
From mirja.kuehlewind@ikr.uni-stuttgart.de  Wed Feb  1 09:18:22 2012
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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
To: Matt Mathis <mattmathis@google.com>
Date: Wed, 1 Feb 2012 18:18:17 +0100
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Subject: [tcpm] Review of PRR
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Hi Matt,

I reviewed the PRR algorithm. Actually I read your IMC paper and not the draft 
but I hope its the same algorithm ;-)

Let me summarize the main points from my understanding and then I have some 
questions/comments:

I think there are 3 changes compared to the current Linux implementation:

1) You removed the limitation that the cwnd can't be larger that fight_size+1 
in recovery
   -> Agreed, this limitation does not make sense and really causes problems

2) Make reduction dependent on congestion control
   -> This is an implementational thing. Currently Linux will reduce 
(hard-coded) on every second ACK which will (eventually) lead to a havling of 
the cwnd. I agree that this decision should be made by the CC algo but its 
mainly Linux not providing an easy interface to do so. 

3) SS like increase in recovery
   -> This follows directly from point 2. There is a way to reset the cwnd 
after recovery to any value but Linux will still limit the cwnd to 
flight_size+max_burst_size. max_burst_size is (hard-coded) 3 in Linux and 
this limitation make sense to avoid large bursts. Here comes your SS idea 
into the game. You try to maintain the min cwnd close to ssthresh in recovery 
to avoid any bursts right after recovery when you set your cwnd to the value 
determined by the congestion control algorithm.

Here my comments:

1) Bursts:
As said above the limitation to flight_size+1 is wrong and causes a too 
conservation behavior of the Linux kernel. But actually the problem is mainly 
because of the +1. I believe there should still be a limitation of the 
max burst size (e.g. 3 as used by Linux). In your algorithm it possible to 
have very arge burst in under certain circumstances, right?
At least there should be:
	At the end of recovery:
	cwnd = min(ssthresh, flight_size + max_burst_size)

2) SS:
The SS thing is a smart idea but I'm not sure if its really needed. You run 
into the SS part if you have large burst losses or you are 
application-limited. I would say both cases are worth to be conservative. As 
soon as you leave recovery and your cwnd is below ssthresh you will be in 
slow start anyway. You paper is saying in your measurement you were in 32% in 
the slow start phase at the beginning of recovery. This seems to be a really 
high value. Do you know what's happening there? Is this due to application 
limitations or burst losses? I would guess its loss. But if you have a lot of 
small transmission which hardly leave slow start, it might also explain this 
number. Maybe both cases even need to be handled differently...? 

3) DeliveredData:
You are using the number of ack'ed bytes. That means you need to have the cwnd 
and ssthresh also in bytes (not sure if this is said explicitly anyway). The 
Linux kernel mostly used number of packets and also assumes that there will 
be an ACK for every new packet in recovery. These assumptions might be not 
correct anymore. But I'm not sure if this should be changed implicit in this 
proposal. This might be an important topic on its own which might have more 
implications (if implemented every correctly). 

And I have two more questions:
2) In the paper you say this algorithm will be in the mainline of the Linux 
kernel. I checked and I dont think it is there yet...? Do you have a patch 
available? 
3) And one completely different question: In the paper you say ECN is disabled 
on Google servers. Why is this?

Regards,
Mirja




From mattmathis@google.com  Wed Feb  1 11:01:52 2012
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From: Matt Mathis <mattmathis@google.com>
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Subject: Re: [tcpm] Review of PRR
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--bcaec521555b41f5eb04b7ebb608
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First the quick answers: PRR is in Linux 3.2-rc1.

The implementation in Linux uses packet counts, not bytes.  Yes this causes
a number of artifacts.

The transactional nature of the web and video (codic) pacing both cause
frequent "application stalls" which cause much of our intuition about how
TCP typically operates to be somewhat inaccurate.

I believe that all of your suggestions were evaluated empirically and
didn't pan out.  Several alternative algorithms were tested but not
described in the paper.   I can speak to the one that surprised me, where
my intuition initially matched yours, but the numbers just didn't agree.

The issue is SS during recovery.    Consider the situation were the
cwnd=3DFlightSize=3D10, and you lose 8 packets.  Without slowstart during t=
he
recovery it would be completed with a window of 2, taking 5 RTTs.
 Furthermore any additional losses are very likely to cause a timeout.
 Compare this to RFC3517, which allows a 3 packet burst, to complete the
recovery in 2 RTTs.

Although this seems like a corner case, it happens frequently enough to be
statistically significant in the measurement data.   For PRR, in this case
each of the two ACKs will cause 2 segments to be sent (4 total), as does
the first ACK of the next RTT (thus pipe=3D5).   Note that both RFC3517 and
PRR send 3 extra packets during recovery, either in a burst or
in response to  three different ACKs.

It turns out that due to the risk of additional losses, PRR's
average performance is  slightly better than RFC3517, even though under
identical conditions PRR transmits exactly the same amount of data but
slightly later.

If the existing standard had not specify something so aggressive, we may
have chosen to to do it as as you suggest, but proposing a
new standard that performs worse than the existing standard was a
non-starter.

I am working on an ID that proposes to re-factor TCP congestion control,
which will bring the many of your observations into focus.

Re: ECN.  Last time we checked it caused a measurable increase in the
number of failed user requests, probably due to return paths that
unconditionally discard packets with ECN marks.  Note that the forward path
must be delivering them, which seems rather odd.    But yes, we are
tinkering with ECN, and would like to help foster its deployment.

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



On Wed, Feb 1, 2012 at 9:18 AM, Mirja K=FChlewind <
mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:

> Hi Matt,
>
> I reviewed the PRR algorithm. Actually I read your IMC paper and not the
> draft
> but I hope its the same algorithm ;-)
>
> Let me summarize the main points from my understanding and then I have so=
me
> questions/comments:
>
> I think there are 3 changes compared to the current Linux implementation:
>
> 1) You removed the limitation that the cwnd can't be larger that
> fight_size+1
> in recovery
>   -> Agreed, this limitation does not make sense and really causes proble=
ms
>
> 2) Make reduction dependent on congestion control
>   -> This is an implementational thing. Currently Linux will reduce
> (hard-coded) on every second ACK which will (eventually) lead to a havlin=
g
> of
> the cwnd. I agree that this decision should be made by the CC algo but it=
s
> mainly Linux not providing an easy interface to do so.
>
> 3) SS like increase in recovery
>   -> This follows directly from point 2. There is a way to reset the cwnd
> after recovery to any value but Linux will still limit the cwnd to
> flight_size+max_burst_size. max_burst_size is (hard-coded) 3 in Linux and
> this limitation make sense to avoid large bursts. Here comes your SS idea
> into the game. You try to maintain the min cwnd close to ssthresh in
> recovery
> to avoid any bursts right after recovery when you set your cwnd to the
> value
> determined by the congestion control algorithm.
>
> Here my comments:
>
> 1) Bursts:
> As said above the limitation to flight_size+1 is wrong and causes a too
> conservation behavior of the Linux kernel. But actually the problem is
> mainly
> because of the +1. I believe there should still be a limitation of the
> max burst size (e.g. 3 as used by Linux). In your algorithm it possible t=
o
> have very arge burst in under certain circumstances, right?
> At least there should be:
>        At the end of recovery:
>        cwnd =3D min(ssthresh, flight_size + max_burst_size)
>
> 2) SS:
> The SS thing is a smart idea but I'm not sure if its really needed. You r=
un
> into the SS part if you have large burst losses or you are
> application-limited. I would say both cases are worth to be conservative.
> As
> soon as you leave recovery and your cwnd is below ssthresh you will be in
> slow start anyway. You paper is saying in your measurement you were in 32=
%
> in
> the slow start phase at the beginning of recovery. This seems to be a
> really
> high value. Do you know what's happening there? Is this due to applicatio=
n
> limitations or burst losses? I would guess its loss. But if you have a lo=
t
> of
> small transmission which hardly leave slow start, it might also explain
> this
> number. Maybe both cases even need to be handled differently...?
>
> 3) DeliveredData:
> You are using the number of ack'ed bytes. That means you need to have the
> cwnd
> and ssthresh also in bytes (not sure if this is said explicitly anyway).
> The
> Linux kernel mostly used number of packets and also assumes that there wi=
ll
> be an ACK for every new packet in recovery. These assumptions might be no=
t
> correct anymore. But I'm not sure if this should be changed implicit in
> this
> proposal. This might be an important topic on its own which might have mo=
re
> implications (if implemented every correctly).
>
> And I have two more questions:
> 2) In the paper you say this algorithm will be in the mainline of the Lin=
ux
> kernel. I checked and I dont think it is there yet...? Do you have a patc=
h
> available?
> 3) And one completely different question: In the paper you say ECN is
> disabled
> on Google servers. Why is this?
>
> Regards,
> Mirja
>
>
>
>

--bcaec521555b41f5eb04b7ebb608
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

First the quick answers: PRR is in Linux 3.2-rc1.<div><br></div><div>The im=
plementation in Linux uses packet counts, not bytes. =A0Yes this causes a n=
umber of artifacts.</div><div><br></div><div>The transactional nature of th=
e web and video (codic) pacing both cause frequent &quot;application stalls=
&quot; which cause much of our intuition about how TCP typically operates t=
o be somewhat inaccurate.<br>
<div><br></div><div>I believe that all of your suggestions were evaluated=
=A0empirically and didn&#39;t pan out. =A0Several alternative algorithms we=
re tested but not described in the paper. =A0 I can speak to the one that=
=A0surprised=A0me, where my intuition=A0initially=A0matched yours, but the =
numbers just didn&#39;t agree.</div>

<div><br></div><div>The issue is SS during recovery. =A0 =A0Consider the si=
tuation were the cwnd=3DFlightSize=3D10, and you lose 8 packets. =A0Without=
 slowstart during the recovery it would be completed with a window of 2, ta=
king 5 RTTs. =A0Furthermore any additional losses are very likely to cause =
a timeout. =A0 =A0Compare this to RFC3517, which allows a 3 packet burst, t=
o complete the recovery in 2 RTTs.</div>

<div><br></div><div>Although this seems like a=A0corner=A0case, it happens =
frequently enough to be statistically significant in the measurement data. =
=A0 For PRR, in this case each of the two ACKs will cause 2 segments to be =
sent (4 total), as does the first ACK of the next RTT (thus pipe=3D5). =A0 =
Note that both RFC3517 and PRR send 3 extra packets during recovery, either=
 in a burst or in=A0response=A0to =A0three different ACKs.</div>
<div><br></div><div>It turns out that due to the risk of additional losses,=
 PRR&#39;s average=A0performance=A0is =A0slightly better than RFC3517, even=
 though=A0under identical conditions PRR=A0transmits exactly the same amoun=
t of data but slightly later.</div>

<div><br></div><div>If the existing standard had not specify something so a=
ggressive, we may have chosen to to do it as as you suggest, but proposing =
a new=A0standard=A0that=A0performs=A0worse than the=A0existing=A0standard w=
as a non-starter.</div>
<div><br></div><div>I am working on an ID that proposes to=A0re-factor TCP =
congestion control, which=A0will bring the many of your observations into f=
ocus.</div><div><br></div><div>Re: ECN. =A0Last time we checked it caused a=
 measurable increase in the number of failed=A0user requests, probably due =
to return paths that unconditionally discard packets with ECN marks. =A0Not=
e that the forward path must be delivering them, which seems rather odd. =
=A0 =A0But yes, we are tinkering with ECN, and would like to help foster it=
s deployment.</div>
<div><br></div><div>Thanks,</div><div>
--MM--<br>The best way to predict the future is to create it. =A0- Alan Kay=
<br><br>
<br><br><div class=3D"gmail_quote">On Wed, Feb 1, 2012 at 9:18 AM, Mirja K=
=FChlewind <span dir=3D"ltr">&lt;<a href=3D"mailto:mirja.kuehlewind@ikr.uni=
-stuttgart.de" target=3D"_blank">mirja.kuehlewind@ikr.uni-stuttgart.de</a>&=
gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi Matt,<br>
<br>
I reviewed the PRR algorithm. Actually I read your IMC paper and not the dr=
aft<br>
but I hope its the same algorithm ;-)<br>
<br>
Let me summarize the main points from my understanding and then I have some=
<br>
questions/comments:<br>
<br>
I think there are 3 changes compared to the current Linux implementation:<b=
r>
<br>
1) You removed the limitation that the cwnd can&#39;t be larger that fight_=
size+1<br>
in recovery<br>
 =A0 -&gt; Agreed, this limitation does not make sense and really causes pr=
oblems<br>
<br>
2) Make reduction dependent on congestion control<br>
 =A0 -&gt; This is an implementational thing. Currently Linux will reduce<b=
r>
(hard-coded) on every second ACK which will (eventually) lead to a havling =
of<br>
the cwnd. I agree that this decision should be made by the CC algo but its<=
br>
mainly Linux not providing an easy interface to do so.<br>
<br>
3) SS like increase in recovery<br>
 =A0 -&gt; This follows directly from point 2. There is a way to reset the =
cwnd<br>
after recovery to any value but Linux will still limit the cwnd to<br>
flight_size+max_burst_size. max_burst_size is (hard-coded) 3 in Linux and<b=
r>
this limitation make sense to avoid large bursts. Here comes your SS idea<b=
r>
into the game. You try to maintain the min cwnd close to ssthresh in recove=
ry<br>
to avoid any bursts right after recovery when you set your cwnd to the valu=
e<br>
determined by the congestion control algorithm.<br>
<br>
Here my comments:<br>
<br>
1) Bursts:<br>
As said above the limitation to flight_size+1 is wrong and causes a too<br>
conservation behavior of the Linux kernel. But actually the problem is main=
ly<br>
because of the +1. I believe there should still be a limitation of the<br>
max burst size (e.g. 3 as used by Linux). In your algorithm it possible to<=
br>
have very arge burst in under certain circumstances, right?<br>
At least there should be:<br>
 =A0 =A0 =A0 =A0At the end of recovery:<br>
 =A0 =A0 =A0 =A0cwnd =3D min(ssthresh, flight_size + max_burst_size)<br>
<br>
2) SS:<br>
The SS thing is a smart idea but I&#39;m not sure if its really needed. You=
 run<br>
into the SS part if you have large burst losses or you are<br>
application-limited. I would say both cases are worth to be conservative. A=
s<br>
soon as you leave recovery and your cwnd is below ssthresh you will be in<b=
r>
slow start anyway. You paper is saying in your measurement you were in 32% =
in<br>
the slow start phase at the beginning of recovery. This seems to be a reall=
y<br>
high value. Do you know what&#39;s happening there? Is this due to applicat=
ion<br>
limitations or burst losses? I would guess its loss. But if you have a lot =
of<br>
small transmission which hardly leave slow start, it might also explain thi=
s<br>
number. Maybe both cases even need to be handled differently...?<br>
<br>
3) DeliveredData:<br>
You are using the number of ack&#39;ed bytes. That means you need to have t=
he cwnd<br>
and ssthresh also in bytes (not sure if this is said explicitly anyway). Th=
e<br>
Linux kernel mostly used number of packets and also assumes that there will=
<br>
be an ACK for every new packet in recovery. These assumptions might be not<=
br>
correct anymore. But I&#39;m not sure if this should be changed implicit in=
 this<br>
proposal. This might be an important topic on its own which might have more=
<br>
implications (if implemented every correctly).<br>
<br>
And I have two more questions:<br>
2) In the paper you say this algorithm will be in the mainline of the Linux=
<br>
kernel. I checked and I dont think it is there yet...? Do you have a patch<=
br>
available?<br>
3) And one completely different question: In the paper you say ECN is disab=
led<br>
on Google servers. Why is this?<br>
<br>
Regards,<br>
Mirja<br>
<br>
<br>
<br>
</blockquote></div><br></div>
</div>

--bcaec521555b41f5eb04b7ebb608--

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Subject: Re: [tcpm] Review of PRR
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On Wed, Feb 1, 2012 at 2:01 PM, Matt Mathis <mattmathis@google.com> wrote:
> First the quick answers: PRR is in Linux 3.2-rc1.
FYI: http://patchwork.ozlabs.org/patch/110857/

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Subject: [tcpm] RFC 6528 on Defending against Sequence Number Attacks
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6528

        Title:      Defending against Sequence Number Attacks 
        Author:     F. Gont, S. Bellovin
        Status:     Standards Track
        Stream:     IETF
        Date:       February 2012
        Mailbox:    fgont@si6networks.com, 
                    bellovin@acm.org
        Pages:      12
        Characters: 26917
        Obsoletes:  RFC1948
        Updates:    RFC0793

        I-D Tag:    draft-ietf-tcpm-rfc1948bis-02.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6528.txt

This document specifies an algorithm for the generation of TCP
Initial Sequence Numbers (ISNs), such that the chances of an off-path
attacker guessing the sequence numbers in use by a target connection
are reduced.  This document revises (and formally obsoletes) RFC
1948, and takes the ISN generation algorithm originally proposed in
that document to Standards Track, formally updating RFC 793.  
[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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Subject: [tcpm] Protocol Action: 'The NewReno Modification to TCP's Fast Recovery	Algorithm' to Proposed Standard (draft-ietf-tcpm-rfc3782-bis-05.txt)
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The IESG has approved the following document:
- 'The NewReno Modification to TCP's Fast Recovery Algorithm'
  (draft-ietf-tcpm-rfc3782-bis-05.txt) as a Proposed Standard

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

The IESG contact persons are Wesley Eddy and David Harrington.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-tcpm-rfc3782-bis/




Technical Summary

>From the abstract:

  RFC 5681 documents the following four intertwined TCP
  congestion control algorithms: slow start, congestion avoidance, fast
  retransmit, and fast recovery.  RFC 5681 explicitly allows
  certain modifications of these algorithms, including modifications
  that use the TCP Selective Acknowledgement (SACK) option (RFC 2883),
  and modifications that respond to "partial acknowledgments" (ACKs
  which cover new data, but not all the data outstanding when loss was
  detected) in the absence of SACK.  This document describes a specific
  algorithm for responding to partial acknowledgments, referred to as
  NewReno.  This response to partial acknowledgments was first proposed
  by Janey Hoe.  This document obsoletes RFC 3782.

Working Group Summary

Nothing exceptional occurred during the working group process for this
document.

Document Quality

This document is an update to TCP NewReno, to address issues that
have been found with existing RFC 3782 implementations, of which
there are several.  In the Acknowledgments section:

  Many thanks to Anil Agarwal, Mark Allman, Armando Caro, Jeffrey Hsu,
  Vern Paxson, Kacheong Poon, Keyur Shah, and Bernie Volz for detailed
  feedback on this document or on its precursor, RFC 2582.  Jeffrey
  Hsu provided clarifications on the handling of the recover variable
  that were applied to RFC 3782 as errata, and now are in Section 8
  of this document.  Yoshifumi Nishida contributed a modification
  to the fast recovery algorithm to account for the case in which
  flightsize is 0 when the TCP sender leaves fast recovery, and the
  TCP receiver uses delayed acknowledgments.  Alexander Zimmermann
  provided several suggestions to improve the clarity of the document. 

Personnel

David Borman (david.borman@windriver.com) is the document shepherd.  
Wes Eddy (wes@mti-systems.com) is the responsible AD.


From mirja.kuehlewind@ikr.uni-stuttgart.de  Fri Feb  3 09:04:02 2012
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From: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Hi Matt,

I agree with a lot of things you said. But regarding your example of cwnd 1=
0=20
and 8 losses I'm not sure. You say its not a corner case. I understand that=
=20
you have many small flows not leaving slow start. But in this case you have=
=20
lost 80% of your packet. I would have expected a max loss of 50% of my=20
packets in SS. For me that means 80% is a really high load situation and I=
=20
should be even more careful...!?!

Mirja

Btw. if this is not a corner case maybe IW of 10 is not a good idea...


