
From micchie@sfc.wide.ad.jp  Sat Oct  2 16:30:24 2010
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From: Michio Honda <micchie@sfc.wide.ad.jp>
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Date: Sun, 3 Oct 2010 00:30:57 +0100
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Cc: Mark Handley <m.handley@cs.ucl.ac.uk>
Subject: [tcpm] Call for contribution to middlebox survey
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Hi,=20

We are surveying middleboxes affecting TCP in the Internet, and we'd =
like you to contribute to this work by running 1 python script at your =
available networks, because we want data of as many paths as possible.   =
=20
This script generates test TCP traffic to a server node, and detects =
various middlebox behavior, for example, it detects how unknown TCP =
options are treated and if sequence number is rewritten. =20

- Overview of script
This generates test TCP traffic by using raw socket or pcap.=20
Destinations of the test traffic are port 80, 443 and 34343 on =
vinson3.sfc.wide.ad.jp, which is located in Japan.=20
The total amount of test traffic is approximately 90 connections (not =
parallel), and each of them uses approximately maximum 2048Byte.   =20

- System requirement
Our script works on Mac OSX 10.5 or 10.6, Linux (kernel 2.6) and FreeBSD =
(7.0 or higher).  This also requires python 2.5 or higher, and libpcap
NOTE. if you try in a virtual machine on Windows, please connect the =
guest OS via not NAT but bridge.  =20

How to run experiment is described below per-OS basis. =20

After the experiment, you will find 3 log files (logxxxxxxxxx.txt) in =
the same directory as the experiment. =20
Please send them to us (micchie@sfc.wide.ad.jp) and tell me your network =
information as much as you know (e.g., product name of the broadband =
router, ISP name, product name of firewall appliance etc...)=20
In addition, let us know if you have hesitation to open these =
information. =20
This experiment doesn't collect traffic information other than those our =
script generated. =20

***** How to run the experiment (Mac OSX) *****

1. Filtering RST TCP segment from OS
Execute a following command by root:
ipfw add 101 deny tcp from any to vinson3.sfc.wide.ad.jp dst-port =
34343,80,443 tcpflags rst

NOTE: if you are already running ipfw, please add equivalent rules
After the experiment, you can revert by "ipfw delete 101"

2. Executing script
Download script from =
http://www.micchie.net/software/tcpexposure/for_distrib.tar.gz, and =
decompress it to anywhere you like (e.g., tar xzf for_distrib.tar.gz by =
command line)

In the for_distrib directory, execute a following command by root:
sh run-bsd2.sh
(This will take approximately 30 min.)


***** How to run the experiment (Linux) *****

1. Filtering RST TCP segment from OS
Execute following command by root:
/sbin/iptables -A OUTPUT -p tcp -d vinson3.sfc.wide.ad.jp --tcp-flags =
RST RST -m multiport --dports 34343,80,443 -j DROP

NOTE: if you are already running iptables, please add equivalent rules
After the experiment, you can revert by opposite commands - using -D =
instead of -A

2. Executing script
Download script from =
http://www.micchie.net/software/tcpexposure/for_distrib.tar.gz, and =
decompress it to anywhere you like (e.g., tar xzf for_distrib.tar.gz)

In the for_distrib directory, execute a following command by root:
sh run-linux2.sh
(This will take approximately 30 min.)


***** How to run the script (FreeBSD) *****

1. Filtering RST TCP segment from OS  =20
If you are using neither ipfw nor pf:=20
 Load pf kernel module with a following command by root:
 kldload /boot/kernel/pf.ko

 Add following 2 lines to /etc/pf.conf (please replace IFNAME to your =
outgoing interface name (e.g., em0):
 pass out all
 block out quick on IFNAME proto tcp to vinson3.sfc.wide.ad.jp port =
{34343,80,443} flags R/R

 Execute following command by root:
 pfctl -e -f /etc/pf.conf

If you are already running pf, please add equivalent rules
 After the experiment, you can revert settings by  cleaning up =
/etc/pf.conf and executing "pfctl -d" by root

If you are already using ipfw:
  Please add a following rule to ipfw configuration:
  deny tcp from any to vinson3.sfc.wide.ad.jp dst-port 34343,80,443 =
tcpflags rst

2. Executing script
Download script from =
http://www.micchie.net/software/tcpexposure/for_distrib.tar.gz, and =
decompress it to anywhere you like (e.g., tar xzf for_distrib.tar.gz)

In the for_distrib directory, execute a following command by root:
sh run-bsd2.sh
(This will take approximately 30 min.)


Best regards,
- Michio


From lars.eggert@nokia.com  Tue Oct  5 01:32:12 2010
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Hi, folks,

where are we with resolving the DISCUSSes on this document? 

Thanks,
Lars
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To: Lars Eggert <lars.eggert@nokia.com>, Jari Arkko <jari.arkko@piuha.net>
Date: Tue, 5 Oct 2010 10:17:28 -0400
Thread-Topic: [tcpm] draft-ietf-tcpm-urgent-data-06
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References: <201009301549.RAA14282@TR-Sys.de> <13205C286662DE4387D9AF3AC30EF456B01839E20B@EMBX01-WF.jnpr.net> <31679F6B-0ACA-421F-8131-DA05B764A4A9@nokia.com>
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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I cleared, assuming that Jari would hold my discuss.

                               Ron


> -----Original Message-----
> From: Lars Eggert [mailto:lars.eggert@nokia.com]
> Sent: Tuesday, October 05, 2010 4:33 AM
> To: Ronald Bonica; Jari Arkko
> Cc: draft-ietf-tcpm-urgent-data.all@tools.ietf.org; tcpm@ietf.org
> Extensions
> Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
>=20
> Hi, folks,
>=20
> where are we with resolving the DISCUSSes on this document?
>=20
> Thanks,
> Lars

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.


	Title           : Increasing TCP's Initial Window
	Author(s)       : J. Chu, et al.
	Filename        : draft-ietf-tcpm-initcwnd-00.txt
	Pages           : 17
	Date            : 2010-10-05

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. Finally, it
outlines a list of concerns to be addressed in future tests.

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

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From alexander.zimmermann@comsys.rwth-aachen.de  Wed Oct  6 02:53:59 2010
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Subject: Re: [tcpm] TCP window max size?
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Hi David,

we had recently also a discussion here in our group about
max window size. What I understand form your example that
a size of 2^31 is not possible since your example stated that it 
would fail. But in theory (regardless the scaling factor) what
is the maximal possible window size so that everything would go ok?
(2^31)-1?

Another question that comes in my mind: TCP says sequence number
x < y <=> 0 < y-x < 2^31. Why do we say here 2^31 and not for example
2^32-20?

Alex

Am 25.08.2010 um 00:07 schrieb David Borman:

> 
> On Aug 24, 2010, at 4:22 PM, David Borman wrote:
> 
>> This reasoning is explained in RFC 1323, section 2.3, at the bottom of
>> page 10 and the top of page 11:
>> 
>>   TCP determines if a data segment is "old" or "new" by testing
>>   whether its sequence number is within 2**31 bytes of the left edge
>>   of the window, and if it is not, discarding the data as "old".  To
>>   insure that new data is never mistakenly considered old and vice-
>>   versa, the left edge of the sender's window has to be at most
>>   2**31 away from the right edge of the receiver's window.
>>   Similarly with the sender's right edge and receiver's left edge.
>>   Since the right and left edges of either the sender's or
>>   receiver's window differ by the window size, and since the sender
>>   and receiver windows can be out of phase by at most the window
>>   size, the above constraints imply that 2 * the max window size
>>   must be less than 2**31, or
>> 
>>        max window < 2**30
>> 
>>   Since the max window is 2**S (where S is the scaling shift count)
>>   times at most 2**16 - 1 (the maximum unscaled window), the maximum
>>   window is guaranteed to be < 2*30 if S <= 14.  Thus, the shift
>>   count must be limited to 14 (which allows windows of 2**30 = 1
>>   Gbyte)...
> 
> And what I forgot to ask was whether you or someone else disagreed with
> this logic, but this is why it is 14.  However, even if the logic is
> faulty, the max value will remain at 14, the only way to change it would
> be via a new TCP option number.
> 
> But mostly the place for trouble is at the boundary conditions, and if
> you had a 2^31 window and the sender/receiver windows were fully skewed,
> any outstanding packet from before left edge of the senders window could
> show up at the receiver and be viewed as new data at the right edge of
> the receivers window, which would be a bad thing.
> 
> 1) Sender has a full 2^31 window of unacknowledged data,
>   receiver is a full 2^31 ahead of the sender.
> 2) Sender retransmits a packet at left edge of it's window.
> 3) In-transit Ack from Receiver arrives, acks all data, and
>   advances window by 2^31.
> 4) Sender sends a new packet.
> 5) New packet gets reordered in transit with the retransmitted packet.
> 6) New packet arrives, and receiver advances window.
> 7) Retransmitted packet arrives, and it now appears to be at the
>   right edge of the receivers window as new data.
> 
> A 2^14 max window keeps things well away from those boundary conditions.
> 
> 

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22222
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


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From steven.sommars@alcatel-lucent.com  Wed Oct  6 08:06:04 2010
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From: "Sommars, Steven E (Steven)" <steven.sommars@alcatel-lucent.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Wed, 6 Oct 2010 10:07:19 -0500
Thread-Topic: missequenced TCP segments
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Subject: [tcpm] missequenced TCP segments
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Vern Paxson suggested this forum for discussing an RFC793 ambiguity.

RFC793 section 3.4 says:
    Reset Generation
    ...
    3.  If the connection is in a synchronized state (ESTABLISHED,
    FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
    any unacceptable segment (out of window sequence number or
    unacceptible acknowledgment number) must elicit only an empty
    acknowledgment segment containing the current send-sequence number
    and an acknowledgment indicating the next sequence number expected
    to be received, and the connection remains in the same state.

The problem comes with a bidirectional TCP flow (both directions carrying p=
ayload) in the presence of IP missequencing.=20

Sender transmits
	Seq 1, Ack 15
	Seq 2, Ack 16
	Seq 3, Ack 17
	Seq 4, Ack 18
Due to packet reordering receiver sees
	Seq 4, Ack 18
	Seq 1, Ack 15  << Ack out of window
	Seq 3, Ack 17  << Ack out of window
	Seq 2, Ack 16  << Ack out of window

The cited RFC793 section suggests that when the Acks are out of window the =
accompanying data segment should not be processed.   That strategy can lead=
 to undesirable TCP stalls.=20

I cannot find a clarification in other RFCs.   Is an erratum appropriate?

From v13@v13.gr  Wed Oct  6 08:31:50 2010
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Subject: Re: [tcpm] missequenced TCP segments
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Hello,

On Wednesday 06 of October 2010, Sommars, Steven E (Steven) wrote:
> Vern Paxson suggested this forum for discussing an RFC793 ambiguity.
> 
> RFC793 section 3.4 says:
>     Reset Generation
>     ...
>     3.  If the connection is in a synchronized state (ESTABLISHED,
>     FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
>     any unacceptable segment (out of window sequence number or
>     unacceptible acknowledgment number) must elicit only an empty
>     acknowledgment segment containing the current send-sequence number
>     and an acknowledgment indicating the next sequence number expected
>     to be received, and the connection remains in the same state.

[...]

> Due to packet reordering receiver sees
> 	Seq 4, Ack 18
> 	Seq 1, Ack 15  << Ack out of window
> 	Seq 3, Ack 17  << Ack out of window
> 	Seq 2, Ack 16  << Ack out of window
> 
> The cited RFC793 section suggests that when the Acks are out of window the
> accompanying data segment should not be processed.   That strategy can
> lead to undesirable TCP stalls.

While I'm not an expert, from what I understand, as you quoted, RFC states 
"unacceptable acknowledgment number" and not "out of window". It also states 
that "the connection remains in the same state" and not that it rejects the 
data. Both of those statements are valid if you just accept the data. In fact, 
assuming that all ACKs with  ACK# >= last_ack-2^31 are valid/acceptable, 
should be OK.

From david.borman@windriver.com  Wed Oct  6 09:32:56 2010
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Subject: Re: [tcpm] missequenced TCP segments
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On Oct 6, 2010, at 10:07 AM, Sommars, Steven E (Steven) wrote:

> Vern Paxson suggested this forum for discussing an RFC793 ambiguity.
>=20
> RFC793 section 3.4 says:
>    Reset Generation
>    ...
>    3.  If the connection is in a synchronized state (ESTABLISHED,
>    FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
>    any unacceptable segment (out of window sequence number or
>    unacceptible acknowledgment number) must elicit only an empty
>    acknowledgment segment containing the current send-sequence number
>    and an acknowledgment indicating the next sequence number expected
>    to be received, and the connection remains in the same state.
>=20
> The problem comes with a bidirectional TCP flow (both directions =
carrying payload) in the presence of IP missequencing.=20
>=20
> Sender transmits
> 	Seq 1, Ack 15
> 	Seq 2, Ack 16
> 	Seq 3, Ack 17
> 	Seq 4, Ack 18
> Due to packet reordering receiver sees
> 	Seq 4, Ack 18
> 	Seq 1, Ack 15  << Ack out of window
> 	Seq 3, Ack 17  << Ack out of window
> 	Seq 2, Ack 16  << Ack out of window
>=20
> The cited RFC793 section suggests that when the Acks are out of window =
the accompanying data segment should not be processed.   That strategy =
can lead to undesirable TCP stalls.

It says unacceptable ACKs, not out of window ACKs.  On page 72:
          If SND.UNA < SEG.ACK =3D< SND.NXT then, set SND.UNA <- =
SEG.ACK.
          Any segments on the retransmission queue which are thereby
          entirely acknowledged are removed.  Users should receive
          positive acknowledgments for buffers which have been SENT and
          fully acknowledged (i.e., SEND buffer should be returned with
          "ok" response).  If the ACK is a duplicate
          (SEG.ACK < SND.UNA), it can be ignored.  If the ACK acks
          something not yet sent (SEG.ACK > SND.NXT) then send an ACK,
          drop the segment, and return.
I.e., duplicate ACKs are acceptable.  The only thing that is =
unacceptable is an ACK for data that hasn't been sent.

I'll also take this opportunity to mention that if an out-of-order =
packet arrives, either due to a gap or duplicate data, the ACK field =
should be processed anyway.  On bi-directional connections, failure to =
process the ACK field in a packet that is purely duplicate data can lead =
to ACK wars on the connection, e.g. in crossing zero window probes.

			-David Borman

>=20
> I cannot find a clarification in other RFCs.   Is an erratum =
appropriate?
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] missequenced TCP segments
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Hi David,

Am 06.10.2010 um 18:33 schrieb David Borman:

>=20
> On Oct 6, 2010, at 10:07 AM, Sommars, Steven E (Steven) wrote:
>=20
>> Vern Paxson suggested this forum for discussing an RFC793 ambiguity.
>>=20
>> RFC793 section 3.4 says:
>>   Reset Generation
>>   ...
>>   3.  If the connection is in a synchronized state (ESTABLISHED,
>>   FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
>>   any unacceptable segment (out of window sequence number or
>>   unacceptible acknowledgment number) must elicit only an empty
>>   acknowledgment segment containing the current send-sequence number
>>   and an acknowledgment indicating the next sequence number expected
>>   to be received, and the connection remains in the same state.
>>=20
>> The problem comes with a bidirectional TCP flow (both directions =
carrying payload) in the presence of IP missequencing.=20
>>=20
>> Sender transmits
>> 	Seq 1, Ack 15
>> 	Seq 2, Ack 16
>> 	Seq 3, Ack 17
>> 	Seq 4, Ack 18
>> Due to packet reordering receiver sees
>> 	Seq 4, Ack 18
>> 	Seq 1, Ack 15  << Ack out of window
>> 	Seq 3, Ack 17  << Ack out of window
>> 	Seq 2, Ack 16  << Ack out of window
>>=20
>> The cited RFC793 section suggests that when the Acks are out of =
window the accompanying data segment should not be processed.   That =
strategy can lead to undesirable TCP stalls.
>=20
> It says unacceptable ACKs, not out of window ACKs.  On page 72:
>          If SND.UNA < SEG.ACK =3D< SND.NXT then, set SND.UNA <- =
SEG.ACK.
>          Any segments on the retransmission queue which are thereby
>          entirely acknowledged are removed.  Users should receive
>          positive acknowledgments for buffers which have been SENT and
>          fully acknowledged (i.e., SEND buffer should be returned with
>          "ok" response).  If the ACK is a duplicate
>          (SEG.ACK < SND.UNA), it can be ignored.  If the ACK acks
>          something not yet sent (SEG.ACK > SND.NXT) then send an ACK,
>          drop the segment, and return.
> I.e., duplicate ACKs are acceptable.  The only thing that is =
unacceptable is an ACK for data that hasn't been sent.

Hmm... IMO an DUPACK is not an acceptable ACK.
An acceptable ACK is an ACK, which acks perviously unacknowledged data.
I like the term "valid" here...

>=20
> I'll also take this opportunity to mention that if an out-of-order =
packet arrives, either due to a gap or duplicate data, the ACK field =
should be processed anyway.  On bi-directional connections, failure to =
process the ACK field in a packet that is purely duplicate data can lead =
to ACK wars on the connection, e.g. in crossing zero window probes.
>=20
> 			-David Borman
>=20
>>=20
>> I cannot find a clarification in other RFCs.   Is an erratum =
appropriate?
>> _______________________________________________
>> 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
// phone: (49-241) 80-21422, fax: (49-241) 80-22222
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


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From david.borman@windriver.com  Wed Oct  6 11:05:21 2010
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Subject: Re: [tcpm] missequenced TCP segments
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On Oct 6, 2010, at 11:57 AM, Alexander Zimmermann wrote:

> Hi David,
>=20
> Am 06.10.2010 um 18:33 schrieb David Borman:
>=20
>>=20
>> On Oct 6, 2010, at 10:07 AM, Sommars, Steven E (Steven) wrote:
>>=20
>>> Vern Paxson suggested this forum for discussing an RFC793 ambiguity.
>>>=20
>>> RFC793 section 3.4 says:
>>>  Reset Generation
>>>  ...
>>>  3.  If the connection is in a synchronized state (ESTABLISHED,
>>>  FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, TIME-WAIT),
>>>  any unacceptable segment (out of window sequence number or
>>>  unacceptible acknowledgment number) must elicit only an empty
>>>  acknowledgment segment containing the current send-sequence number
>>>  and an acknowledgment indicating the next sequence number expected
>>>  to be received, and the connection remains in the same state.
>>>=20
>>> The problem comes with a bidirectional TCP flow (both directions =
carrying payload) in the presence of IP missequencing.=20
>>>=20
>>> Sender transmits
>>> 	Seq 1, Ack 15
>>> 	Seq 2, Ack 16
>>> 	Seq 3, Ack 17
>>> 	Seq 4, Ack 18
>>> Due to packet reordering receiver sees
>>> 	Seq 4, Ack 18
>>> 	Seq 1, Ack 15  << Ack out of window
>>> 	Seq 3, Ack 17  << Ack out of window
>>> 	Seq 2, Ack 16  << Ack out of window
>>>=20
>>> The cited RFC793 section suggests that when the Acks are out of =
window the accompanying data segment should not be processed.   That =
strategy can lead to undesirable TCP stalls.
>>=20
>> It says unacceptable ACKs, not out of window ACKs.  On page 72:
>>         If SND.UNA < SEG.ACK =3D< SND.NXT then, set SND.UNA <- =
SEG.ACK.
>>         Any segments on the retransmission queue which are thereby
>>         entirely acknowledged are removed.  Users should receive
>>         positive acknowledgments for buffers which have been SENT and
>>         fully acknowledged (i.e., SEND buffer should be returned with
>>         "ok" response).  If the ACK is a duplicate
>>         (SEG.ACK < SND.UNA), it can be ignored.  If the ACK acks
>>         something not yet sent (SEG.ACK > SND.NXT) then send an ACK,
>>         drop the segment, and return.
>> I.e., duplicate ACKs are acceptable.  The only thing that is =
unacceptable is an ACK for data that hasn't been sent.
>=20
> Hmm... IMO an DUPACK is not an acceptable ACK.
> An acceptable ACK is an ACK, which acks perviously unacknowledged =
data.
> I like the term "valid" here...

