect the design of the=20
>> signalling?  If so, it may be premature to commit on the signalling.
>> (e.g. for low threshold ECN (aka DCTCP) isolated ECN marks=20
>> would be extremely unlikely and it would probably be ok to=20
>> miss a few.),
>>=20
>> Thanks,
>> --MM--
>> The best way to predict the future is to create it.  - Alan Kay
>>=20
>>=20
>> On Fri, Aug 3, 2012 at 1:58 PM, Scheffenegger, Richard=20
>> <rs@netapp.com> wrote:
>>>=20
>>> Hi John,
>>>=20
>>>=20
>>> Thanks for your comments so far!
>>>=20
>>>>> I would argue that the current ECE/CWR mechanism is perfectly=20
>>>>> reliable
>>>>=20
>>>>   I might take the other side of that argument...
>>>=20
>>> Could you please explain? (I should have perhaps added,=20
>> RFC3168 is perfectly reliable for at least, and up to, one CE=20
>> mark per RTT; The sender gets the information that at least=20
>> one CE mark was received in the RTT under all circumstances).
>>>=20
>>>=20
>>>>   Actually, there are two counters, packets and (payload) bytes.
>>>=20
>>> Correct; but as soon as not every data segment is ACKed=20
>> (and the ACK not lost), the delineation (for figuring out=20
>> which payload bytes got marked) becomes fuzzy... Even when=20
>> the sender keeps state as to the segment boundaries, it's not=20
>> clear which of multiple segments were marked (except in the=20
>> two extreme cases), when delayed ACKs (or even ACK=20
>> compression) are in play...
>>>=20
>>> Which brings up another question that I haven't put in this forum:
>>>=20
>>> Will there be a conceivable benefit when the sender knows=20
>> the exact order or markings?
>>>=20
>>> The way the receiver ECN state-machine of DCTCP is=20
>> described allows the sender to reconstruct the exact order of=20
>> markings (the "CE stream"), provided no (relevant) ACKs are=20
>> lost. With the complication that the sender rarely knows,=20
>> when and if a relevant ACK got dropped...
>>>=20
>>> This level of precision is lost, when the receiver uses=20
>> some kind of summary signal; But making such a signal=20
>> reliable requires even more bits, and some more or less=20
>> complex encoding scheme...
>>>=20
>>>=20
>>> But I am not sure if such an extreme precision will be necessary to=20=

>>> TCP at least (and other transports do provide means to=20
>> encode that...)
>>>=20
>>>=20
>>> To have a proper counter (congested bytes), I think that=20
>> would need to be explicit; Mirja did some experiments using=20
>> implicit ACKs, but to get them work properly, you end up with=20
>> sending out more ACKs than strictly necessary (like what=20
>> DCTCP receivers do), and single ACK loss can still result in=20
>> wrong signals at the sender....
>>>=20
>>>=20
>>>=20
>>>=20
>>> Richard Scheffenegger
>>>=20
>>>=20
>>>=20
>>>=20
>>>> -----Original Message-----
>>>> From: John Leslie [mailto:john@jlc.net]
>>>> Sent: Freitag, 03. August 2012 11:34
>>>> To: Scheffenegger, Richard
>>>> Cc: tcpm@ietf.org
>>>> Subject: Re: [tcpm] Consenus on adding a new TCPM charter=20
>> milestone=20
>>>> for accurate feedback
>>>>=20
>>>> Scheffenegger, Richard <rs@netapp.com> wrote:
>>>>>=20
>>>>> Hi John,
>>>>>=20
>>>>> Am I right in assuming that with "greater precision" you=20
>> imply that=20
>>>>> this doesn't necessarily mean "exact" feedback?
>>>>=20
>>>>   Yes, but... This is mainly to avoid arguing what=20
>> "exact" might mean.
>>>>=20
>>>>> (Originally we discussed using "more accurate" in the=20
>> title of the=20
>>>>> draft; that meant lower precision than "exact", but the=20
>> "more" got=20
>>>>> lost along the way, leaving "accurate").
>>>>=20
>>>>   (I must admit I'm at least as anxious to avoid arguing=20
>> what "accurate"
>>>> might mean.)
>>>>=20
>>>>> I would argue that the current ECE/CWR mechanism is perfectly=20
>>>>> reliable
>>>>=20
>>>>   I might take the other side of that argument...
>>>>=20
>>>>> - up to a certain precision limited by the number of=20
>> bits set aside=20
>>>>> for the signal back from the receiver to the sender.=20
>> What would be=20
>>>>> more reliable?
>>>>=20
>>>>   "Reliable" in TCP means that everything not=20
>> acknowledged will be=20
>>>> retransmitted. That would imply repeating each ECE until that=20
>>>> specific ECE is acknowledged -- and clarifying to the=20
>> sender how that=20
>>>> CWR was understood.
>>>>=20
>>>>   But I'm not trying to argue either way whether=20
>> "reliable" would be=20
>>>> "better" than "unreliable".
>>>>=20
>>>>> (ECE is a counter that can only increase, and remains at the=20
>>>>> maximumi value until being reset; CWR provides the reset signal).
>>>>=20
>>>>   Umm... "remains at maximum" implies some loss of precision...
>>>>=20
>>>>> OTOH, if the feedback signal is encoded with enough bits,=20
>>>>> reliability can be brought arbitrarily close to "perfect", even=20
>>>>> without an explicit forward signal (CWR).
>>>>=20
>>>>   Indeed, one could argue "reliability" by unacknowledged=20
>> repetition.
>>>> (I'd prefer some other term, but it certainly would serve for the=20=