On Wednesday 01 February 2012 20:01:49 Matt Mathis wrote:
> First the quick answers: PRR is in Linux 3.2-rc1.
>
> The implementation in Linux uses packet counts, not bytes.  Yes this caus=
es
> a number of artifacts.
>
> The transactional nature of the web and video (codic) pacing both cause
> frequent "application stalls" which cause much of our intuition about how
> TCP typically operates to be somewhat inaccurate.
>
> I believe that all of your suggestions were evaluated empirically and
> didn't pan out.  Several alternative algorithms were tested but not
> described in the paper.   I can speak to the one that surprised me, where
> my intuition initially matched yours, but the numbers just didn't agree.
>
> The issue is SS during recovery.    Consider the situation were the
> cwnd=3DFlightSize=3D10, and you lose 8 packets.  Without slowstart during=
 the
> recovery it would be completed with a window of 2, taking 5 RTTs.
>  Furthermore any additional losses are very likely to cause a timeout.
>  Compare this to RFC3517, which allows a 3 packet burst, to complete the
> recovery in 2 RTTs.
>
> Although this seems like a corner case, it happens frequently enough to be
> statistically significant in the measurement data.   For PRR, in this case
> each of the two ACKs will cause 2 segments to be sent (4 total), as does
> the first ACK of the next RTT (thus pipe=3D5).   Note that both RFC3517 a=
nd
> PRR send 3 extra packets during recovery, either in a burst or
> in response to  three different ACKs.
>
> It turns out that due to the risk of additional losses, PRR's
> average performance is  slightly better than RFC3517, even though under
> identical conditions PRR transmits exactly the same amount of data but
> slightly later.
>
> If the existing standard had not specify something so aggressive, we may
> have chosen to to do it as as you suggest, but proposing a
> new standard that performs worse than the existing standard was a
> non-starter.
>
> I am working on an ID that proposes to re-factor TCP congestion control,
> which will bring the many of your observations into focus.
>
> Re: ECN.  Last time we checked it caused a measurable increase in the
> number of failed user requests, probably due to return paths that
> unconditionally discard packets with ECN marks.  Note that the forward pa=
th
> must be delivering them, which seems rather odd.    But yes, we are
> tinkering with ECN, and would like to help foster its deployment.
>
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
>
>
> On Wed, Feb 1, 2012 at 9:18 AM, Mirja K=FChlewind <
>
> mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:
> > Hi Matt,
> >
> > I reviewed the PRR algorithm. Actually I read your IMC paper and not the
> > draft
> > but I hope its the same algorithm ;-)
> >
> > Let me summarize the main points from my understanding and then I have
> > some questions/comments:
> >
> > I think there are 3 changes compared to the current Linux implementatio=
n:
> >
> > 1) You removed the limitation that the cwnd can't be larger that
> > fight_size+1
> > in recovery
> >   -> Agreed, this limitation does not make sense and really causes
> > problems
> >
> > 2) Make reduction dependent on congestion control
> >   -> This is an implementational thing. Currently Linux will reduce
> > (hard-coded) on every second ACK which will (eventually) lead to a
> > havling of
> > the cwnd. I agree that this decision should be made by the CC algo but
> > its mainly Linux not providing an easy interface to do so.
> >
> > 3) SS like increase in recovery
> >   -> This follows directly from point 2. There is a way to reset the cw=
nd
> > after recovery to any value but Linux will still limit the cwnd to
> > flight_size+max_burst_size. max_burst_size is (hard-coded) 3 in Linux a=
nd
> > this limitation make sense to avoid large bursts. Here comes your SS id=
ea
> > into the game. You try to maintain the min cwnd close to ssthresh in
> > recovery
> > to avoid any bursts right after recovery when you set your cwnd to the
> > value
> > determined by the congestion control algorithm.
> >
> > Here my comments:
> >
> > 1) Bursts:
> > As said above the limitation to flight_size+1 is wrong and causes a too
> > conservation behavior of the Linux kernel. But actually the problem is
> > mainly
> > because of the +1. I believe there should still be a limitation of the
> > max burst size (e.g. 3 as used by Linux). In your algorithm it possible
> > to have very arge burst in under certain circumstances, right?
> > At least there should be:
> >        At the end of recovery:
> >        cwnd =3D min(ssthresh, flight_size + max_burst_size)
> >
> > 2) SS:
> > The SS thing is a smart idea but I'm not sure if its really needed. You
> > run into the SS part if you have large burst losses or you are
> > application-limited. I would say both cases are worth to be conservativ=
e.
> > As
> > soon as you leave recovery and your cwnd is below ssthresh you will be =
in
> > slow start anyway. You paper is saying in your measurement you were in
> > 32% in
> > the slow start phase at the beginning of recovery. This seems to be a
> > really
> > high value. Do you know what's happening there? Is this due to
> > application limitations or burst losses? I would guess its loss. But if
> > you have a lot of
> > small transmission which hardly leave slow start, it might also explain
> > this
> > number. Maybe both cases even need to be handled differently...?
> >
> > 3) DeliveredData:
> > You are using the number of ack'ed bytes. That means you need to have t=
he
> > cwnd
> > and ssthresh also in bytes (not sure if this is said explicitly anyway).
> > The
> > Linux kernel mostly used number of packets and also assumes that there
> > will be an ACK for every new packet in recovery. These assumptions might
> > be not correct anymore. But I'm not sure if this should be changed
> > implicit in this
> > proposal. This might be an important topic on its own which might have
> > more implications (if implemented every correctly).
> >
> > And I have two more questions:
> > 2) In the paper you say this algorithm will be in the mainline of the
> > Linux kernel. I checked and I dont think it is there yet...? Do you have
> > a patch available?
> > 3) And one completely different question: In the paper you say ECN is
> > disabled
> > on Google servers. Why is this?
> >
> > Regards,
> > Mirja



=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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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
To: Yuchung Cheng <ycheng@google.com>
Date: Fri, 3 Feb 2012 18:04:13 +0100
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Subject: Re: [tcpm] Review of PRR
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thx.

On Wednesday 01 February 2012 22:19:59 Yuchung Cheng wrote:
> On Wed, Feb 1, 2012 at 2:01 PM, Matt Mathis <mattmathis@google.com> wrote:
> > First the quick answers: PRR is in Linux 3.2-rc1.
>
> FYI: http://patchwork.ozlabs.org/patch/110857/



From mattmathis@google.com  Fri Feb  3 12:44:38 2012
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From: Matt Mathis <mattmathis@google.com>
To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Subject: Re: [tcpm] Review of PRR
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I should have been clearer.   The "not a corner case" I was referring to
was losses exceeding the chosen window reduction, which is the situation
that causes 3517 to send bursts and PRR to invoke slowstarts during
recovery.   We observe that this happens in 32% of the recovery events.

You are correct that a single connection into a single bottleneck should
not cause more than %50 losses.   The most likely cause are bursts on other
connections.  Note that depending on how the packets from two
concurrent slowstarts are interleaved, you expect the losses from to be
slightly asymmetric.    If the receiver is doing "fast ACKs" to speed
slowstart, then you would expect more than 50% losses to
be reasonably common.

However, the primary cause for our data was probably QUBIC itself, which
only invokes a 30% reduction.  This plus fast ACK plus drop tail is likely
cause losses to frequently exceed the target reduction.

We do not have data on how frequently connections experience 80% losses

And you are correct, IW10 could be contributing.  We tried to quantify
that, but couldn't.  What is really needed is  a TCP header study
from within an ISP, just upstream of the primary bottleneck, for all sorts
of types of bottleneck gear, clients and content sources.

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



On Fri, Feb 3, 2012 at 9:03 AM, Mirja Kuehlewind <
mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:

> Hi Matt,
>
> I agree with a lot of things you said. But regarding your example of cwnd
> 10
> and 8 losses I'm not sure. You say its not a corner case. I understand th=
at
> you have many small flows not leaving slow start. But in this case you ha=
ve
> lost 80% of your packet. I would have expected a max loss of 50% of my
> packets in SS. For me that means 80% is a really high load situation and =
I
> should be even more careful...!?!
>
> Mirja
>
> Btw. if this is not a corner case maybe IW of 10 is not a good idea...
>
>
> On Wednesday 01 February 2012 20:01:49 Matt Mathis wrote:
> > First the quick answers: PRR is in Linux 3.2-rc1.
> >
> > The implementation in Linux uses packet counts, not bytes.  Yes this
> causes
> > a number of artifacts.
> >
> > The transactional nature of the web and video (codic) pacing both cause
> > frequent "application stalls" which cause much of our intuition about h=
ow
> > TCP typically operates to be somewhat inaccurate.
> >
> > I believe that all of your suggestions were evaluated empirically and
> > didn't pan out.  Several alternative algorithms were tested but not
> > described in the paper.   I can speak to the one that surprised me, whe=
re
> > my intuition initially matched yours, but the numbers just didn't agree=
.
> >
> > The issue is SS during recovery.    Consider the situation were the
> > cwnd=3DFlightSize=3D10, and you lose 8 packets.  Without slowstart duri=
ng the
> > recovery it would be completed with a window of 2, taking 5 RTTs.
> >  Furthermore any additional losses are very likely to cause a timeout.
> >  Compare this to RFC3517, which allows a 3 packet burst, to complete th=
e
> > recovery in 2 RTTs.
> >
> > Although this seems like a corner case, it happens frequently enough to
> be
> > statistically significant in the measurement data.   For PRR, in this
> case
> > each of the two ACKs will cause 2 segments to be sent (4 total), as doe=
s
> > the first ACK of the next RTT (thus pipe=3D5).   Note that both RFC3517=
 and
> > PRR send 3 extra packets during recovery, either in a burst or
> > in response to  three different ACKs.
> >
> > It turns out that due to the risk of additional losses, PRR's
> > average performance is  slightly better than RFC3517, even though under
> > identical conditions PRR transmits exactly the same amount of data but
> > slightly later.
> >
> > If the existing standard had not specify something so aggressive, we ma=
y
> > have chosen to to do it as as you suggest, but proposing a
> > new standard that performs worse than the existing standard was a
> > non-starter.
> >
> > I am working on an ID that proposes to re-factor TCP congestion control=
,
> > which will bring the many of your observations into focus.
> >
> > Re: ECN.  Last time we checked it caused a measurable increase in the
> > number of failed user requests, probably due to return paths that
> > unconditionally discard packets with ECN marks.  Note that the forward
> path
> > must be delivering them, which seems rather odd.    But yes, we are
> > tinkering with ECN, and would like to help foster its deployment.
> >
> > Thanks,
> > --MM--
> > The best way to predict the future is to create it.  - Alan Kay
> >
> >
> >
> > On Wed, Feb 1, 2012 at 9:18 AM, Mirja K=FChlewind <
> >
> > mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:
> > > Hi Matt,
> > >
> > > I reviewed the PRR algorithm. Actually I read your IMC paper and not
> the
> > > draft
> > > but I hope its the same algorithm ;-)
> > >
> > > Let me summarize the main points from my understanding and then I hav=
e
> > > some questions/comments:
> > >
> > > I think there are 3 changes compared to the current Linux
> implementation:
> > >
> > > 1) You removed the limitation that the cwnd can't be larger that
> > > fight_size+1
> > > in recovery
> > >   -> Agreed, this limitation does not make sense and really causes
> > > problems
> > >
> > > 2) Make reduction dependent on congestion control
> > >   -> This is an implementational thing. Currently Linux will reduce
> > > (hard-coded) on every second ACK which will (eventually) lead to a
> > > havling of
> > > the cwnd. I agree that this decision should be made by the CC algo bu=
t
> > > its mainly Linux not providing an easy interface to do so.
> > >
> > > 3) SS like increase in recovery
> > >   -> This follows directly from point 2. There is a way to reset the
> cwnd
> > > after recovery to any value but Linux will still limit the cwnd to
> > > flight_size+max_burst_size. max_burst_size is (hard-coded) 3 in Linux
> and
> > > this limitation make sense to avoid large bursts. Here comes your SS
> idea
> > > into the game. You try to maintain the min cwnd close to ssthresh in
> > > recovery
> > > to avoid any bursts right after recovery when you set your cwnd to th=
e
> > > value
> > > determined by the congestion control algorithm.
> > >
> > > Here my comments:
> > >
> > > 1) Bursts:
> > > As said above the limitation to flight_size+1 is wrong and causes a t=
oo
> > > conservation behavior of the Linux kernel. But actually the problem i=
s
> > > mainly
> > > because of the +1. I believe there should still be a limitation of th=
e
> > > max burst size (e.g. 3 as used by Linux). In your algorithm it possib=
le
> > > to have very arge burst in under certain circumstances, right?
> > > At least there should be:
> > >        At the end of recovery:
> > >        cwnd =3D min(ssthresh, flight_size + max_burst_size)
> > >
> > > 2) SS:
> > > The SS thing is a smart idea but I'm not sure if its really needed. Y=
ou
> > > run into the SS part if you have large burst losses or you are
> > > application-limited. I would say both cases are worth to be
> conservative.
> > > As
> > > soon as you leave recovery and your cwnd is below ssthresh you will b=
e
> in
> > > slow start anyway. You paper is saying in your measurement you were i=
n
> > > 32% in
> > > the slow start phase at the beginning of recovery. This seems to be a
> > > really
> > > high value. Do you know what's happening there? Is this due to
> > > application limitations or burst losses? I would guess its loss. But =
if
> > > you have a lot of
> > > small transmission which hardly leave slow start, it might also expla=
in
> > > this
> > > number. Maybe both cases even need to be handled differently...?
> > >
> > > 3) DeliveredData:
> > > You are using the number of ack'ed bytes. That means you need to have
> the
> > > cwnd
> > > and ssthresh also in bytes (not sure if this is said explicitly
> anyway).
> > > The
> > > Linux kernel mostly used number of packets and also assumes that ther=
e
> > > will be an ACK for every new packet in recovery. These assumptions
> might
> > > be not correct anymore. But I'm not sure if this should be changed
> > > implicit in this
> > > proposal. This might be an important topic on its own which might hav=
e
> > > more implications (if implemented every correctly).
> > >
> > > And I have two more questions:
> > > 2) In the paper you say this algorithm will be in the mainline of the
> > > Linux kernel. I checked and I dont think it is there yet...? Do you
> have
> > > a patch available?
> > > 3) And one completely different question: In the paper you say ECN is
> > > disabled
> > > on Google servers. Why is this?
> > >
> > > Regards,
> > > Mirja
>
>
>
> --
> -------------------------------------------------------------------
> 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
> -------------------------------------------------------------------
>

--0015174478f2576c6d04b81561d9
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I should have been clearer. =A0 The &quot;not a corner case&quot; I was ref=
erring to was losses exceeding the=A0chosen=A0window reduction, which is th=
e situation that causes 3517 to send bursts and PRR to invoke slowstarts du=
ring recovery. =A0 We observe that this happens in 32% of the recovery even=
ts.<div>

<br></div><div>You are correct that a single connection into a single bottl=
eneck should not cause more than %50 losses. =A0 The most likely cause are =
bursts on other connections. =A0Note that depending on how the packets from=
 two concurrent=A0slowstarts=A0are interleaved, you expect the losses from =
to be slightly=A0asymmetric. =A0=A0=A0If the receiver is doing &quot;fast A=
CKs&quot; to speed slowstart, then you would expect more than 50% losses to=
 be=A0reasonably=A0common.</div>

<div><br></div><div>However, the primary cause for our data was probably QU=
BIC itself, which only invokes a 30% reduction. =A0This=A0plus fast ACK plu=
s drop tail is likely cause losses to=A0frequently=A0exceed=A0the target=A0=
reduction.</div>

<div><br></div><div>We do not have data on how frequently connections exper=
ience 80% losses</div><div><br></div><div>And you are correct, IW10 could b=
e contributing. =A0We tried to quantify that, but couldn&#39;t. =A0What is =
really needed is =A0a TCP header study from=A0within=A0an ISP, just upstrea=
m of the=A0primary=A0bottleneck, for all sorts of types of bottleneck gear,=
 clients and content sources.</div>

<div><br></div><div>Thanks,</div><div><div>--MM--<br>The best way to predic=
t the future is to create it. =A0- Alan Kay<br><br>
<br><br><div class=3D"gmail_quote">On Fri, Feb 3, 2012 at 9:03 AM, Mirja Ku=
ehlewind <span dir=3D"ltr">&lt;<a href=3D"mailto:mirja.kuehlewind@ikr.uni-s=
tuttgart.de" target=3D"_blank">mirja.kuehlewind@ikr.uni-stuttgart.de</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Matt,<br>
<br>
I agree with a lot of things you said. But regarding your example of cwnd 1=
0<br>
and 8 losses I&#39;m not sure. You say its not a corner case. I understand =
that<br>
you have many small flows not leaving slow start. But in this case you have=
<br>
lost 80% of your packet. I would have expected a max loss of 50% of my<br>
packets in SS. For me that means 80% is a really high load situation and I<=
br>
should be even more careful...!?!<br>
<br>
Mirja<br>
<br>
Btw. if this is not a corner case maybe IW of 10 is not a good idea...<br>
<div><div><br>
<br>
On Wednesday 01 February 2012 20:01:49 Matt Mathis wrote:<br>
&gt; First the quick answers: PRR is in Linux 3.2-rc1.<br>
&gt;<br>
&gt; The implementation in Linux uses packet counts, not bytes. =A0Yes this=
 causes<br>
&gt; a number of artifacts.<br>
&gt;<br>
&gt; The transactional nature of the web and video (codic) pacing both caus=
e<br>
&gt; frequent &quot;application stalls&quot; which cause much of our intuit=
ion about how<br>
&gt; TCP typically operates to be somewhat inaccurate.<br>
&gt;<br>
&gt; I believe that all of your suggestions were evaluated empirically and<=
br>
&gt; didn&#39;t pan out. =A0Several alternative algorithms were tested but =
not<br>
&gt; described in the paper. =A0 I can speak to the one that surprised me, =
where<br>
&gt; my intuition initially matched yours, but the numbers just didn&#39;t =
agree.<br>
&gt;<br>
&gt; The issue is SS during recovery. =A0 =A0Consider the situation were th=
e<br>
&gt; cwnd=3DFlightSize=3D10, and you lose 8 packets. =A0Without slowstart d=
uring the<br>
&gt; recovery it would be completed with a window of 2, taking 5 RTTs.<br>
&gt; =A0Furthermore any additional losses are very likely to cause a timeou=
t.<br>
&gt; =A0Compare this to RFC3517, which allows a 3 packet burst, to complete=
 the<br>
&gt; recovery in 2 RTTs.<br>
&gt;<br>
&gt; Although this seems like a corner case, it happens frequently enough t=
o be<br>
&gt; statistically significant in the measurement data. =A0 For PRR, in thi=
s case<br>
&gt; each of the two ACKs will cause 2 segments to be sent (4 total), as do=
es<br>
&gt; the first ACK of the next RTT (thus pipe=3D5). =A0 Note that both RFC3=
517 and<br>
&gt; PRR send 3 extra packets during recovery, either in a burst or<br>
&gt; in response to =A0three different ACKs.<br>
&gt;<br>
&gt; It turns out that due to the risk of additional losses, PRR&#39;s<br>
&gt; average performance is =A0slightly better than RFC3517, even though un=
der<br>
&gt; identical conditions PRR transmits exactly the same amount of data but=
<br>
&gt; slightly later.<br>
&gt;<br>
&gt; If the existing standard had not specify something so aggressive, we m=
ay<br>
&gt; have chosen to to do it as as you suggest, but proposing a<br>
&gt; new standard that performs worse than the existing standard was a<br>
&gt; non-starter.<br>
&gt;<br>
&gt; I am working on an ID that proposes to re-factor TCP congestion contro=
l,<br>
&gt; which will bring the many of your observations into focus.<br>
&gt;<br>
&gt; Re: ECN. =A0Last time we checked it caused a measurable increase in th=
e<br>
&gt; number of failed user requests, probably due to return paths that<br>
&gt; unconditionally discard packets with ECN marks. =A0Note that the forwa=
rd path<br>
&gt; must be delivering them, which seems rather odd. =A0 =A0But yes, we ar=
e<br>
&gt; tinkering with ECN, and would like to help foster its deployment.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; --MM--<br>
&gt; The best way to predict the future is to create it. =A0- Alan Kay<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Wed, Feb 1, 2012 at 9:18 AM, Mirja K=FChlewind &lt;<br>
&gt;<br>
&gt; <a href=3D"mailto:mirja.kuehlewind@ikr.uni-stuttgart.de" target=3D"_bl=
ank">mirja.kuehlewind@ikr.uni-stuttgart.de</a>&gt; wrote:<br>
&gt; &gt; Hi Matt,<br>
&gt; &gt;<br>
&gt; &gt; I reviewed the PRR algorithm. Actually I read your IMC paper and =
not the<br>
&gt; &gt; draft<br>
&gt; &gt; but I hope its the same algorithm ;-)<br>
&gt; &gt;<br>
&gt; &gt; Let me summarize the main points from my understanding and then I=
 have<br>
&gt; &gt; some questions/comments:<br>
&gt; &gt;<br>
&gt; &gt; I think there are 3 changes compared to the current Linux impleme=
ntation:<br>
&gt; &gt;<br>
&gt; &gt; 1) You removed the limitation that the cwnd can&#39;t be larger t=
hat<br>
&gt; &gt; fight_size+1<br>
&gt; &gt; in recovery<br>
&gt; &gt; =A0 -&gt; Agreed, this limitation does not make sense and really =
causes<br>
&gt; &gt; problems<br>
&gt; &gt;<br>
&gt; &gt; 2) Make reduction dependent on congestion control<br>
&gt; &gt; =A0 -&gt; This is an implementational thing. Currently Linux will=
 reduce<br>
&gt; &gt; (hard-coded) on every second ACK which will (eventually) lead to =
a<br>
&gt; &gt; havling of<br>
&gt; &gt; the cwnd. I agree that this decision should be made by the CC alg=
o but<br>
&gt; &gt; its mainly Linux not providing an easy interface to do so.<br>
&gt; &gt;<br>
&gt; &gt; 3) SS like increase in recovery<br>
&gt; &gt; =A0 -&gt; This follows directly from point 2. There is a way to r=
eset the cwnd<br>
&gt; &gt; after recovery to any value but Linux will still limit the cwnd t=
o<br>
&gt; &gt; flight_size+max_burst_size. max_burst_size is (hard-coded) 3 in L=
inux and<br>
&gt; &gt; this limitation make sense to avoid large bursts. Here comes your=
 SS idea<br>
&gt; &gt; into the game. You try to maintain the min cwnd close to ssthresh=
 in<br>
&gt; &gt; recovery<br>
&gt; &gt; to avoid any bursts right after recovery when you set your cwnd t=
o the<br>
&gt; &gt; value<br>
&gt; &gt; determined by the congestion control algorithm.<br>
&gt; &gt;<br>
&gt; &gt; Here my comments:<br>
&gt; &gt;<br>
&gt; &gt; 1) Bursts:<br>
&gt; &gt; As said above the limitation to flight_size+1 is wrong and causes=
 a too<br>
&gt; &gt; conservation behavior of the Linux kernel. But actually the probl=
em is<br>
&gt; &gt; mainly<br>
&gt; &gt; because of the +1. I believe there should still be a limitation o=
f the<br>
&gt; &gt; max burst size (e.g. 3 as used by Linux). In your algorithm it po=
ssible<br>
&gt; &gt; to have very arge burst in under certain circumstances, right?<br=
>
&gt; &gt; At least there should be:<br>
&gt; &gt; =A0 =A0 =A0 =A0At the end of recovery:<br>
&gt; &gt; =A0 =A0 =A0 =A0cwnd =3D min(ssthresh, flight_size + max_burst_siz=
e)<br>
&gt; &gt;<br>
&gt; &gt; 2) SS:<br>
&gt; &gt; The SS thing is a smart idea but I&#39;m not sure if its really n=
eeded. You<br>
&gt; &gt; run into the SS part if you have large burst losses or you are<br=
>
&gt; &gt; application-limited. I would say both cases are worth to be conse=
rvative.<br>
&gt; &gt; As<br>
&gt; &gt; soon as you leave recovery and your cwnd is below ssthresh you wi=
ll be in<br>
&gt; &gt; slow start anyway. You paper is saying in your measurement you we=
re in<br>
&gt; &gt; 32% in<br>
&gt; &gt; the slow start phase at the beginning of recovery. This seems to =
be a<br>
&gt; &gt; really<br>
&gt; &gt; high value. Do you know what&#39;s happening there? Is this due t=
o<br>
&gt; &gt; application limitations or burst losses? I would guess its loss. =
But if<br>
&gt; &gt; you have a lot of<br>
&gt; &gt; small transmission which hardly leave slow start, it might also e=
xplain<br>
&gt; &gt; this<br>
&gt; &gt; number. Maybe both cases even need to be handled differently...?<=
br>
&gt; &gt;<br>
&gt; &gt; 3) DeliveredData:<br>
&gt; &gt; You are using the number of ack&#39;ed bytes. That means you need=
 to have the<br>
&gt; &gt; cwnd<br>
&gt; &gt; and ssthresh also in bytes (not sure if this is said explicitly a=
nyway).<br>
&gt; &gt; The<br>
&gt; &gt; Linux kernel mostly used number of packets and also assumes that =
there<br>
&gt; &gt; will be an ACK for every new packet in recovery. These assumption=
s might<br>
&gt; &gt; be not correct anymore. But I&#39;m not sure if this should be ch=
anged<br>
&gt; &gt; implicit in this<br>
&gt; &gt; proposal. This might be an important topic on its own which might=
 have<br>
&gt; &gt; more implications (if implemented every correctly).<br>
&gt; &gt;<br>
&gt; &gt; And I have two more questions:<br>
&gt; &gt; 2) In the paper you say this algorithm will be in the mainline of=
 the<br>
&gt; &gt; Linux kernel. I checked and I dont think it is there yet...? Do y=
ou have<br>
&gt; &gt; a patch available?<br>
&gt; &gt; 3) And one completely different question: In the paper you say EC=
N is<br>
&gt; &gt; disabled<br>
&gt; &gt; on Google servers. Why is this?<br>
&gt; &gt;<br>
&gt; &gt; Regards,<br>
&gt; &gt; Mirja<br>
<br>
<br>
<br>
</div></div><span><font color=3D"#888888">--<br>
-------------------------------------------------------------------<br>
Dipl.-Ing. Mirja K=FChlewind<br>
Institute of Communication Networks and Computer Engineering (IKR)<br>
University of Stuttgart, Germany<br>
Pfaffenwaldring 47, D-70569 Stuttgart<br>
<br>
tel: <a href=3D"tel:%2B49%280%29711%2F685-67973" value=3D"+4971168567973" t=
arget=3D"_blank">+49(0)711/685-67973</a><br>
email: <a href=3D"mailto:mirja.kuehlewind@ikr.uni-stuttgart.de" target=3D"_=
blank">mirja.kuehlewind@ikr.uni-stuttgart.de</a><br>
web: <a href=3D"http://www.ikr.uni-stuttgart.de" target=3D"_blank">www.ikr.=
uni-stuttgart.de</a><br>
-------------------------------------------------------------------<br>
</font></span></blockquote></div><br></div></div>