Ok, if you are going to wordsmith, look at it this way: an acceptable =
ACK is one that the TCP state machine knows how to properly process, and =
that includes duplicate ACKs.  It is only ACKs for data that hasn't been =
sent that TCP can't deal with, since that situation shouldn't occur, =
hence those are the unacceptable ACKs, the ones that cause the packet to =
be dropped, an ACK to be sent back to the other side, and the state of =
the connection is left unchanged.

Certainly you aren't arguing that it is unacceptable to receive =
duplicate ACKs. :-)

			-David Borman

>=20
>>=20
>> I'll also take this opportunity to mention that if an out-of-order =
packet arrives, either due to a gap or duplicate data, the ACK field =
should be processed anyway.  On bi-directional connections, failure to =
process the ACK field in a packet that is purely duplicate data can lead =
to ACK wars on the connection, e.g. in crossing zero window probes.
>>=20
>> 			-David Borman
>>=20
>>>=20
>>> I cannot find a clarification in other RFCs.   Is an erratum =
appropriate?
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22222
> // email: zimmermann@cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>=20


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Cc: "tcpm@ietf.org WG" <tcpm@ietf.org>, David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] TCP window max size?
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   Alexander,

   Despite I'm not an expert, I commented the second question:

2010/10/6 Alexander Zimmermann <alexander.zimmermann@comsys.rwth-aachen.de>

> Hi David,
>
> we had recently also a discussion here in our group about
> max window size. What I understand form your example that
> a size of 2^31 is not possible since your example stated that it
> would fail. But in theory (regardless the scaling factor) what
> is the maximal possible window size so that everything would go ok?
> (2^31)-1?
>
> Another question that comes in my mind: TCP says sequence number
> x < y <=> 0 < y-x < 2^31. Why do we say here 2^31 and not for example
> 2^32-20?

 The TCP uses 2's complement arithmetic to verify if a number y is greater
than x. This way, y must be at most 2^31 away from x to be considered a new
data, otherwise it will be considered older (duplicated).


The 2's complement (Modular Arithmetic) is used to handle the wrap around
problem of a finite sequence number space (32 bits).


You can find a good explanation in the section 24.7 TCP Sequence Numbers
from Stevens, W., and Gary Wright. *TCP/IP Illustrated, Vol. 2: The
Implementation*. Addison-Wesley Pub. Co, 1994.



> Alex
>
> Am 25.08.2010 um 00:07 schrieb David Borman:
>
> >
> > On Aug 24, 2010, at 4:22 PM, David Borman wrote:
> >
> >> This reasoning is explained in RFC 1323, section 2.3, at the bottom of
> >> page 10 and the top of page 11:
> >>
> >>   TCP determines if a data segment is "old" or "new" by testing
> >>   whether its sequence number is within 2**31 bytes of the left edge
> >>   of the window, and if it is not, discarding the data as "old".  To
> >>   insure that new data is never mistakenly considered old and vice-
> >>   versa, the left edge of the sender's window has to be at most
> >>   2**31 away from the right edge of the receiver's window.
> >>   Similarly with the sender's right edge and receiver's left edge.
> >>   Since the right and left edges of either the sender's or
> >>   receiver's window differ by the window size, and since the sender
> >>   and receiver windows can be out of phase by at most the window
> >>   size, the above constraints imply that 2 * the max window size
> >>   must be less than 2**31, or
> >>
> >>        max window < 2**30
> >>
> >>   Since the max window is 2**S (where S is the scaling shift count)
> >>   times at most 2**16 - 1 (the maximum unscaled window), the maximum
> >>   window is guaranteed to be < 2*30 if S <= 14.  Thus, the shift
> >>   count must be limited to 14 (which allows windows of 2**30 = 1
> >>   Gbyte)...
> >
> > And what I forgot to ask was whether you or someone else disagreed with
> > this logic, but this is why it is 14.  However, even if the logic is
> > faulty, the max value will remain at 14, the only way to change it would
> > be via a new TCP option number.
> >
> > But mostly the place for trouble is at the boundary conditions, and if
> > you had a 2^31 window and the sender/receiver windows were fully skewed,
> > any outstanding packet from before left edge of the senders window could
> > show up at the receiver and be viewed as new data at the right edge of
> > the receivers window, which would be a bad thing.
> >
> > 1) Sender has a full 2^31 window of unacknowledged data,
> >   receiver is a full 2^31 ahead of the sender.
> > 2) Sender retransmits a packet at left edge of it's window.
> > 3) In-transit Ack from Receiver arrives, acks all data, and
> >   advances window by 2^31.
> > 4) Sender sends a new packet.
> > 5) New packet gets reordered in transit with the retransmitted packet.
> > 6) New packet arrives, and receiver advances window.
> > 7) Retransmitted packet arrives, and it now appears to be at the
> >   right edge of the receivers window as new data.
> >
> > A 2^14 max window keeps things well away from those boundary conditions.
> >
> >
>
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22222
> // email: zimmermann@cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>
>

--000e0cd3287a1be0c1049200003c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br>=A0=A0 Alexander,<br><br>=A0=A0 Despite I&#39;m not an expert, I commen=
ted the second question:<br><br><div class=3D"gmail_quote">2010/10/6 Alexan=
der Zimmermann <span dir=3D"ltr">&lt;<a href=3D"mailto:alexander.zimmermann=
@comsys.rwth-aachen.de">alexander.zimmermann@comsys.rwth-aachen.de</a>&gt;<=
/span><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi David,<br>
<br>
we had recently also a discussion here in our group about<br>
max window size. What I understand form your example that<br>
a size of 2^31 is not possible since your example stated that it<br>
would fail. But in theory (regardless the scaling factor) what<br>
is the maximal possible window size so that everything would go ok?<br>
(2^31)-1?<br>
<br>
Another question that comes in my mind: TCP says sequence number<br>
x &lt; y &lt;=3D&gt; 0 &lt; y-x &lt; 2^31. Why do we say here 2^31 and not =
for example<br>
2^32-20?=A0
=09
	</blockquote><div><style type=3D"text/css">
	<!--
		@page { margin: 0.79in }
		P { margin-bottom: 0.08in }
	-->
	</style>


<p class=3D"western" style=3D"margin-bottom: 0in;">
</p>


<p class=3D"western" style=3D"margin-bottom: 0in;">The TCP uses 2&#39;s
complement arithmetic to verify if a number y is greater than x. This
way, y must be at most 2^31 away from x to be considered a new data,
otherwise it will be considered older (duplicated).</p><p class=3D"western"=
 style=3D"margin-bottom: 0in;"><br></p><p class=3D"western" style=3D"margin=
-bottom: 0in;">The 2&#39;s complement
(Modular Arithmetic) is used to handle the wrap around problem of a
finite sequence number space (32 bits).</p><p class=3D"western" style=3D"ma=
rgin-bottom: 0in;"><br></p>
<p class=3D"western" style=3D"margin-bottom: 0in;">You can find a good
explanation in the section 24.7 TCP Sequence Numbers from Stevens, W., and =
Gary
Wright. <em><span style=3D"font-style: normal;">TCP/IP Illustrated, Vol.
2: The Implementation</span></em>. Addison-Wesley Pub. Co, 1994. <br></p><p=
 class=3D"western" style=3D"margin-bottom: 0in;"><br></p>
</div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb=
(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
Alex<br>
<br>
Am 25.08.2010 um 00:07 schrieb David Borman:<br>
<div><div></div><div class=3D"h5"><br>
&gt;<br>
&gt; On Aug 24, 2010, at 4:22 PM, David Borman wrote:<br>
&gt;<br>
&gt;&gt; This reasoning is explained in RFC 1323, section 2.3, at the botto=
m of<br>
&gt;&gt; page 10 and the top of page 11:<br>
&gt;&gt;<br>
&gt;&gt; =A0 TCP determines if a data segment is &quot;old&quot; or &quot;n=
ew&quot; by testing<br>
&gt;&gt; =A0 whether its sequence number is within 2**31 bytes of the left =
edge<br>
&gt;&gt; =A0 of the window, and if it is not, discarding the data as &quot;=
old&quot;. =A0To<br>
&gt;&gt; =A0 insure that new data is never mistakenly considered old and vi=
ce-<br>
&gt;&gt; =A0 versa, the left edge of the sender&#39;s window has to be at m=
ost<br>
&gt;&gt; =A0 2**31 away from the right edge of the receiver&#39;s window.<b=
r>
&gt;&gt; =A0 Similarly with the sender&#39;s right edge and receiver&#39;s =
left edge.<br>
&gt;&gt; =A0 Since the right and left edges of either the sender&#39;s or<b=
r>
&gt;&gt; =A0 receiver&#39;s window differ by the window size, and since the=
 sender<br>
&gt;&gt; =A0 and receiver windows can be out of phase by at most the window=
<br>
&gt;&gt; =A0 size, the above constraints imply that 2 * the max window size=
<br>
&gt;&gt; =A0 must be less than 2**31, or<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0max window &lt; 2**30<br>
&gt;&gt;<br>
&gt;&gt; =A0 Since the max window is 2**S (where S is the scaling shift cou=
nt)<br>
&gt;&gt; =A0 times at most 2**16 - 1 (the maximum unscaled window), the max=
imum<br>
&gt;&gt; =A0 window is guaranteed to be &lt; 2*30 if S &lt;=3D 14. =A0Thus,=
 the shift<br>
&gt;&gt; =A0 count must be limited to 14 (which allows windows of 2**30 =3D=
 1<br>
&gt;&gt; =A0 Gbyte)...<br>
&gt;<br>
&gt; And what I forgot to ask was whether you or someone else disagreed wit=
h<br>
&gt; this logic, but this is why it is 14. =A0However, even if the logic is=
<br>
&gt; faulty, the max value will remain at 14, the only way to change it wou=
ld<br>
&gt; be via a new TCP option number.<br>
&gt;<br>
&gt; But mostly the place for trouble is at the boundary conditions, and if=
<br>
&gt; you had a 2^31 window and the sender/receiver windows were fully skewe=
d,<br>
&gt; any outstanding packet from before left edge of the senders window cou=
ld<br>
&gt; show up at the receiver and be viewed as new data at the right edge of=
<br>
&gt; the receivers window, which would be a bad thing.<br>
&gt;<br>
&gt; 1) Sender has a full 2^31 window of unacknowledged data,<br>
&gt; =A0 receiver is a full 2^31 ahead of the sender.<br>
&gt; 2) Sender retransmits a packet at left edge of it&#39;s window.<br>
&gt; 3) In-transit Ack from Receiver arrives, acks all data, and<br>
&gt; =A0 advances window by 2^31.<br>
&gt; 4) Sender sends a new packet.<br>
&gt; 5) New packet gets reordered in transit with the retransmitted packet.=
<br>
&gt; 6) New packet arrives, and receiver advances window.<br>
&gt; 7) Retransmitted packet arrives, and it now appears to be at the<br>
&gt; =A0 right edge of the receivers window as new data.<br>
&gt;<br>
&gt; A 2^14 max window keeps things well away from those boundary condition=
s.<br>
&gt;<br>
&gt;<br>
<br>
</div></div>//<br>
// Dipl.-Inform. Alexander Zimmermann<br>
// Department of Computer Science, Informatik 4<br>
// RWTH Aachen University<br>
// Ahornstr. 55, 52056 Aachen, Germany<br>
// phone: (49-241) 80-21422, fax: (49-241) 80-22222<br>
// email: <a href=3D"mailto:zimmermann@cs.rwth-aachen.de">zimmermann@cs.rwt=
h-aachen.de</a><br>
// web: <a href=3D"http://www.umic-mesh.net" target=3D"_blank">http://www.u=
mic-mesh.net</a><br>
//<br>
<br>
</blockquote></div><br>

--000e0cd3287a1be0c1049200003c--

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Thread-Topic: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: <tcpm@ietf.org>, "Matthew Mathis" <mattmathis@google.com>, <mallman@icir.org>
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Subject: [tcpm]  end-of-stream loss recovery (RFC2018 SACK errata?)
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Hello members of the TCPM group, Matt, Mark,


I'm currently looking into the loss recovery behavior of TCP SACK -
especially at the end-of-stream.


End-of-stream (case 2) of the small congestion window, as per RFC5827)
is of particular interest, as there are synchronous IO loads over TCP,
that will only deliver additional data to the socket once an application
level acknowledgement has been received. To be precise, single threaded,
synchronous NFS read/write IOs. However that application layer Ack will
only be sent by the receiver, once the complete App IO was received,
thus any stalls can tremendeously delay the IO on the application.=20

The analogy would be TCP (with a large MTU) within TCP (with a small
MTU) - which is known to behave sub-optimal.



It appears that RFC2018 has codified behavior, which makes TCP SACK
perform less optimal compared with TCP NewReno, when loss occurs
directly at the last segments of the stream.

"
5. Interpreting the Sack Option and Retransmission Strategy: Data
   Sender Behavior
:
:
   After the SACKed bit is turned on (as the result of processing a
   received SACK option), the data sender will skip that segment during
   any later retransmission.  Any segment that has the SACKed bit turned
   off and is less than the highest SACKed segment is available for
   retransmission.
"

This definition leaves out the range [snd_fack .. snd_nxt (at loss
recovery initiation time)]. There is no explanation as to why that
omission was done (other than snd_fack not being defined at the time).=20


As RFC5827 is only concerned about detection and entering loss recovery,
it too gives no advice in this regard.

Consider a sequence of 4 Segments sent, S1 .. S4; S2 and S4 get dropped.

Early Retransmit (RFC5827) will allow entering loss recovery.=20

Depending on TCP SACK, or TCP NewReno, the behavior will be different:

NewReno will retransmit S2;
The ACK for S3 will trigger retransmission of S4 (and nobody got hurt)

SACK will retransmit S2;
The ACK for S3 will *not* trigger a retransmit of S4 (it falls outside
the boundary defined in RFC2018).
Only a RTO will recover S4, and the stream might continue (as
applications resume operation).

Of course, S4 could be any number of packets, just prior to the
end-of-stream. And the same sequence of events can happen without Early
Retransmit and large cwnds...

With infinite data to send, after SACK finished loss recovery - at the
highest SACKed (snd_fack) block - the sender would resume sending
starting from snd_nxt (and most likely in slow start). As soon as 3 new
segments would be received, this missing segment chain (snd_fack ..
Former snd_nxt) could be detected and recovered. Of course, counting
this as one additional congestion event (unlike what NewReno would do).


Or are SACK implementations which silently ignore the hole between
snd_fack and snd_nxt (when loss recovery is entered) broken?
=20



I would like to learn what you think would be the proper behavior. From
my point of view, this appears to have been a simple oversight in
RFC2018, and changing the statement to=20


"
5. Interpreting the Sack Option and Retransmission Strategy: Data
   Sender Behavior
:
:
   After the SACKed bit is turned on (as the result of processing a
   received SACK option), the data sender will skip that segment during
   any later retransmission.  Any segment that has the SACKed bit turned
   off and is less than the highest sent segment at the beginning of the

   SACK recovery episode is available for retransmission.
"

The fix would be simply add an implicit hole (or an implicit SACK block)
up to snd_nxt. If none of these tailing segments was lost, they will be
covered in an SACK eventually. That would happen before the sender could
retransmit those segments, as cwnd is also reduced thus limiting the
sending rate.=20




Richard Scheffenegger

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Hi,

On 2010-10-5, at 17:17, Ronald Bonica wrote:
> I cleared, assuming that Jari would hold my discuss.

thanks, Ron.

Editors, chairs, WG: are we clear on what needs to happen to the ID to =
address the discuss? When will we see a revision?

Thanks,
Lars=

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Hi, Lars,

Thanks so much for following this one closely. Comments inline....

On 11/10/2010 05:17 a.m., Lars Eggert wrote:
> Hi,
> 
> On 2010-10-5, at 17:17, Ronald Bonica wrote:
>> I cleared, assuming that Jari would hold my discuss.
> 
> thanks, Ron.
> 
> Editors, chairs, WG: are we clear on what needs to happen to the ID
> to address the discuss? When will we see a revision?

It's clear that there are two different positions on this DISCUSS:
David's, and Jari/Ron's (probably shared by Joe, too). But it's not
clear to me whether I should simply modify the document as indicated by
Jari/Ron, or whether there's a middle-ground that could possibly make
both parties happy (I'm assuming that sticking to David's position is
not an option).

(FWIW, me, I agree with David's position... but for obvious reasons I
might be biased... hence my request for advice on how to proceed).

Thoughts?

Thanks!

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





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Hi,

On 2010-10-11, at 17:27, Fernando Gont wrote:
> It's clear that there are two different positions on this DISCUSS:
> David's, and Jari/Ron's (probably shared by Joe, too). But it's not
> clear to me whether I should simply modify the document as indicated =
by
> Jari/Ron, or whether there's a middle-ground that could possibly make
> both parties happy (I'm assuming that sticking to David's position is
> not an option).

if you can find a middle ground that addresses the discusses, that would =
obviously also be a way forward. Please suggest one.

> (FWIW, me, I agree with David's position... but for obvious reasons I
> might be biased... hence my request for advice on how to proceed).