>>>> "precision" I was seeking.)
>>>>=20
>>>>> As soon as the counter (and signal for it) cover all the packets=20=

>>>>> individually in a window, the ECN feedback can be exact;
>>>>=20
>>>>   Actually, there are two counters, packets and (payload) bytes.
>>>>=20
>>>>> The question becomes, can the reset mechanism be abandoned when=20
>>>>> "enough" bits are available for the signal, or must the=20
>> principal=20
>>>>> architecture (a counter increasing up to some point, until it is
>>>>> reset) be maintained...
>>>>=20
>>>>   I don't choose to express an opinion on that yet.
>>>>=20
>>>>> BTW: I found that ECN for SCTP also failed to discuss this, but=20
>>>>> there a 32 bit counter is used (plus a Reset signal/CWR).
>>>>=20
>>>>   :^)
>>>>=20
>>>>> To jump forward a bit, I suspect the delineation is somewhere=20
>>>>> around
>>>>> 5 bits (2.3 bits in the codepoint draft; 3 bits in=20
>> reECN; 8 bits in=20
>>>>> the TCP Option draft). Using 5 bits or less, a handshare=20
>> mechanism=20
>>>>> is required; using more than 5, and a simple counter suffices to=20=

>>>>> maintain exactness (until TCP suffers from loss of the=20
>> ACK clock etc).
>>>>=20
>>>>   No comment (yet)...
>>>>=20
>>>>> The signaling is another question; Perhaps a certain loss of=20
>>>>> granularity is acceptable for the currently discussed=20
>> protocols (i.e.
>>>>> Conex; slight overestimating congestion is not as bad as=20
>>>>> underestimating). But other (future) uses (e.g. DCTCP) would=20
>>>>> probably require the full precision - with other words,=20
>> "exact" signaling...
>>>>=20
>>>>   "Exact" _is_ a reasonable goal...
>>>>=20
>>>>> However, we (authors), while discussing this internally,=20
>> would also=20
>>>>> be glad of any opinion regarding this!
>>>>=20
>>>>   I will note that current ECN processing in TCP is=20
>> awfully specialized:
>>>> I can imagine congestion-control algorithms it would serve poorly.=20=

>>>> :^( But leaving that aside, ECN processing for other transport=20
>>>> protocols almost certainly deserves a different treatment.
>>>>=20
>>>>   It would be nice if our "greater precision" effort=20
>> could serve the=20
>>>> needs of yet-unseen future transport protocols.
>>>>=20
>>>> --
>>>> John Leslie <john@jlc.net>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From john@jlc.net  Tue Aug 28 07:37:25 2012
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From: John Leslie <john@jlc.net>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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Scharf, Michael (Michael) <michael.scharf@alcatel-lucent.com> wrote:
> 
> So, let's discuss wordsmithing. What about simply saying "more detailed"?
> 
>  "Sep 2013 	Submit document on more detailed ECN feedback in TCP to the IESG for publication as an Experimental RFC"

   Sounds better to me...