--0015174478f2576c6d04b81561d9--

From pasi.sarolahti@iki.fi  Wed Feb  8 05:01:18 2012
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Subject: [tcpm] WGLC for draft-ietf-tcpm-3517bis-01
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Hello,

The authors of "A Conservative Selective Acknowledgment (SACK)-based =
Loss Recovery Algorithm for TCP" (draft-ietf-tcpm-3517bis-01) think the =
latest version addresses all comments made on the draft. Therefore, this =
mail starts Working Group Last Call for the draft, intended to be =
published as Proposed Standard. The WGLC ends on Friday, February 24, =
i.e., within a bit over two weeks.=20

Please read the draft and send any comments to the TCPM mailing list by =
the above deadline. If you think the draft is ok and ready to be =
published without further changes, indicating that would also be =
helpful.

The draft is available at =
http://tools.ietf.org/html/draft-ietf-tcpm-3517bis-01

- Pasi


From abartomeu@swissms.ch  Wed Feb  8 04:15:38 2012
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Hello,

I recently faced the problem of passively estimating the packet loss of =
TCP connection. After a bit of investigating the literature I found that =
there are tens of papers addressing this issue but I couldn't find any =
perfect solution.=20
Since this is an important issue for network operators I have been =
thinking that there would be a very easy solution by just making the two =
TCP connection end points to account for its sent segments and report it =
to the other end point in the FIN segment.
This approach could be used as well for other measurable parameters =
along the TCP session without the need for any other offline method and =
It would be very helpful for detecting failures along networks.=20

Any comments?

Thanks for your time.=

From perfgeek@mac.com  Wed Feb  8 08:32:59 2012
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From: rick jones <perfgeek@mac.com>
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On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:

> Hello,
>=20
> I recently faced the problem of passively estimating the packet loss =
of TCP connection. After a bit of investigating the literature I found =
that there are tens of papers addressing this issue but I couldn't find =
any perfect solution.=20
> Since this is an important issue for network operators I have been =
thinking that there would be a very easy solution by just making the two =
TCP connection end points to account for its sent segments and report it =
to the other end point in the FIN segment.
> This approach could be used as well for other measurable parameters =
along the TCP session without the need for any other offline method and =
It would be very helpful for detecting failures along networks.=20
>=20
> Any comments?

If the count is reported in a FIN segment, isn't that rather like =
closing the barn door after the horses have left?  At least as far as =
doing anything for that connection is concerned.  So, already we are =
talking about operating on timescales larger than connection lifetimes.  =
And what would the side receiving the count-containing FIN do with this =
information? Sure, now it is known on both ends, but the problem still =
lies somewhere in the middle (not always, but often).  Is it expected =
that points along the middle will be sniffing this?  What would they do =
in response?  What could they (reasonably) do in response with that =
information just from TCP vs any other transports?

When a sending TCP sends the first FIN with a count at the end of a MB =
window would it retransmit the FIN, even if the first was received, to =
include retransmissions in that last window's-worth of data?  How long =
would the receiving TCP wait for this "final tally, really" FIN to =
arrive after it sent the ACK to SEQfin to then pass-along the count to a =
higher layer, somehow?

I think we are already asking for enough angels to dance on the TCP =
pin-header.

rick jones=

From ycheng@google.com  Wed Feb  8 08:43:32 2012
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I second that. Even TCP end host can not get this 100%: spurious
retransmission can not be perfectly detected w/ ack path loss for
instance.

On Wed, Feb 8, 2012 at 8:32 AM, rick jones <perfgeek@mac.com> wrote:
>
> On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
>
>> Hello,
>>
>> I recently faced the problem of passively estimating the packet loss of =
TCP connection. After a bit of investigating the literature I found that th=
ere are tens of papers addressing this issue but I couldn't find any perfec=
t solution.
>> Since this is an important issue for network operators I have been think=
ing that there would be a very easy solution by just making the two TCP con=
nection end points to account for its sent segments and report it to the ot=
her end point in the FIN segment.
>> This approach could be used as well for other measurable parameters alon=
g the TCP session without the need for any other offline method and It woul=
d be very helpful for detecting failures along networks.
>>
>> Any comments?
>
> If the count is reported in a FIN segment, isn't that rather like closing=
 the barn door after the horses have left? =A0At least as far as doing anyt=
hing for that connection is concerned. =A0So, already we are talking about =
operating on timescales larger than connection lifetimes. =A0And what would=
 the side receiving the count-containing FIN do with this information? Sure=
, now it is known on both ends, but the problem still lies somewhere in the=
 middle (not always, but often). =A0Is it expected that points along the mi=
ddle will be sniffing this? =A0What would they do in response? =A0What coul=
d they (reasonably) do in response with that information just from TCP vs a=
ny other transports?
>
> When a sending TCP sends the first FIN with a count at the end of a MB wi=
ndow would it retransmit the FIN, even if the first was received, to includ=
e retransmissions in that last window's-worth of data? =A0How long would th=
e receiving TCP wait for this "final tally, really" FIN to arrive after it =
sent the ACK to SEQfin to then pass-along the count to a higher layer, some=
how?
>
> I think we are already asking for enough angels to dance on the TCP pin-h=
eader.
>
> rick jones
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From touch@isi.edu  Wed Feb  8 09:46:29 2012
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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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On 2/8/2012 4:15 AM, Ángel Bartomeu Bonillo wrote:
> Hello,
>
> I recently faced the problem of passively estimating the packet loss
> of TCP connection. After a bit of investigating the literature I
> found that there are tens of papers addressing this issue but I
> couldn't find any perfect solution. Since this is an important issue
> for network operators I have been thinking that there would be a very
> easy solution by just making the two TCP connection end points to
> account for its sent segments and report it to the other end point in
> the FIN segment. This approach could be used as well for other
> measurable parameters along the TCP session without the need for any
> other offline method and It would be very helpful for detecting
> failures along networks.
>
> Any comments?

Regardless of when you do this - either at the end of a connection, or 
during a connection - the ONLY information you can provide is "here is a 
list of segments I have not yet received".

However, that list is irrelevant. Those segments might be in the network 
and may still arrive - after you report, after the connection is over, etc.

There is no way to KNOW that a segment is lost vs. late.

Even when you retransmit a segment (e.g., because it is basically 
overdue, either because your side's timer expired or the other side sent 
either a SACK or ACK indicating a gap), you don't know that the first 
segment was lost - it could arrive after the timer expires.

Even though segments have TTLs, and TTLs are *supposed* to be 
decremented once every second or once every hop (whichever is greater), 
there are devices that queue packets for long periods (e.g., when a link 
is interrupted) and send them on later (even though they should have 
been dropped).

So there is no such information as "lost segments"; the best you know is 
retransmissions.

However, keep in mind that there are lots of TCP connections that are 
terminated without a FIN (e.g., using a RST - I don't think it's 
appropriate in many cases, but the method is in widespread use).

Joe

From abartomeu@swissms.ch  Wed Feb  8 09:53:03 2012
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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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Hi Rick,=20
El 08/02/2012, a las 17:32, rick jones escribi=F3:

>=20
> On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
>=20
>> Hello,
>>=20
>> I recently faced the problem of passively estimating the packet loss =
of TCP connection. After a bit of investigating the literature I found =
that there are tens of papers addressing this issue but I couldn't find =
any perfect solution.=20
>> Since this is an important issue for network operators I have been =
thinking that there would be a very easy solution by just making the two =
TCP connection end points to account for its sent segments and report it =
to the other end point in the FIN segment.
>> This approach could be used as well for other measurable parameters =
along the TCP session without the need for any other offline method and =
It would be very helpful for detecting failures along networks.=20
>>=20
>> Any comments?
>=20
> If the count is reported in a FIN segment, isn't that rather like =
closing the barn door after the horses have left?  At least as far as =
doing anything for that connection is concerned.  So, already we are =
talking about operating on timescales larger than connection lifetimes.  =
And what would the side receiving the count-containing FIN do with this =
information? Sure, now it is known on both ends, but the problem still =
lies somewhere in the middle (not always, but often).  Is it expected =
that points along the middle will be sniffing this?  What would they do =
in response?  What could they (reasonably) do in response with that =
information just from TCP vs any other transports?
>=20

I'm talking about timescales larger than the connection lifetimes. This =
would be a mechanism so TCP can report how many packets it sent out. If =
both ends send at the end of the session the number of packets they sent =
out the other peer and anyone in the middle would be able to estimate de =
loss in both directions.

The idea is to enable TCP so it can report a little but valuable =
information about its performance once the connection finished. This is =
not a mechanism so TCP can adapt to losses in the middle of a =
communication but rather a long term mechanism for network operators and =
such. Nowadays many of the end user services use TCP and the end to end =
packet loss is many times impossible to calculate mainly because only =
the sending node knows the real number of packets sent out. For example =
in the uplink direction on a mobile network.
I think that adding just a couple of bytes (as a TCP option) to the FIN =
segment would give a priceless information for network operators.

> When a sending TCP sends the first FIN with a count at the end of a MB =
window would it retransmit the FIN, even if the first was received, to =
include retransmissions in that last window's-worth of data?  How long =
would the receiving TCP wait for this "final tally, really" FIN to =
arrive after it sent the ACK to SEQfin to then pass-along the count to a =
higher layer, somehow?

The idea much simpler: the peer adds the num of packets sent during the =
connection as an option in the FIN. There is no need to change TCP =
behavior.=20

>=20
> I think we are already asking for enough angels to dance on the TCP =
pin-header.
>=20
> rick jones


From Donald.Smith@CenturyLink.com  Wed Feb  8 10:06:06 2012
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From: "Smith, Donald" <Donald.Smith@CenturyLink.com>
To: "'Joe Touch'" <touch@isi.edu>, =?iso-8859-1?Q?=27=C1ngel_Bartomeu_Bonillo=27?= <abartomeu@swissms.ch>
Date: Wed, 8 Feb 2012 11:05:55 -0700
Thread-Topic: [tcpm] Aid passive estimate of tcp performance
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Just agreeing with Joe, Many applications uses resets instead of a FIN hand=
shake to terminate connections to speed things up especially browsers.


http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf



When packets collide the controllers cease transmission AND wait a random t=
ime before retransmission (mostly)!
Donald.Smith@CenturyLink.com


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Joe Touch
> Sent: Wednesday, February 08, 2012 10:46 AM
> To: =C1ngel Bartomeu Bonillo
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>
>
>
> On 2/8/2012 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
> > Hello,
> >
> > I recently faced the problem of passively estimating the packet loss
> > of TCP connection. After a bit of investigating the literature I
> found
> > that there are tens of papers addressing this issue but I couldn't
> > find any perfect solution. Since this is an important issue for
> > network operators I have been thinking that there would be a very
> easy
> > solution by just making the two TCP connection end points to account
> > for its sent segments and report it to the other end point in the FIN
> > segment. This approach could be used as well for other measurable
> > parameters along the TCP session without the need for any other
> > offline method and It would be very helpful for detecting failures
> > along networks.
> >
> > Any comments?
>
> Regardless of when you do this - either at the end of a connection, or
> during a connection - the ONLY information you can provide is "here is
> a list of segments I have not yet received".
>
> However, that list is irrelevant. Those segments might be in the
> network and may still arrive - after you report, after the connection
> is over, etc.
>
> There is no way to KNOW that a segment is lost vs. late.
>
> Even when you retransmit a segment (e.g., because it is basically
> overdue, either because your side's timer expired or the other side
> sent either a SACK or ACK indicating a gap), you don't know that the
> first segment was lost - it could arrive after the timer expires.
>
> Even though segments have TTLs, and TTLs are *supposed* to be
> decremented once every second or once every hop (whichever is greater),
> there are devices that queue packets for long periods (e.g., when a
> link is interrupted) and send them on later (even though they should
> have been dropped).
>
> So there is no such information as "lost segments"; the best you know
> is retransmissions.
>
> However, keep in mind that there are lots of TCP connections that are
> terminated without a FIN (e.g., using a RST - I don't think it's
> appropriate in many cases, but the method is in widespread use).
>
> Joe
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

This communication is the property of CenturyLink and may contain confident=
ial or privileged information. Unauthorized use of this communication is st=
rictly
prohibited and may be unlawful.  If you have received this communication
in error, please immediately notify the sender by reply e-mail and destroy
all copies of the communication and any attachments.

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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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On Feb 8, 2012, at 9:52, =C1ngel Bartomeu Bonillo wrote:
> I'm talking about timescales larger than the connection lifetimes. =
This would be a mechanism so TCP can report how many packets it sent =
out. If both ends send at the end of the session the number of packets =
they sent out the other peer and anyone in the middle would be able to =
estimate de loss in both directions.

You might want to take a look at what the CONEX WG is trying to do: =
https://datatracker.ietf.org/wg/conex/charter/

Lars=

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From mallman@icir.org  Wed Feb  8 10:23:48 2012
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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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I agree with Rick and the others.  This doesn't seem terribly useful.

> > When a sending TCP sends the first FIN with a count at the end of a
> > MB window would it retransmit the FIN, even if the first was
> > received, to include retransmissions in that last window's-worth of
> > data?  How long would the receiving TCP wait for this "final tally,
> > really" FIN to arrive after it sent the ACK to SEQfin to then
> > pass-along the count to a higher layer, somehow? 
> 
> The idea much simpler: the peer adds the num of packets sent during
> the connection as an option in the FIN. There is no need to change TCP
> behavior.  

I believe what Rick was saying was that you send a ton of packets and
the last one has the FIN bit lit up.  At that point these packets have
no been ACKed.  So, lets say the last window is 100 packets, the 100th
packet has the FIN, and this makes 1000 packets in total the sender has
transmitted.  So, in the FIN we send "1000".  Now, say 10 of the final
100 packets are lost and we have to retransmit those 10 packets.  Now,
the real total number of transmissions is 1010 and the number we stuffed
in the FIN is wrong.  If we want to get that number right then maybe we
hold off on sending the FIN until the final 100 packets are ACKed
(including needed retransmits).  But, that is a change to TCP behavior.
I think that is a relevant example.

Several additional thoughts popped into my head ...

  - This is useless for the TCP connection.  So, what is the incentive
    to bother stuffing it in there?  Some warm and fuzzy lets help the
    operators notion?  I doubt it.

  - This is useless for the TCP connection and so it doesn't make a lot
    of sense to me to add complexity to TCP for no tangible benefit.

  - You are right that passive loss estimators are not "perfect".  But,
    I believe they get you reasonably close.  How "perfect" do you need
    to be for operational purposes?  My strong hunch is that the answer
    is "not perfect at all, the right order would be good".  I.e.,
    operators are going to make different decision when faced with loss
    rates of .1%, 1% and 10%.  But, they are likely going to do the same
    thing with loss rates of .8%, 1%, 1.5% and 2%.  Loss estimators are
    good enough to get you into that sort of ballpark.

  - If the endpoint will give you some information its for sure less
    work than doing the passive monitoring necessary.  But, I am not
    sure reducing some operators' complexity by adding complexity (even
    modestly) to TCP is the right tradeoff.

allman




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To: "mallman@icir.org" <mallman@icir.org>
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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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I just want to add that what the top stack sees today can be much different f=
rom what the network carries... just think of offloading engines that fool t=
he OS declaring a mss MTU of 16kB, then chop the segments into many smaller o=
nes. Or some nasty nat box, or some load balancer in front of some server fa=
rm...
that is, passive tools will have a better picture in those cases...

Inviato da iPad

Il giorno 08/feb/2012, alle ore 19:08, Mark Allman <mallman@icir.org> ha scr=
itto:

>=20
> I agree with Rick and the others.  This doesn't seem terribly useful.
>=20
>>> When a sending TCP sends the first FIN with a count at the end of a
>>> MB window would it retransmit the FIN, even if the first was
>>> received, to include retransmissions in that last window's-worth of
>>> data?  How long would the receiving TCP wait for this "final tally,
>>> really" FIN to arrive after it sent the ACK to SEQfin to then
>>> pass-along the count to a higher layer, somehow?=20
>>=20
>> The idea much simpler: the peer adds the num of packets sent during
>> the connection as an option in the FIN. There is no need to change TCP
>> behavior. =20
>=20
> I believe what Rick was saying was that you send a ton of packets and
> the last one has the FIN bit lit up.  At that point these packets have
> no been ACKed.  So, lets say the last window is 100 packets, the 100th
> packet has the FIN, and this makes 1000 packets in total the sender has
> transmitted.  So, in the FIN we send "1000".  Now, say 10 of the final
> 100 packets are lost and we have to retransmit those 10 packets.  Now,
> the real total number of transmissions is 1010 and the number we stuffed
> in the FIN is wrong.  If we want to get that number right then maybe we
> hold off on sending the FIN until the final 100 packets are ACKed
> (including needed retransmits).  But, that is a change to TCP behavior.
> I think that is a relevant example.
>=20
> Several additional thoughts popped into my head ...
>=20
>  - This is useless for the TCP connection.  So, what is the incentive
>    to bother stuffing it in there?  Some warm and fuzzy lets help the
>    operators notion?  I doubt it.
>=20
>  - This is useless for the TCP connection and so it doesn't make a lot
>    of sense to me to add complexity to TCP for no tangible benefit.
>=20
>  - You are right that passive loss estimators are not "perfect".  But,
>    I believe they get you reasonably close.  How "perfect" do you need
>    to be for operational purposes?  My strong hunch is that the answer
>    is "not perfect at all, the right order would be good".  I.e.,
>    operators are going to make different decision when faced with loss
>    rates of .1%, 1% and 10%.  But, they are likely going to do the same
>    thing with loss rates of .8%, 1%, 1.5% and 2%.  Loss estimators are
>    good enough to get you into that sort of ballpark.
>=20
>  - If the endpoint will give you some information its for sure less
>    work than doing the passive monitoring necessary.  But, I am not
>    sure reducing some operators' complexity by adding complexity (even
>    modestly) to TCP is the right tradeoff.
>=20
> allman
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From touch@isi.edu  Wed Feb  8 10:58:06 2012
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On 2/8/2012 10:05 AM, Smith, Donald wrote:
> Just agreeing with Joe, Many applications uses resets instead of a FIN handshake to terminate connections to speed things up especially browsers.
>
>
> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf

It doesn't just "speed" things; it's reckless and dangerous, esp. since 
the parties tossing around these RSTs typically don't maintain TIME-WAIT 
before reusing those connection IDs.