I'm mostly agreeing with David as well, but a significant fraction of =
the IESG does not.=20

Lars=

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From wesley.m.eddy@nasa.gov  Mon Oct 11 17:28:28 2010
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From: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
To: Lars Eggert <lars.eggert@nokia.com>, Fernando Gont <fernando@gont.com.ar>
Date: Mon, 11 Oct 2010 19:24:40 -0500
Thread-Topic: [tcpm] draft-ietf-tcpm-urgent-data-06
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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For what it's worth, I also agree with the position that David presented.  =
I don't think it's necessarily worth spending a lot of time and energy to d=
ebate, though.

________________________________________
From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] On Behalf Of Lars Egger=
t [lars.eggert@nokia.com]
Sent: Monday, October 11, 2010 11:51 AM
To: Fernando Gont
Cc: draft-ietf-tcpm-urgent-data.all@tools.ietf.org; tcpm@ietf.org Extension=
s
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06

...

> (FWIW, me, I agree with David's position... but for obvious reasons I
> might be biased... hence my request for advice on how to proceed).

I'm mostly agreeing with David as well, but a significant fraction of the I=
ESG does not.

Lars=

From touch@isi.edu  Mon Oct 11 19:48:53 2010
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Hi, all,

I can't speak for the IESG, but I can explain my reasoning:

The doc presents two pieces of info:

- update the spec
- change use to SHOULD NOT

Given we don't have apps breaking all over the place, the second one 
seems to be more dramatic in its impact to the community. The document 
presents the second one in an almost ancillary way.

IMO, if this doc is really about updating the spec, and we don't really 
have issue with new uses of the feature, then the SHOULD NOT should be 
dropped.

If the SHOULD NOT is kept, I can't see how it's not more prominent (and 
thus warrants doc restructuring as suggested).

Joe

On 10/11/2010 5:24 PM, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP] 
wrote:
> For what it's worth, I also agree with the position that David presented.  I don't think it's necessarily worth spending a lot of time and energy to debate, though.
>
> ________________________________________
> From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] On Behalf Of Lars Eggert [lars.eggert@nokia.com]
> Sent: Monday, October 11, 2010 11:51 AM
> To: Fernando Gont
> Cc: draft-ietf-tcpm-urgent-data.all@tools.ietf.org; tcpm@ietf.org Extensions
> Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
>
> ...
>
>> (FWIW, me, I agree with David's position... but for obvious reasons I
>> might be biased... hence my request for advice on how to proceed).
>
> I'm mostly agreeing with David as well, but a significant fraction of the IESG does not.
>
> Lars
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From bruno.mongazon-cazavet@alcatel-lucent.com  Tue Oct 12 07:40:47 2010
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From: Bruno Mongazon-Cazavet <bruno.mongazon-cazavet@alcatel-lucent.com>
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Subject: [tcpm] I-D Action:draft-mongazon-tcpm-tcp-rehash-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : TCP Rehash
	Author(s)       : B. Mongazon-Cazavet
	Filename        : draft-mongazon-tcpm-tcp-rehash-00.txt
	Pages           : 13
	Date            : 2010-10-11

The present specification describes a light extension of the TCP
protocol [RFC793] that allows a TCP transport connection to be
maintained operational whenever the underlying IP network address of
its end-points changes.  This includes situations where a single-
interface host changes its IP address or where a multiple-interface
host diverts a connection from one interface to another.  The
mechanism used to maintain a transport connection in such situations
is based on the capability of both end-points to "Rehash" a TCP
connection (rebuild its lookup key) with a new IP address in a fast
and reliable manner.

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

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

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.
ftp://ftp.ietf.org/internet-drafts/draft-mongazon-tcpm-tcp-rehash-00.txt


From david.borman@windriver.com  Tue Oct 12 07:57:47 2010
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To: Joe Touch <touch@isi.edu>, Fernando Gont <fernando@gont.com.ar>
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Cc: draft-ietf-tcpm-urgent-data.all@tools.ietf.org, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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On Oct 11, 2010, at 9:49 PM, Joe Touch wrote:

> Hi, all,
>=20
> I can't speak for the IESG, but I can explain my reasoning:
>=20
> The doc presents two pieces of info:
>=20
> - update the spec
> - change use to SHOULD NOT
>=20
> Given we don't have apps breaking all over the place, the second one =
seems to be more dramatic in its impact to the community. The document =
presents the second one in an almost ancillary way.
>=20
> IMO, if this doc is really about updating the spec, and we don't =
really have issue with new uses of the feature, then the SHOULD NOT =
should be dropped.

=46rom my viewpoint, the issue is to 1) make application developers =
aware of the problems cause by the potential of OS's having different =
definitions of the urgent pointer, and 2) tell them NOT to use the =
broken bsd socket() OOB mechanism, by setting SO_OOBINLINE.  The goal =
isn't to say that applications can't use the Urgent pointer, but to warn =
of the potential issues in a generic environment.

As you said, we don't have applications breaking all over the place.  =
Also, the goal of this document isn't to change all the current Urgent =
pointer implementations, but to match the spec to the majority of =
implementations.

The WG did not put forth this document to deprecate the Urgent pointer, =
but to clarify it and warn of the interoperability issues.

So perhaps the "SHOULD NOT" should be changed to a "MAY", meaning you =
can still use it, but be aware of the issues and make sure you design =
around them.  It is entirely possible to reliably make use of the urgent =
pointer in an application, even with different OS definitions.  The =
application can the test urgent pointer at the beginning of the =
connection by sending a known piece of data and observing how it gets =
delivered to the receiver, the application can then adjust accordingly =
and use it reliably.

Perhaps combine section 5 & 6 into a single section like:

5.  Advice on using and implementing the TCP urgent mechanism

  As a result of the issues discussed in Section 3.2 and Section 3.4,
  using the TCP urgent mechanism has the potential to be unreliable.
  New applications MAY use the TCP urgent mechanism, but should be
  robust against these issues.  In addition, applications MUST set
  the SO_OOBINLINE socket option, such that "urgent data" is delivered
  inline, as intended by the IETF specifications.

  TCP implementations MUST include support for the urgent mechanism
  such that existing applications can still use it.

			-David Borman

>=20
> If the SHOULD NOT is kept, I can't see how it's not more prominent =
(and thus warrants doc restructuring as suggested).
>=20
> Joe
>=20
> On 10/11/2010 5:24 PM, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP] =
wrote:
>> For what it's worth, I also agree with the position that David =
presented.  I don't think it's necessarily worth spending a lot of time =
and energy to debate, though.
>>=20
>> ________________________________________
>> From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] On Behalf Of Lars =
Eggert [lars.eggert@nokia.com]
>> Sent: Monday, October 11, 2010 11:51 AM
>> To: Fernando Gont
>> Cc: draft-ietf-tcpm-urgent-data.all@tools.ietf.org; tcpm@ietf.org =
Extensions
>> Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
>>=20
>> ...
>>=20
>>> (FWIW, me, I agree with David's position... but for obvious reasons =
I
>>> might be biased... hence my request for advice on how to proceed).
>>=20
>> I'm mostly agreeing with David as well, but a significant fraction of =
the IESG does not.
>>=20
>> Lars
>> _______________________________________________
>> 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 touch@isi.edu  Tue Oct 12 09:12:53 2010
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Hi, Dave,

On 10/12/2010 7:58 AM, David Borman wrote:

> As you said, we don't have applications breaking all over the place.
> Also, the goal of this document isn't to change all the current Urgent
> pointer implementations, but to match the spec to the majority of
> implementations.
>
> The WG did not put forth this document to deprecate the Urgent
> pointer, but to clarify it and warn of the interoperability issues.
>
> So perhaps the "SHOULD NOT" should be changed to a "MAY", meaning
> you can still use it, but be aware of the issues and make sure you
> design around them.

It's sufficient to delete Sec 5 then. It's already a "MAY" for users in 
RFC793 (though 2119 language isn't used, the intent is there). We 
definitely should not restate requirements already sufficiently stated.

> It is entirely possible to reliably make use of
> the urgent pointer in an application, even with different OS
> definitions. The application can the test urgent pointer at the
> beginning of the connection by sending a known piece of data and
> observing how it gets delivered to the receiver, the application can
> then adjust accordingly and use it reliably.

That's not in the doc anywhere. No current app I've ever heard of does 
this, and it doesn't seem practical for most apps (esp. since it's not 
backward compatible; the test data would appear in the data stream). I'm 
not sure this is a useful consideration for this discussion at all.

> Perhaps combine section 5& 6 into a single section like:
> 5.  Advice on using and implementing the TCP urgent mechanism
>
>    As a result of the issues discussed in Section 3.2 and Section 3.4,
>    using the TCP urgent mechanism has the potential to be unreliable.
>    New applications MAY use the TCP urgent mechanism, but should be
>    robust against these issues.

The above simply isn't required; it's already part of RFC793.

>    In addition, applications MUST set
>    the SO_OOBINLINE socket option, such that "urgent data" is delivered
>    inline, as intended by the IETF specifications.

SO_OOBINLINE and its behavior isn't defined in RFC793.

The fact that urgent data is inline needs to be defined and explained. 
Then the requirement stated that urgent pointer data is inline, as is 
all TCP data.

It may be useful to note that the way to do this in some implementations 
is to set this socket option, and that this particular option should 
default to ON (and note that it isn't, so apps should set it), but 2119 
language isn't appropriate. The discussion shouldn't use 2119 language 
in reference to an implementation of the API that isn't required.

>    TCP implementations MUST include support for the urgent mechanism
>    such that existing applications can still use it.

That's not necessary either; it's in 793, and it's already required 
therein. Again, I don't think this doc should restate existing requirements.

Joe


From touch@isi.edu  Tue Oct 12 09:32:45 2010
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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PS - to summarize my suggestions to wrap this up:

- delete section 5

- revise section 6 to explain what inline vs. out-of-band data is, and 
remind users that urgent is NOT an out-of-band signal -- that the data 
indicated by the urgent pointer MUST NOT be extricated from the data 
byte stream presented to the user via means indicated in RFC793 (e.g., 
read calls). This would also note that in some implementations this is 
requires overriding the default, and setting SO_OOBINLINE=1.

Joe

On 10/12/2010 9:12 AM, Joe Touch wrote:
> Hi, Dave,
>
> On 10/12/2010 7:58 AM, David Borman wrote:
>
>> As you said, we don't have applications breaking all over the place.
>> Also, the goal of this document isn't to change all the current Urgent
>> pointer implementations, but to match the spec to the majority of
>> implementations.
>>
>> The WG did not put forth this document to deprecate the Urgent
>> pointer, but to clarify it and warn of the interoperability issues.
>>
>> So perhaps the "SHOULD NOT" should be changed to a "MAY", meaning
>> you can still use it, but be aware of the issues and make sure you
>> design around them.
>
> It's sufficient to delete Sec 5 then. It's already a "MAY" for users in
> RFC793 (though 2119 language isn't used, the intent is there). We
> definitely should not restate requirements already sufficiently stated.
>
>> It is entirely possible to reliably make use of
>> the urgent pointer in an application, even with different OS
>> definitions. The application can the test urgent pointer at the
>> beginning of the connection by sending a known piece of data and
>> observing how it gets delivered to the receiver, the application can
>> then adjust accordingly and use it reliably.
>
> That's not in the doc anywhere. No current app I've ever heard of does
> this, and it doesn't seem practical for most apps (esp. since it's not
> backward compatible; the test data would appear in the data stream). I'm
> not sure this is a useful consideration for this discussion at all.
>
>> Perhaps combine section 5& 6 into a single section like:
>> 5. Advice on using and implementing the TCP urgent mechanism
>>
>> As a result of the issues discussed in Section 3.2 and Section 3.4,
>> using the TCP urgent mechanism has the potential to be unreliable.
>> New applications MAY use the TCP urgent mechanism, but should be
>> robust against these issues.
>
> The above simply isn't required; it's already part of RFC793.
>
>> In addition, applications MUST set
>> the SO_OOBINLINE socket option, such that "urgent data" is delivered
>> inline, as intended by the IETF specifications.
>
> SO_OOBINLINE and its behavior isn't defined in RFC793.
>
> The fact that urgent data is inline needs to be defined and explained.
> Then the requirement stated that urgent pointer data is inline, as is
> all TCP data.
>
> It may be useful to note that the way to do this in some implementations
> is to set this socket option, and that this particular option should
> default to ON (and note that it isn't, so apps should set it), but 2119
> language isn't appropriate. The discussion shouldn't use 2119 language
> in reference to an implementation of the API that isn't required.
>
>> TCP implementations MUST include support for the urgent mechanism
>> such that existing applications can still use it.
>
> That's not necessary either; it's in 793, and it's already required
> therein. Again, I don't think this doc should restate existing
> requirements.
>
> Joe

From jari.arkko@piuha.net  Fri Oct 15 14:02:57 2010
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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All,

We discussed this yesterday in the IESG conference call. I would like to 
summarize my thinking about it and suggest a way forward. (There are a 
few other ADs who have similar but perhaps not exactly the same 
thoughts, but I'm the one who representing these thoughts with the Discuss.)

The main concern is that at least in my view, urgent data seems pretty 
badly broken. It has potential interop problems if out of band mode is 
used, mismatch of recommendations wrt real world implementations that 
mostly use out of band mode as the default, etc. As far as I understand, 
when out of band mode is used the two end points have to agree about the 
semantics of the urgent pointer. It is reassuring that most 
implementations that you tested actually use the same semantics, but we 
an implementation that we do not know about or a middlebox could disrupt 
an application's understanding of the correct order and place of TCP 
data from the other end. With this background, I felt that the document 
stopped short of making clear enough recommendations about the use of 
this feature. In the IESG we talked about what the right level of 
"health warning" should be, and there were different opinions ranging 
from text at the beginning of the document to formal deprecation of the 
feature.

I have a proposal below that takes the warning text approach. Note that 
I'm not really changing anything substantive from what the document was 
already saying, but if we publish a new RFC on urgent indications, it 
seems reasonable to make it very clear that there are issues. If these 
changes were made to the document I would be happy to clear my Discuss, 
though I do know that Ron would probably would like to see an even more 
drastic action. What does the working group think?

Jari

Change Abstract:

OLD:
   This document analyzes how current TCP implementations process TCP
   urgent indications, and how the behavior of some widely-deployed
   middle-boxes affect how urgent indications are processed by end
   systems.  This document updates the relevant specifications such that
   they accommodate current practice in processing TCP urgent
   indications, provides advice to applications that make use of the
   urgent mechanism, and raises awareness about the reliability of TCP
   urgent indications in the current Internet.
NEW:
   This document analyzes how current TCP implementations process TCP
   urgent indications, and how the behavior of some widely-deployed
   middle-boxes affect how urgent indications are processed by end
   systems. This document updates the relevant specifications such that
   they accommodate current practice in processing TCP urgent
   indications, raises awareness about the reliability of TCP
   urgent indications in the Internet and recommends against the use of the
   urgent indications (but provides advice to applications in case that 
they do).

Change Introduction:

OLD:
   This document analyzes how some current TCP implementations process
   TCP urgent indications, and how the behavior of some widely-deployed
   middle-boxes affect the processing of urgent indications by hosts.
   This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
   1122 [RFC1122] such that they accommodate current practice in
   processing TCP urgent indications, provides advice to applications
   using the urgent mechanism, and raises awareness about the
   reliability of TCP urgent indications in the current Internet.
NEW:
   This document analyzes how some current TCP implementations process
   TCP urgent indications, and how the behavior of some widely-deployed
   middle-boxes affect the processing of urgent indications by hosts.
   This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
   1122 [RFC1122] such that they accommodate current practice in
   processing TCP urgent indications, provides advice to applications
   using the urgent mechanism, and raises awareness about the
   reliability of TCP urgent indications in the current Internet.

   Given the above issues and potential interoperability issues with 
respect to the
   currently common default mode operation, it is strongly recommended that
   applications do not employ urgent indications. Nevertheless, urgent 
indications
   are still retained as a mandatory part of the TCP protocol to support the
   few legacy applications that employ them. However, it is expected 
that even
   these applications will have difficulties in environments with 
middle-boxes.


From rbonica@juniper.net  Fri Oct 15 14:07:48 2010
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From: Ronald Bonica <rbonica@juniper.net>
To: Jari Arkko <jari.arkko@piuha.net>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Date: Fri, 15 Oct 2010 17:07:25 -0400
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Looks good to me.

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Jari Arkko
> Sent: Friday, October 15, 2010 5:04 PM
> To: tcpm@ietf.org Extensions
> Cc: David Borman; Fernando Gont; Joe Touch
> Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
>=20
> All,
>=20
> We discussed this yesterday in the IESG conference call. I would like
> to
> summarize my thinking about it and suggest a way forward. (There are a
> few other ADs who have similar but perhaps not exactly the same
> thoughts, but I'm the one who representing these thoughts with the
> Discuss.)
>=20
> The main concern is that at least in my view, urgent data seems pretty
> badly broken. It has potential interop problems if out of band mode is
> used, mismatch of recommendations wrt real world implementations that
> mostly use out of band mode as the default, etc. As far as I
> understand,
> when out of band mode is used the two end points have to agree about
> the
> semantics of the urgent pointer. It is reassuring that most
> implementations that you tested actually use the same semantics, but we
> an implementation that we do not know about or a middlebox could
> disrupt
> an application's understanding of the correct order and place of TCP
> data from the other end. With this background, I felt that the document
> stopped short of making clear enough recommendations about the use of
> this feature. In the IESG we talked about what the right level of
> "health warning" should be, and there were different opinions ranging
> from text at the beginning of the document to formal deprecation of the
> feature.
>=20
> I have a proposal below that takes the warning text approach. Note that
> I'm not really changing anything substantive from what the document was
> already saying, but if we publish a new RFC on urgent indications, it
> seems reasonable to make it very clear that there are issues. If these
> changes were made to the document I would be happy to clear my Discuss,
> though I do know that Ron would probably would like to see an even more
> drastic action. What does the working group think?
>=20
> Jari
>=20
> Change Abstract:
>=20
> OLD:
>    This document analyzes how current TCP implementations process TCP
>    urgent indications, and how the behavior of some widely-deployed
>    middle-boxes affect how urgent indications are processed by end
>    systems.  This document updates the relevant specifications such
> that
>    they accommodate current practice in processing TCP urgent
>    indications, provides advice to applications that make use of the
>    urgent mechanism, and raises awareness about the reliability of TCP
>    urgent indications in the current Internet.
> NEW:
>    This document analyzes how current TCP implementations process TCP
>    urgent indications, and how the behavior of some widely-deployed
>    middle-boxes affect how urgent indications are processed by end
>    systems. This document updates the relevant specifications such that
>    they accommodate current practice in processing TCP urgent
>    indications, raises awareness about the reliability of TCP
>    urgent indications in the Internet and recommends against the use of
> the
>    urgent indications (but provides advice to applications in case that
> they do).
>=20
> Change Introduction:
>=20
> OLD:
>    This document analyzes how some current TCP implementations process
>    TCP urgent indications, and how the behavior of some widely-deployed
>    middle-boxes affect the processing of urgent indications by hosts.
>    This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>    1122 [RFC1122] such that they accommodate current practice in
>    processing TCP urgent indications, provides advice to applications
>    using the urgent mechanism, and raises awareness about the
>    reliability of TCP urgent indications in the current Internet.
> NEW:
>    This document analyzes how some current TCP implementations process
>    TCP urgent indications, and how the behavior of some widely-deployed
>    middle-boxes affect the processing of urgent indications by hosts.
>    This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>    1122 [RFC1122] such that they accommodate current practice in
>    processing TCP urgent indications, provides advice to applications
>    using the urgent mechanism, and raises awareness about the
>    reliability of TCP urgent indications in the current Internet.
>=20
>    Given the above issues and potential interoperability issues with
> respect to the
>    currently common default mode operation, it is strongly recommended
> that
>    applications do not employ urgent indications. Nevertheless, urgent
> indications
>    are still retained as a mandatory part of the TCP protocol to
> support the
>    few legacy applications that employ them. However, it is expected
> that even
>    these applications will have difficulties in environments with
> middle-boxes.
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From stpeter@stpeter.im  Fri Oct 15 14:11:42 2010
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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+1 -- that seems to be an accurate summary of what folks think this
document is saying or should be saying.