--
John Leslie <john@jlc.net>

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Subject: Re: [tcpm] Consenus on adding a new TCPM charter milestone for accurate feedback
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On Tue, Aug 28, 2012 at 12:27 AM, Scharf, Michael (Michael)
<michael.scharf@alcatel-lucent.com> wrote:
> So, let's discuss wordsmithing. What about simply saying "more detailed"?
>
>  "Sep 2013      Submit document on more detailed ECN feedback in TCP to the IESG for publication as an Experimental RFC"
+1
>
> Comments are welcome.
>
> Michael
>
>
>
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
>> Behalf Of Matt Mathis
>> Sent: Sunday, August 05, 2012 10:02 PM
>> To: Scheffenegger, Richard
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] Consenus on adding a new TCPM charter
>> milestone for accurate feedback
>>
>> I agree that "accurate" is a little bit too strong of term.
>> Perhaps "counted", "proportional", "quantified", "measured", etc.
>>
>> And yes, TCPM is probably the right place, since all CC
>> standards track docs are explicitly in scope, even as applied
>> to other protocols.
>>
>> If we separate signalling from action, some proposed actions clearly
>> need to mature in ICCRG for a bit.   Here is a tough question though:
>> does the choice of alternate actions effect the design of the
>> signalling?  If so, it may be premature to commit on the signalling.
>> (e.g. for low threshold ECN (aka DCTCP) isolated ECN marks
>> would be extremely unlikely and it would probably be ok to
>> miss a few.),
>>
>> Thanks,
>> --MM--
>> The best way to predict the future is to create it.  - Alan Kay
>>
>>
>> On Fri, Aug 3, 2012 at 1:58 PM, Scheffenegger, Richard
>> <rs@netapp.com> wrote:
>> >
>> > Hi John,
>> >
>> >
>> > Thanks for your comments so far!
>> >
>> >> > I would argue that the current ECE/CWR mechanism is perfectly
>> >> > reliable
>> >>
>> >>    I might take the other side of that argument...
>> >
>> > Could you please explain? (I should have perhaps added,
>> RFC3168 is perfectly reliable for at least, and up to, one CE
>> mark per RTT; The sender gets the information that at least
>> one CE mark was received in the RTT under all circumstances).
>> >
>> >
>> >>    Actually, there are two counters, packets and (payload) bytes.
>> >
>> > Correct; but as soon as not every data segment is ACKed
>> (and the ACK not lost), the delineation (for figuring out
>> which payload bytes got marked) becomes fuzzy... Even when
>> the sender keeps state as to the segment boundaries, it's not
>> clear which of multiple segments were marked (except in the
>> two extreme cases), when delayed ACKs (or even ACK
>> compression) are in play...
>> >
>> > Which brings up another question that I haven't put in this forum:
>> >
>> > Will there be a conceivable benefit when the sender knows
>> the exact order or markings?
>> >
>> > The way the receiver ECN state-machine of DCTCP is
>> described allows the sender to reconstruct the exact order of
>> markings (the "CE stream"), provided no (relevant) ACKs are
>> lost. With the complication that the sender rarely knows,
>> when and if a relevant ACK got dropped...
>> >
>> > This level of precision is lost, when the receiver uses
>> some kind of summary signal; But making such a signal
>> reliable requires even more bits, and some more or less
>> complex encoding scheme...
>> >
>> >
>> > But I am not sure if such an extreme precision will be necessary to
>> > TCP at least (and other transports do provide means to
>> encode that...)
>> >
>> >
>> > To have a proper counter (congested bytes), I think that
>> would need to be explicit; Mirja did some experiments using
>> implicit ACKs, but to get them work properly, you end up with
>> sending out more ACKs than strictly necessary (like what
>> DCTCP receivers do), and single ACK loss can still result in
>> wrong signals at the sender....
>> >
>> >
>> >
>> >
>> > Richard Scheffenegger
>> >
>> >
>> >
>> >
>> >> -----Original Message-----
>> >> From: John Leslie [mailto:john@jlc.net]
>> >> Sent: Freitag, 03. August 2012 11:34
>> >> To: Scheffenegger, Richard
>> >> Cc: tcpm@ietf.org
>> >> Subject: Re: [tcpm] Consenus on adding a new TCPM charter
>> milestone
>> >> for accurate feedback
>> >>
>> >> Scheffenegger, Richard <rs@netapp.com> wrote:
>> >> >
>> >> > Hi John,
>> >> >
>> >> > Am I right in assuming that with "greater precision" you
>> imply that
>> >> > this doesn't necessarily mean "exact" feedback?
>> >>
>> >>    Yes, but... This is mainly to avoid arguing what
>> "exact" might mean.
>> >>
>> >> > (Originally we discussed using "more accurate" in the
>> title of the
>> >> > draft; that meant lower precision than "exact", but the
>> "more" got
>> >> > lost along the way, leaving "accurate").
>> >>
>> >>    (I must admit I'm at least as anxious to avoid arguing
>> what "accurate"
>> >> might mean.)
>> >>
>> >> > I would argue that the current ECE/CWR mechanism is perfectly
>> >> > reliable
>> >>
>> >>    I might take the other side of that argument...
>> >>
>> >> > - up to a certain precision limited by the number of
>> bits set aside
>> >> > for the signal back from the receiver to the sender.
>> What would be
>> >> > more reliable?
>> >>
>> >>    "Reliable" in TCP means that everything not
>> acknowledged will be
>> >> retransmitted. That would imply repeating each ECE until that
>> >> specific ECE is acknowledged -- and clarifying to the
>> sender how that
>> >> CWR was understood.
>> >>
>> >>    But I'm not trying to argue either way whether
>> "reliable" would be
>> >> "better" than "unreliable".
>> >>
>> >> > (ECE is a counter that can only increase, and remains at the
>> >> > maximumi value until being reset; CWR provides the reset signal).
>> >>
>> >>    Umm... "remains at maximum" implies some loss of precision...
>> >>
>> >> > OTOH, if the feedback signal is encoded with enough bits,
>> >> > reliability can be brought arbitrarily close to "perfect", even
>> >> > without an explicit forward signal (CWR).
>> >>
>> >>    Indeed, one could argue "reliability" by unacknowledged
>> repetition.
>> >> (I'd prefer some other term, but it certainly would serve for the
>> >> "precision" I was seeking.)
>> >>
>> >> > As soon as the counter (and signal for it) cover all the packets
>> >> > individually in a window, the ECN feedback can be exact;
>> >>
>> >>    Actually, there are two counters, packets and (payload) bytes.
>> >>
>> >> > The question becomes, can the reset mechanism be abandoned when
>> >> > "enough" bits are available for the signal, or must the
>> principal
>> >> > architecture (a counter increasing up to some point, until it is
>> >> > reset) be maintained...
>> >>
>> >>    I don't choose to express an opinion on that yet.
>> >>
>> >> > BTW: I found that ECN for SCTP also failed to discuss this, but
>> >> > there a 32 bit counter is used (plus a Reset signal/CWR).
>> >>
>> >>    :^)
>> >>
>> >> > To jump forward a bit, I suspect the delineation is somewhere
>> >> > around
>> >> > 5 bits (2.3 bits in the codepoint draft; 3 bits in
>> reECN; 8 bits in
>> >> > the TCP Option draft). Using 5 bits or less, a handshare
>> mechanism
>> >> > is required; using more than 5, and a simple counter suffices to
>> >> > maintain exactness (until TCP suffers from loss of the
>> ACK clock etc).
>> >>
>> >>    No comment (yet)...
>> >>
>> >> > The signaling is another question; Perhaps a certain loss of
>> >> > granularity is acceptable for the currently discussed
>> protocols (i.e.
>> >> > Conex; slight overestimating congestion is not as bad as
>> >> > underestimating). But other (future) uses (e.g. DCTCP) would
>> >> > probably require the full precision - with other words,
>> "exact" signaling...
>> >>
>> >>    "Exact" _is_ a reasonable goal...
>> >>
>> >> > However, we (authors), while discussing this internally,
>> would also
>> >> > be glad of any opinion regarding this!
>> >>
>> >>    I will note that current ECN processing in TCP is
>> awfully specialized:
>> >> I can imagine congestion-control algorithms it would serve poorly.
>> >> :^( But leaving that aside, ECN processing for other transport
>> >> protocols almost certainly deserves a different treatment.
>> >>
>> >>    It would be nice if our "greater precision" effort
>> could serve the
>> >> needs of yet-unseen future transport protocols.
>> >>
>> >> --
>> >> John Leslie <john@jlc.net>
>> > _______________________________________________
>> > tcpm mailing list
>> > tcpm@ietf.org
>> > https://www.ietf.org/mailman/listinfo/tcpm
>> _______________________________________________
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>>
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Subject: [tcpm] Question and comments about draft-dukkipati-tcpm-tcp-loss-probe
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Hi,