Joe

>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
>> Joe Touch
>> Sent: Wednesday, February 08, 2012 10:46 AM
>> To: Ángel Bartomeu Bonillo
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>
>>
>>
>> On 2/8/2012 4:15 AM, Ángel Bartomeu Bonillo wrote:
>>> Hello,
>>>
>>> I recently faced the problem of passively estimating the packet loss
>>> of TCP connection. After a bit of investigating the literature I
>> found
>>> that there are tens of papers addressing this issue but I couldn't
>>> find any perfect solution. Since this is an important issue for
>>> network operators I have been thinking that there would be a very
>> easy
>>> solution by just making the two TCP connection end points to account
>>> for its sent segments and report it to the other end point in the FIN
>>> segment. This approach could be used as well for other measurable
>>> parameters along the TCP session without the need for any other
>>> offline method and It would be very helpful for detecting failures
>>> along networks.
>>>
>>> Any comments?
>>
>> Regardless of when you do this - either at the end of a connection, or
>> during a connection - the ONLY information you can provide is "here is
>> a list of segments I have not yet received".
>>
>> However, that list is irrelevant. Those segments might be in the
>> network and may still arrive - after you report, after the connection
>> is over, etc.
>>
>> There is no way to KNOW that a segment is lost vs. late.
>>
>> Even when you retransmit a segment (e.g., because it is basically
>> overdue, either because your side's timer expired or the other side
>> sent either a SACK or ACK indicating a gap), you don't know that the
>> first segment was lost - it could arrive after the timer expires.
>>
>> Even though segments have TTLs, and TTLs are *supposed* to be
>> decremented once every second or once every hop (whichever is greater),
>> there are devices that queue packets for long periods (e.g., when a
>> link is interrupted) and send them on later (even though they should
>> have been dropped).
>>
>> So there is no such information as "lost segments"; the best you know
>> is retransmissions.
>>
>> However, keep in mind that there are lots of TCP connections that are
>> terminated without a FIN (e.g., using a RST - I don't think it's
>> appropriate in many cases, but the method is in widespread use).
>>
>> Joe
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>
> This communication is the property of CenturyLink and may contain confidential or privileged information. Unauthorized use of this communication is strictly
> prohibited and may be unlawful.  If you have received this communication
> in error, please immediately notify the sender by reply e-mail and destroy
> all copies of the communication and any attachments.

From touch@isi.edu  Wed Feb  8 11:00:38 2012
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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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Hi, all,

On 2/8/2012 11:01 AM, Marco Mellia wrote:
> I just want to add that what the top stack sees today can be much different from what the network carries... just think of offloading engines that fool the OS declaring a mss MTU of 16kB, then chop the segments into many smaller ones. Or some nasty nat box, or some load balancer in front of some server farm...
> that is, passive tools will have a better picture in those cases...

It's impossible to account for forgers; if a proxy terminates a 
connection for an IP address it doesn't own, all bets are off for active 
*and* passive analysis.

Joe

>
> Inviato da iPad
>
> Il giorno 08/feb/2012, alle ore 19:08, Mark Allman<mallman@icir.org>  ha scritto:
>
>>
>> I agree with Rick and the others.  This doesn't seem terribly useful.
>>
>>>> When a sending TCP sends the first FIN with a count at the end of a
>>>> MB window would it retransmit the FIN, even if the first was
>>>> received, to include retransmissions in that last window's-worth of
>>>> data?  How long would the receiving TCP wait for this "final tally,
>>>> really" FIN to arrive after it sent the ACK to SEQfin to then
>>>> pass-along the count to a higher layer, somehow?
>>>
>>> The idea much simpler: the peer adds the num of packets sent during
>>> the connection as an option in the FIN. There is no need to change TCP
>>> behavior.
>>
>> I believe what Rick was saying was that you send a ton of packets and
>> the last one has the FIN bit lit up.  At that point these packets have
>> no been ACKed.  So, lets say the last window is 100 packets, the 100th
>> packet has the FIN, and this makes 1000 packets in total the sender has
>> transmitted.  So, in the FIN we send "1000".  Now, say 10 of the final
>> 100 packets are lost and we have to retransmit those 10 packets.  Now,
>> the real total number of transmissions is 1010 and the number we stuffed
>> in the FIN is wrong.  If we want to get that number right then maybe we
>> hold off on sending the FIN until the final 100 packets are ACKed
>> (including needed retransmits).  But, that is a change to TCP behavior.
>> I think that is a relevant example.
>>
>> Several additional thoughts popped into my head ...
>>
>>   - This is useless for the TCP connection.  So, what is the incentive
>>     to bother stuffing it in there?  Some warm and fuzzy lets help the
>>     operators notion?  I doubt it.
>>
>>   - This is useless for the TCP connection and so it doesn't make a lot
>>     of sense to me to add complexity to TCP for no tangible benefit.
>>
>>   - You are right that passive loss estimators are not "perfect".  But,
>>     I believe they get you reasonably close.  How "perfect" do you need
>>     to be for operational purposes?  My strong hunch is that the answer
>>     is "not perfect at all, the right order would be good".  I.e.,
>>     operators are going to make different decision when faced with loss
>>     rates of .1%, 1% and 10%.  But, they are likely going to do the same
>>     thing with loss rates of .8%, 1%, 1.5% and 2%.  Loss estimators are
>>     good enough to get you into that sort of ballpark.
>>
>>   - If the endpoint will give you some information its for sure less
>>     work than doing the passive monitoring necessary.  But, I am not
>>     sure reducing some operators' complexity by adding complexity (even
>>     modestly) to TCP is the right tradeoff.
>>
>> 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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On 02/08/2012 01:56 PM, Joe Touch wrote:
>
>
> On 2/8/2012 10:05 AM, Smith, Donald wrote:
>> Just agreeing with Joe, Many applications uses resets instead of a
>> FIN handshake to terminate connections to speed things up especially
>> browsers.
>>
>>
>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
>
> It doesn't just "speed" things; it's reckless and dangerous, esp.
> since the parties tossing around these RSTs typically don't maintain
> TIME-WAIT before reusing those connection IDs.

Joe, browsers have done this forever.

If you don't like it, then it's (long past time) to make the case that
they should change their behaviour.
                        - Jim

>
> Joe
>
>>> -----Original Message-----
>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
>>> Joe Touch
>>> Sent: Wednesday, February 08, 2012 10:46 AM
>>> To: Ángel Bartomeu Bonillo
>>> Cc: tcpm@ietf.org
>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>>
>>>
>>>
>>> On 2/8/2012 4:15 AM, Ángel Bartomeu Bonillo wrote:
>>>> Hello,
>>>>
>>>> I recently faced the problem of passively estimating the packet loss
>>>> of TCP connection. After a bit of investigating the literature I
>>> found
>>>> that there are tens of papers addressing this issue but I couldn't
>>>> find any perfect solution. Since this is an important issue for
>>>> network operators I have been thinking that there would be a very
>>> easy
>>>> solution by just making the two TCP connection end points to account
>>>> for its sent segments and report it to the other end point in the FIN
>>>> segment. This approach could be used as well for other measurable
>>>> parameters along the TCP session without the need for any other
>>>> offline method and It would be very helpful for detecting failures
>>>> along networks.
>>>>
>>>> Any comments?
>>>
>>> Regardless of when you do this - either at the end of a connection, or
>>> during a connection - the ONLY information you can provide is "here is
>>> a list of segments I have not yet received".
>>>
>>> However, that list is irrelevant. Those segments might be in the
>>> network and may still arrive - after you report, after the connection
>>> is over, etc.
>>>
>>> There is no way to KNOW that a segment is lost vs. late.
>>>
>>> Even when you retransmit a segment (e.g., because it is basically
>>> overdue, either because your side's timer expired or the other side
>>> sent either a SACK or ACK indicating a gap), you don't know that the
>>> first segment was lost - it could arrive after the timer expires.
>>>
>>> Even though segments have TTLs, and TTLs are *supposed* to be
>>> decremented once every second or once every hop (whichever is greater),
>>> there are devices that queue packets for long periods (e.g., when a
>>> link is interrupted) and send them on later (even though they should
>>> have been dropped).
>>>
>>> So there is no such information as "lost segments"; the best you know
>>> is retransmissions.
>>>
>>> However, keep in mind that there are lots of TCP connections that are
>>> terminated without a FIN (e.g., using a RST - I don't think it's
>>> appropriate in many cases, but the method is in widespread use).
>>>
>>> Joe
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>> This communication is the property of CenturyLink and may contain
>> confidential or privileged information. Unauthorized use of this
>> communication is strictly
>> prohibited and may be unlawful.  If you have received this communication
>> in error, please immediately notify the sender by reply e-mail and
>> destroy
>> all copies of the communication and any attachments.
> _______________________________________________
> tcpm mailing list
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> https://www.ietf.org/mailman/listinfo/tcpm


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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Jim,

Routers have been over-buffering forever too. Yet you continue to want 
to get them to change things and correctly implement the standards.

That's what I want for endpoints with TCP too.

Ignorance or deliberate defiance is not an excuse. These are errors that 
need to be corrected, since they have real implications throughout the 
network, both for users on those systems and those that share those 
addresses (e.g., behind shared NATs or proxies).

Joe

On 2/8/2012 11:03 AM, Jim Gettys wrote:
> On 02/08/2012 01:56 PM, Joe Touch wrote:
>>
>>
>> On 2/8/2012 10:05 AM, Smith, Donald wrote:
>>> Just agreeing with Joe, Many applications uses resets instead of a
>>> FIN handshake to terminate connections to speed things up especially
>>> browsers.
>>>
>>>
>>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
>>
>> It doesn't just "speed" things; it's reckless and dangerous, esp.
>> since the parties tossing around these RSTs typically don't maintain
>> TIME-WAIT before reusing those connection IDs.
>
> Joe, browsers have done this forever.
>
> If you don't like it, then it's (long past time) to make the case that
> they should change their behaviour.
>                          - Jim
>
>>
>> Joe
>>
>>>> -----Original Message-----
>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
>>>> Joe Touch
>>>> Sent: Wednesday, February 08, 2012 10:46 AM
>>>> To: Ángel Bartomeu Bonillo
>>>> Cc: tcpm@ietf.org
>>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>>>
>>>>
>>>>
>>>> On 2/8/2012 4:15 AM, Ángel Bartomeu Bonillo wrote:
>>>>> Hello,
>>>>>
>>>>> I recently faced the problem of passively estimating the packet loss
>>>>> of TCP connection. After a bit of investigating the literature I
>>>> found
>>>>> that there are tens of papers addressing this issue but I couldn't
>>>>> find any perfect solution. Since this is an important issue for
>>>>> network operators I have been thinking that there would be a very
>>>> easy
>>>>> solution by just making the two TCP connection end points to account
>>>>> for its sent segments and report it to the other end point in the FIN
>>>>> segment. This approach could be used as well for other measurable
>>>>> parameters along the TCP session without the need for any other
>>>>> offline method and It would be very helpful for detecting failures
>>>>> along networks.
>>>>>
>>>>> Any comments?
>>>>
>>>> Regardless of when you do this - either at the end of a connection, or
>>>> during a connection - the ONLY information you can provide is "here is
>>>> a list of segments I have not yet received".
>>>>
>>>> However, that list is irrelevant. Those segments might be in the
>>>> network and may still arrive - after you report, after the connection
>>>> is over, etc.
>>>>
>>>> There is no way to KNOW that a segment is lost vs. late.
>>>>
>>>> Even when you retransmit a segment (e.g., because it is basically
>>>> overdue, either because your side's timer expired or the other side
>>>> sent either a SACK or ACK indicating a gap), you don't know that the
>>>> first segment was lost - it could arrive after the timer expires.
>>>>
>>>> Even though segments have TTLs, and TTLs are *supposed* to be
>>>> decremented once every second or once every hop (whichever is greater),
>>>> there are devices that queue packets for long periods (e.g., when a
>>>> link is interrupted) and send them on later (even though they should
>>>> have been dropped).
>>>>
>>>> So there is no such information as "lost segments"; the best you know
>>>> is retransmissions.
>>>>
>>>> However, keep in mind that there are lots of TCP connections that are
>>>> terminated without a FIN (e.g., using a RST - I don't think it's
>>>> appropriate in many cases, but the method is in widespread use).
>>>>
>>>> Joe
>>>> _______________________________________________
>>>> tcpm mailing list
>>>> tcpm@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>> This communication is the property of CenturyLink and may contain
>>> confidential or privileged information. Unauthorized use of this
>>> communication is strictly
>>> prohibited and may be unlawful.  If you have received this communication
>>> in error, please immediately notify the sender by reply e-mail and
>>> destroy
>>> all copies of the communication and any attachments.
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm

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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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On 02/08/2012 02:08 PM, Joe Touch wrote:
> Jim,
>
> Routers have been over-buffering forever too. Yet you continue to want
> to get them to change things and correctly implement the standards.
>
> That's what I want for endpoints with TCP too.
>
> Ignorance or deliberate defiance is not an excuse. These are errors
> that need to be corrected, since they have real implications
> throughout the network, both for users on those systems and those that
> share those addresses (e.g., behind shared NATs or proxies).

Over the years, I have noted a distinct lack of communications between
those who understand how the bits flow, and how they are used....  For
example, the transient bufferbloat phenomena is something that is pretty
obvious to those who understand TCP, and completely unobvious to most
who do web browsers...

I believe both groups need to talk to each other, strangely.  Sometimes
you may find that the guys who do the applications actually have needs
not being met by standards, that are well worth understanding; and
conversely, the guys who do the apps may find that life is better for
both them and others if they understand the standards.

Just go in with an open mind and say: "We observe the browsers/servers
doing the following thing that we believe is much less than optimal, why
are you doing this?" rather than "You are disobeying the rules, fix it",
and I think you'll get a warm welcome.
                - Jim


>
> Joe
>
> On 2/8/2012 11:03 AM, Jim Gettys wrote:
>> On 02/08/2012 01:56 PM, Joe Touch wrote:
>>>
>>>
>>> On 2/8/2012 10:05 AM, Smith, Donald wrote:
>>>> Just agreeing with Joe, Many applications uses resets instead of a
>>>> FIN handshake to terminate connections to speed things up especially
>>>> browsers.
>>>>
>>>>
>>>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
>>>
>>> It doesn't just "speed" things; it's reckless and dangerous, esp.
>>> since the parties tossing around these RSTs typically don't maintain
>>> TIME-WAIT before reusing those connection IDs.
>>
>> Joe, browsers have done this forever.
>>
>> If you don't like it, then it's (long past time) to make the case that
>> they should change their behaviour.
>>                          - Jim
>>
>>>
>>> Joe
>>>
>>>>> -----Original Message-----
>>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
>>>>> Behalf Of
>>>>> Joe Touch
>>>>> Sent: Wednesday, February 08, 2012 10:46 AM
>>>>> To: Ángel Bartomeu Bonillo
>>>>> Cc: tcpm@ietf.org
>>>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>>>>
>>>>>
>>>>>
>>>>> On 2/8/2012 4:15 AM, Ángel Bartomeu Bonillo wrote:
>>>>>> Hello,
>>>>>>
>>>>>> I recently faced the problem of passively estimating the packet loss
>>>>>> of TCP connection. After a bit of investigating the literature I
>>>>> found
>>>>>> that there are tens of papers addressing this issue but I couldn't
>>>>>> find any perfect solution. Since this is an important issue for
>>>>>> network operators I have been thinking that there would be a very
>>>>> easy
>>>>>> solution by just making the two TCP connection end points to account
>>>>>> for its sent segments and report it to the other end point in the
>>>>>> FIN
>>>>>> segment. This approach could be used as well for other measurable
>>>>>> parameters along the TCP session without the need for any other
>>>>>> offline method and It would be very helpful for detecting failures
>>>>>> along networks.
>>>>>>
>>>>>> Any comments?
>>>>>
>>>>> Regardless of when you do this - either at the end of a
>>>>> connection, or
>>>>> during a connection - the ONLY information you can provide is
>>>>> "here is
>>>>> a list of segments I have not yet received".
>>>>>
>>>>> However, that list is irrelevant. Those segments might be in the
>>>>> network and may still arrive - after you report, after the connection
>>>>> is over, etc.
>>>>>
>>>>> There is no way to KNOW that a segment is lost vs. late.
>>>>>
>>>>> Even when you retransmit a segment (e.g., because it is basically
>>>>> overdue, either because your side's timer expired or the other side
>>>>> sent either a SACK or ACK indicating a gap), you don't know that the
>>>>> first segment was lost - it could arrive after the timer expires.
>>>>>
>>>>> Even though segments have TTLs, and TTLs are *supposed* to be
>>>>> decremented once every second or once every hop (whichever is
>>>>> greater),
>>>>> there are devices that queue packets for long periods (e.g., when a
>>>>> link is interrupted) and send them on later (even though they should
>>>>> have been dropped).
>>>>>
>>>>> So there is no such information as "lost segments"; the best you know
>>>>> is retransmissions.
>>>>>
>>>>> However, keep in mind that there are lots of TCP connections that are
>>>>> terminated without a FIN (e.g., using a RST - I don't think it's
>>>>> appropriate in many cases, but the method is in widespread use).
>>>>>
>>>>> Joe
>>>>> _______________________________________________
>>>>> tcpm mailing list
>>>>> tcpm@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>>
>>>> This communication is the property of CenturyLink and may contain
>>>> confidential or privileged information. Unauthorized use of this
>>>> communication is strictly
>>>> prohibited and may be unlawful.  If you have received this
>>>> communication
>>>> in error, please immediately notify the sender by reply e-mail and
>>>> destroy
>>>> all copies of the communication and any attachments.
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm


From ananth@cisco.com  Wed Feb  8 11:39:18 2012
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Thread-Topic: [tcpm] Aid passive estimate of tcp performance
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References: <920E9EEF-0BD8-4CB4-B385-149FCBC799E1@swissms.ch><4F32B4C7.6070102@isi.edu><B01905DA0C7CDC478F42870679DF0F1010EABF0A49@qtdenexmbm24.AD.QINTRA.COM><4F32C573.6090606@isi.edu> <4F32C6EB.5030404@freedesktop.org> <4F32C843.3010405@isi.edu>
From: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
To: "Joe Touch" <touch@isi.edu>, "Jim Gettys" <jg@freedesktop.org>
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Cc: tcpm@ietf.org
Subject: Re: [tcpm] Aid passive estimate of tcp performance
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Well,
   Some assumptions that existed long ago, are no longer serious or may =
be not hold good for the modern internet. Lets take the example of =
TIMEWAIT state which you mention below.  RFC recommends 2*MSL which =
translates to 4 mins. Many OS'es (Windows, Linux ) and Cisco IOS etc., =
have configurable knobs to tune this value to anywhere between 30 secs =
to 240 secs.
   Lets take up the initial window case, it was 1 segment 30 years ago, =
got increased to 3 segments and now it's on it way to hit the 10 =
segments mark.  So things keep changing and the "degree of conservatism" =
changes to adopt to the current internet.
   So, you just can't say these are erroneous or mis-interpretations, it =
is just the way things have moved on (IMO for good in most cases)
-Anantha=20

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf =
Of
> Joe Touch
> Sent: Wednesday, February 08, 2012 11:09 AM
> To: Jim Gettys
> Cc: 'tcpm@ietf.org'
> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>=20
> Jim,
>=20
> Routers have been over-buffering forever too. Yet you continue to want
> to get them to change things and correctly implement the standards.
>=20
> That's what I want for endpoints with TCP too.
>=20
> Ignorance or deliberate defiance is not an excuse. These are errors
> that
> need to be corrected, since they have real implications throughout the
> network, both for users on those systems and those that share those
> addresses (e.g., behind shared NATs or proxies).
>=20
> Joe
>=20
> On 2/8/2012 11:03 AM, Jim Gettys wrote:
> > On 02/08/2012 01:56 PM, Joe Touch wrote:
> >>
> >>
> >> On 2/8/2012 10:05 AM, Smith, Donald wrote:
> >>> Just agreeing with Joe, Many applications uses resets instead of a
> >>> FIN handshake to terminate connections to speed things up
> especially
> >>> browsers.
> >>>
> >>>
> >>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
> >>
> >> It doesn't just "speed" things; it's reckless and dangerous, esp.
> >> since the parties tossing around these RSTs typically don't =
maintain
> >> TIME-WAIT before reusing those connection IDs.
> >
> > Joe, browsers have done this forever.
> >
> > If you don't like it, then it's (long past time) to make the case
> that
> > they should change their behaviour.
> >                          - Jim
> >
> >>
> >> Joe
> >>
> >>>> -----Original Message-----
> >>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
> Behalf Of
> >>>> Joe Touch
> >>>> Sent: Wednesday, February 08, 2012 10:46 AM
> >>>> To: =C1ngel Bartomeu Bonillo
> >>>> Cc: tcpm@ietf.org
> >>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
> >>>>
> >>>>
> >>>>
> >>>> On 2/8/2012 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
> >>>>> Hello,
> >>>>>
> >>>>> I recently faced the problem of passively estimating the packet
> loss
> >>>>> of TCP connection. After a bit of investigating the literature I
> >>>> found
> >>>>> that there are tens of papers addressing this issue but I
> couldn't
> >>>>> find any perfect solution. Since this is an important issue for
> >>>>> network operators I have been thinking that there would be a =
very
> >>>> easy
> >>>>> solution by just making the two TCP connection end points to
> account
> >>>>> for its sent segments and report it to the other end point in =
the
> FIN
> >>>>> segment. This approach could be used as well for other =
measurable
> >>>>> parameters along the TCP session without the need for any other
> >>>>> offline method and It would be very helpful for detecting
> failures
> >>>>> along networks.
> >>>>>
> >>>>> Any comments?
> >>>>
> >>>> Regardless of when you do this - either at the end of a
> connection, or
> >>>> during a connection - the ONLY information you can provide is
> "here is
> >>>> a list of segments I have not yet received".
> >>>>
> >>>> However, that list is irrelevant. Those segments might be in the
> >>>> network and may still arrive - after you report, after the
> connection
> >>>> is over, etc.
> >>>>
> >>>> There is no way to KNOW that a segment is lost vs. late.
> >>>>
> >>>> Even when you retransmit a segment (e.g., because it is basically
> >>>> overdue, either because your side's timer expired or the other
> side
> >>>> sent either a SACK or ACK indicating a gap), you don't know that
> the
> >>>> first segment was lost - it could arrive after the timer expires.
> >>>>
> >>>> Even though segments have TTLs, and TTLs are *supposed* to be
> >>>> decremented once every second or once every hop (whichever is
> greater),
> >>>> there are devices that queue packets for long periods (e.g., when
> a
> >>>> link is interrupted) and send them on later (even though they
> should
> >>>> have been dropped).
> >>>>
> >>>> So there is no such information as "lost segments"; the best you
> know
> >>>> is retransmissions.
> >>>>
> >>>> However, keep in mind that there are lots of TCP connections that
> are
> >>>> terminated without a FIN (e.g., using a RST - I don't think it's
> >>>> appropriate in many cases, but the method is in widespread use).
> >>>>
> >>>> Joe
> >>>> _______________________________________________
> >>>> tcpm mailing list
> >>>> tcpm@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/tcpm
> >>>
> >>> This communication is the property of CenturyLink and may contain
> >>> confidential or privileged information. Unauthorized use of this
> >>> communication is strictly
> >>> prohibited and may be unlawful.  If you have received this
> communication
> >>> in error, please immediately notify the sender by reply e-mail and
> >>> destroy
> >>> all copies of the communication and any attachments.
> >> _______________________________________________
> >> tcpm mailing list
> >> tcpm@ietf.org
> >> https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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From: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
To: "Jim Gettys" <jg@freedesktop.org>, "Joe Touch" <touch@isi.edu>
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+ 1

> Just go in with an open mind and say: "We observe the browsers/servers
> doing the following thing that we believe is much less than optimal,
> why
> are you doing this?" rather than "You are disobeying the rules, fix
> it",
> and I think you'll get a warm welcome.
>                 - Jim

We have been through this before it is sad that some folks use the
argument of "there is a high bar for changing TCP" and try to bury ideas
which have not been standaridized and have not given a chance to get
standardized. Linux TCP has many things (good) which are not IETF
standards. Of course other OS'es as well.