On 10/15/10 3:04 PM, Jari Arkko wrote:
> All,
>=20
> We discussed this yesterday in the IESG conference call. I would like t=
o
> summarize my thinking about it and suggest a way forward. (There are a
> few other ADs who have similar but perhaps not exactly the same
> thoughts, but I'm the one who representing these thoughts with the
> Discuss.)
>=20
> The main concern is that at least in my view, urgent data seems pretty
> badly broken. It has potential interop problems if out of band mode is
> used, mismatch of recommendations wrt real world implementations that
> mostly use out of band mode as the default, etc. As far as I understand=
,
> when out of band mode is used the two end points have to agree about th=
e
> semantics of the urgent pointer. It is reassuring that most
> implementations that you tested actually use the same semantics, but we=

> an implementation that we do not know about or a middlebox could disrup=
t
> an application's understanding of the correct order and place of TCP
> data from the other end. With this background, I felt that the document=

> stopped short of making clear enough recommendations about the use of
> this feature. In the IESG we talked about what the right level of
> "health warning" should be, and there were different opinions ranging
> from text at the beginning of the document to formal deprecation of the=

> feature.
>=20
> I have a proposal below that takes the warning text approach. Note that=

> I'm not really changing anything substantive from what the document was=

> already saying, but if we publish a new RFC on urgent indications, it
> seems reasonable to make it very clear that there are issues. If these
> changes were made to the document I would be happy to clear my Discuss,=

> though I do know that Ron would probably would like to see an even more=

> drastic action. What does the working group think?
>=20
> Jari
>=20
> Change Abstract:
>=20
> OLD:
>   This document analyzes how current TCP implementations process TCP
>   urgent indications, and how the behavior of some widely-deployed
>   middle-boxes affect how urgent indications are processed by end
>   systems.  This document updates the relevant specifications such that=

>   they accommodate current practice in processing TCP urgent
>   indications, provides advice to applications that make use of the
>   urgent mechanism, and raises awareness about the reliability of TCP
>   urgent indications in the current Internet.
> NEW:
>   This document analyzes how current TCP implementations process TCP
>   urgent indications, and how the behavior of some widely-deployed
>   middle-boxes affect how urgent indications are processed by end
>   systems. This document updates the relevant specifications such that
>   they accommodate current practice in processing TCP urgent
>   indications, raises awareness about the reliability of TCP
>   urgent indications in the Internet and recommends against the use of =
the
>   urgent indications (but provides advice to applications in case that
> they do).
>=20
> Change Introduction:
>=20
> OLD:
>   This document analyzes how some current TCP implementations process
>   TCP urgent indications, and how the behavior of some widely-deployed
>   middle-boxes affect the processing of urgent indications by hosts.
>   This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>   1122 [RFC1122] such that they accommodate current practice in
>   processing TCP urgent indications, provides advice to applications
>   using the urgent mechanism, and raises awareness about the
>   reliability of TCP urgent indications in the current Internet.
> NEW:
>   This document analyzes how some current TCP implementations process
>   TCP urgent indications, and how the behavior of some widely-deployed
>   middle-boxes affect the processing of urgent indications by hosts.
>   This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>   1122 [RFC1122] such that they accommodate current practice in
>   processing TCP urgent indications, provides advice to applications
>   using the urgent mechanism, and raises awareness about the
>   reliability of TCP urgent indications in the current Internet.
>=20
>   Given the above issues and potential interoperability issues with
> respect to the
>   currently common default mode operation, it is strongly recommended t=
hat
>   applications do not employ urgent indications. Nevertheless, urgent
> indications
>   are still retained as a mandatory part of the TCP protocol to support=
 the
>   few legacy applications that employ them. However, it is expected tha=
t
> even
>   these applications will have difficulties in environments with
> middle-boxes.


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From david.borman@windriver.com  Fri Oct 15 14:40:34 2010
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To: Jari Arkko <jari.arkko@piuha.net>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Jari,
This seems to be a reasonable compromise for the differing points of =
view, and a way to get past this issue.  Thanks.
			-David

On Oct 15, 2010, at 4:04 PM, Jari Arkko wrote:

> All,
>=20
> We discussed this yesterday in the IESG conference call. I would like =
to summarize my thinking about it and suggest a way forward. (There are =
a few other ADs who have similar but perhaps not exactly the same =
thoughts, but I'm the one who representing these thoughts with the =
Discuss.)
>=20
> The main concern is that at least in my view, urgent data seems pretty =
badly broken. It has potential interop problems if out of band mode is =
used, mismatch of recommendations wrt real world implementations that =
mostly use out of band mode as the default, etc. As far as I understand, =
when out of band mode is used the two end points have to agree about the =
semantics of the urgent pointer. It is reassuring that most =
implementations that you tested actually use the same semantics, but we =
an implementation that we do not know about or a middlebox could disrupt =
an application's understanding of the correct order and place of TCP =
data from the other end. With this background, I felt that the document =
stopped short of making clear enough recommendations about the use of =
this feature. In the IESG we talked about what the right level of =
"health warning" should be, and there were different opinions ranging =
from text at the beginning of the document to formal deprecation of the =
feature.
>=20
> I have a proposal below that takes the warning text approach. Note =
that I'm not really changing anything substantive from what the document =
was already saying, but if we publish a new RFC on urgent indications, =
it seems reasonable to make it very clear that there are issues. If =
these changes were made to the document I would be happy to clear my =
Discuss, though I do know that Ron would probably would like to see an =
even more drastic action. What does the working group think?
>=20
> Jari
>=20
> Change Abstract:
>=20
> OLD:
>  This document analyzes how current TCP implementations process TCP
>  urgent indications, and how the behavior of some widely-deployed
>  middle-boxes affect how urgent indications are processed by end
>  systems.  This document updates the relevant specifications such that
>  they accommodate current practice in processing TCP urgent
>  indications, provides advice to applications that make use of the
>  urgent mechanism, and raises awareness about the reliability of TCP
>  urgent indications in the current Internet.
> NEW:
>  This document analyzes how current TCP implementations process TCP
>  urgent indications, and how the behavior of some widely-deployed
>  middle-boxes affect how urgent indications are processed by end
>  systems. This document updates the relevant specifications such that
>  they accommodate current practice in processing TCP urgent
>  indications, raises awareness about the reliability of TCP
>  urgent indications in the Internet and recommends against the use of =
the
>  urgent indications (but provides advice to applications in case that =
they do).
>=20
> Change Introduction:
>=20
> OLD:
>  This document analyzes how some current TCP implementations process
>  TCP urgent indications, and how the behavior of some widely-deployed
>  middle-boxes affect the processing of urgent indications by hosts.
>  This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>  1122 [RFC1122] such that they accommodate current practice in
>  processing TCP urgent indications, provides advice to applications
>  using the urgent mechanism, and raises awareness about the
>  reliability of TCP urgent indications in the current Internet.
> NEW:
>  This document analyzes how some current TCP implementations process
>  TCP urgent indications, and how the behavior of some widely-deployed
>  middle-boxes affect the processing of urgent indications by hosts.
>  This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>  1122 [RFC1122] such that they accommodate current practice in
>  processing TCP urgent indications, provides advice to applications
>  using the urgent mechanism, and raises awareness about the
>  reliability of TCP urgent indications in the current Internet.
>=20
>  Given the above issues and potential interoperability issues with =
respect to the
>  currently common default mode operation, it is strongly recommended =
that
>  applications do not employ urgent indications. Nevertheless, urgent =
indications
>  are still retained as a mandatory part of the TCP protocol to support =
the
>  few legacy applications that employ them. However, it is expected =
that even
>  these applications will have difficulties in environments with =
middle-boxes.
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From touch@isi.edu  Fri Oct 15 14:48:07 2010
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Hi, all,

FWIW, this skirts the key questions:

1) should this doc include any particular 2119 language on the urgent 
option?

	- IMO, it should not repeat or restate existing requirements
	- based on the note from Jari, I can't tell whether the
	doc is recommending SHOULD NOT for new apps or not

	if it is, then the doc should definitely be revised to
	lead with that issue; if not, then such revision isn't needed

2) how should this doc address the in/out of band issue?
Right now, it explains it solely by reference to the implementation of a 
Unix API. That doesn't seem sufficient, as per the note I already posted.

It'd be useful to understand what the note below says to these two points.

Joe

On 10/15/2010 2:41 PM, David Borman wrote:
> Jari,
> This seems to be a reasonable compromise for the differing points of view, and a way to get past this issue.  Thanks.
> 			-David
>
> On Oct 15, 2010, at 4:04 PM, Jari Arkko wrote:
>
>> All,
>>
>> We discussed this yesterday in the IESG conference call. I would like to summarize my thinking about it and suggest a way forward. (There are a few other ADs who have similar but perhaps not exactly the same thoughts, but I'm the one who representing these thoughts with the Discuss.)
>>
>> The main concern is that at least in my view, urgent data seems pretty badly broken. It has potential interop problems if out of band mode is used, mismatch of recommendations wrt real world implementations that mostly use out of band mode as the default, etc. As far as I understand, when out of band mode is used the two end points have to agree about the semantics of the urgent pointer. It is reassuring that most implementations that you tested actually use the same semantics, but we an implementation that we do not know about or a middlebox could disrupt an application's understanding of the correct order and place of TCP data from the other end. With this background, I felt that the document stopped short of making clear enough recommendations about the use of this feature. In the IESG we talked about what the right level of "health warning" should be, and there were different opinions ranging from text at the beginning of the document to formal deprecation of the feat
ure.
>>
>> I have a proposal below that takes the warning text approach. Note that I'm not really changing anything substantive from what the document was already saying, but if we publish a new RFC on urgent indications, it seems reasonable to make it very clear that there are issues. If these changes were made to the document I would be happy to clear my Discuss, though I do know that Ron would probably would like to see an even more drastic action. What does the working group think?
>>
>> Jari
>>
>> Change Abstract:
>>
>> OLD:
>>   This document analyzes how current TCP implementations process TCP
>>   urgent indications, and how the behavior of some widely-deployed
>>   middle-boxes affect how urgent indications are processed by end
>>   systems.  This document updates the relevant specifications such that
>>   they accommodate current practice in processing TCP urgent
>>   indications, provides advice to applications that make use of the
>>   urgent mechanism, and raises awareness about the reliability of TCP
>>   urgent indications in the current Internet.
>> NEW:
>>   This document analyzes how current TCP implementations process TCP
>>   urgent indications, and how the behavior of some widely-deployed
>>   middle-boxes affect how urgent indications are processed by end
>>   systems. This document updates the relevant specifications such that
>>   they accommodate current practice in processing TCP urgent
>>   indications, raises awareness about the reliability of TCP
>>   urgent indications in the Internet and recommends against the use of the
>>   urgent indications (but provides advice to applications in case that they do).
>>
>> Change Introduction:
>>
>> OLD:
>>   This document analyzes how some current TCP implementations process
>>   TCP urgent indications, and how the behavior of some widely-deployed
>>   middle-boxes affect the processing of urgent indications by hosts.
>>   This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>>   1122 [RFC1122] such that they accommodate current practice in
>>   processing TCP urgent indications, provides advice to applications
>>   using the urgent mechanism, and raises awareness about the
>>   reliability of TCP urgent indications in the current Internet.
>> NEW:
>>   This document analyzes how some current TCP implementations process
>>   TCP urgent indications, and how the behavior of some widely-deployed
>>   middle-boxes affect the processing of urgent indications by hosts.
>>   This document updates RFC 793 [RFC0793], RFC 1011 [RFC1011], and RFC
>>   1122 [RFC1122] such that they accommodate current practice in
>>   processing TCP urgent indications, provides advice to applications
>>   using the urgent mechanism, and raises awareness about the
>>   reliability of TCP urgent indications in the current Internet.
>>
>>   Given the above issues and potential interoperability issues with respect to the
>>   currently common default mode operation, it is strongly recommended that
>>   applications do not employ urgent indications. Nevertheless, urgent indications
>>   are still retained as a mandatory part of the TCP protocol to support the
>>   few legacy applications that employ them. However, it is expected that even
>>   these applications will have difficulties in environments with middle-boxes.
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>

From jari.arkko@piuha.net  Fri Oct 15 14:58:45 2010
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, David Borman <david.borman@windriver.com>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Joe,

Just a quick response on one point:

>     - based on the note from Jari, I can't tell whether the
>     doc is recommending SHOULD NOT for new apps or not

My text did not change that aspect in any way. The new text is in the 
introduction and has no keywords. The formal requirement is later in the 
document. IIRC the current text there says SHOULD NOT for new apps.

Jari


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.


	Title           : On the implementation of the TCP urgent mechanism
	Author(s)       : F. Gont, A. Yourtchenko
	Filename        : draft-ietf-tcpm-urgent-data-07.txt
	Pages           : 14
	Date            : 2010-10-15

This document analyzes how current TCP implementations process TCP
urgent indications, and how the behavior of some widely-deployed
middle-boxes affect how urgent indications are processed by end
systems.  This document updates the relevant specifications such that
they accommodate current practice in processing TCP urgent
indications, raises awareness about the reliability of TCP urgent
indications in the Internet and recommends against the use of the
urgent indications (but provides advice to applications in case that
they do).

A URL for this Internet-Draft is:
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As Markku mentioned on the mailing list some time ago, and discussed in Ana=
heim, the authors of draft-ietf-tcpm-sack-recovery and RFC 3517 have discus=
sed merging the changes that the sack-recovery draft introduces into an upd=
ate to RFC 3517.  This makes a lot of sense, and simplifies the document tr=
ee we create a bit.

At this point, they're almost ready to submit the merged document, and want=
 to confirm this path with the working group so they can figure out what to=
 name it (draft-ietf-tcpm, or draft-lastname-tcpm).  We know that in Anahei=
m, Lars was concerned with the timeframe this merging might take, but there=
 are no errata on 3517, and the sack-recovery work is felt to be complete b=
y the authors, so the merging has good odds of completing quickly.

If there are any objections to the merged document in preparation replacing=
 the sack-recovery working group draft (as a working group item), please br=
ing them up in the next couple weeks.=

From rs@netapp.com  Mon Oct 18 09:54:05 2010
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>, <tcpm@ietf.org>
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Subject: Re: [tcpm] 3517 update w/ SACK-entry draft
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The only comment I would like to make is about my recent find, where TCP
SACK loss recovery implementations currently don't recover from loss of
last segement in the stream.=20

It might be a good idea to add some paragraph talking about the expected
behavior so that SACK will never exhibit worst performance than NewReno?


Regards,=20

Richard Scheffenegger

=20

> -----Original Message-----
> From: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]=20
> [mailto:wesley.m.eddy@nasa.gov]=20
> Sent: Sonntag, 17. Oktober 2010 22:06
> To: tcpm@ietf.org
> Subject: [tcpm] 3517 update w/ SACK-entry draft
>=20
> As Markku mentioned on the mailing list some time ago, and=20
> discussed in Anaheim, the authors of=20
> draft-ietf-tcpm-sack-recovery and RFC 3517 have discussed=20
> merging the changes that the sack-recovery draft introduces=20
> into an update to RFC 3517.  This makes a lot of sense, and=20
> simplifies the document tree we create a bit.
>=20
> At this point, they're almost ready to submit the merged=20
> document, and want to confirm this path with the working=20
> group so they can figure out what to name it=20
> (draft-ietf-tcpm, or draft-lastname-tcpm).  We know that in=20
> Anaheim, Lars was concerned with the timeframe this merging=20
> might take, but there are no errata on 3517, and the=20
> sack-recovery work is felt to be complete by the authors, so=20
> the merging has good odds of completing quickly.
>=20
> If there are any objections to the merged document in=20
> preparation replacing the sack-recovery working group draft=20
> (as a working group item), please bring them up in the next=20
> couple weeks.
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>=20

From eblanton@cs.ohiou.edu  Mon Oct 18 10:22:39 2010
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To: "Scheffenegger, Richard" <rs@netapp.com>
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Scheffenegger, Richard spake unto us the following wisdom:
> The only comment I would like to make is about my recent find, where TCP
> SACK loss recovery implementations currently don't recover from loss of
> last segement in the stream.=20

RFC3517 SACK certainly handles this recovery, it simply requires an
RTO. This is a well-known limitation, and stems from the strict
observation of conservation of packets which is made possible by the
pipe algorithm.

Note that this behavior is described *both* in RFC 2018 (where it is
not prescriptive) and in RFC 3517 (where it is prescriptive for any
TCP implementating purporting to implement 3517), and that our present
draft of 3517bis does not change this behavior.

This situation is a corner case, and while it may have real-world
implications, I am fairly sure that we do not want to tackle it in
what is otherwise a *very* minor update.  Triggering this corner case
requires that *two* segments in the same window of data are lost, and
that they are separated by at least three segments (required to
trigger Fast Retransmit, in the absence of ER).  As an author of
3517bis, I strongly prefer to address this issue in a different
document, if at all.

I also don't think this bears on whether or not SACK-based entry to
loss recovery should be accepted as a WG document or not.  The
SACK-based entry document forwarded by Markku and Ilpo is already an
accepted WG document, and the only question is whether its new status
as 3517bis is also accepted.  Our sense (as authors) is that it was
generally accepted by the WG that 3517bis is a good path forward on
this front based on previous discussions.