In an effort to address the CPM chairs' suggestion to clarify how =
draft-hurtig-tcpm-rtorestart related to =
draft-dukkipati-tcpm-tcp-loss-probe, I finally got around to reading the =
loss-probe draft - and I'm missing a key bit of information that would =
help me make the distinction (and would probably be good to include in =
your draft anyway - and: sorry if I missed it! but I think it's really =
not there): what is the rationale for your choice of "2 * SRTT" in the =
calculation of the probe timeout (PTO)? I mean, why not 3*RTT, 1.5*RTT, =
whatever? If this is just the result of measurements, it would be good =
to report that where you report the others.

On a side note, I also think that it would be helpful for the reader to =
know how you obtained the first results that you present, regarding the =
large difference between RTO and RTT, in the introduction (par. starting =
with "To get a sense...". I assume that this is the effect of a queue =
growing on a short flow, giving a large variance in the RTO calculation, =
but if that's what it is it would be good to say so.

I also noticed an inconsistency: your draft is called "TCP Loss Probe", =
whereas the slides at =
http://www.ietf.org/proceedings/84/slides/slides-84-tcpm-14.pdf
call it "Tail Loss Probe". Personally I prefer the latter term.

Finally, I think that this draft raises the obvious question: if it's =
beneficial to send an extra packet, why not 2, just to be safe? Or 3? Or =
...   =3D> surely there is a point of where things get unacceptable. It =
would be nice to see this trade-off quantified and evaluated, leading to =
a conclusion that explains why 1 probe packet is a good choice (or maybe =
even leading to a different number?).

Cheers,
Michael