I am afraid all these are a result of the latter attitude. I thought a
few times before writing this email, but alas I have a better feeling
now even though in some sense it is a repeat and a viewpoint which is
shared by a considerable number of folks.

-Anantha

From touch@isi.edu  Wed Feb  8 12:55:09 2012
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From: Joe Touch <touch@isi.edu>
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On 2/8/2012 11:39 AM, Anantha Ramaiah (ananth) wrote:
> Well,
> Some assumptions that existed long ago, are no longer serious or may
> be not hold good for the modern internet.

Where that's true, it's worth re-examining.

 > Lets take the example of
> TIMEWAIT state which you mention below. RFC recommends 2*MSL which
> translates to 4 mins. Many OS'es (Windows, Linux ) and Cisco IOS etc.,
> have configurable knobs to tune this value to anywhere between 30 secs
> to 240 secs.

It's worth understanding why. It's not with a sense that this was safe 
per-se; it was a sense of improving performance. That's a bad reason to 
make a change, especially when it's ignorant of the rationale for the 
safety mechanism being avoided.

> Lets take up the initial window case, it was 1 segment 30 years ago,
> got increased to 3 segments and now it's on it way to hit the 10
> segments mark. So things keep changing and the "degree of conservatism"
> changes to adopt to the current internet.

Yes, and that was done by changing the specs.

> So, you just can't say these are erroneous or mis-interpretations,
> it is just the way things have moved on (IMO for good in most cases)

My specific experience is with the interaction between TCP and HTTP. 
HTTP was designed because of the ignorance of the rationale of the 
design of FTP - notably that it wasn't useful to open two connections to 
get a file when only one file per location would be useful (this is 
paraphrased from Berners-Lee's original papers). HTTP incurred long 
(200ms) delays when first deployed because of a similar ignorance about 
how application protocols interact with TCP - that interactive protocols 
with multi-byte exchanges must have Nagle off.

There is a similar issue with RSTs. They're useful to terminate 
connections when that is the desired intent - an abort. However, they're 
overused by browsers for other reasons - e.g., when switching pages.

There are reasons to change TCP, and to have it evolve; there are also 
reasons to find and declare variations "non-compliant" bugs and fix 
them. The distinction should be made with a full understanding of what 
TCP is *already* capable of, and the implications of these changes.

Joe

>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
>> Joe Touch
>> Sent: Wednesday, February 08, 2012 11:09 AM
>> To: Jim Gettys
>> Cc: 'tcpm@ietf.org'
>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>
>> Jim,
>>
>> Routers have been over-buffering forever too. Yet you continue to want
>> to get them to change things and correctly implement the standards.
>>
>> That's what I want for endpoints with TCP too.
>>
>> Ignorance or deliberate defiance is not an excuse. These are errors
>> that
>> need to be corrected, since they have real implications throughout the
>> network, both for users on those systems and those that share those
>> addresses (e.g., behind shared NATs or proxies).
>>
>> Joe
>>
>> On 2/8/2012 11:03 AM, Jim Gettys wrote:
>>> On 02/08/2012 01:56 PM, Joe Touch wrote:
>>>>
>>>>
>>>> On 2/8/2012 10:05 AM, Smith, Donald wrote:
>>>>> Just agreeing with Joe, Many applications uses resets instead of a
>>>>> FIN handshake to terminate connections to speed things up
>> especially
>>>>> browsers.
>>>>>
>>>>>
>>>>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
>>>>
>>>> It doesn't just "speed" things; it's reckless and dangerous, esp.
>>>> since the parties tossing around these RSTs typically don't maintain
>>>> TIME-WAIT before reusing those connection IDs.
>>>
>>> Joe, browsers have done this forever.
>>>
>>> If you don't like it, then it's (long past time) to make the case
>> that
>>> they should change their behaviour.
>>>                           - Jim
>>>
>>>>
>>>> Joe
>>>>
>>>>>> -----Original Message-----
>>>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
>> Behalf Of
>>>>>> Joe Touch
>>>>>> Sent: Wednesday, February 08, 2012 10:46 AM
>>>>>> To: Ángel Bartomeu Bonillo
>>>>>> Cc: tcpm@ietf.org
>>>>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 2/8/2012 4:15 AM, Ángel Bartomeu Bonillo wrote:
>>>>>>> Hello,
>>>>>>>
>>>>>>> I recently faced the problem of passively estimating the packet
>> loss
>>>>>>> of TCP connection. After a bit of investigating the literature I
>>>>>> found
>>>>>>> that there are tens of papers addressing this issue but I
>> couldn't
>>>>>>> find any perfect solution. Since this is an important issue for
>>>>>>> network operators I have been thinking that there would be a very
>>>>>> easy
>>>>>>> solution by just making the two TCP connection end points to
>> account
>>>>>>> for its sent segments and report it to the other end point in the
>> FIN
>>>>>>> segment. This approach could be used as well for other measurable
>>>>>>> parameters along the TCP session without the need for any other
>>>>>>> offline method and It would be very helpful for detecting
>> failures
>>>>>>> along networks.
>>>>>>>
>>>>>>> Any comments?
>>>>>>
>>>>>> Regardless of when you do this - either at the end of a
>> connection, or
>>>>>> during a connection - the ONLY information you can provide is
>> "here is
>>>>>> a list of segments I have not yet received".
>>>>>>
>>>>>> However, that list is irrelevant. Those segments might be in the
>>>>>> network and may still arrive - after you report, after the
>> connection
>>>>>> is over, etc.
>>>>>>
>>>>>> There is no way to KNOW that a segment is lost vs. late.
>>>>>>
>>>>>> Even when you retransmit a segment (e.g., because it is basically
>>>>>> overdue, either because your side's timer expired or the other
>> side
>>>>>> sent either a SACK or ACK indicating a gap), you don't know that
>> the
>>>>>> first segment was lost - it could arrive after the timer expires.
>>>>>>
>>>>>> Even though segments have TTLs, and TTLs are *supposed* to be
>>>>>> decremented once every second or once every hop (whichever is
>> greater),
>>>>>> there are devices that queue packets for long periods (e.g., when
>> a
>>>>>> link is interrupted) and send them on later (even though they
>> should
>>>>>> have been dropped).
>>>>>>
>>>>>> So there is no such information as "lost segments"; the best you
>> know
>>>>>> is retransmissions.
>>>>>>
>>>>>> However, keep in mind that there are lots of TCP connections that
>> are
>>>>>> terminated without a FIN (e.g., using a RST - I don't think it's
>>>>>> appropriate in many cases, but the method is in widespread use).
>>>>>>
>>>>>> Joe
>>>>>> _______________________________________________
>>>>>> tcpm mailing list
>>>>>> tcpm@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>>>
>>>>> This communication is the property of CenturyLink and may contain
>>>>> confidential or privileged information. Unauthorized use of this
>>>>> communication is strictly
>>>>> prohibited and may be unlawful.  If you have received this
>> communication
>>>>> in error, please immediately notify the sender by reply e-mail and
>>>>> destroy
>>>>> all copies of the communication and any attachments.
>>>> _______________________________________________
>>>> 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 Donald.Smith@CenturyLink.com  Wed Feb  8 13:50:43 2012
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From: "Smith, Donald" <Donald.Smith@CenturyLink.com>
To: "'Joe Touch'" <touch@isi.edu>, "'Anantha Ramaiah (ananth)'" <ananth@cisco.com>
Date: Wed, 8 Feb 2012 14:50:31 -0700
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When packets collide the controllers cease transmission AND wait a random t=
ime before retransmission (mostly)!
Donald.Smith@CenturyLink.com


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Joe Touch
> Sent: Wednesday, February 08, 2012 1:54 PM
> To: Anantha Ramaiah (ananth)
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>
>
>
> On 2/8/2012 11:39 AM, Anantha Ramaiah (ananth) wrote:
> > Well,
> > Some assumptions that existed long ago, are no longer serious or may
> > be not hold good for the modern internet.
>
> Where that's true, it's worth re-examining.
>
>  > Lets take the example of
> > TIMEWAIT state which you mention below. RFC recommends 2*MSL which
> > translates to 4 mins. Many OS'es (Windows, Linux ) and Cisco IOS
> etc.,
> > have configurable knobs to tune this value to anywhere between 30
> secs
> > to 240 secs.
>
> It's worth understanding why. It's not with a sense that this was safe
> per-se; it was a sense of improving performance. That's a bad reason to
> make a change, especially when it's ignorant of the rationale for the
> safety mechanism being avoided.
It depends on what you mean by safe. If your trying to survive a ddos attac=
k making those timers very low helps quite a bit and doesn't seem to affect=
 the clients.

In what transport today does a open take more than a second or two?
Same for a close with FIN those normally complete in a second or two.
I realize some of these recommendations were written when a 4800baud was a =
high speed link.

I can see where a satalite bounce might take a few seconds but even that is=
n't measured in minutes.

If we made new recommendations (not changing ratios just changing the value=
s by a factor of 10 or so) maybe we could convince the browser vendors and =
web server vendors to not use RESETS the way so many of them do today.


>
> > Lets take up the initial window case, it was 1 segment 30 years ago,
> > got increased to 3 segments and now it's on it way to hit the 10
> > segments mark. So things keep changing and the "degree of
> conservatism"
> > changes to adopt to the current internet.
>
> Yes, and that was done by changing the specs.
>
> > So, you just can't say these are erroneous or mis-interpretations, it
> > is just the way things have moved on (IMO for good in most cases)
>
> My specific experience is with the interaction between TCP and HTTP.
> HTTP was designed because of the ignorance of the rationale of the
> design of FTP - notably that it wasn't useful to open two connections
> to get a file when only one file per location would be useful (this is
> paraphrased from Berners-Lee's original papers). HTTP incurred long
> (200ms) delays when first deployed because of a similar ignorance about
> how application protocols interact with TCP - that interactive
> protocols with multi-byte exchanges must have Nagle off.
>
> There is a similar issue with RSTs. They're useful to terminate
> connections when that is the desired intent - an abort. However,
> they're overused by browsers for other reasons - e.g., when switching
> pages.
>
> There are reasons to change TCP, and to have it evolve; there are also
> reasons to find and declare variations "non-compliant" bugs and fix
> them. The distinction should be made with a full understanding of what
> TCP is *already* capable of, and the implications of these changes.
>
> Joe
>
> >> -----Original Message-----
> >> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf
> >> Of Joe Touch
> >> Sent: Wednesday, February 08, 2012 11:09 AM
> >> To: Jim Gettys
> >> Cc: 'tcpm@ietf.org'
> >> Subject: Re: [tcpm] Aid passive estimate of tcp performance
> >>
> >> Jim,
> >>
> >> Routers have been over-buffering forever too. Yet you continue to
> >> want to get them to change things and correctly implement the
> standards.
> >>
> >> That's what I want for endpoints with TCP too.
> >>
> >> Ignorance or deliberate defiance is not an excuse. These are errors
> >> that need to be corrected, since they have real implications
> >> throughout the network, both for users on those systems and those
> >> that share those addresses (e.g., behind shared NATs or proxies).
> >>
> >> Joe
> >>
> >> On 2/8/2012 11:03 AM, Jim Gettys wrote:
> >>> On 02/08/2012 01:56 PM, Joe Touch wrote:
> >>>>
> >>>>
> >>>> On 2/8/2012 10:05 AM, Smith, Donald wrote:
> >>>>> Just agreeing with Joe, Many applications uses resets instead of
> a
> >>>>> FIN handshake to terminate connections to speed things up
> >> especially
> >>>>> browsers.
> >>>>>
> >>>>>
> >>>>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
> >>>>
> >>>> It doesn't just "speed" things; it's reckless and dangerous, esp.
> >>>> since the parties tossing around these RSTs typically don't
> >>>> maintain TIME-WAIT before reusing those connection IDs.
> >>>
> >>> Joe, browsers have done this forever.
> >>>
> >>> If you don't like it, then it's (long past time) to make the case
> >> that
> >>> they should change their behaviour.
> >>>                           - Jim
> >>>
> >>>>
> >>>> Joe
> >>>>
> >>>>>> -----Original Message-----
> >>>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
> >> Behalf Of
> >>>>>> Joe Touch
> >>>>>> Sent: Wednesday, February 08, 2012 10:46 AM
> >>>>>> To: =C1ngel Bartomeu Bonillo
> >>>>>> Cc: tcpm@ietf.org
> >>>>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> On 2/8/2012 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
> >>>>>>> Hello,
> >>>>>>>
> >>>>>>> I recently faced the problem of passively estimating the packet
> >> loss
> >>>>>>> of TCP connection. After a bit of investigating the literature
> I
> >>>>>> found
> >>>>>>> that there are tens of papers addressing this issue but I
> >> couldn't
> >>>>>>> find any perfect solution. Since this is an important issue for
> >>>>>>> network operators I have been thinking that there would be a
> >>>>>>> very
> >>>>>> easy
> >>>>>>> solution by just making the two TCP connection end points to
> >> account
> >>>>>>> for its sent segments and report it to the other end point in
> >>>>>>> the
> >> FIN
> >>>>>>> segment. This approach could be used as well for other
> >>>>>>> measurable parameters along the TCP session without the need
> for
> >>>>>>> any other offline method and It would be very helpful for
> >>>>>>> detecting
> >> failures
> >>>>>>> along networks.
> >>>>>>>
> >>>>>>> Any comments?
> >>>>>>
> >>>>>> Regardless of when you do this - either at the end of a
> >> connection, or
> >>>>>> during a connection - the ONLY information you can provide is
> >> "here is
> >>>>>> a list of segments I have not yet received".
> >>>>>>
> >>>>>> However, that list is irrelevant. Those segments might be in the
> >>>>>> network and may still arrive - after you report, after the
> >> connection
> >>>>>> is over, etc.
> >>>>>>
> >>>>>> There is no way to KNOW that a segment is lost vs. late.
> >>>>>>
> >>>>>> Even when you retransmit a segment (e.g., because it is
> basically
> >>>>>> overdue, either because your side's timer expired or the other
> >> side
> >>>>>> sent either a SACK or ACK indicating a gap), you don't know that
> >> the
> >>>>>> first segment was lost - it could arrive after the timer
> expires.
> >>>>>>
> >>>>>> Even though segments have TTLs, and TTLs are *supposed* to be
> >>>>>> decremented once every second or once every hop (whichever is
> >> greater),
> >>>>>> there are devices that queue packets for long periods (e.g.,
> when
> >> a
> >>>>>> link is interrupted) and send them on later (even though they
> >> should
> >>>>>> have been dropped).
> >>>>>>
> >>>>>> So there is no such information as "lost segments"; the best you
> >> know
> >>>>>> is retransmissions.
> >>>>>>
> >>>>>> However, keep in mind that there are lots of TCP connections
> that
> >> are
> >>>>>> terminated without a FIN (e.g., using a RST - I don't think it's
> >>>>>> appropriate in many cases, but the method is in widespread use).
> >>>>>>
> >>>>>> Joe
> >>>>>> _______________________________________________
> >>>>>> tcpm mailing list
> >>>>>> tcpm@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/tcpm
> >>>>>
> >>>>> This communication is the property of CenturyLink and may contain
> >>>>> confidential or privileged information. Unauthorized use of this
> >>>>> communication is strictly prohibited and may be unlawful.  If you
> >>>>> have received this
> >> communication
> >>>>> in error, please immediately notify the sender by reply e-mail
> and
> >>>>> destroy all copies of the communication and any attachments.
> >>>> _______________________________________________
> >>>> tcpm mailing list
> >>>> tcpm@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/tcpm
> >> _______________________________________________
> >> tcpm mailing list
> >> tcpm@ietf.org
> >> https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

This communication is the property of CenturyLink and may contain confident=
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From: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
To: "Joe Touch" <touch@isi.edu>
X-OriginalArrivalTime: 08 Feb 2012 21:57:01.0126 (UTC) FILETIME=[95F0EE60:01CCE6AC]
Cc: tcpm@ietf.org
Subject: Re: [tcpm] Aid passive estimate of tcp performance
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> -----Original Message-----
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Wednesday, February 08, 2012 12:54 PM
> To: Anantha Ramaiah (ananth)
> Cc: Jim Gettys; tcpm@ietf.org
> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>=20
>=20
>=20
> On 2/8/2012 11:39 AM, Anantha Ramaiah (ananth) wrote:
> > Well,
> > Some assumptions that existed long ago, are no longer serious or may
> > be not hold good for the modern internet.
>=20
> Where that's true, it's worth re-examining.
>=20
>  > Lets take the example of
> > TIMEWAIT state which you mention below. RFC recommends 2*MSL which
> > translates to 4 mins. Many OS'es (Windows, Linux ) and Cisco IOS
> etc.,
> > have configurable knobs to tune this value to anywhere between 30
> secs
> > to 240 secs.
>=20
> It's worth understanding why. It's not with a sense that this was safe
> per-se; it was a sense of improving performance. That's a bad reason
to
> make a change, especially when it's ignorant of the rationale for the
> safety mechanism being avoided.

Many changes proposed to TCP are in the areas of improving TCP
performance, this one is no different. The reason you want to change
initial window to a larger value is to have a better throughput. The
reason you want to decrease TIMEWAIT value is to have more connections
per second without port contention issues. The rationale is similar in
both cases. Regarding your safety argument: having a timewait value of X
was considered safe ages ago, now having a lesser value is considered
safe, it is parameter change and every parameter change that affects TCP
and/or has other side effects needs to evaluated carefully. I don't see
any difference.

>=20
> There is a similar issue with RSTs. They're useful to terminate
> connections when that is the desired intent - an abort. However,
> they're
> overused by browsers for other reasons - e.g., when switching pages.

Well, I thought RST's original intent was to detect and recover from
half open connections (crash etc.,). RFC 793 does list abort as the user
interface to TCP, but it doesn't tell you when you should not use
abort's. (I can see that that are supplement documents that discuss
RST's like RFC 3360, which mainly talks about middleboxes inappropriate
usage of RST's, but I don't remember anybody mandating its (abort API)
use in one way or the other). Abort API evolved to be used in other
scenarios. You can't blame the implementations for that.

>=20
> There are reasons to change TCP, and to have it evolve; there are also
> reasons to find and declare variations "non-compliant" bugs and fix
> them. The distinction should be made with a full understanding of what
> TCP is *already* capable of, and the implications of these changes.

Agree.

-Anantha
>=20
> Joe
>=20

<snip>

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Subject: [tcpm] TCP and applications [was Re: Aid passive estimate of tcp performance]
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On Wed, Feb 8, 2012 at 11:20 AM, Jim Gettys <jg@freedesktop.org> wrote:
> Over the years, I have noted a distinct lack of communications between
> those who understand how the bits flow, and how they are used.... =A0For
> example, the transient bufferbloat phenomena is something that is pretty
> obvious to those who understand TCP, and completely unobvious to most
> who do web browsers...
>
> I believe both groups need to talk to each other, strangely. =A0Sometimes
> you may find that the guys who do the applications actually have needs
> not being met by standards, that are well worth understanding; and
> conversely, the guys who do the apps may find that life is better for
> both them and others if they understand the standards.
+1

Can't agree more. We need a venue for two-way communications.

app dev: TCP sucks because blah blah blah
TCP dudes: app sucks because blah blah blah
[I am in a company where I have to hear both  :(  ]

Nothing gets done until both parties try to help each other by
listening to each other.