Ethan

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From rs@netapp.com  Mon Oct 18 10:32:44 2010
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Ethan Blanton" <eblanton@cs.ohiou.edu>
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Ethan,

Thank you for your feedback!

I agree, that particular case is a corner case; from a conservation of
packets point of view, a full ACK (no SACK entries) below SND.MAX would
be a valid indication, IMHO - just like in NewReno. At least to address
this corner case without any undue superflous retransmits (if additional
segments are sent out, the 2nd loss in the window will eventually
recovered by the information gathered from SACK entries. This case SACK
can already cover.).

I'm already trying to prepare an I-D with this (and the lost
retransmission detection) as a "less conservative" SACK recovery method.
So I'm fine with not merging this with the pending changes of 3517bis,
and split it off into an (expirimental?) draft.

Richard Scheffenegger
=20

> -----Original Message-----
> From: Ethan Blanton [mailto:eblanton@cs.ohiou.edu]=20
> Sent: Montag, 18. Oktober 2010 13:24
> To: Scheffenegger, Richard
> Cc: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]; tcpm@ietf.org
> Subject: Re: [tcpm] 3517 update w/ SACK-entry draft
>=20
> Scheffenegger, Richard spake unto us the following wisdom:
> > The only comment I would like to make is about my recent=20
> find, where=20
> > TCP SACK loss recovery implementations currently don't recover from=20
> > loss of last segement in the stream.
>=20
> RFC3517 SACK certainly handles this recovery, it simply=20
> requires an RTO. This is a well-known limitation, and stems=20
> from the strict observation of conservation of packets which=20
> is made possible by the pipe algorithm.
>=20
> Note that this behavior is described *both* in RFC 2018=20
> (where it is not prescriptive) and in RFC 3517 (where it is=20
> prescriptive for any TCP implementating purporting to=20
> implement 3517), and that our present draft of 3517bis does=20
> not change this behavior.
>=20
> This situation is a corner case, and while it may have=20
> real-world implications, I am fairly sure that we do not want=20
> to tackle it in what is otherwise a *very* minor update. =20
> Triggering this corner case requires that *two* segments in=20
> the same window of data are lost, and that they are separated=20
> by at least three segments (required to trigger Fast=20
> Retransmit, in the absence of ER).  As an author of 3517bis,=20
> I strongly prefer to address this issue in a different=20
> document, if at all.
>=20
> I also don't think this bears on whether or not SACK-based=20
> entry to loss recovery should be accepted as a WG document or=20
> not.  The SACK-based entry document forwarded by Markku and=20
> Ilpo is already an accepted WG document, and the only=20
> question is whether its new status as 3517bis is also=20
> accepted.  Our sense (as authors) is that it was generally=20
> accepted by the WG that 3517bis is a good path forward on=20
> this front based on previous discussions.
>=20
> Ethan
>=20

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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 18 Oct 2010 18:14:18 -0700
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To: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
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Subject: Re: [tcpm] 3517 update w/ SACK-entry draft
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The updates look good to me.

It would be better if the bis explicitly defines FlightSize or
explicitly cites the RFC5681's definition:
FLIGHT SIZE: The amount of data that has been sent but not yet
_cumulatively_ acknowledged.

Some readers (at least myself) mistakenly think that the FlightSize is
the estimated amount of
outstanding packets/bytes in the network (=3D=3D the pipe var. in the RFC).

Cheers,

Yuchung

On Sun, Oct 17, 2010 at 7:06 PM, Eddy, Wesley M. (GRC-MS00)[ASRC
AEROSPACE CORP] <wesley.m.eddy@nasa.gov> wrote:
>
> As Markku mentioned on the mailing list some time ago, and discussed in A=
naheim, the authors of draft-ietf-tcpm-sack-recovery and RFC 3517 have disc=
ussed merging the changes that the sack-recovery draft introduces into an u=
pdate to RFC 3517. =A0This makes a lot of sense, and simplifies the documen=
t tree we create a bit.
>
> At this point, they're almost ready to submit the merged document, and wa=
nt to confirm this path with the working group so they can figure out what =
to name it (draft-ietf-tcpm, or draft-lastname-tcpm). =A0We know that in An=
aheim, Lars was concerned with the timeframe this merging might take, but t=
here are no errata on 3517, and the sack-recovery work is felt to be comple=
te by the authors, so the merging has good odds of completing quickly.
>
> If there are any objections to the merged document in preparation replaci=
ng the sack-recovery working group draft (as a working group item), please =
bring them up in the next couple weeks.
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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On 10/15/2010 3:00 PM, Jari Arkko wrote:
> Joe,
>
> Just a quick response on one point:
>
>> - based on the note from Jari, I can't tell whether the
>> doc is recommending SHOULD NOT for new apps or not
>
> My text did not change that aspect in any way. The new text is in the
> introduction and has no keywords. The formal requirement is later in the
> document. IIRC the current text there says SHOULD NOT for new apps.

Correct. In that case, the note was not responsive to the thread I 
raised discussing the point that this is a new and substantial change, 
one that I feel suggests that the title, abstract, and overall structure 
should be inverted to focus on SHOULD NOT as the key point.

It's hard to understand a doc that basically focuses on a change to the 
spec, and includes the SHOULD NOT almost parenthetically.

Joe

From jari.arkko@piuha.net  Tue Oct 19 15:13:16 2010
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Joe, I think I agree with you. I was not planning on holding on to a 
Discuss for that part, however.

Jari

Joe Touch kirjoitti:
>
>
> On 10/15/2010 3:00 PM, Jari Arkko wrote:
>> Joe,
>>
>> Just a quick response on one point:
>>
>>> - based on the note from Jari, I can't tell whether the
>>> doc is recommending SHOULD NOT for new apps or not
>>
>> My text did not change that aspect in any way. The new text is in the
>> introduction and has no keywords. The formal requirement is later in the
>> document. IIRC the current text there says SHOULD NOT for new apps.
>
> Correct. In that case, the note was not responsive to the thread I 
> raised discussing the point that this is a new and substantial change, 
> one that I feel suggests that the title, abstract, and overall 
> structure should be inverted to focus on SHOULD NOT as the key point.
>
> It's hard to understand a doc that basically focuses on a change to 
> the spec, and includes the SHOULD NOT almost parenthetically.
>
> Joe
>


From touch@isi.edu  Tue Oct 19 15:21:03 2010
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Hmm. That was the issue that the discuss I thought that Ron was holding 
was focused on (see attached)

Ron?

Joe

On 10/19/2010 3:14 PM, Jari Arkko wrote:
> Joe, I think I agree with you. I was not planning on holding on to a
> Discuss for that part, however.
>
> Jari
>
> Joe Touch kirjoitti:
>>
>>
>> On 10/15/2010 3:00 PM, Jari Arkko wrote:
>>> Joe,
>>>
>>> Just a quick response on one point:
>>>
>>>> - based on the note from Jari, I can't tell whether the
>>>> doc is recommending SHOULD NOT for new apps or not
>>>
>>> My text did not change that aspect in any way. The new text is in the
>>> introduction and has no keywords. The formal requirement is later in the
>>> document. IIRC the current text there says SHOULD NOT for new apps.
>>
>> Correct. In that case, the note was not responsive to the thread I
>> raised discussing the point that this is a new and substantial change,
>> one that I feel suggests that the title, abstract, and overall
>> structure should be inverted to focus on SHOULD NOT as the key point.
>>
>> It's hard to understand a doc that basically focuses on a change to
>> the spec, and includes the SHOULD NOT almost parenthetically.
>>
>> Joe
>>

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From: Ronald Bonica <rbonica@juniper.net>
To: Joe Touch <touch@isi.edu>, John Heffner <johnwheffner@gmail.com>
CC: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Mon, 27 Sep 2010 15:49:05 -0400
Subject: RE: [tcpm] draft-ietf-tcpm-urgent-data-06
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Joe,

I think that you have hit the nail on the head. Reading through Sections 1-5, I thought that we were trying to fix the feature. Only in Section 6 did I realize that we really wanted to drive a stake through its heart.

Maybe if we changed the title and added some test to the abstract and introduction, this would be better. Otherwise, people may spend time and money fixing the TCP Urgent option instead of just avoiding it.

                                                         

> -----Original Message-----
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Thursday, September 23, 2010 10:18 PM
> To: John Heffner
> Cc: Ronald Bonica; tcpm@ietf.org
> Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
> 
> 
> 
> On 9/23/2010 6:00 PM, John Heffner wrote:
> > My understanding is that deprecating urgent data is exactly what this
> > document is doing.  In section 5: "... new applications SHOULD NOT
> > employ the TCP urgent mechanism."  The rest of the document is
> largely
> > a justification for deprecating the mechanism.
> 
> And a clarification as to what byte the urgent data actually starts at.
> The spec doesn't match any of the implementations, and since there's no
> benefit to either variant, the WG decided to change the spec to match
> the extant implementations. See section 4.
> 
> This is NOT an enhancement, and is NOT intended to make future apps
> more
> likely to use the urgent mechanism. See section 6, which deprecates the
> feature.
> 
> Is there a question that, e.g., the title be updated to reflect the
> primary intent, i.e.,"Deprecating TCP's Urgent Mechanism", with a few
> other related changes (e.g., update the abstract/intro, move section 6
> forward, and then explain that the remainder of the doc is to document
> how existing apps work?)?
> 
> I could see how the current doc reads more like "we're updating the
> spec
> first. oh yeah, and don't use this", which seems inverted; the WG
> conclusion was more like "deprecate this. and oh yeah, here's how it
> really works in the few apps that we know of that actually use it"
> 
> Joe
> 
> >
> >    -John
> >
> >
> > On Thu, Sep 23, 2010 at 5:10 PM, Ronald Bonica<rbonica@juniper.net>
> wrote:
> >> Folks,
> >>
> >> The purpose of this email is to encourage the TCPM working group to
> deprecate the feature described in draft-ietf-tcpm-urgent-data-06.
> >>
> >> I understand the following statements to be true:
> >>
> >> - The WG discourages new applications from using the feature.
> >> - Many middle-boxes defeat the feature by removing the urgent data
> indicator. Therefore, when an application indicates that data is
> urgent, it is never sure that that indication will reach the remote
> TCP.
> >> - Only three applications have been identified as using the feature.
> These are a proprietary application, telnet and ftp. More elegant
> mechanisms are available to Telnet and FTP.
> >> - The feature is rarely implemented as specified.
> >>
> >> Finally, there are more elegant ways to bring urgent data to
> applications.
> >>
> >> IMHO, this feature should be deprecated, not enhanced. Let's not
> pass up this opportunity to simplify the TCP stack.
> >>
> >>                                                            Ron
> >>
> >> _______________________________________________
> >> 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

--------------040206020807040400090603--

From rbonica@juniper.net  Tue Oct 19 17:47:26 2010
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From: Ronald Bonica <rbonica@juniper.net>
To: Joe Touch <touch@isi.edu>, Jari Arkko <jari.arkko@piuha.net>
Date: Tue, 19 Oct 2010 20:45:58 -0400
Thread-Topic: [tcpm] draft-ietf-tcpm-urgent-data-06
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I cleared, letting Jari hold the DISCUSS.

             Ron


> -----Original Message-----
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Tuesday, October 19, 2010 6:22 PM
> To: Jari Arkko
> Cc: David Borman; tcpm@ietf.org Extensions; Fernando Gont
> Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
>=20
> Hmm. That was the issue that the discuss I thought that Ron was holding
> was focused on (see attached)
>=20
> Ron?
>=20
> Joe
>=20
> On 10/19/2010 3:14 PM, Jari Arkko wrote:
> > Joe, I think I agree with you. I was not planning on holding on to a
> > Discuss for that part, however.
> >
> > Jari
> >
> > Joe Touch kirjoitti:
> >>
> >>
> >> On 10/15/2010 3:00 PM, Jari Arkko wrote:
> >>> Joe,
> >>>
> >>> Just a quick response on one point:
> >>>
> >>>> - based on the note from Jari, I can't tell whether the doc is
> >>>> recommending SHOULD NOT for new apps or not
> >>>
> >>> My text did not change that aspect in any way. The new text is in
> >>> the introduction and has no keywords. The formal requirement is
> >>> later in the document. IIRC the current text there says SHOULD NOT
> for new apps.
> >>
> >> Correct. In that case, the note was not responsive to the thread I
> >> raised discussing the point that this is a new and substantial
> >> change, one that I feel suggests that the title, abstract, and
> >> overall structure should be inverted to focus on SHOULD NOT as the
> key point.
> >>
> >> It's hard to understand a doc that basically focuses on a change to
> >> the spec, and includes the SHOULD NOT almost parenthetically.
> >>
> >> Joe
> >>

From david.borman@windriver.com  Tue Oct 19 19:17:24 2010
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To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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On Oct 19, 2010, at 5:10 PM, Joe Touch wrote:

>=20
>=20
> On 10/15/2010 3:00 PM, Jari Arkko wrote:
>> Joe,
>>=20
>> Just a quick response on one point:
>>=20
>>> - based on the note from Jari, I can't tell whether the
>>> doc is recommending SHOULD NOT for new apps or not
>>=20
>> My text did not change that aspect in any way. The new text is in the
>> introduction and has no keywords. The formal requirement is later in =
the
>> document. IIRC the current text there says SHOULD NOT for new apps.
>=20
> Correct. In that case, the note was not responsive to the thread I =
raised discussing the point that this is a new and substantial change, =
one that I feel suggests that the title, abstract, and overall structure =
should be inverted to focus on SHOULD NOT as the key point.
>=20
> It's hard to understand a doc that basically focuses on a change to =
the spec, and includes the SHOULD NOT almost parenthetically.

We're not trying to deprecate the URGENT pointer.  The goal is to first =
align the spec with the implementations, and then point out issues with =
the URGENT pointer, which is how the doc is currently structured.  I =
don't want the "SHOULD NOT" to be the main point of this document, it is =
a secondary point due to the issues listed.  The SHOULD NOT applies to =
new applications, and it means "If you really want to make use of the =
URGENT pointer, make sure you understand and have addressed all the =
issues pointed out in this document."

The SHOULD NOT for application writers using the URGENT pointer, not for =
the URGENT pointer support in the TCP implementations.  I have no =
objections to the additional highlighting of this, but that doesn't make =
it the key point of the document.  I say leave the document as it is.

			-David Borman

>=20
> Joe


From touch@isi.edu  Tue Oct 19 21:42:39 2010
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Hi, Dave,

One additional point (while I wait to see if Ron agrees with your 
summary); the current discussion of in-line data still needs to be 
revised to avoid relying primarily on the definition of a particular 
interface (SO_OOBINLINE).

As to your point, IMO *every* aspect of a protocol is SHOULD NOT by your 
definition, i.e., every capability should not be used unless its 
implications, side-effects, and nuances are understood.

To me, SHOULD NOT on an existing, required protocol capability 
effectively relegates it to "deprecated", i.e., there to support legacy 
apps. There's no point in a MUST NOT on such a capability - MUST NOT 
would mistakenly suggest that the capability need not be implemented 
anymore.

Joe

On 10/19/2010 7:18 PM, David Borman wrote:
>
> On Oct 19, 2010, at 5:10 PM, Joe Touch wrote:
>
>>
>>
>> On 10/15/2010 3:00 PM, Jari Arkko wrote:
>>> Joe,
>>>
>>> Just a quick response on one point:
>>>
>>>> - based on the note from Jari, I can't tell whether the
>>>> doc is recommending SHOULD NOT for new apps or not
>>>
>>> My text did not change that aspect in any way. The new text is in the
>>> introduction and has no keywords. The formal requirement is later in the
>>> document. IIRC the current text there says SHOULD NOT for new apps.
>>
>> Correct. In that case, the note was not responsive to the thread I raised discussing the point that this is a new and substantial change, one that I feel suggests that the title, abstract, and overall structure should be inverted to focus on SHOULD NOT as the key point.
>>
>> It's hard to understand a doc that basically focuses on a change to the spec, and includes the SHOULD NOT almost parenthetically.
>
> We're not trying to deprecate the URGENT pointer.  The goal is to first align the spec with the implementations, and then point out issues with the URGENT pointer, which is how the doc is currently structured.  I don't want the "SHOULD NOT" to be the main point of this document, it is a secondary point due to the issues listed.  The SHOULD NOT applies to new applications, and it means "If you really want to make use of the URGENT pointer, make sure you understand and have addressed all the issues pointed out in this document."
>
> The SHOULD NOT for application writers using the URGENT pointer, not for the URGENT pointer support in the TCP implementations.  I have no objections to the additional highlighting of this, but that doesn't make it the key point of the document.  I say leave the document as it is.
>
> 			-David Borman
>
>>
>> Joe
>

From jari.arkko@piuha.net  Tue Oct 19 23:15:39 2010
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References: <201009301549.RAA14282@TR-Sys.de>	<13205C286662DE4387D9AF3AC30EF456B01839E20B@EMBX01-WF.jnpr.net>	<31679F6B-0ACA-421F-8131-DA05B764A4A9@nokia.com>	<13205C286662DE4387D9AF3AC30EF456B0201C2D1C@EMBX01-WF.jnpr.net>	<FCF0200B-A1AF-4902-A110-D7EC08C39FEC@nokia.com>	<4CB31ECF.6040307@gont.com.ar>, <1D4E3C6E-B0BF-4CAC-A01E-EC9CDAC14EE0@nokia.com>	<C304DB494AC0C04C87C6A6E2FF5603DB481FB1C732@NDJSSCC01.ndc.nasa.gov>	<4CB3CCBB.1030400@isi.edu>	<675AC39A-0E6D-4014-850B-2A05389577C4@windriver.com>	<4CB488F2.1070800@isi.edu> <4CB48DCE.7010606@isi.edu> <4CB8C1CF.1020106@piuha.net> <30410F46-8F64-4988-B6ED-73ECFB7F59E9@windriver.com> <4CB8CC43.8080305@isi.edu> <4CB8CEE1.8070408@piuha.net> <4CBE1739.6040203@isi.edu> <4CBE1853.9020502@piuha.net> <4CBE1A1B.6040306@isi.edu>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, David Borman <david.borman@windriver.com>, Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] draft-ietf-tcpm-urgent-data-06
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Joe,

Yes, there were several concerned ADs. After a discussion on the list 
and in the IESG it was my opinion that the changes that I suggested were 
a sufficient compromise to make the document acceptable. I would 
personally (and Ron too) wanted to go further, indeed make the SHOULD 
NOT a major point in the document. But I chose not to make that a part 
of a blocking comment. It is of course always tricky when one AD is 
representing concerns from others as well as himself. It was my 
interpretation that what we did is an acceptable way forward, but it 
could of course happen that Ron or others feel strongly and want to 
Discuss further.

Jari


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From: Matt Mathis <mattmathis@google.com>
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Cc: TCP Maintenance and Minor Extensions WG <tcpm@ietf.org>
Subject: Re: [tcpm] 3517 update w/ SACK-entry draft
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A couple of comments: =A0There is an errata against 2018 about the
requirement that timeouts MUST clear the score board. =A0 From the
errata:

"At least one OS (Linux) violates the MUST to good effect: Even when timeou=
t
driven, it keeps old SACK data so it can avoid retransmitting data already =
at
the receiver. Thus even under severe bandwidth exhaustion, 100% of the data
delivered to the receiver causes forward progress and the system is not sub=
ject
to classical congestion collapse (that is, congestion collapse from
unnecessarily-retransmitted packets)."

It would be better to update section 5.1 of 3517bis to reflect this.
Note that the errata introduces several caveats and and additional
issues that need to be addressed.

Also there is another issue that should be introduced in the security
considerations section:

Traditional recovery mechanism serve to protect the Internet from
implementers who change congestion control to make it too aggressive,
because doing so tends to increase the probability of timeouts, and
often reduces overall throuthput.  This creates a strong disincentive
to deploying extra aggressive congestion control.

As recovery algorithms improve, this disincentive is reduced.  In the
limit an ideal recovery mechanism might never suffer from a timeout
except after losing an entire window of data or ACKs.  Such a recovery
algorithm might create an incentive to deploy arbitrarily aggressive
congestion control.  The algorithms presented in this document are are
believed to reduce the negative consequence of making congestion
control more aggressive than permitted by RFC 5681.

--
On other issue: would this be an appropriate time (and doc) to
introduce some other ideas from our rate halving work from  ages ago?

Thanks,
--MM--
The best way to predict the future is to create it. =A0- Alan Kay




On Sun, Oct 17, 2010 at 10:06 PM, Eddy, Wesley M. (GRC-MS00)[ASRC
AEROSPACE CORP] <wesley.m.eddy@nasa.gov> wrote:
> As Markku mentioned on the mailing list some time ago, and discussed in A=
naheim, the authors of draft-ietf-tcpm-sack-recovery and RFC 3517 have disc=
ussed merging the changes that the sack-recovery draft introduces into an u=
pdate to RFC 3517. =A0This makes a lot of sense, and simplifies the documen=
t tree we create a bit.
>
> At this point, they're almost ready to submit the merged document, and wa=
nt to confirm this path with the working group so they can figure out what =
to name it (draft-ietf-tcpm, or draft-lastname-tcpm). =A0We know that in An=
aheim, Lars was concerned with the timeframe this merging might take, but t=
here are no errata on 3517, and the sack-recovery work is felt to be comple=
te by the authors, so the merging has good odds of completing quickly.
>
> If there are any objections to the merged document in preparation replaci=
ng the sack-recovery working group draft (as a working group item), please =
bring them up in the next couple weeks.
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From rs@netapp.com  Thu Oct 21 15:21:04 2010
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Matt Mathis" <mattmathis@google.com>
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Cc: TCP Maintenance and Minor Extensions WG <tcpm@ietf.org>
Subject: Re: [tcpm] 3517 update w/ SACK-entry draft
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=20

> -----Original Message-----
> From: Matt Mathis [mailto:mattmathis@google.com]=20
>=20
> As recovery algorithms improve, this disincentive is reduced.=20
>  In the limit an ideal recovery mechanism might never suffer=20
> from a timeout except after losing an entire window of data=20
> or ACKs.  Such a recovery algorithm might create an incentive=20
> to deploy arbitrarily aggressive congestion control.  The=20
> algorithms presented in this document are are believed to=20
> reduce the negative consequence of making congestion control=20
> more aggressive than permitted by RFC 5681.

Based on this statement, what does that mean for lost retransmission
detection, as implemented and deployed by Linux?

Would such a more ideal recovery mechanism be considered unfit for
adoption as an RFC then?


> --
> On other issue: would this be an appropriate time (and doc)=20
> to introduce some other ideas from our rate halving work from=20
>  ages ago?
>=20

As you know, my viewpoint is, that rate halving (which is implemented in
some form in Linux and also current BSD stacks) should at least be
(informally) documented - just like lost retransmission detection
(either the way Linux is doing it, or in an idealized model).

Following up on my recent email, I'm currently investigating the special
case of loss recovery close to the end of stream, where SACK (and LRD)
fail to prevent RTOs reliably. Using SACK and timestamp information, LRD
can be made to work near optimal (no superfluous retransmits), improving
upon what Linux is already doing...

However, I would welcome any guidance the members of this group would
like to provide, before submitting my I-D on end-of-stream SACK (LRD)
behavior.

Thanks a lot,
   Richard

From ycheng@google.com  Thu Oct 21 18:12:39 2010
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From: Yuchung Cheng <ycheng@google.com>
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Cc: tcpm@ietf.org, Matthew Mathis <mattmathis@google.com>, mallman@icir.org
Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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On Fri, Oct 8, 2010 at 7:36 AM, Scheffenegger, Richard <rs@netapp.com> wrote:
>
> Consider a sequence of 4 Segments sent, S1 .. S4; S2 and S4 get dropped.
>
> Early Retransmit (RFC5827) will allow entering loss recovery.
>
> Depending on TCP SACK, or TCP NewReno, the behavior will be different:
>
> NewReno will retransmit S2;
> The ACK for S3 will trigger retransmission of S4 (and nobody got hurt)

NewReno will not retransmit S4 because the ACK elicited by S3 is a
DUP-ACK, not a partial ACK (Step 5 in RFC 2582).

W/o additional intelligence, I don't think it's a good idea to
retransmit S4 because the DUP-ACK sacking S4 often come right after
the ACK of S2.

Cheers,

Yuchung

From rs@netapp.com  Thu Oct 21 19:54:45 2010
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Date: Fri, 22 Oct 2010 03:56:13 +0100
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Thread-Topic: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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References: <DE08556E79FD1649AEE892A8C353EF611348C9477C@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <A1FD783C-6EDB-41D4-A0DE-9C77A486395D@windriver.com> <5254FC29-AD81-4B83-8AC9-7C03A7F4E74C@comsys.rwth-aachen.de> <49576D00-CF67-4A91-A00A-9B6CD37EEA46@windriver.com> <5FDC413D5FA246468C200652D63E627A0AC76DED@LDCMVEXC1-PRD.hq.netapp.com> <AANLkTi=4FCZQsw-KFXznhbFrhFkkEjxWscaE9j=HY0QO@mail.gmail.com>
From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Yuchung Cheng" <ycheng@google.com>
X-OriginalArrivalTime: 22 Oct 2010 02:56:18.0837 (UTC) FILETIME=[B35AEC50:01CB7194]
Cc: tcpm@ietf.org, Matthew Mathis <mattmathis@google.com>, mallman@icir.org
Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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Yuchung,

> > Consider a sequence of 4 Segments sent, S1 .. S4; S2 and S4=20
> > get dropped.=20

sorry for the appreviated example.

A more complete example would have been S1..S6 with S2 and S6 getting =
lost, and a window of >=3D6 when recovery is triggered, to avoid any =
confusion.


> > NewReno will retransmit S2;
> > The ACK for S3 will trigger retransmission of S4 (and=20
> > nobody got hurt)
>=20
> NewReno will not retransmit S4 because the ACK elicited by S3=20
> is a DUP-ACK, not a partial ACK (Step 5 in RFC 2582).

The initial S3 will cause a DUP-ACK (of S1)

> W/o additional intelligence, I don't think it's a good idea=20
> to retransmit S4 because the DUP-ACK sacking S4 often come=20
> right after the ACK of S2.

The retransmission of S2 will cause a Full Ack of S3, that will trigger =
the retransmission of S4 (with NewReno). There won't be any ACK for S2; =
only S1;S1;S3 (and eventually S4).


Let me paste two traces, both between the same hosts, same network, same =
drop pattern (93001 (newreno)/95001(sack) and 99001 are dropped at first =
transmit). Only difference is NewReno vs. SACK. These were recorded on =
the receiver side, and timed such that the sender would end up in =
recovery mode until after the last segment of the stream.=20

Note the time-delta when 99001 is retransmitted....

NewReno IS retransmitting the S4 in the original example, while SACK is =
not...


NewReno:
0.000009 A -> B TCP  93001 1    5888 48709 > 5001 [ACK] Seq=3D93001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736607 [drop]
0.000005 A -> B TCP  94001 1    5888 48709 > 5001 [ACK] Seq=3D94001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736607
0.000012 B -> A TCP  1 93001  990528 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D990528 Len=3D0 TSval=3D15736608 TSecr=3D375481555
0.000029 A -> B TCP  95001 1    5888 48709 > 5001 [ACK] Seq=3D95001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736607
0.000012 B -> A TCP  1 93001  990528 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D990528 Len=3D0 TSval=3D15736608 TSecr=3D375481555 [TCP =
Dup ACK 150#1]
0.000044 A -> B TCP  96001 1    5888 48709 > 5001 [ACK] Seq=3D96001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736607
0.000011 B -> A TCP  1 93001  990528 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D990528 Len=3D0 TSval=3D15736608 TSecr=3D375481555 [TCP =
Dup ACK 150#2]=20
0.000078 A -> B TCP  97001 1    5888 48709 > 5001 [ACK] Seq=3D97001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736608
0.000012 B -> A TCP  1 93001  990528 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D990528 Len=3D0 TSval=3D15736608 TSecr=3D375481555 [TCP =
Dup ACK 150#3]=20
0.000029 A -> B TCP  98001 1    5888 48709 > 5001 [ACK] Seq=3D98001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736608
0.000011 B -> A TCP  1 93001  990528 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D990528 Len=3D0 TSval=3D15736608 TSecr=3D375481555 [TCP =
Dup ACK 150#4]=20
0.000029 A -> B TCP  99001 1    5888 48709 > 5001 [FIN, PSH, ACK] =
Seq=3D99001 Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 =
TSecr=3D15736608 [drop]
0.000099 B -> A TCP  1 93001  998720 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D998720 Len=3D0 TSval=3D15736608 TSecr=3D375481555 [TCP =
Window Update]
0.000151 B -> A TCP  1 93001 1006912 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D93001 Win=3D1006912 Len=3D0 TSval=3D15736608 TSecr=3D375481555 =
[TCP Window Update]=20
0.000126 A -> B TCP  93001 1    5888 48709 > 5001 [ACK] Seq=3D93001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 TSecr=3D15736608 [TCP =
Out-Of-Order]
0.000017 B -> A TCP  1 99001 1000960 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D99001 Win=3D1000960 Len=3D0 TSval=3D15736609 TSecr=3D375481555
0.000116 B -> A TCP  1 99001 1009152 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D99001 Win=3D1009152 Len=3D0 TSval=3D15736609 TSecr=3D375481555 =
[TCP Window Update]
0.000151 B -> A TCP  1 99001 1017344 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D99001 Win=3D1017344 Len=3D0 TSval=3D15736609 TSecr=3D375481555 =
[TCP Window Update]
0.000119 A -> B TCP  99001 1    5888 48709 > 5001 [FIN, PSH, ACK] =
Seq=3D99001 Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D375481555 =
TSecr=3D15736609 [TCP Out-Of-Order]
0.000016 B -> A TCP 1 100002 1016320 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D100002 Win=3D1016320 Len=3D0 TSval=3D15736609 TSecr=3D375481555
0.000125 B -> A TCP 1 100002 1024512 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D100002 Win=3D1024512 Len=3D0 TSval=3D15736609 TSecr=3D375481555 =
[TCP Window Update]=20
0.000150 B -> A TCP 1 100002 1032704 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D100002 Win=3D1032704 Len=3D0 TSval=3D15736609 TSecr=3D375481555 =
[TCP Window Update]=20
0.000148 B -> A TCP 1 100002 1040896 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D100002 Win=3D1040896 Len=3D0 TSval=3D15736609 TSecr=3D375481555 =
[TCP Window Update]=20
0.000166 B -> A TCP 1 100002 1048960 5001 > 48709 [ACK] Seq=3D1 =
Ack=3D100002 Win=3D1048960 Len=3D0 TSval=3D15736610 TSecr=3D375481555 =
[TCP Window Update]=20
0.000206 B -> A TCP 1 100002 1048960 5001 > 48709 [FIN, ACK] Seq=3D1 =
Ack=3D100002 Win=3D1048960 Len=3D0 TSval=3D15736610 TSecr=3D375481555
0.001338 A -> B TCP 100002 2    5888 48709 > 5001 [ACK] Seq=3D100002 =
Ack=3D2 Win=3D5888 Len=3D0 TSval=3D375481556 TSecr=3D15736610


SACK:

0.000006 A -> B TCP  95001 1    5888 60934 > 5001 [ACK] Seq=3D95001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716224 TSecr=3D3613608261 =
[drop]
0.000006 A -> B TCP  96001 1    5888 60934 > 5001 [ACK] Seq=3D96001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716224 TSecr=3D3613608261
0.000005 A -> B TCP  97001 1    5888 60934 > 5001 [ACK] Seq=3D97001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716224 TSecr=3D3613608261
0.000006 A -> B TCP  98001 1    5888 60934 > 5001 [ACK] Seq=3D98001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716224 TSecr=3D3613608261
0.000066 A -> B TCP  99001 1    5888 60934 > 5001 [FIN, PSH, ACK] =
Seq=3D99001 Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716224 =
TSecr=3D3613608261 [drop]
0.003322 B -> A TCP  1 90001 1046976 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D90001 Win=3D1046976 Len=3D0 TSval=3D3613608265 TSecr=3D367716224
0.000078 B -> A TCP  1 92001 1044992 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D92001 Win=3D1044992 Len=3D0 TSval=3D3613608265 TSecr=3D367716224
0.000032 B -> A TCP  1 94001 1042944 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D94001 Win=3D1042944 Len=3D0 TSval=3D3613608265 TSecr=3D367716224
0.000032 B -> A TCP  1 95001 1041984 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D95001 Win=3D1041984 Len=3D0 TSval=3D3613608265 TSecr=3D367716224 =
SLE=3D96001 SRE=3D97001
0.000032 B -> A TCP  1 95001 1041984 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D95001 Win=3D1041984 Len=3D0 TSval=3D3613608265 TSecr=3D367716224 =
SLE=3D96001 SRE=3D98001 [TCP Dup ACK 163#1]=20
0.000032 B -> A TCP  1 95001 1041984 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D95001 Win=3D1041984 Len=3D0 TSval=3D3613608265 TSecr=3D367716224 =
SLE=3D96001 SRE=3D99001 [TCP Dup ACK 163#2]=20
0.000032 B -> A TCP  1 95001 1048960 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D95001 Win=3D1048960 Len=3D0 TSval=3D3613608265 TSecr=3D367716224 =
SLE=3D96001 SRE=3D99001 [TCP Window Update]=20
0.000434 A -> B TCP  95001 1    5888 60934 > 5001 [ACK] Seq=3D95001 =
Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716225 TSecr=3D3613608265 [TCP =
Retransmission]=20
0.003328 B -> A TCP  1 99001 1044992 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D99001 Win=3D1044992 Len=3D0 TSval=3D3613608269 TSecr=3D367716225
0.000084 B -> A TCP  1 99001 1048960 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D99001 Win=3D1048960 Len=3D0 TSval=3D3613608269 TSecr=3D367716225 =
[TCP Window Update]=20
0.202595 A -> B TCP  99001 1    5888 60934 > 5001 [FIN, PSH, ACK] =
Seq=3D99001 Ack=3D1 Win=3D5888 Len=3D1000 TSval=3D367716277 =
TSecr=3D3613608269 [TCP Retransmission]
0.003322 B -> A TCP 1 100002 1048000 5001 > 60934 [ACK] Seq=3D1 =
Ack=3D100002 Win=3D1048000 Len=3D0 TSval=3D3613608475 TSecr=3D367716277
0.000078 B -> A TCP 1 100002 1048960 5001 > 60934 [FIN, ACK] Seq=3D1 =
Ack=3D100002 Win=3D1048960 Len=3D0 TSval=3D3613608475 TSecr=3D367716277
0.000473 A -> B TCP 100002 2    5888 60934 > 5001 [ACK] Seq=3D100002 =
Ack=3D2 Win=3D5888 Len=3D0 TSval=3D367716277 TSecr=3D3613608475


Richard Scheffenegger
Storage Infrastructure Architect
Area Deutsche Telekom=20
NetApp
rs@netapp.com
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20

EURO PLAZA
Geb=E4ude G, Stiege 7, 3.OG
Am Euro Platz 2
A-1120 Wien

=20

> -----Original Message-----
> From: Yuchung Cheng [mailto:ycheng@google.com]=20
> Sent: Donnerstag, 21. Oktober 2010 21:14
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org; Matthew Mathis; mallman@icir.org
> Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
>=20
> On Fri, Oct 8, 2010 at 7:36 AM, Scheffenegger, Richard=20
> <rs@netapp.com> wrote:
> >
> > Consider a sequence of 4 Segments sent, S1 .. S4; S2 and S4=20
> get dropped.
> >
> > Early Retransmit (RFC5827) will allow entering loss recovery.
> >
> > Depending on TCP SACK, or TCP NewReno, the behavior will be=20
> different:
> >
> > NewReno will retransmit S2;
> > The ACK for S3 will trigger retransmission of S4 (and=20
> nobody got hurt)
>=20
> NewReno will not retransmit S4 because the ACK elicited by S3=20
> is a DUP-ACK, not a partial ACK (Step 5 in RFC 2582).
>=20
> W/o additional intelligence, I don't think it's a good idea=20
> to retransmit S4 because the DUP-ACK sacking S4 often come=20
> right after the ACK of S2.
>=20
> Cheers,
>=20
> Yuchung
>=20

From yoshifumi.nishida@gmail.com  Fri Oct 22 01:41:50 2010
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Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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Hello Richard,

I think this discussion should be based on rfc3517 since it describes
a loss recovery algorithm for TCP SACK.