>
> Just go in with an open mind and say: "We observe the browsers/servers
> doing the following thing that we believe is much less than optimal, why
> are you doing this?" rather than "You are disobeying the rules, fix it",
> and I think you'll get a warm welcome.
> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- Jim
>
>
>>
>> Joe
>>
>> On 2/8/2012 11:03 AM, Jim Gettys wrote:
>>> On 02/08/2012 01:56 PM, Joe Touch wrote:
>>>>
>>>>
>>>> On 2/8/2012 10:05 AM, Smith, Donald wrote:
>>>>> Just agreeing with Joe, Many applications uses resets instead of a
>>>>> FIN handshake to terminate connections to speed things up especially
>>>>> browsers.
>>>>>
>>>>>
>>>>> http://pages.cpsc.ucalgary.ca/~carey/papers/2005/TCP-Resets.pdf
>>>>
>>>> It doesn't just "speed" things; it's reckless and dangerous, esp.
>>>> since the parties tossing around these RSTs typically don't maintain
>>>> TIME-WAIT before reusing those connection IDs.
>>>
>>> Joe, browsers have done this forever.
>>>
>>> If you don't like it, then it's (long past time) to make the case that
>>> they should change their behaviour.
>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- Jim
>>>
>>>>
>>>> Joe
>>>>
>>>>>> -----Original Message-----
>>>>>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
>>>>>> Behalf Of
>>>>>> Joe Touch
>>>>>> Sent: Wednesday, February 08, 2012 10:46 AM
>>>>>> To: =C1ngel Bartomeu Bonillo
>>>>>> Cc: tcpm@ietf.org
>>>>>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 2/8/2012 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
>>>>>>> Hello,
>>>>>>>
>>>>>>> I recently faced the problem of passively estimating the packet los=
s
>>>>>>> of TCP connection. After a bit of investigating the literature I
>>>>>> found
>>>>>>> that there are tens of papers addressing this issue but I couldn't
>>>>>>> find any perfect solution. Since this is an important issue for
>>>>>>> network operators I have been thinking that there would be a very
>>>>>> easy
>>>>>>> solution by just making the two TCP connection end points to accoun=
t
>>>>>>> for its sent segments and report it to the other end point in the
>>>>>>> FIN
>>>>>>> segment. This approach could be used as well for other measurable
>>>>>>> parameters along the TCP session without the need for any other
>>>>>>> offline method and It would be very helpful for detecting failures
>>>>>>> along networks.
>>>>>>>
>>>>>>> Any comments?
>>>>>>
>>>>>> Regardless of when you do this - either at the end of a
>>>>>> connection, or
>>>>>> during a connection - the ONLY information you can provide is
>>>>>> "here is
>>>>>> a list of segments I have not yet received".
>>>>>>
>>>>>> However, that list is irrelevant. Those segments might be in the
>>>>>> network and may still arrive - after you report, after the connectio=
n
>>>>>> is over, etc.
>>>>>>
>>>>>> There is no way to KNOW that a segment is lost vs. late.
>>>>>>
>>>>>> Even when you retransmit a segment (e.g., because it is basically
>>>>>> overdue, either because your side's timer expired or the other side
>>>>>> sent either a SACK or ACK indicating a gap), you don't know that the
>>>>>> first segment was lost - it could arrive after the timer expires.
>>>>>>
>>>>>> Even though segments have TTLs, and TTLs are *supposed* to be
>>>>>> decremented once every second or once every hop (whichever is
>>>>>> greater),
>>>>>> there are devices that queue packets for long periods (e.g., when a
>>>>>> link is interrupted) and send them on later (even though they should
>>>>>> have been dropped).
>>>>>>
>>>>>> So there is no such information as "lost segments"; the best you kno=
w
>>>>>> is retransmissions.
>>>>>>
>>>>>> However, keep in mind that there are lots of TCP connections that ar=
e
>>>>>> terminated without a FIN (e.g., using a RST - I don't think it's
>>>>>> appropriate in many cases, but the method is in widespread use).
>>>>>>
>>>>>> Joe
>>>>>> _______________________________________________
>>>>>> tcpm mailing list
>>>>>> tcpm@ietf.org
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> Can't agree more. We need a venue for two-way communications.
> 
> app dev: TCP sucks because blah blah blah
> TCP dudes: app sucks because blah blah blah

Wellllll..... that is one reason everyone rearranges all the furniture
in a big damn hotel three times a year.  But, alas, you can lead a horse
to water ...

allman




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From abartomeu@swissms.ch  Thu Feb  9 00:16:16 2012
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Subject: Re: [tcpm] TCP and applications [was Re: Aid passive estimate of tcp performance]
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Well, the case that I had in mind was calculating the packet loss =
passively from the end user. The estimation for packet loss would be the =
number of retransmissions minus the number of spurious retransmission. =
In the Uplink traffic you know the real number of retransmission and you =
can estimate the spurious ones with the info in the DSAck. However in =
the downlink it is impossible since you know the spurious but not the =
real number of packets sent out by the other peer any approach would =
fail in cases like having the remote peer in slow start and being all =
the packets dropped. You never know if you do not get packets because =
either there are no packets to send or because they were lost.

There is obviously no point in changing the TCP just to aid this case =
but I thought that maybe some of you would find a more generic usage for =
this idea that I considered quite simple and I thought I could be =
useful. I see now because of your comments that I was wrong since I =
didn't see all this consideration you just showed.=20

Thanks anyways for your interesting comments.

El 09/02/2012, a las 04:31, Mark Allman escribi=F3:

>=20
>> Can't agree more. We need a venue for two-way communications.
>>=20
>> app dev: TCP sucks because blah blah blah
>> TCP dudes: app sucks because blah blah blah
>=20
> Wellllll..... that is one reason everyone rearranges all the furniture
> in a big damn hotel three times a year.  But, alas, you can lead a =
horse
> to water ...
>=20
> allman
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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On 2/9/2012 3:15 AM, Ángel Bartomeu Bonillo wrote:
> Well, the case that I had in mind was calculating the packet loss passively from the end user. The estimation for packet loss would be the number of retransmissions minus the number of spurious retransmission. In the Uplink traffic you know the real number of retransmission and you can estimate the spurious ones with the info in the DSAck. However in the downlink it is impossible since you know the spurious but not the real number of packets sent out by the other peer any approach would fail in cases like having the remote peer in slow start and being all the packets dropped. You never know if you do not get packets because either there are no packets to send or because they were lost.
> 

Even from the sender-side where you know about all of the
transmissions, you can still only compute an estimate due
to sources of error like lost ACKs carrying DSACK information
(as one easy example).

There is (in my opinion) a good description of some algorithms
for different types of connections (DSACK-supported, SACK but
no DSACK, and no SACK) and the associated sources of error in:

http://www.icir.org/mallman/papers/least-per-dec2003.ps
Mark Allman, Wesley Eddy, Shawn Ostermann.
Estimating Loss Rates With TCP.
ACM Performance Evaluation Review, 31(3), December 2003.

-- 
Wes Eddy
MTI Systems

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Subject: Re: [tcpm] TCP and applications [was Re: Aid passive estimate of tcp performance]
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El 09/02/2012, a las 15:44, Wesley Eddy escribi=F3:

> On 2/9/2012 3:15 AM, =C1ngel Bartomeu Bonillo wrote:
>> Well, the case that I had in mind was calculating the packet loss =
passively from the end user. The estimation for packet loss would be the =
number of retransmissions minus the number of spurious retransmission. =
In the Uplink traffic you know the real number of retransmission and you =
can estimate the spurious ones with the info in the DSAck. However in =
the downlink it is impossible since you know the spurious but not the =
real number of packets sent out by the other peer any approach would =
fail in cases like having the remote peer in slow start and being all =
the packets dropped. You never know if you do not get packets because =
either there are no packets to send or because they were lost.
>>=20
>=20
> Even from the sender-side where you know about all of the
> transmissions, you can still only compute an estimate due
> to sources of error like lost ACKs carrying DSACK information
> (as one easy example).
>=20
> There is (in my opinion) a good description of some algorithms
> for different types of connections (DSACK-supported, SACK but
> no DSACK, and no SACK) and the associated sources of error in:
>=20
> http://www.icir.org/mallman/papers/least-per-dec2003.ps
> Mark Allman, Wesley Eddy, Shawn Ostermann.
> Estimating Loss Rates With TCP.
> ACM Performance Evaluation Review, 31(3), December 2003.


Yeah Wesley I have already read that and a few others. That's why I am =
interested in the downlink. AFAIK the downlink is not covered in any =
other paper.
Thank you.

>=20
> --=20
> Wes Eddy
> MTI Systems


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I believe the draft is indeed ready for publication.=20

Thanks again to all the authors to pick up my vague suggestion which now=20
ended up as Rescue Retransmission!

Richard Scheffenegger


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Pasi Sarolahti
> Sent: Mittwoch, 08. Februar 2012 14:01
> To: tcpm@ietf.org
> Subject: [tcpm] WGLC for draft-ietf-tcpm-3517bis-01
>=20
> Hello,
>=20
> The authors of "A Conservative Selective Acknowledgment (SACK)-based
> Loss Recovery Algorithm for TCP" (draft-ietf-tcpm-3517bis-01) think the
> latest version addresses all comments made on the draft. Therefore, this
> mail starts Working Group Last Call for the draft, intended to be publish=
ed as
> Proposed Standard. The WGLC ends on Friday, February 24, i.e., within a b=
it
> over two weeks.
>=20
> Please read the draft and send any comments to the TCPM mailing list by t=
he
> above deadline. If you think the draft is ok and ready to be published wi=
thout
> further changes, indicating that would also be helpful.
>=20
> The draft is available at http://tools.ietf.org/html/draft-ietf-tcpm-3517=
bis-01
>=20
> - Pasi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From pasi.sarolahti@iki.fi  Fri Feb 17 02:43:32 2012
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Subject: [tcpm] Agenda requests for Paris meeting
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Hi,

We have requested a TCPM meeting slot in Paris. Send tcpm-chairs your =
requests for topics to be presented in the TCPM meeting, with the =
following information:

* Presentation Title and draft name
* Speaker
* Amount of time requested

Please send your requests as early as possible, to help us in agenda =
planning.

- Pasi


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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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----- Original Message -----
From: "Ángel Bartomeu Bonillo" <abartomeu@swissms.ch>
To: "rick jones" <perfgeek@mac.com>
Cc: <tcpm@ietf.org>
Sent: Wednesday, February 08, 2012 6:52 PM
Hi Rick,
El 08/02/2012, a las 17:32, rick jones escribió:

>
> On Feb 8, 2012, at 4:15 AM, Ángel Bartomeu Bonillo wrote:
>
>> Hello,
>>
>> I recently faced the problem of passively estimating the packet loss of TCP
connection. After a bit of investigating the literature I found that there are
tens of papers addressing this issue but I couldn't find any perfect solution.
>> Since this is an important issue for network operators I have been thinking
that there would be a very easy solution by just making the two TCP connection
end points to account for its sent segments and report it to the other end point
in the FIN segment.
>> This approach could be used as well for other measurable parameters along the
TCP session without the need for any other offline method and It would be very
helpful for detecting failures along networks.
>>
>> Any comments?
>
> If the count is reported in a FIN segment, isn't that rather like closing the
barn door after the horses have left?  At least as far as doing anything for
that connection is concerned.  So, already we are talking about operating on
timescales larger than connection lifetimes.  And what would the side receiving
the count-containing FIN do with this information? Sure, now it is known on both
ends, but the problem still lies somewhere in the middle (not always, but
often).  Is it expected that points along the middle will be sniffing this?
What would they do in response?  What could they (reasonably) do in response
with that information just from TCP vs any other transports?
>

I'm talking about timescales larger than the connection lifetimes. This would be
a mechanism so TCP can report how many packets it sent out. If both ends send at
the end of the session the number of packets they sent out the other peer and
anyone in the middle would be able to estimate de loss in both directions.

The idea is to enable TCP so it can report a little but valuable information
about its performance once the connection finished. This is not a mechanism so
TCP can adapt to losses in the middle of a communication but rather a long term
mechanism for network operators and such. Nowadays many of the end user services
use TCP and the end to end packet loss is many times impossible to calculate
mainly because only the sending node knows the real number of packets sent out.
For example in the uplink direction on a mobile network.
I think that adding just a couple of bytes (as a TCP option) to the FIN segment
would give a priceless information for network operators.

<tp>
Apologies for being late to the party but I think that it is a lovely idea.  As
has been pointed out, the figures that you would collect are not what you want,
but they would be a lot closer than anything available at the moment, and that
is engineering.  Ditto comments about RST; if there is a positive benefit to
shutting down properly and getting these figures, then that could discourage
RSTs.

Tom Petch
<tp>






> When a sending TCP sends the first FIN with a count at the end of a MB window
would it retransmit the FIN, even if the first was received, to include
retransmissions in that last window's-worth of data?  How long would the
receiving TCP wait for this "final tally, really" FIN to arrive after it sent
the ACK to SEQfin to then pass-along the count to a higher layer, somehow?

The idea much simpler: the peer adds the num of packets sent during the
connection as an option in the FIN. There is no need to change TCP behavior.

>
> I think we are already asking for enough angels to dance on the TCP
pin-header.
>
> rick jones

_______________________________________________
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https://www.ietf.org/mailman/listinfo/tcpm



From internet-drafts@ietf.org  Sat Feb 18 10:48:53 2012
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-fastopen-00.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 =
Working Group of the IETF.

	Title           : TCP Fast Open
	Author(s)       : Yuchung Cheng
                          Jerry Chu
                          Sivasankar Radhakrishnan
                          Arvind Jain
	Filename        : draft-ietf-tcpm-fastopen-00.txt
	Pages           : 21
	Date            : 2012-02-17

   TCP Fast Open (TFO) allows data to be carried in the SYN and SYN-ACK
   packets and consumed by the receiving end during the initial
   connection handshake, thus providing a saving of up to one full round
   trip time (RTT) compared to standard TCP requiring a three-way
   handshake (3WHS) to complete before data can be exchanged.

Terminology

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-fastopen-00.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-fastopen-00.txt


From Michael.Scharf@alcatel-lucent.com  Mon Feb 20 02:20:09 2012
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Thread-Topic: TCPM rechartering discussion
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Dear all,

The current TCPM charter https://datatracker.ietf.org/wg/tcpm/charter/
includes some outdated text. Most notably, the "Specific Goals"
mentioned in the charter are not up to date. Also, the current charter
requires IESG approval of every new milestone.

Therefore, the chairs think that it would make sense to recharter TCPM
in order to clean up the completed goals and to make it easier to manage
future milestones. Please find below a first draft for a new charter
text. The intention is a minor modification that does not to affect the
overall scope of the working group.

We would highly appreciate comments or suggestions.

Best regards

Michael (TCPM co-chair)




TCP Maintenance and Minor Extensions (tcpm)
-------------------------------------------

Description of Working Group: ***DRAFT***

TCP is currently the Internet's predominant transport protocol.=20
To maintain TCP's utility the IETF has regularly updated both=20
the protocol itself and the congestion control algorithms=20
implemented by the protocol that are crucial for the stability=20
of the Internet. These changes reflect our evolving=20
understanding of transport protocols, congestion control and new=20
needs presented by an ever-changing network. The TCPM WG provides
a venue within the IETF to work on these issues. The WG=20
serves several purposes:=20

* The WG mostly focuses on maintenance issues (e.g., bug=20
fixes) and modest changes to the protocol and algorithms=20
that maintain TCP's utility, including incremental enhancements
of the congestion control mechanisms.

* The WG is a venue for moving current TCP specifications=20
along the standards track (as community energy is available=20
for such efforts).=20

* The focus of the working group are minor extensions to TCP
algorithms and mechanisms. In cases where these are
directly applicable to other transports (e.g. SCTP or DCCP),
the mappings to other transports may be specified alongside
that for TCP, but significant additions and changes to other
transports are not in scope.

TCPM is the working group within the IETF that handles small TCP
changes, or minor extensions to TCP algorithms and mechanisms. While
fundamental changes to TCP or its congestion control algorithms
(e.g., departure from loss-based congestion control) should be brought
through TCPM, it is expected that such large changes will be handled
by other working groups or require additional reviews, in particular by
the IRTF Congestion Control Research Group (ICCRG).

TCP's congestion control algorithms are the model followed by=20
alternate transports (e.g., SCTP and (in some cases) DCCP),
which are handled for instance by the Transport Area WG (tsvwg).
The IETF has recently worked on several documents=20
about algorithms that are specified for multiple protocols=20
(e.g., TCP and SCTP) in the same document. Which WG shepherds=20
such documents will determined on a case-by-case=20
basis. In any case, the TCPM WG will remain in close contact=20
with other relevant WGs working on these protocols to ensure=20
openness and stringent review from all angles.=20

New TCPM milestones that fall within the scope specified within
the charter can be added after approval of the working group and the
responsible Area Director.


From Michael.Scharf@alcatel-lucent.com  Mon Feb 20 03:34:53 2012
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From: "SCHARF, Michael" <Michael.Scharf@alcatel-lucent.com>
To: "Joe Touch" <touch@isi.edu>
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Joe,

The wording on the nonce length in draft-touch-tcpm-experimental-options
may be a little bit unclear ("it MAY be as few as 16 bits if desired").

Does that statement intend to say something like "MUST NOT be shorter
than 16 bits long"? Or shall even a nonce of length 0 be allowed,
provided that there are additional security mechanism such as checksums
or digital signatures? In the latter case, existing use of the
codepoints without nonce could be compliant to the recommendation,
provided that there are other means to detect false positives.

As a side note, I guess that the numbers "245 and 255" in the second
paragraph are wrong. This should refer to 253 and 254, right?

Michael (as individual)

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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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Hi Tom.
What you say was my original point. However I have been overwhelmed by =
the amount of drawbacks pointed out by the other folks.=20
My idea was to enable TCP to give a little report at the end of the =
session. This report should be concise and simple. Just accounting for =
any interesting parameters that are either difficult or impossible to be =
obtained without an explicit procedure. I was thinking about the =
estimate of the packet loss but I was hopping that other (more expert =
people) would find any other interesting usages.=20
However, as I said, I think there have been too much negative feedback =
on this idea so far.=20

What does everyone think?
Should we give up with this issue?

Thank you all.

El 17/02/2012, a las 16:46, t.petch escribi=F3:

> ----- Original Message -----
> From: "=C1ngel Bartomeu Bonillo" <abartomeu@swissms.ch>
> To: "rick jones" <perfgeek@mac.com>
> Cc: <tcpm@ietf.org>
> Sent: Wednesday, February 08, 2012 6:52 PM
> Hi Rick,
> El 08/02/2012, a las 17:32, rick jones escribi=F3:
>=20
>>=20
>> On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
>>=20
>>> Hello,
>>>=20
>>> I recently faced the problem of passively estimating the packet loss =
of TCP
> connection. After a bit of investigating the literature I found that =
there are
> tens of papers addressing this issue but I couldn't find any perfect =
solution.
>>> Since this is an important issue for network operators I have been =
thinking
> that there would be a very easy solution by just making the two TCP =
connection
> end points to account for its sent segments and report it to the other =
end point
> in the FIN segment.
>>> This approach could be used as well for other measurable parameters =
along the
> TCP session without the need for any other offline method and It would =
be very
> helpful for detecting failures along networks.
>>>=20
>>> Any comments?
>>=20
>> If the count is reported in a FIN segment, isn't that rather like =
closing the
> barn door after the horses have left?  At least as far as doing =
anything for
> that connection is concerned.  So, already we are talking about =
operating on
> timescales larger than connection lifetimes.  And what would the side =
receiving
> the count-containing FIN do with this information? Sure, now it is =
known on both
> ends, but the problem still lies somewhere in the middle (not always, =
but
> often).  Is it expected that points along the middle will be sniffing =
this?
> What would they do in response?  What could they (reasonably) do in =
response
> with that information just from TCP vs any other transports?
>>=20
>=20
> I'm talking about timescales larger than the connection lifetimes. =
This would be
> a mechanism so TCP can report how many packets it sent out. If both =
ends send at
> the end of the session the number of packets they sent out the other =
peer and
> anyone in the middle would be able to estimate de loss in both =
directions.
>=20
> The idea is to enable TCP so it can report a little but valuable =
information
> about its performance once the connection finished. This is not a =
mechanism so
> TCP can adapt to losses in the middle of a communication but rather a =
long term
> mechanism for network operators and such. Nowadays many of the end =
user services
> use TCP and the end to end packet loss is many times impossible to =
calculate
> mainly because only the sending node knows the real number of packets =
sent out.
> For example in the uplink direction on a mobile network.
> I think that adding just a couple of bytes (as a TCP option) to the =
FIN segment
> would give a priceless information for network operators.
>=20
> <tp>
> Apologies for being late to the party but I think that it is a lovely =
idea.  As
> has been pointed out, the figures that you would collect are not what =
you want,
> but they would be a lot closer than anything available at the moment, =
and that
> is engineering.  Ditto comments about RST; if there is a positive =
benefit to
> shutting down properly and getting these figures, then that could =
discourage
> RSTs.
>=20
> Tom Petch
> <tp>
>=20
>=20
>=20
>=20
>=20
>=20
>> When a sending TCP sends the first FIN with a count at the end of a =
MB window
> would it retransmit the FIN, even if the first was received, to =
include
> retransmissions in that last window's-worth of data?  How long would =
the
> receiving TCP wait for this "final tally, really" FIN to arrive after =
it sent
> the ACK to SEQfin to then pass-along the count to a higher layer, =
somehow?
>=20
> The idea much simpler: the peer adds the num of packets sent during =
the
> connection as an option in the FIN. There is no need to change TCP =
behavior.
>=20
>>=20
>> I think we are already asking for enough angels to dance on the TCP
> pin-header.
>>=20
>> rick jones
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>=20
>=20



From Michael.Scharf@alcatel-lucent.com  Mon Feb 20 09:19:08 2012
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From: "SCHARF, Michael" <Michael.Scharf@alcatel-lucent.com>
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Hi Angel,

> What you say was my original point. However I have been=20
> overwhelmed by the amount of drawbacks pointed out by the=20
> other folks.=20

Well, please note that TCP is not a trivial protocol. Even ideas that =
appear to be trivial at first sight may require a careful analysis. This =
is what the TCPM working group is about.

> My idea was to enable TCP to give a little report at the end=20
> of the session. This report should be concise and simple.=20
> Just accounting for any interesting parameters that are=20
> either difficult or impossible to be obtained without an=20
> explicit procedure. I was thinking about the estimate of the=20
> packet loss but I was hopping that other (more expert people)=20
> would find any other interesting usages.=20
> However, as I said, I think there have been too much negative=20
> feedback on this idea so far.=20
>=20
> What does everyone think?

My own view is that a gap analysis of the existing solutions would be a =
good starting point.

For instance, Section 5.10.14 in RFC 5102 apparently mentions a way to =
export the total number of dropped packets by IPFIX, on a per-flow basis =
(I am not familiar with IPFIX, so sorry if I miss something =
fundamental). Potential related work in CONEX has already been mentioned =
as well. We probably should better understand if another TCP option =
offers a significant benefit compared to that, and in what cases.

> Should we give up with this issue?

Just to mention something obvious: The other alternative would be to put =
together an Internet Draft that addresses the issues already mentioned =
on the list.

Michael


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Angel:

This is just my opinion. I wrote something earlier then withdrew it =
based on a lot of email churn on this topic.

I agree that this is not a bad idea. However to make it work, you do =
need some TCP changes like :-
- having a TCP option to agree upon this reporting between the =
connection end points.
- at the end of the TCP connection ( be it RST or FIN) you need some =
clear cut ways to exchange the report.
This would need some changes to TCP.=20
- Atleast in the RST case, (nothing prevents from the sender to put some =
information in the RST data portion) FWIW, RFC 793 says RST segments can =
contain data. RST becomes a best effort since RST segments can be lost.=20
- FIN case, you need to account for data in the flight and send the =
report only when all the data has been acknowledged.