Thanks,
--
Yoshifumi Nishida
nishida@sfc.wide.ad.jp

2010/10/8 Scheffenegger, Richard <rs@netapp.com>:
>
> Hello members of the TCPM group, Matt, Mark,
>
>
> I'm currently looking into the loss recovery behavior of TCP SACK -
> especially at the end-of-stream.
>
>
> End-of-stream (case 2) of the small congestion window, as per RFC5827)
> is of particular interest, as there are synchronous IO loads over TCP,
> that will only deliver additional data to the socket once an application
> level acknowledgement has been received. To be precise, single threaded,
> synchronous NFS read/write IOs. However that application layer Ack will
> only be sent by the receiver, once the complete App IO was received,
> thus any stalls can tremendeously delay the IO on the application.
>
> The analogy would be TCP (with a large MTU) within TCP (with a small
> MTU) - which is known to behave sub-optimal.
>
>
>
> It appears that RFC2018 has codified behavior, which makes TCP SACK
> perform less optimal compared with TCP NewReno, when loss occurs
> directly at the last segments of the stream.
>
> "
> 5. Interpreting the Sack Option and Retransmission Strategy: Data
> =A0 Sender Behavior
> :
> :
> =A0 After the SACKed bit is turned on (as the result of processing a
> =A0 received SACK option), the data sender will skip that segment during
> =A0 any later retransmission. =A0Any segment that has the SACKed bit turn=
ed
> =A0 off and is less than the highest SACKed segment is available for
> =A0 retransmission.
> "
>
> This definition leaves out the range [snd_fack .. snd_nxt (at loss
> recovery initiation time)]. There is no explanation as to why that
> omission was done (other than snd_fack not being defined at the time).
>
>
> As RFC5827 is only concerned about detection and entering loss recovery,
> it too gives no advice in this regard.
>
> Consider a sequence of 4 Segments sent, S1 .. S4; S2 and S4 get dropped.
>
> Early Retransmit (RFC5827) will allow entering loss recovery.
>
> Depending on TCP SACK, or TCP NewReno, the behavior will be different:
>
> NewReno will retransmit S2;
> The ACK for S3 will trigger retransmission of S4 (and nobody got hurt)
>
> SACK will retransmit S2;
> The ACK for S3 will *not* trigger a retransmit of S4 (it falls outside
> the boundary defined in RFC2018).
> Only a RTO will recover S4, and the stream might continue (as
> applications resume operation).
>
> Of course, S4 could be any number of packets, just prior to the
> end-of-stream. And the same sequence of events can happen without Early
> Retransmit and large cwnds...
>
> With infinite data to send, after SACK finished loss recovery - at the
> highest SACKed (snd_fack) block - the sender would resume sending
> starting from snd_nxt (and most likely in slow start). As soon as 3 new
> segments would be received, this missing segment chain (snd_fack ..
> Former snd_nxt) could be detected and recovered. Of course, counting
> this as one additional congestion event (unlike what NewReno would do).
>
>
> Or are SACK implementations which silently ignore the hole between
> snd_fack and snd_nxt (when loss recovery is entered) broken?
>
>
>
>
> I would like to learn what you think would be the proper behavior. From
> my point of view, this appears to have been a simple oversight in
> RFC2018, and changing the statement to
>
>
> "
> 5. Interpreting the Sack Option and Retransmission Strategy: Data
> =A0 Sender Behavior
> :
> :
> =A0 After the SACKed bit is turned on (as the result of processing a
> =A0 received SACK option), the data sender will skip that segment during
> =A0 any later retransmission. =A0Any segment that has the SACKed bit turn=
ed
> =A0 off and is less than the highest sent segment at the beginning of the
>
> =A0 SACK recovery episode is available for retransmission.
> "
>
> The fix would be simply add an implicit hole (or an implicit SACK block)
> up to snd_nxt. If none of these tailing segments was lost, they will be
> covered in an SACK eventually. That would happen before the sender could
> retransmit those segments, as cwnd is also reduced thus limiting the
> sending rate.
>
>
>
>
> Richard Scheffenegger
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From wwwrun@core3.amsl.com  Fri Oct 22 06:56:06 2010
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Subject: [tcpm] Protocol Action: 'On the implementation of the TCP urgent mechanism' to Proposed Standard
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The IESG has approved the following document:

- 'On the implementation of the TCP urgent mechanism '
   <draft-ietf-tcpm-urgent-data-07.txt> as a Proposed Standard


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

The IESG contact person is Lars Eggert.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-urgent-data-07.txt

Technical Summary

  This document analyzes how current TCP implementations process TCP
  urgent indications, and how the behavior of some widely-deployed
  middle-boxes affect how urgent indications are processed by end
  systems.  This document updates the relevant specifications such that
  they accommodate current practice in processing TCP urgent
  indications, provides advice to applications that make use of the
  urgent mechanism, and raises awareness about the reliability of TCP
  urgent indications in the current Internet.

Working Group Summary

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

Document Quality

  This document specifically is aimed at bringing the specification and
  existing implementations into line clearly with each other.  The
  implementation practices in this document have been adopted in multiple
  implementations and this document provides guidance to aid in future
  compatibility with applications in new and old TCP implementations.

Personnel

  Wesley Eddy (Wesley.M.Eddy@nasa.gov) is the document shepherd.
  Lars Eggert (lars.eggert@nokia.com) reviewed this document for the IESG.


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Thread-Topic: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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References: <DE08556E79FD1649AEE892A8C353EF611348C9477C@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><A1FD783C-6EDB-41D4-A0DE-9C77A486395D@windriver.com><5254FC29-AD81-4B83-8AC9-7C03A7F4E74C@comsys.rwth-aachen.de><49576D00-CF67-4A91-A00A-9B6CD37EEA46@windriver.com><5FDC413D5FA246468C200652D63E627A0AC76DED@LDCMVEXC1-PRD.hq.netapp.com> <AANLkTincZy08rQq-fCP5PrLQn5JeooZFkq=mLjJ4zibg@mail.gmail.com>
From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Yoshifumi Nishida" <nishida@sfc.wide.ad.jp>
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Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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Thank you Yoshifumi-san,

I agree; it's the combination of SACK signalling (as defined in
RFC2018), which segments are elegibe for SACK-based retransmission, and
RFC3517 (how to actually behave during SACK loss recovery).

Unfortunately, neiter of which covers the corner case of loosing the
last segment(s) of a stream [while already in loss recovery].

With NewReno, every FullAck below SND.MAX causes the retransmission of
one segment (after the ACK), before continuing with the next segment
after SND.MAX...

But my findings so far indicate, that such a FullACK (no SACK entries)
below SND.MAX doesn't trigger the retransmission of the next segment.

Obviously, with a continous stream of data, this is not a big issue, as
eventually you will have a SACK entry again, making the lost, previously
last, segment elegible for retransmission under RFC2018/3517.



Would there be any support by the group for an I-D, which addresses this
particular corner case?=20


Richard Scheffenegger

> -----Original Message-----
> From: Yoshifumi Nishida [mailto:nishida@sfc.wide.ad.jp]=20
> Sent: Freitag, 22. Oktober 2010 04:43
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
>=20
> Hello Richard,
>=20
> I think this discussion should be based on rfc3517 since it=20
> describes a loss recovery algorithm for TCP SACK.
>=20
> Thanks,
> --
> Yoshifumi Nishida
> nishida@sfc.wide.ad.jp
>=20

From mattmathis@google.com  Fri Oct 22 12:20:26 2010
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Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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I think it would be better to try to get it into 3517bis, but if that
doesn't work, a separate ID will do.

My rational:  If the very last segment is lost, the partial ACK for
the preceding segment will not include a SACK block because the
receiver mistakenly thinks the data is complete.  I would characterize
this situation as leaving SACK recovery prematurely, which why it
should be in scope for 3517bis.

Thanks,
--MM--
The best way to predict the future is to create it. =A0- Alan Kay

On Fri, Oct 22, 2010 at 11:25 AM, Scheffenegger, Richard <rs@netapp.com> wr=
ote:
>
> Thank you Yoshifumi-san,
>
> I agree; it's the combination of SACK signalling (as defined in
> RFC2018), which segments are elegibe for SACK-based retransmission, and
> RFC3517 (how to actually behave during SACK loss recovery).
>
> Unfortunately, neiter of which covers the corner case of loosing the
> last segment(s) of a stream [while already in loss recovery].
>
> With NewReno, every FullAck below SND.MAX causes the retransmission of
> one segment (after the ACK), before continuing with the next segment
> after SND.MAX...
>
> But my findings so far indicate, that such a FullACK (no SACK entries)
> below SND.MAX doesn't trigger the retransmission of the next segment.
>
> Obviously, with a continous stream of data, this is not a big issue, as
> eventually you will have a SACK entry again, making the lost, previously
> last, segment elegible for retransmission under RFC2018/3517.
>
>
>
> Would there be any support by the group for an I-D, which addresses this
> particular corner case?
>
>
> Richard Scheffenegger
>
>> -----Original Message-----
>> From: Yoshifumi Nishida [mailto:nishida@sfc.wide.ad.jp]
>> Sent: Freitag, 22. Oktober 2010 04:43
>> To: Scheffenegger, Richard
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
>>
>> Hello Richard,
>>
>> I think this discussion should be based on rfc3517 since it
>> describes a loss recovery algorithm for TCP SACK.
>>
>> Thanks,
>> --
>> Yoshifumi Nishida
>> nishida@sfc.wide.ad.jp
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 22 Oct 2010 14:14:14 -0700
Message-ID: <AANLkTikkcC4rBoV0Pu+OHbcg9EQ=W+RO-zriOoye5d1k@mail.gmail.com>
To: Matt Mathis <mattmathis@google.com>
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Cc: tcpm@ietf.org
Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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The example Richard gave makes lots of sense; it should improve loss
recovery latency for web traffic as well. I also suggest revising RFC3517
about this case. My rational is simply TCP has too many RFCs on similar
subjects. It's nightmare for developers (at least me) to follow up.

but I disagree it's receiver's mistake: the receiver has no way to tell in
this case! unless we put SND_NXT in the sender packets.

On Fri, Oct 22, 2010 at 12:21 PM, Matt Mathis <mattmathis@google.com> wrote:

> I think it would be better to try to get it into 3517bis, but if that
> doesn't work, a separate ID will do.
>
> My rational:  If the very last segment is lost, the partial ACK for
> the preceding segment will not include a SACK block because the
> receiver mistakenly thinks the data is complete.  I would characterize
> this situation as leaving SACK recovery prematurely, which why it
> should be in scope for 3517bis.
>
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
> On Fri, Oct 22, 2010 at 11:25 AM, Scheffenegger, Richard <rs@netapp.com>
> wrote:
> >
> > Thank you Yoshifumi-san,
> >
> > I agree; it's the combination of SACK signalling (as defined in
> > RFC2018), which segments are elegibe for SACK-based retransmission, and
> > RFC3517 (how to actually behave during SACK loss recovery).
> >
> > Unfortunately, neiter of which covers the corner case of loosing the
> > last segment(s) of a stream [while already in loss recovery].
> >
> > With NewReno, every FullAck below SND.MAX causes the retransmission of
> > one segment (after the ACK), before continuing with the next segment
> > after SND.MAX...
> >
> > But my findings so far indicate, that such a FullACK (no SACK entries)
> > below SND.MAX doesn't trigger the retransmission of the next segment.
> >
> > Obviously, with a continous stream of data, this is not a big issue, as
> > eventually you will have a SACK entry again, making the lost, previously
> > last, segment elegible for retransmission under RFC2018/3517.
> >
> >
> >
> > Would there be any support by the group for an I-D, which addresses this
> > particular corner case?
> >
> >
> > Richard Scheffenegger
> >
> >> -----Original Message-----
> >> From: Yoshifumi Nishida [mailto:nishida@sfc.wide.ad.jp]
> >> Sent: Freitag, 22. Oktober 2010 04:43
> >> To: Scheffenegger, Richard
> >> Cc: tcpm@ietf.org
> >> Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
> >>
> >> Hello Richard,
> >>
> >> I think this discussion should be based on rfc3517 since it
> >> describes a loss recovery algorithm for TCP SACK.
> >>
> >> Thanks,
> >> --
> >> Yoshifumi Nishida
> >> nishida@sfc.wide.ad.jp
> >>
> > _______________________________________________
> > 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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The example Richard gave makes lots of sense; it should improve loss recove=
ry latency for web traffic as well.=A0I also suggest revising RFC3517 about=
 this case. My rational=A0is simply TCP has too many RFCs on similar subjec=
ts. It&#39;s nightmare for developers (at least me) to follow up.<div>

<br></div><div>but I disagree it&#39;s receiver&#39;s mistake: the receiver=
 has no way to tell in this case! unless we put SND_NXT in the sender packe=
ts.</div><div><br><div class=3D"gmail_quote">On Fri, Oct 22, 2010 at 12:21 =
PM, Matt Mathis <span dir=3D"ltr">&lt;<a href=3D"mailto:mattmathis@google.c=
om">mattmathis@google.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">I think it would be better to try to get it=
 into 3517bis, but if that<br>
doesn&#39;t work, a separate ID will do.<br>
<br>
My rational: =A0If the very last segment is lost, the partial ACK for<br>
the preceding segment will not include a SACK block because the<br>
receiver mistakenly thinks the data is complete. =A0I would characterize<br=
>
this situation as leaving SACK recovery prematurely, which why it<br>
should be in scope for 3517bis.<br>
<br>
Thanks,<br>
--MM--<br>
The best way to predict the future is to create it. =A0- Alan Kay<br>
<div><div></div><div class=3D"h5"><br>
On Fri, Oct 22, 2010 at 11:25 AM, Scheffenegger, Richard &lt;<a href=3D"mai=
lto:rs@netapp.com">rs@netapp.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Thank you Yoshifumi-san,<br>
&gt;<br>
&gt; I agree; it&#39;s the combination of SACK signalling (as defined in<br=
>
&gt; RFC2018), which segments are elegibe for SACK-based retransmission, an=
d<br>
&gt; RFC3517 (how to actually behave during SACK loss recovery).<br>
&gt;<br>
&gt; Unfortunately, neiter of which covers the corner case of loosing the<b=
r>
&gt; last segment(s) of a stream [while already in loss recovery].<br>
&gt;<br>
&gt; With NewReno, every FullAck below SND.MAX causes the retransmission of=
<br>
&gt; one segment (after the ACK), before continuing with the next segment<b=
r>
&gt; after SND.MAX...<br>
&gt;<br>
&gt; But my findings so far indicate, that such a FullACK (no SACK entries)=
<br>
&gt; below SND.MAX doesn&#39;t trigger the retransmission of the next segme=
nt.<br>
&gt;<br>
&gt; Obviously, with a continous stream of data, this is not a big issue, a=
s<br>
&gt; eventually you will have a SACK entry again, making the lost, previous=
ly<br>
&gt; last, segment elegible for retransmission under RFC2018/3517.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Would there be any support by the group for an I-D, which addresses th=
is<br>
&gt; particular corner case?<br>
&gt;<br>
&gt;<br>
&gt; Richard Scheffenegger<br>
&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: Yoshifumi Nishida [mailto:<a href=3D"mailto:nishida@sfc.wide=
.ad.jp">nishida@sfc.wide.ad.jp</a>]<br>
&gt;&gt; Sent: Freitag, 22. Oktober 2010 04:43<br>
&gt;&gt; To: Scheffenegger, Richard<br>
&gt;&gt; Cc: <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt;&gt; Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK erra=
ta?)<br>
&gt;&gt;<br>
&gt;&gt; Hello Richard,<br>
&gt;&gt;<br>
&gt;&gt; I think this discussion should be based on rfc3517 since it<br>
&gt;&gt; describes a loss recovery algorithm for TCP SACK.<br>
&gt;&gt;<br>
&gt;&gt; Thanks,<br>
&gt;&gt; --<br>
&gt;&gt; Yoshifumi Nishida<br>
&gt;&gt; <a href=3D"mailto:nishida@sfc.wide.ad.jp">nishida@sfc.wide.ad.jp</=
a><br>
&gt;&gt;<br>
&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt;<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div>

--20cf3040e76a1de78e04933b219b--

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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Yuchung Cheng" <ycheng@google.com>, "Matt Mathis" <mattmathis@google.com>, "Ethan Blanton" <eblanton@cs.ohiou.edu>
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Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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Hi Ethan, Mattis, Yuchung and Yoshifuma,
=20
Let me state the conclusion of the below thoughts first:
=20
A Full ACK (no SACK blocks) of SND.FACK should be a strong enough
trigger to start retransmit of all segments between SND.FACK and
SND.MAX. With the nice side effect that no additional state needs to be
kept or saved in the senders TCP when entering SACK loss recovery...=20
=20
=20
=20
=20
Some thoughts w.r.t. conservation of packets when such a problem
happens; please correct me, if any of these assumptions don't hold:
=20
o) worst case scenario would be making one window minus 4 (initial loss,
3 delivered to trigger sack loss recovery, remaining segments heavily
delayed) segments elegible for retransmissions.
=20
o) would require delay (reordering) of all of these segments for at
least one RTT (delivery after S2 is received on the receiver side).
=20
o) due to the loss of the ACK clock, no more than at most dupthresh
segments would be sent spurious after receiving an FullACK, while the
remaining segments are
still delayed.
=20
o) depending on implementation details, up to (cwnd-4)*7/12 segments may
be retransmitted unneccessarly, but nicely clocked out by ACKs. (With
delayed ACKs, that might only be (cwnd-4)*3/12 segments).
=20
=20
However, if the reordering delay would be higher than 1 RTT, even RTO
based loss recovery would sent spurious retransmits.... but potentially
even more than allowing SACK loss recovery to continue: (cwnd-4) - and
by doing a slow start.
=20
=20
Overall, for low RTT environment, I would argue that the benefit of
addressing this RTO far outweights the potential, worst case cost.=20
=20
For high RTT environment, the current behavior can already be worse
(slow-starting (bursts) after RTO, bursting out potentially delayed
segments) than the worst case behavior of addressing this corner case -
provided my back of the envelope sketch above holds any merit.
=20
The scenario to end up with worst case violation of conservation of
packets seem pretty extreme (LIFO delivery, exactly timed, adjusted to
the cwnd....), so in a normal heavy reordering scenario, only very few
segments may be retransmitted unnecessarily...
=20
=20
A Full ACK (no SACK blocks) of SND.FACK should be a strong enough
trigger to start retransmit of all segments between SND.FACK and
SND.MAX. With the nice side effect that no additional state needs to be
kept or saved when entering SACK loss recovery...=20
=20
ACKs with (higher) SACK entries received after starting the
retransmission of that SND.FACK ..  SND.MAX range may only cause a
reduction of unnecessarily transmitted segments (IIRC, ACKs with SACK
entries that don't ack any previously unacked segment, are not counting
against the rate-halving scheme, or the rfc3517 pipe calculation...).
=20

Richard Scheffenegger



________________________________

	From: Yuchung Cheng [mailto:ycheng@google.com]=20
	Sent: Freitag, 22. Oktober 2010 17:14
	To: Matt Mathis
	Cc: Scheffenegger, Richard; tcpm@ietf.org
	Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK
errata?)
=09
=09
	The example Richard gave makes lots of sense; it should improve
loss recovery latency for web traffic as well. I also suggest revising
RFC3517 about this case. My rational is simply TCP has too many RFCs on
similar subjects. It's nightmare for developers (at least me) to follow
up.=20

	but I disagree it's receiver's mistake: the receiver has no way
to tell in this case! unless we put SND_NXT in the sender packets.