Also questions like :-
- do you want to send this report during the end OR anytime during the =
lifetime of a connection?
- do you want the active closer to send this report OR it could be sent =
by a both ends?
- There are many existing mechanisms that can be useful to obtain the =
similar information which you are looking for, like :-
 In cases where it is feasible, active performance methods like the one =
outlined in RFC 6349 can be used (several programs like TTCP, iPerf =
etc.,  are there and can be useful), SNMP is often used to monitor the =
connection characteristics, IPSLA (TWAMP is the equivalent IETF =
proposal) is often used by network operators to measure network =
characteristics. Poor man's version is to have a socket option which =
would make TCP dump the connection block before the TCB gets freed =
(which should include everything other than just packet counts), which =
is purely local but useful.

I just saw an email from Michael suggesting to write an ID, may be that =
is a good starting point if you are planning to take this effort =
seriously. Atleast the draft should have the use case clearly outlined =
and the reasons why the existing mechanisms aren't good enough.=20

Thanks,
-Anantha

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf =
Of
> =C1ngel Bartomeu Bonillo
> Sent: Monday, February 20, 2012 8:31 AM
> To: t.petch
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>=20
> Hi Tom.
> What you say was my original point. However I have been overwhelmed by
> the amount of drawbacks pointed out by the other folks.
> My idea was to enable TCP to give a little report at the end of the
> session. This report should be concise and simple. Just accounting for
> any interesting parameters that are either difficult or impossible to
> be obtained without an explicit procedure. I was thinking about the
> estimate of the packet loss but I was hopping that other (more expert
> people) would find any other interesting usages.
> However, as I said, I think there have been too much negative feedback
> on this idea so far.
>=20
> What does everyone think?
> Should we give up with this issue?
>=20
> Thank you all.
>=20
> El 17/02/2012, a las 16:46, t.petch escribi=F3:
>=20
> > ----- Original Message -----
> > From: "=C1ngel Bartomeu Bonillo" <abartomeu@swissms.ch>
> > To: "rick jones" <perfgeek@mac.com>
> > Cc: <tcpm@ietf.org>
> > Sent: Wednesday, February 08, 2012 6:52 PM
> > Hi Rick,
> > El 08/02/2012, a las 17:32, rick jones escribi=F3:
> >
> >>
> >> On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
> >>
> >>> Hello,
> >>>
> >>> I recently faced the problem of passively estimating the packet
> loss of TCP
> > connection. After a bit of investigating the literature I found that
> there are
> > tens of papers addressing this issue but I couldn't find any perfect
> solution.
> >>> Since this is an important issue for network operators I have been
> thinking
> > that there would be a very easy solution by just making the two TCP
> connection
> > end points to account for its sent segments and report it to the
> other end point
> > in the FIN segment.
> >>> This approach could be used as well for other measurable =
parameters
> along the
> > TCP session without the need for any other offline method and It
> would be very
> > helpful for detecting failures along networks.
> >>>
> >>> Any comments?
> >>
> >> If the count is reported in a FIN segment, isn't that rather like
> closing the
> > barn door after the horses have left?  At least as far as doing
> anything for
> > that connection is concerned.  So, already we are talking about
> operating on
> > timescales larger than connection lifetimes.  And what would the =
side
> receiving
> > the count-containing FIN do with this information? Sure, now it is
> known on both
> > ends, but the problem still lies somewhere in the middle (not =
always,
> but
> > often).  Is it expected that points along the middle will be =
sniffing
> this?
> > What would they do in response?  What could they (reasonably) do in
> response
> > with that information just from TCP vs any other transports?
> >>
> >
> > I'm talking about timescales larger than the connection lifetimes.
> This would be
> > a mechanism so TCP can report how many packets it sent out. If both
> ends send at
> > the end of the session the number of packets they sent out the other
> peer and
> > anyone in the middle would be able to estimate de loss in both
> directions.
> >
> > The idea is to enable TCP so it can report a little but valuable
> information
> > about its performance once the connection finished. This is not a
> mechanism so
> > TCP can adapt to losses in the middle of a communication but rather =
a
> long term
> > mechanism for network operators and such. Nowadays many of the end
> user services
> > use TCP and the end to end packet loss is many times impossible to
> calculate
> > mainly because only the sending node knows the real number of =
packets
> sent out.
> > For example in the uplink direction on a mobile network.
> > I think that adding just a couple of bytes (as a TCP option) to the
> FIN segment
> > would give a priceless information for network operators.
> >
> > <tp>
> > Apologies for being late to the party but I think that it is a =
lovely
> idea.  As
> > has been pointed out, the figures that you would collect are not =
what
> you want,
> > but they would be a lot closer than anything available at the =
moment,
> and that
> > is engineering.  Ditto comments about RST; if there is a positive
> benefit to
> > shutting down properly and getting these figures, then that could
> discourage
> > RSTs.
> >
> > Tom Petch
> > <tp>
> >
> >
> >
> >
> >
> >
> >> When a sending TCP sends the first FIN with a count at the end of a
> MB window
> > would it retransmit the FIN, even if the first was received, to
> include
> > retransmissions in that last window's-worth of data?  How long would
> the
> > receiving TCP wait for this "final tally, really" FIN to arrive =
after
> it sent
> > the ACK to SEQfin to then pass-along the count to a higher layer,
> somehow?
> >
> > The idea much simpler: the peer adds the num of packets sent during
> the
> > connection as an option in the FIN. There is no need to change TCP
> behavior.
> >
> >>
> >> I think we are already asking for enough angels to dance on the TCP
> > pin-header.
> >>
> >> rick jones
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
> >
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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On Feb 20, 2012, at 9:39 AM, Anantha Ramaiah (ananth) wrote:
> - There are many existing mechanisms that can be useful to obtain the similar information which you are looking for, like :-
> In cases where it is feasible, active performance methods like the one outlined in RFC 6349 can be used (several programs like TTCP, iPerf etc.,  are there and can be useful),

FWIW, netperf (2.5.0 and later) can, under Linux, report the retransmission statistics from a TCP_INFO getsockopt() and will do so for either end.  In this case I have Linux only at the far end, so I run a TCP_MAERTS test and report the "remote_transport_retrans"  using the output selectors (*)

$ ./netperf -l 20 -H <linuxbox> -t TCP_MAERTS -- -O throughput,remote_transport_retrans -s 1M
MIGRATED TCP MAERTS TEST from (null) (0.0.0.0) port 0 AF_INET to <linuxbox> () port 0 AF_INET : demo
Throughput Remote          
           Transport       
           Retransmissions 
                           
4.94       13              

happy benchmarking,

rick jones

(*) http://www.netperf.org/svn/netperf2/tags/netperf-2.5.0/doc/netperf.html#Omni-Output-Selection

MAERTS is STREAM spelled backwards and is a test where the netserver side sends to the netperf.

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Am 20.02.2012 um 18:53 schrieb rick jones:

>=20
> On Feb 20, 2012, at 9:39 AM, Anantha Ramaiah (ananth) wrote:
>> - There are many existing mechanisms that can be useful to obtain the =
similar information which you are looking for, like :-
>> In cases where it is feasible, active performance methods like the =
one outlined in RFC 6349 can be used (several programs like TTCP, iPerf =
etc.,  are there and can be useful),
>=20
> FWIW, netperf (2.5.0 and later) can, under Linux, report the =
retransmission statistics from a TCP_INFO getsockopt() and will do so =
for either end.

For the sake of completeness flowgrind (https://launchpad.net/flowgrind) =
can do this too :-)
See http://www.umic-mesh.net/research/flowgrind/output.html

Alex

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// mobile: (49-176) 32820027, fax: (49-241) 80-22222
// email: zimmermann@cs.rwth-aachen.de
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Hi Michael.

El 20/02/2012, a las 18:19, SCHARF, Michael escribi=F3:

> Hi Angel,
>=20
>> What you say was my original point. However I have been=20
>> overwhelmed by the amount of drawbacks pointed out by the=20
>> other folks.=20
>=20
> Well, please note that TCP is not a trivial protocol. Even ideas that =
appear to be trivial at first sight may require a careful analysis. This =
is what the TCPM working group is about.

Sure thing :)


>=20
>> My idea was to enable TCP to give a little report at the end=20
>> of the session. This report should be concise and simple.=20
>> Just accounting for any interesting parameters that are=20
>> either difficult or impossible to be obtained without an=20
>> explicit procedure. I was thinking about the estimate of the=20
>> packet loss but I was hopping that other (more expert people)=20
>> would find any other interesting usages.=20
>> However, as I said, I think there have been too much negative=20
>> feedback on this idea so far.=20
>>=20
>> What does everyone think?
>=20
> My own view is that a gap analysis of the existing solutions would be =
a good starting point.

I agree here too but, first of all, I would evaluate the the possible =
benefits of making TCP have its own report mechanism.=20
Let's imagine there was a group of measurable parameters that are =
configurable in the handshake using an option. If both peer support them =
and are able to report them (or are configured to do so)  they would add =
this information in the FIN/RST, or maybe in some other point (I'm not =
sure about this). =20

Now here is in my opinion the key questions:=20

- would TCP benefit from the use of this in the long term?=20
- If this report mechanism becomes usual, could it help, for example, to =
ease the research on future enhancements or changes to TCP?


>=20
> For instance, Section 5.10.14 in RFC 5102 apparently mentions a way to =
export the total number of dropped packets by IPFIX, on a per-flow basis =
(I am not familiar with IPFIX, so sorry if I miss something =
fundamental). Potential related work in CONEX has already been mentioned =
as well. We probably should better understand if another TCP option =
offers a significant benefit compared to that, and in what cases.
>=20
>> Should we give up with this issue?
>=20
> Just to mention something obvious: The other alternative would be to =
put together an Internet Draft that addresses the issues already =
mentioned on the list.

Since I'm a newbie here I will rely on the community for this.


>=20
> Michael
>=20



From perfgeek@mac.com  Mon Feb 20 11:26:31 2012
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From: rick jones <perfgeek@mac.com>
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To: Alexander Zimmermann <alexander.zimmermann@comsys.rwth-aachen.de>
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Cc: tcpm@ietf.org, "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>, "t.petch" <ietfa@btconnect.com>
Subject: Re: [tcpm] Aid passive estimate of tcp performance
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On Feb 20, 2012, at 10:20 AM, Alexander Zimmermann wrote:

> 
> Am 20.02.2012 um 18:53 schrieb rick jones:
> 
>> 
>> On Feb 20, 2012, at 9:39 AM, Anantha Ramaiah (ananth) wrote:
>>> - There are many existing mechanisms that can be useful to obtain the similar information which you are looking for, like :-
>>> In cases where it is feasible, active performance methods like the one outlined in RFC 6349 can be used (several programs like TTCP, iPerf etc.,  are there and can be useful),
>> 
>> FWIW, netperf (2.5.0 and later) can, under Linux, report the retransmission statistics from a TCP_INFO getsockopt() and will do so for either end.
> 
> For the sake of completeness flowgrind (https://launchpad.net/flowgrind) can do this too :-)
> See http://www.umic-mesh.net/research/flowgrind/output.html

Excellent.  Looks like it goes rather farther than netperf in that area - reporting the TCP_INFO on the intervals.  Netperf only does it at the end of the test, and then only for retransmissions.

rick jones
there is no rest for the wicked, yet the virtuous have no pillows


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Subject: Re: [tcpm] Aid passive estimate of tcp performance
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Hi Anantha.

El 20/02/2012, a las 18:39, Anantha Ramaiah (ananth) escribi=F3:

> Angel:
>=20
> This is just my opinion. I wrote something earlier then withdrew it =
based on a lot of email churn on this topic.
>=20
> I agree that this is not a bad idea. However to make it work, you do =
need some TCP changes like :-
> - having a TCP option to agree upon this reporting between the =
connection end points.

Yes. Like with the SACK option.

> - at the end of the TCP connection ( be it RST or FIN) you need some =
clear cut ways to exchange the report.
> This would need some changes to TCP.=20
> - Atleast in the RST case, (nothing prevents from the sender to put =
some information in the RST data portion) FWIW, RFC 793 says RST =
segments can contain data. RST becomes a best effort since RST segments =
can be lost.=20
> - FIN case, you need to account for data in the flight and send the =
report only when all the data has been acknowledged.
>=20

I think that a approach could be next:=20
Every peer reports the number of data segments sent out in the FIN =
segment.
Every peer reports the number of data segments received in the ACK to =
the FIN segment.

As for the RST, AFAIK it does not indicate a normal connection end so I =
think we should not consider that case. Also, as  Tom pointed out, it =
could discourage its use for ending connections.


> Also questions like :-
> - do you want to send this report during the end OR anytime during the =
lifetime of a connection?
> - do you want the active closer to send this report OR it could be =
sent by a both ends?
> - There are many existing mechanisms that can be useful to obtain the =
similar information which you are looking for, like :-
> In cases where it is feasible, active performance methods like the one =
outlined in RFC 6349 can be used (several programs like TTCP, iPerf =
etc.,  are there and can be useful), SNMP is often used to monitor the =
connection characteristics, IPSLA (TWAMP is the equivalent IETF =
proposal) is often used by network operators to measure network =
characteristics. Poor man's version is to have a socket option which =
would make TCP dump the connection block before the TCB gets freed =
(which should include everything other than just packet counts), which =
is purely local but useful.

The problem in this tools you mentioned is that they do active tests and =
you need to control both ends. If TCP would have it own report mechanism =
these tests would be much easier to do in a passive and more wide way.

>=20
> I just saw an email from Michael suggesting to write an ID, may be =
that is a good starting point if you are planning to take this effort =
seriously. Atleast the draft should have the use case clearly outlined =
and the reasons why the existing mechanisms aren't good enough.=20

After hearing the community comments I'm not sure that just this would =
be worth the effort.  Also I'm not sure if Operating System would =
implement it by default. If not then it would become another iperf that =
you have to configure in both ends.

Do you think such a little tool would be worth it? Not just the effort =
but the fact of adding complexity to TCP.

However as I mentioned in a previous mail I was also thinking in a more =
generic and extensible report system in case we agree that it could =
benefit aid the monitor of TCP in a more generic way.


>=20
> Thanks,
> -Anantha
>=20
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf =
Of
>> =C1ngel Bartomeu Bonillo
>> Sent: Monday, February 20, 2012 8:31 AM
>> To: t.petch
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] Aid passive estimate of tcp performance
>>=20
>> Hi Tom.
>> What you say was my original point. However I have been overwhelmed =
by
>> the amount of drawbacks pointed out by the other folks.
>> My idea was to enable TCP to give a little report at the end of the
>> session. This report should be concise and simple. Just accounting =
for
>> any interesting parameters that are either difficult or impossible to
>> be obtained without an explicit procedure. I was thinking about the
>> estimate of the packet loss but I was hopping that other (more expert
>> people) would find any other interesting usages.
>> However, as I said, I think there have been too much negative =
feedback
>> on this idea so far.
>>=20
>> What does everyone think?
>> Should we give up with this issue?
>>=20
>> Thank you all.
>>=20
>> El 17/02/2012, a las 16:46, t.petch escribi=F3:
>>=20
>>> ----- Original Message -----
>>> From: "=C1ngel Bartomeu Bonillo" <abartomeu@swissms.ch>
>>> To: "rick jones" <perfgeek@mac.com>
>>> Cc: <tcpm@ietf.org>
>>> Sent: Wednesday, February 08, 2012 6:52 PM
>>> Hi Rick,
>>> El 08/02/2012, a las 17:32, rick jones escribi=F3:
>>>=20
>>>>=20
>>>> On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
>>>>=20
>>>>> Hello,
>>>>>=20
>>>>> I recently faced the problem of passively estimating the packet
>> loss of TCP
>>> connection. After a bit of investigating the literature I found that
>> there are
>>> tens of papers addressing this issue but I couldn't find any perfect
>> solution.
>>>>> Since this is an important issue for network operators I have been
>> thinking
>>> that there would be a very easy solution by just making the two TCP
>> connection
>>> end points to account for its sent segments and report it to the
>> other end point
>>> in the FIN segment.
>>>>> This approach could be used as well for other measurable =
parameters
>> along the
>>> TCP session without the need for any other offline method and It
>> would be very
>>> helpful for detecting failures along networks.
>>>>>=20
>>>>> Any comments?
>>>>=20
>>>> If the count is reported in a FIN segment, isn't that rather like
>> closing the
>>> barn door after the horses have left?  At least as far as doing
>> anything for
>>> that connection is concerned.  So, already we are talking about
>> operating on
>>> timescales larger than connection lifetimes.  And what would the =
side
>> receiving
>>> the count-containing FIN do with this information? Sure, now it is
>> known on both
>>> ends, but the problem still lies somewhere in the middle (not =
always,
>> but
>>> often).  Is it expected that points along the middle will be =
sniffing
>> this?
>>> What would they do in response?  What could they (reasonably) do in
>> response
>>> with that information just from TCP vs any other transports?
>>>>=20
>>>=20
>>> I'm talking about timescales larger than the connection lifetimes.
>> This would be
>>> a mechanism so TCP can report how many packets it sent out. If both
>> ends send at
>>> the end of the session the number of packets they sent out the other
>> peer and
>>> anyone in the middle would be able to estimate de loss in both
>> directions.
>>>=20
>>> The idea is to enable TCP so it can report a little but valuable
>> information
>>> about its performance once the connection finished. This is not a
>> mechanism so
>>> TCP can adapt to losses in the middle of a communication but rather =
a
>> long term
>>> mechanism for network operators and such. Nowadays many of the end
>> user services
>>> use TCP and the end to end packet loss is many times impossible to
>> calculate
>>> mainly because only the sending node knows the real number of =
packets
>> sent out.
>>> For example in the uplink direction on a mobile network.
>>> I think that adding just a couple of bytes (as a TCP option) to the
>> FIN segment
>>> would give a priceless information for network operators.
>>>=20
>>> <tp>
>>> Apologies for being late to the party but I think that it is a =
lovely
>> idea.  As
>>> has been pointed out, the figures that you would collect are not =
what
>> you want,
>>> but they would be a lot closer than anything available at the =
moment,
>> and that
>>> is engineering.  Ditto comments about RST; if there is a positive
>> benefit to
>>> shutting down properly and getting these figures, then that could
>> discourage
>>> RSTs.
>>>=20
>>> Tom Petch
>>> <tp>
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>> When a sending TCP sends the first FIN with a count at the end of a
>> MB window
>>> would it retransmit the FIN, even if the first was received, to
>> include
>>> retransmissions in that last window's-worth of data?  How long would
>> the
>>> receiving TCP wait for this "final tally, really" FIN to arrive =
after
>> it sent
>>> the ACK to SEQfin to then pass-along the count to a higher layer,
>> somehow?
>>>=20
>>> The idea much simpler: the peer adds the num of packets sent during
>> the
>>> connection as an option in the FIN. There is no need to change TCP
>> behavior.
>>>=20
>>>>=20
>>>> I think we are already asking for enough angels to dance on the TCP
>>> pin-header.
>>>>=20
>>>> rick jones
>>>=20
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>=20
>>>=20
>>=20
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm



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Interesting tool Alexander.


El 20/02/2012, a las 19:20, Alexander Zimmermann escribi=F3:

>=20
> Am 20.02.2012 um 18:53 schrieb rick jones:
>=20
>>=20
>> On Feb 20, 2012, at 9:39 AM, Anantha Ramaiah (ananth) wrote:
>>> - There are many existing mechanisms that can be useful to obtain =
the similar information which you are looking for, like :-
>>> In cases where it is feasible, active performance methods like the =
one outlined in RFC 6349 can be used (several programs like TTCP, iPerf =
etc.,  are there and can be useful),
>>=20
>> FWIW, netperf (2.5.0 and later) can, under Linux, report the =
retransmission statistics from a TCP_INFO getsockopt() and will do so =
for either end.
>=20
> For the sake of completeness flowgrind =
(https://launchpad.net/flowgrind) can do this too :-)
> See http://www.umic-mesh.net/research/flowgrind/output.html
>=20
> Alex
>=20
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // mobile: (49-176) 32820027, fax: (49-241) 80-22222
> // email: zimmermann@cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div>Interesting tool Alexander.</div><div><br></div><br><div><div>El =
20/02/2012, a las 19:20, Alexander Zimmermann escribi=F3:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div><br>Am =
20.02.2012 um 18:53 schrieb rick jones:<br><br><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">On Feb 20, =
2012, at 9:39 AM, Anantha Ramaiah (ananth) =
wrote:<br></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">- There are many existing mechanisms that can be useful to =
obtain the similar information which you are looking for, like =
:-<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">In cases where it is feasible, active performance methods =
like the one outlined in RFC 6349 can be used (several programs like =
TTCP, iPerf etc., &nbsp;are there and can be =
useful),<br></blockquote></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">FWIW, netperf =
(2.5.0 and later) can, under Linux, report the retransmission statistics =
from a TCP_INFO getsockopt() and will do so for either =
end.<br></blockquote><br>For the sake of completeness flowgrind (<a =
href=3D"https://launchpad.net/flowgrind">https://launchpad.net/flowgrind</=
a>) can do this too :-)<br>See <a =
href=3D"http://www.umic-mesh.net/research/flowgrind/output.html">http://ww=
w.umic-mesh.net/research/flowgrind/output.html</a><br><br>Alex<br><br>//<b=
r>// Dipl.-Inform. Alexander Zimmermann<br>// Department of Computer =
Science, Informatik 4<br>// RWTH Aachen University<br>// Ahornstr. 55, =
52056 Aachen, Germany<br>// mobile: (49-176) 32820027, fax: (49-241) =
80-22222<br>// email: <a =
href=3D"mailto:zimmermann@cs.rwth-aachen.de">zimmermann@cs.rwth-aachen.de<=
/a><br>// web: <a =
href=3D"http://www.umic-mesh.net">http://www.umic-mesh.net</a><br>//<br><b=
r>_______________________________________________<br>tcpm mailing =
list<br><a =
href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>https://www.ietf.org/ma=
ilman/listinfo/tcpm<br></div></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div><p =
style=3D"margin-bottom: 0cm; "><font color=3D"#333333"><font =
face=3D"Arial, sans-serif"><font><i>Este mensaje contiene informaci=F3n =
reservada y confidencial destinada exclusivamente al destinatario. Si =
usted no es el destinatario no est=E1 autorizado a copiar, reproducir o =
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style=3D"margin-bottom: 0cm; "><i style=3D"font-family: Arial; color: =
rgb(51, 51, 51); ">The information in this e-mail is confidential and =
may be legally privileged. It is intended solely for the addressee(s). =
It may contain privileged and confidential information. If you are not =
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class=3D"Apple-interchange-newline">
</div>
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From: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
To: =?iso-8859-1?Q?=C1ngel_Bartomeu_Bonillo?= <abartomeu@swissms.ch>
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Cc: tcpm@ietf.org, "t.petch" <ietfa@btconnect.com>
Subject: Re: [tcpm] Aid passive estimate of tcp performance
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>=20
> Do you think such a little tool would be worth it? Not just the effort
> but the fact of adding complexity to TCP.
>=20
> However as I mentioned in a previous mail I was also thinking in a =
more
> generic and extensible report system in case we agree that it could
> benefit aid the monitor of TCP in a more generic way.


RFC 6349 was about TCP performance benchmarking, the same rationale can =
be applied to this effort since it is related.. but the key difference =
is that this calls for some minor additions to TCP. I agree that it =
would be a useful thing if TCPM wants to look into it.

OTOH, I would think such an effort could fall into the jurisdiction of =
IPPM WG, but if it has TCP changes then it would anyways need to come to =
TCPM.=20

-Anantha
>=20
>=20
> >
> > Thanks,
> > -Anantha
> >
> >> -----Original Message-----
> >> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On =
Behalf
> Of
> >> =C1ngel Bartomeu Bonillo
> >> Sent: Monday, February 20, 2012 8:31 AM
> >> To: t.petch
> >> Cc: tcpm@ietf.org
> >> Subject: Re: [tcpm] Aid passive estimate of tcp performance
> >>
> >> Hi Tom.
> >> What you say was my original point. However I have been overwhelmed
> by
> >> the amount of drawbacks pointed out by the other folks.
> >> My idea was to enable TCP to give a little report at the end of the
> >> session. This report should be concise and simple. Just accounting
> for
> >> any interesting parameters that are either difficult or impossible
> to
> >> be obtained without an explicit procedure. I was thinking about the
> >> estimate of the packet loss but I was hopping that other (more
> expert
> >> people) would find any other interesting usages.
> >> However, as I said, I think there have been too much negative
> feedback
> >> on this idea so far.
> >>
> >> What does everyone think?
> >> Should we give up with this issue?
> >>
> >> Thank you all.
> >>
> >> El 17/02/2012, a las 16:46, t.petch escribi=F3:
> >>
> >>> ----- Original Message -----
> >>> From: "=C1ngel Bartomeu Bonillo" <abartomeu@swissms.ch>
> >>> To: "rick jones" <perfgeek@mac.com>
> >>> Cc: <tcpm@ietf.org>
> >>> Sent: Wednesday, February 08, 2012 6:52 PM
> >>> Hi Rick,
> >>> El 08/02/2012, a las 17:32, rick jones escribi=F3:
> >>>
> >>>>
> >>>> On Feb 8, 2012, at 4:15 AM, =C1ngel Bartomeu Bonillo wrote:
> >>>>
> >>>>> Hello,
> >>>>>
> >>>>> I recently faced the problem of passively estimating the packet
> >> loss of TCP
> >>> connection. After a bit of investigating the literature I found
> that
> >> there are
> >>> tens of papers addressing this issue but I couldn't find any
> perfect
> >> solution.
> >>>>> Since this is an important issue for network operators I have
> been
> >> thinking
> >>> that there would be a very easy solution by just making the two =
TCP
> >> connection
> >>> end points to account for its sent segments and report it to the
> >> other end point
> >>> in the FIN segment.
> >>>>> This approach could be used as well for other measurable
> parameters
> >> along the
> >>> TCP session without the need for any other offline method and It
> >> would be very
> >>> helpful for detecting failures along networks.
> >>>>>
> >>>>> Any comments?
> >>>>
> >>>> If the count is reported in a FIN segment, isn't that rather like
> >> closing the
> >>> barn door after the horses have left?  At least as far as doing
> >> anything for
> >>> that connection is concerned.  So, already we are talking about
> >> operating on
> >>> timescales larger than connection lifetimes.  And what would the
> side
> >> receiving
> >>> the count-containing FIN do with this information? Sure, now it is
> >> known on both
> >>> ends, but the problem still lies somewhere in the middle (not
> always,
> >> but
> >>> often).  Is it expected that points along the middle will be
> sniffing
> >> this?
> >>> What would they do in response?  What could they (reasonably) do =
in
> >> response
> >>> with that information just from TCP vs any other transports?
> >>>>
> >>>
> >>> I'm talking about timescales larger than the connection lifetimes.
> >> This would be
> >>> a mechanism so TCP can report how many packets it sent out. If =
both
> >> ends send at
> >>> the end of the session the number of packets they sent out the
> other
> >> peer and
> >>> anyone in the middle would be able to estimate de loss in both
> >> directions.
> >>>
> >>> The idea is to enable TCP so it can report a little but valuable
> >> information
> >>> about its performance once the connection finished. This is not a
> >> mechanism so
> >>> TCP can adapt to losses in the middle of a communication but =
rather
> a
> >> long term
> >>> mechanism for network operators and such. Nowadays many of the end
> >> user services
> >>> use TCP and the end to end packet loss is many times impossible to
> >> calculate
> >>> mainly because only the sending node knows the real number of
> packets
> >> sent out.
> >>> For example in the uplink direction on a mobile network.
> >>> I think that adding just a couple of bytes (as a TCP option) to =
the
> >> FIN segment
> >>> would give a priceless information for network operators.
> >>>
> >>> <tp>
> >>> Apologies for being late to the party but I think that it is a
> lovely
> >> idea.  As
> >>> has been pointed out, the figures that you would collect are not
> what
> >> you want,
> >>> but they would be a lot closer than anything available at the
> moment,
> >> and that
> >>> is engineering.  Ditto comments about RST; if there is a positive
> >> benefit to
> >>> shutting down properly and getting these figures, then that could
> >> discourage
> >>> RSTs.
> >>>
> >>> Tom Petch
> >>> <tp>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>> When a sending TCP sends the first FIN with a count at the end of
> a
> >> MB window
> >>> would it retransmit the FIN, even if the first was received, to
> >> include
> >>> retransmissions in that last window's-worth of data?  How long
> would
> >> the
> >>> receiving TCP wait for this "final tally, really" FIN to arrive
> after
> >> it sent
> >>> the ACK to SEQfin to then pass-along the count to a higher layer,
> >> somehow?
> >>>
> >>> The idea much simpler: the peer adds the num of packets sent =
during
> >> the
> >>> connection as an option in the FIN. There is no need to change TCP
> >> behavior.
> >>>
> >>>>
> >>>> I think we are already asking for enough angels to dance on the
> TCP
> >>> pin-header.
> >>>>
> >>>> rick jones
> >>>
> >>> _______________________________________________
> >>> 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


From internet-drafts@ietf.org  Fri Feb 24 14:20:04 2012
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-proportional-rate-reduction-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the TCP Maintenance and Minor Extensions =
Working Group of the IETF.

	Title           : Proportional Rate Reduction for TCP
	Author(s)       : Matt Mathis
                          Nandita Dukkipati
                          Yuchung Cheng
	Filename        : draft-ietf-tcpm-proportional-rate-reduction-01.txt
	Pages           : 16
	Date            : 2012-02-24

   This document describes an experimental algorithm, Proportional Rate
   Reduction (PPR) to improve the accuracy of the amount of data sent by
   TCP during loss recovery.  Standard Congestion Control requires that
   TCP and other protocols reduce their congestion window in response to
   losses.  This window reduction naturally occurs in the same round
   trip as the data retransmissions to repair the losses, and is
   implemented by choosing not to transmit any data in response to some
   ACKs arriving from the receiver.  Two widely deployed algorithms are
   used to implement this window reduction: Fast Recovery and Rate
   Halving.  Both algorithms are needlessly fragile under a number of
   conditions, particularly when there is a burst of losses that such
   that the number of ACKs returning to the sender is small.
   Proportional Rate Reduction minimizes these excess window reductions
   such that at the end of recovery the actual window size will be as
   close as possible to ssthresh, the window size determined by the
   congestion control algorithm.  It is patterned after Rate Halving,
   but using the fraction that is appropriate for target window chosen
   by the congestion control algorithm.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-proportional-rate-reduc=
tion-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-proportional-rate-reduct=
ion-01.txt


From internet-drafts@ietf.org  Sun Feb 26 23:43:55 2012
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Date: Sun, 26 Feb 2012 23:43:54 -0800
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-initcwnd-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the TCP Maintenance and Minor Extensions =
Working Group of the IETF.

	Title           : Increasing TCP's Initial Window
	Author(s)       : Jerry Chu
                          Nandita Dukkipati
                          Yuchung Cheng
                          Matt Mathis
	Filename        : draft-ietf-tcpm-initcwnd-03.txt
	Pages           : 22
	Date            : 2012-02-26

   This document proposes an increase in the permitted TCP initial
   window (IW) from between 2 and 4 segments, as specified in RFC 3390,
   to 10 segments. It discusses the motivation behind the increase, the
   advantages and disadvantages of the higher initial window, and
   presents results from several large scale experiments showing that
   the higher initial window improves the overall performance of many
   web services without risking congestion collapse. The document closes
   with a discussion of usage and deployment recommended by the IETF TCP
   Maintenance and Minor Extensions (TCPM) working group.

Terminology

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-initcwnd-03.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-initcwnd-03.txt


From lars@netapp.com  Mon Feb 27 01:08:32 2012
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From: "Eggert, Lars" <lars@netapp.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: draft-williams-overlaypath-ip-tcp-rfc-00.txt
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Hi,

from draft-williams-overlaypath-ip-tcp-rfc-00.txt:

>    8.  IANA Considerations
>=20
>    The assignment of numbers 26 for IPv4, 11 for IPv6, and 30 for TCP
>    are provisional pending official IANA approval.  The Type values
>    could change if IANA assigns different numbers.
>=20
>    It may be important to note that some early implementations of this
>    specification use TCP option numbers 28 and 29 due to the fact that
>    these numbers were available at the time of the implementation.
>    These early implementations will require updates after an official
>    option number has been assigned.

We should instruct IANA to indicate in the options number registry that =
options 28 (TCP-UTO) and 29 (TCP-AO) have also been inappropriately used =
for other then the assigned TCP extensions.

(This is esp. unfortunate for AO! We had the clash on 253/254, and now =
we have another one. Sigh.)

Lars=

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Hi,

The working group last call has now ended. During the WGLC there was one =
comment from Richard (thanks!), who supported the draft, and no requests =
for changes. We will therefore pass the draft forward to ADs for =
publication.

- Pasi


On Feb 8, 2012, at 3:01 PM, Pasi Sarolahti wrote:

> Hello,
>=20
> The authors of "A Conservative Selective Acknowledgment (SACK)-based =
Loss Recovery Algorithm for TCP" (draft-ietf-tcpm-3517bis-01) think the =
latest version addresses all comments made on the draft. Therefore, this =
mail starts Working Group Last Call for the draft, intended to be =
published as Proposed Standard. The WGLC ends on Friday, February 24, =
i.e., within a bit over two weeks.=20
>=20
> Please read the draft and send any comments to the TCPM mailing list =
by the above deadline. If you think the draft is ok and ready to be =
published without further changes, indicating that would also be =
helpful.
>=20
> The draft is available at =
http://tools.ietf.org/html/draft-ietf-tcpm-3517bis-01
>=20
> - Pasi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From alexander.zimmermann@comsys.rwth-aachen.de  Mon Feb 27 06:02:17 2012
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-3517bis-01
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Hi Pasi,

I also support the doc. My comments are already incorporated in a =
previous version of the draft.

Alex

Am 27.02.2012 um 14:53 schrieb Pasi Sarolahti:

> Hi,
>=20
> The working group last call has now ended. During the WGLC there was =
one comment from Richard (thanks!), who supported the draft, and no =
requests for changes. We will therefore pass the draft forward to ADs =
for publication.
>=20
> - Pasi
>=20
>=20
> On Feb 8, 2012, at 3:01 PM, Pasi Sarolahti wrote:
>=20
>> Hello,
>>=20
>> The authors of "A Conservative Selective Acknowledgment (SACK)-based =
Loss Recovery Algorithm for TCP" (draft-ietf-tcpm-3517bis-01) think the =
latest version addresses all comments made on the draft. Therefore, this =
mail starts Working Group Last Call for the draft, intended to be =
published as Proposed Standard. The WGLC ends on Friday, February 24, =
i.e., within a bit over two weeks.=20
>>=20
>> Please read the draft and send any comments to the TCPM mailing list =
by the above deadline. If you think the draft is ok and ready to be =
published without further changes, indicating that would also be =
helpful.
>>=20
>> The draft is available at =
http://tools.ietf.org/html/draft-ietf-tcpm-3517bis-01
>>=20
>> - Pasi
>>=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

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// mobile: (49-176) 32820027, fax: (49-241) 80-22222
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
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From touch@isi.edu  Mon Feb 27 09:55:33 2012
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc-00.txt
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Agreed.

It's not an issue for AO, though - so long as 29 is interpreted as AO 
for compliant implementations, the keying should end up just dumping 
anyone else's segments that use 29 as "malformed" or incorrectly keyed.

Joe

On 2/27/2012 1:07 AM, Eggert, Lars wrote:
> Hi,
>
> from draft-williams-overlaypath-ip-tcp-rfc-00.txt:
>
>>     8.  IANA Considerations
>>
>>     The assignment of numbers 26 for IPv4, 11 for IPv6, and 30 for TCP
>>     are provisional pending official IANA approval.  The Type values
>>     could change if IANA assigns different numbers.
>>
>>     It may be important to note that some early implementations of this
>>     specification use TCP option numbers 28 and 29 due to the fact that
>>     these numbers were available at the time of the implementation.
>>     These early implementations will require updates after an official
>>     option number has been assigned.
>
> We should instruct IANA to indicate in the options number registry that options 28 (TCP-UTO) and 29 (TCP-AO) have also been inappropriately used for other then the assigned TCP extensions.
>
> (This is esp. unfortunate for AO! We had the clash on 253/254, and now we have another one. Sigh.)
>
> Lars
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From nishida@sfc.wide.ad.jp  Tue Feb 28 12:14:56 2012
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Subject: Re: [tcpm] Minimum nonce length in draft-touch-tcpm-experimental-options
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Hi,

On Mon, Feb 20, 2012 at 3:34 AM, SCHARF, Michael
<Michael.Scharf@alcatel-lucent.com> wrote:
> Joe,
>
> The wording on the nonce length in draft-touch-tcpm-experimental-options
> may be a little bit unclear ("it MAY be as few as 16 bits if desired").
>
> Does that statement intend to say something like "MUST NOT be shorter
> than 16 bits long"? Or shall even a nonce of length 0 be allowed,
> provided that there are additional security mechanism such as checksums
> or digital signatures? In the latter case, existing use of the
> codepoints without nonce could be compliant to the recommendation,
> provided that there are other means to detect false positives.

I don't have strong opinion on this yet, but I personally think we
don't have to mention minimum length specifically in the draft.
However, I might prefer to put a little bit more stress in this regards, like,
   o It's implementor's responsibility to find good balance between
the overhead of magic number and the overhead to handle false
positive.
   o the draft doesn't guarantee anything even if an implementor
chooses longer magic number

Thanks,
--
Yoshifumi Nishida

From pasi.sarolahti@iki.fi  Wed Feb 29 12:08:34 2012
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Subject: Re: [tcpm] Minimum nonce length in draft-touch-tcpm-experimental-options
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On Feb 28, 2012, at 10:14 PM, Yoshifumi Nishida wrote:

>> Does that statement intend to say something like "MUST NOT be shorter
>> than 16 bits long"? Or shall even a nonce of length 0 be allowed,
>> provided that there are additional security mechanism such as =
checksums
>> or digital signatures? In the latter case, existing use of the
>> codepoints without nonce could be compliant to the recommendation,
>> provided that there are other means to detect false positives.
>=20
> I don't have strong opinion on this yet, but I personally think we
> don't have to mention minimum length specifically in the draft.
> However, I might prefer to put a little bit more stress in this =
regards, like,
>   o It's implementor's responsibility to find good balance between
> the overhead of magic number and the overhead to handle false
> positive.
>   o the draft doesn't guarantee anything even if an implementor
> chooses longer magic number

I agree. Given the scarcity of TCP option space, you do not want to use =
any more space for the magic number than necessary. Maybe one possible =
option could be to use a 32-bit magic number in the SYN handshake, to =
declare that for the rest of that connection option types 253 and 254 =
are used without magic number for feature X (or perhaps with a shorter =
magic number, to allow simultaneous experimental features). This could =
be useful when the option space is expected to get tight.

- Pasi


From Michael.Scharf@alcatel-lucent.com  Wed Feb 29 12:29:41 2012
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Thread-Topic: [tcpm] Minimum nonce length in draft-touch-tcpm-experimental-options
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From: "SCHARF, Michael" <Michael.Scharf@alcatel-lucent.com>
To: "Pasi Sarolahti" <pasi.sarolahti@iki.fi>, "Yoshifumi Nishida" <nishida@sfc.wide.ad.jp>
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Cc: tcpm@ietf.org, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] Minimum nonce length in draft-touch-tcpm-experimental-options
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> >> Does that statement intend to say something like "MUST NOT=20
> be shorter=20
> >> than 16 bits long"? Or shall even a nonce of length 0 be allowed,=20
> >> provided that there are additional security mechanism such as=20
> >> checksums or digital signatures? In the latter case,=20
> existing use of=20
> >> the codepoints without nonce could be compliant to the=20
> >> recommendation, provided that there are other means to=20
> detect false positives.
> >=20
> > I don't have strong opinion on this yet, but I personally think we=20
> > don't have to mention minimum length specifically in the draft.
> > However, I might prefer to put a little bit more stress in=20
> this regards, like,
> >   o It's implementor's responsibility to find good balance=20
> between the=20
> > overhead of magic number and the overhead to handle false positive.
> >   o the draft doesn't guarantee anything even if an implementor=20
> > chooses longer magic number
>=20
> I agree. Given the scarcity of TCP option space, you do not=20
> want to use any more space for the magic number than=20
> necessary. Maybe one possible option could be to use a 32-bit=20
> magic number in the SYN handshake, to declare that for the=20
> rest of that connection option types 253 and 254 are used=20
> without magic number for feature X (or perhaps with a shorter=20
> magic number, to allow simultaneous experimental features).=20
> This could be useful when the option space is expected to get tight.

Option space is particularly scarce in SYNs. Afterwards one could try to
piggyback a long option to an empty ACK instead. Thus, this only helps
if a rather long option must indeed be present in a data segment, right?

Michael

From touch@isi.edu  Wed Feb 29 13:15:36 2012
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Subject: Re: [tcpm] Minimum nonce length in draft-touch-tcpm-experimental-options
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On 2/29/2012 12:29 PM, SCHARF, Michael wrote:
>>>> Does that statement intend to say something like "MUST NOT
>> be shorter
>>>> than 16 bits long"? Or shall even a nonce of length 0 be allowed,
>>>> provided that there are additional security mechanism such as
>>>> checksums or digital signatures? In the latter case,
>> existing use of
>>>> the codepoints without nonce could be compliant to the
>>>> recommendation, provided that there are other means to
>> detect false positives.
>>>
>>> I don't have strong opinion on this yet, but I personally think we
>>> don't have to mention minimum length specifically in the draft.
>>> However, I might prefer to put a little bit more stress in
>> this regards, like,
>>>    o It's implementor's responsibility to find good balance
>> between the
>>> overhead of magic number and the overhead to handle false positive.
>>>    o the draft doesn't guarantee anything even if an implementor
>>> chooses longer magic number
>>
>> I agree. Given the scarcity of TCP option space, you do not
>> want to use any more space for the magic number than
>> necessary. Maybe one possible option could be to use a 32-bit
>> magic number in the SYN handshake, to declare that for the
>> rest of that connection option types 253 and 254 are used
>> without magic number for feature X (or perhaps with a shorter
>> magic number, to allow simultaneous experimental features).
>> This could be useful when the option space is expected to get tight.
>
> Option space is particularly scarce in SYNs. Afterwards one could try to
> piggyback a long option to an empty ACK instead. Thus, this only helps
> if a rather long option must indeed be present in a data segment, right?
>
> Michael

There could be multiple instances of option 253 or 254, and the only way 
to know they're distinct is the magic number. Otherwise, any shortcut 
you use could conflict with someone else's.

So there's no shortcut here; as you note, this isn't helping the more 
critical location (SYNs) anyway.

Joe



From mcmanus@ducksong.com  Wed Feb 29 15:02:13 2012
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On Wed, 2012-02-08 at 22:31 -0500, Mark Allman wrote:
> > Can't agree more. We need a venue for two-way communications.
> > 
> > app dev: TCP sucks because blah blah blah
> > TCP dudes: app sucks because blah blah blah
> 
> Wellllll..... that is one reason everyone rearranges all the furniture
> in a big damn hotel three times a year.  But, alas, you can lead a horse
> to water ...

[Apologies for appending to an old thread. I read tcpm in batches every
few weeks.]

Hi all.

I develop network gunk for firefox - if you think there is something we
could be doing to make the web more capable, more open, and more awesome
please reach out to me.

A few of you have worked on some topic with me in the past, and I like
to think we're both better for it. I'm a reasonable guy who loves to
talk about these interactions and what is realistic to do about them. My
counterparts at other browsers, at least the ones I know, are good
people too - all trying to make a better Internet while balancing many
needs and constraints.

So that's something concrete for you - a person to engage. I'll be in
Paris too.

BTW - I'm not trying to just throw this over the wall and say "call me
if you need me", I think tcpm is part of my community and try and keep
up and contribute. My community is just a big world (from HTML5 to
ethernet), so stuff slips through and the needs may not necessarily
align.

Congenially,
Patrick
aka pmcmanus@mozilla.com