	On Fri, Oct 22, 2010 at 12:21 PM, Matt Mathis
<mattmathis@google.com> wrote:
=09

		I think it would be better to try to get it into
3517bis, but if that
		doesn't work, a separate ID will do.
	=09
		My rational:  If the very last segment is lost, the
partial ACK for
		the preceding segment will not include a SACK block
because the
		receiver mistakenly thinks the data is complete.  I
would characterize
		this situation as leaving SACK recovery prematurely,
which why it
		should be in scope for 3517bis.
	=09
		Thanks,
		--MM--
		The best way to predict the future is to create it.  -
Alan Kay
	=09

		On Fri, Oct 22, 2010 at 11:25 AM, Scheffenegger, Richard
<rs@netapp.com> wrote:
		>
		> Thank you Yoshifumi-san,
		>
		> I agree; it's the combination of SACK signalling (as
defined in
		> RFC2018), which segments are elegibe for SACK-based
retransmission, and
		> RFC3517 (how to actually behave during SACK loss
recovery).
		>
		> Unfortunately, neiter of which covers the corner case
of loosing the
		> last segment(s) of a stream [while already in loss
recovery].
		>
		> With NewReno, every FullAck below SND.MAX causes the
retransmission of
		> one segment (after the ACK), before continuing with
the next segment
		> after SND.MAX...
		>
		> But my findings so far indicate, that such a FullACK
(no SACK entries)
		> below SND.MAX doesn't trigger the retransmission of
the next segment.
		>
		> Obviously, with a continous stream of data, this is
not a big issue, as
		> eventually you will have a SACK entry again, making
the lost, previously
		> last, segment elegible for retransmission under
RFC2018/3517.
		>
		>
		>
		> Would there be any support by the group for an I-D,
which addresses this
		> particular corner case?
		>
		>
		> Richard Scheffenegger
		>
		>> -----Original Message-----
		>> From: Yoshifumi Nishida
[mailto:nishida@sfc.wide.ad.jp]
		>> Sent: Freitag, 22. Oktober 2010 04:43
		>> To: Scheffenegger, Richard
		>> Cc: tcpm@ietf.org
		>> Subject: Re: [tcpm] end-of-stream loss recovery
(RFC2018 SACK errata?)
		>>
		>> Hello Richard,
		>>
		>> I think this discussion should be based on rfc3517
since it
		>> describes a loss recovery algorithm for TCP SACK.
		>>
		>> Thanks,
		>> --
		>> Yoshifumi Nishida
		>> nishida@sfc.wide.ad.jp
		>>
		> _______________________________________________
		> 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
	=09



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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D248144221-22102010>Hi Ethan, Mattis, Yuchung and=20
Yoshifuma,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D248144221-22102010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>Let me state the conclusion of the below thoughts=20
first:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>A&nbsp;Full ACK (no SACK blocks)&nbsp;of SND.FACK =
should be a=20
strong enough trigger to start retransmit of all segments&nbsp;between =
SND.FACK=20
and SND.MAX. With the nice side effect that no additional state needs to =
be kept=20
or saved&nbsp;in the senders TCP when entering&nbsp;SACK loss =
recovery...=20
</FONT></SPAN></DIV></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>Some thoughts&nbsp;w.r.t. conservation of =
packets&nbsp;when=20
such a problem happens; please correct me, if any of these assumptions =
don't=20
hold:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>o) worst case scenario would be&nbsp;making one =
window minus 4=20
(initial loss, 3 delivered to trigger sack loss recovery, remaining =
segments=20
heavily delayed) segments elegible for =
retransmissions.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>o) would require delay (reordering) of&nbsp;all of =
these=20
segments&nbsp;for&nbsp;at least one&nbsp;RTT (delivery after S2 is =
received on=20
the receiver side).</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>o) due to the loss of the ACK clock, no more than =
at most=20
dupthresh segments would be sent spurious after receiving an FullACK, =
while the=20
remaining segments are</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>still delayed.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>o) depending on implementation details, up to =
(cwnd-4)*7/12=20
segments may be retransmitted unneccessarly, but nicely clocked out by =
ACKs.=20
(With delayed ACKs, that might only be (cwnd-4)*3/12=20
segments).</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>However, if the reordering delay would be higher =
than 1 RTT,=20
even RTO based loss recovery would sent spurious retransmits.... but =
potentially=20
even more than allowing SACK loss recovery to continue: (cwnd-4) - and =
by doing=20
a slow start.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>Overall, for&nbsp;low RTT environment, I would =
argue that the=20
benefit of addressing this RTO far outweights the potential, worst case =
cost.=20
</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>For high RTT environment, the current behavior can =
already be=20
worse (slow-starting (bursts) after RTO, bursting out potentially =
delayed=20
segments)&nbsp;than the worst case behavior of addressing this corner =
case -=20
provided my back of the envelope sketch above holds any=20
merit.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>The scenario to end up with worst case violation =
of=20
conservation of packets seem pretty extreme (LIFO delivery, exactly =
timed,=20
adjusted to the cwnd....), so in a normal heavy reordering scenario, =
only very=20
few segments may be retransmitted unnecessarily...</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>A&nbsp;Full ACK (no SACK blocks)&nbsp;of SND.FACK =
should be a=20
strong enough trigger to start retransmit of all segments&nbsp;between =
SND.FACK=20
and SND.MAX. With the nice side effect that no additional state needs to =
be kept=20
or saved&nbsp;when entering&nbsp;SACK loss recovery... =
</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D248144221-22102010><FONT =
color=3D#0000ff=20
size=3D2 face=3DArial>ACKs with (higher) SACK entries received after =
starting the=20
retransmission of that SND.FACK ..&nbsp; SND.MAX range may only cause a=20
reduction of unnecessarily transmitted segments (IIRC, ACKs with SACK =
entries=20
that don't ack any previously unacked segment, are not counting against =
the=20
rate-halving scheme, or the rfc3517 pipe =
calculation...).</FONT></SPAN></DIV>
<DIV><SPAN class=3D248144221-22102010><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV><!-- Converted from text/rtf =
format -->
<P><B><SPAN lang=3Dde><FONT size=3D2 face=3DArial>Richard=20
Scheffenegger</FONT></SPAN></B><SPAN lang=3Dde><BR></SPAN></P><BR>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; PADDING-LEFT: 5px; MARGIN-LEFT: =
5px; MARGIN-RIGHT: 0px"=20
dir=3Dltr>
  <DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT size=3D2 face=3DTahoma><B>From:</B> Yuchung Cheng =
[mailto:ycheng@google.com]=20
  <BR><B>Sent:</B> Freitag, 22. Oktober 2010 17:14<BR><B>To:</B> Matt=20
  Mathis<BR><B>Cc:</B> Scheffenegger, Richard; =
tcpm@ietf.org<BR><B>Subject:</B>=20
  Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK=20
  errata?)<BR></FONT><BR></DIV>
  <DIV></DIV>The example Richard gave makes lots of sense; it should =
improve=20
  loss recovery latency for web traffic as well.&nbsp;I also suggest =
revising=20
  RFC3517 about this case. My rational&nbsp;is simply TCP has too many =
RFCs on=20
  similar subjects. It's nightmare for developers (at least me) to =
follow up.
  <DIV><BR></DIV>
  <DIV>but I disagree it's receiver's mistake: the receiver has no way =
to tell=20
  in this case! unless we put SND_NXT in the sender packets.</DIV>
  <DIV><BR>
  <DIV class=3Dgmail_quote>On Fri, Oct 22, 2010 at 12:21 PM, Matt Mathis =
<SPAN=20
  dir=3Dltr>&lt;<A=20
  =
href=3D"mailto:mattmathis@google.com">mattmathis@google.com</A>&gt;</SPAN=
>=20
  wrote:<BR>
  <BLOCKQUOTE=20
  style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
  class=3Dgmail_quote>I think it would be better to try to get it into =
3517bis,=20
    but if that<BR>doesn't work, a separate ID will do.<BR><BR>My =
rational:=20
    &nbsp;If the very last segment is lost, the partial ACK for<BR>the =
preceding=20
    segment will not include a SACK block because the<BR>receiver =
mistakenly=20
    thinks the data is complete. &nbsp;I would characterize<BR>this =
situation as=20
    leaving SACK recovery prematurely, which why it<BR>should be in =
scope for=20
    3517bis.<BR><BR>Thanks,<BR>--MM--<BR>The best way to predict the =
future is=20
    to create it. &nbsp;- Alan Kay<BR>
    <DIV>
    <DIV></DIV>
    <DIV class=3Dh5><BR>On Fri, Oct 22, 2010 at 11:25 AM, Scheffenegger, =
Richard=20
    &lt;<A href=3D"mailto:rs@netapp.com">rs@netapp.com</A>&gt;=20
    wrote:<BR>&gt;<BR>&gt; Thank you Yoshifumi-san,<BR>&gt;<BR>&gt; I =
agree;=20
    it's the combination of SACK signalling (as defined in<BR>&gt; =
RFC2018),=20
    which segments are elegibe for SACK-based retransmission, =
and<BR>&gt;=20
    RFC3517 (how to actually behave during SACK loss =
recovery).<BR>&gt;<BR>&gt;=20
    Unfortunately, neiter of which covers the corner case of loosing =
the<BR>&gt;=20
    last segment(s) of a stream [while already in loss=20
    recovery].<BR>&gt;<BR>&gt; With NewReno, every FullAck below SND.MAX =
causes=20
    the retransmission of<BR>&gt; one segment (after the ACK), before =
continuing=20
    with the next segment<BR>&gt; after SND.MAX...<BR>&gt;<BR>&gt; But =
my=20
    findings so far indicate, that such a FullACK (no SACK =
entries)<BR>&gt;=20
    below SND.MAX doesn't trigger the retransmission of the next=20
    segment.<BR>&gt;<BR>&gt; Obviously, with a continous stream of data, =
this is=20
    not a big issue, as<BR>&gt; eventually you will have a SACK entry =
again,=20
    making the lost, previously<BR>&gt; last, segment elegible for=20
    retransmission under RFC2018/3517.<BR>&gt;<BR>&gt;<BR>&gt;<BR>&gt; =
Would=20
    there be any support by the group for an I-D, which addresses =
this<BR>&gt;=20
    particular corner case?<BR>&gt;<BR>&gt;<BR>&gt; Richard=20
    Scheffenegger<BR>&gt;<BR>&gt;&gt; -----Original =
Message-----<BR>&gt;&gt;=20
    From: Yoshifumi Nishida [mailto:<A=20
    =
href=3D"mailto:nishida@sfc.wide.ad.jp">nishida@sfc.wide.ad.jp</A>]<BR>&gt=
;&gt;=20
    Sent: Freitag, 22. Oktober 2010 04:43<BR>&gt;&gt; To: Scheffenegger, =

    Richard<BR>&gt;&gt; Cc: <A=20
    href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</A><BR>&gt;&gt; Subject: =
Re:=20
    [tcpm] end-of-stream loss recovery (RFC2018 SACK=20
    errata?)<BR>&gt;&gt;<BR>&gt;&gt; Hello =
Richard,<BR>&gt;&gt;<BR>&gt;&gt; I=20
    think this discussion should be based on rfc3517 since =
it<BR>&gt;&gt;=20
    describes a loss recovery algorithm for TCP =
SACK.<BR>&gt;&gt;<BR>&gt;&gt;=20
    Thanks,<BR>&gt;&gt; --<BR>&gt;&gt; Yoshifumi Nishida<BR>&gt;&gt; <A=20
    =
href=3D"mailto:nishida@sfc.wide.ad.jp">nishida@sfc.wide.ad.jp</A><BR>&gt;=
&gt;<BR>&gt;=20
    _______________________________________________<BR>&gt; tcpm mailing =

    list<BR>&gt; <A =
href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</A><BR>&gt; <A=20
    href=3D"https://www.ietf.org/mailman/listinfo/tcpm"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/tcpm</A><BR>&gt;<BR=
>_______________________________________________<BR>tcpm=20
    mailing list<BR><A =
href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</A><BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/tcpm"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/tcpm</A><BR></DIV><=
/DIV></BLOCKQUOTE></DIV><BR></DIV></BLOCKQUOTE></BODY></HTML>

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To: Matt Mathis <mattmathis@google.com>
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Subject: Re: [tcpm] 3517 update w/ SACK-entry draft
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Hi, Matt.

Matt Mathis spake unto us the following wisdom:
> A couple of comments: =A0There is an errata against 2018 about the
> requirement that timeouts MUST clear the score board. =A0 From the
> errata:
>=20
> "At least one OS (Linux) violates the MUST to good effect: Even when time=
out
> driven, it keeps old SACK data so it can avoid retransmitting data alread=
y at
> the receiver. Thus even under severe bandwidth exhaustion, 100% of the da=
ta
> delivered to the receiver causes forward progress and the system is not s=
ubject
> to classical congestion collapse (that is, congestion collapse from
> unnecessarily-retransmitted packets)."

We have discussed this, actually.  We prefer *not* to address it as
part of 3517bis for a number of reasons (my coauthors will hopefully
correct me if I misrepresent anything):

1) 3517 really only addresses the "fast recovery" portion of loss
   recovery, and has little to say about the RTO and what follows.
   The text in Section 5.1 dealing with the RTO is more or less just
   reiterating what 2018 says, and then pointing out some obvious ways
   to deal with incoming SACK data.  Note that it doesn't even specify
   precisely how a sender uses SACK data during slow start, except to
   point out that NextSeg () isn't really appropriate.

2) 3517bis is a *tiny* delta from 3517.  Look at Section 9 to see
   this.  Introducing larger changes at this point is really not
   something we want to see.  We would much rather see larger changes
   standardized separately, their kinks worked out, and then roll them
   int o 3517bis-bis if appropriate.  3517 is subtle, and we really
   feel like we got it Right in a lot of ways.  We do not want to
   introduce bugs in what is supposed to be a straightforward rev to
   improve a single, well-understood point (entry into loss recovery).

3) We do not envision RFC 3517 as being the *only* SACK algorithm out
   there.  In particular, we recognize that FACK is already in use, as
   well as variants of both 3517 and FACK.  We envision 3517 as *one*,
   sane and well-understood, way to deal with SACK information.

> It would be better to update section 5.1 of 3517bis to reflect this.
> Note that the errata introduces several caveats and and additional
> issues that need to be addressed.

Things like these caveats and additional issues are precisely what I
want to avoid!

> Also there is another issue that should be introduced in the security
> considerations section:
>=20
> Traditional recovery mechanism serve to protect the Internet from
> implementers who change congestion control to make it too aggressive,
> because doing so tends to increase the probability of timeouts, and
> often reduces overall throuthput.  This creates a strong disincentive
> to deploying extra aggressive congestion control.
>=20
> As recovery algorithms improve, this disincentive is reduced.  In the
> limit an ideal recovery mechanism might never suffer from a timeout
> except after losing an entire window of data or ACKs.  Such a recovery
> algorithm might create an incentive to deploy arbitrarily aggressive
> congestion control.  The algorithms presented in this document are are
> believed to reduce the negative consequence of making congestion
> control more aggressive than permitted by RFC 5681.

OK; I agree with your point above entirely.  We will have to think
about this with respect to 3517bis.  It is absolutely the case that
smarter loss recovery makes it more feasible to back off aggressively.
I'm not completely sure this is 3517's problem, though.

> --
> On other issue: would this be an appropriate time (and doc) to
> introduce some other ideas from our rate halving work from  ages ago?

I would prefer to see this in an additional document, and I think my
coauthors would agree.

Ethan

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Matt Mathis spake unto us the following wisdom:
> I think it would be better to try to get it into 3517bis, but if that
> doesn't work, a separate ID will do.

See my response to the 3517bis thread for my response to this, too.
;-)  I prefer to see this in separate ID.

> My rational:  If the very last segment is lost, the partial ACK for
> the preceding segment will not include a SACK block because the
> receiver mistakenly thinks the data is complete.  I would characterize
> this situation as leaving SACK recovery prematurely, which why it
> should be in scope for 3517bis.

I largely agree with this characterization.  There may be some
potential for confusion if the segment immediately preceding the
incoming cumulative ACK was retransmitted, where loss recovery is
indeed complete and the next segment *should not* be retransmitted.
Due to this and other possible issues, I'd much rather see this
hammered out carefully and integrated later, while 3517bis is pushed
through and finished in the meantime.

It sounds to me like there's room for a document (or maybe two
documents) which says:

1) There's a corner case where a specific loss pattern which involves
   at least two segment losses in a single window of data separated by
   at least three intervening segments causes SACK to perform worse
   than NewReno.

2) There is a related set of corner cases with general losses at the
   end of the window where, under some circumstances, it may be
   beneficial to retransmit a packet a bit "early".

--- possible document break

3) There are better ways to handle SACK data on RTO than flushing it
   blindly.  In particular, Josh Blanton documented as part of his
   Masters thesis a scheme which reliably disambiguates reneging on
   SACK information from other behaviors.


Then, at some later point in time, it may make sense to roll these
algorithms into the RFC 3517 lineage.  In particular, I believe that
points 1) and 2) are more subtle than this thread suggests.  I think
3) is probably more straightforward.

Ethan

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From eblanton@cs.ohiou.edu  Sat Oct 23 06:24:07 2010
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From: Ethan Blanton <eblanton@cs.ohiou.edu>
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Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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Richard, I'm pulling this back to the list, because I suspect you're
not the only reader who didn't follow ... my original example was too
terse.

Scheffenegger, Richard spake unto us the following wisdom:
> Unfortunately, I'm afraid I am not following what you wanted to describe
> here:

It requires three losses, and your change of rules.

> > I largely agree with this characterization.  There may be some
> > potential for confusion if the segment immediately preceding the
> > incoming cumulative ACK was retransmitted, where loss recovery is
> > indeed complete and the next segment *should not* be retransmitted.
> > Due to this and other possible issues, I'd much rather see this
> > hammered out carefully and integrated later, while 3517bis is pushed
> > through and finished in the meantime.

[snip your example]

S1 ... S7 are sent.  S1, S6, and S7 are lost.  The sender retransmits
S1, gets a cumulative ACK for S6, retransmits S6, then gets a
cumulative ACK for S7.  Should it retransmit S7?  Did the original
transmission or the retransmission trigger the cum ACK for S7?
Consider the case were S6 was not lost in the first place, but delayed
in the network for ~1RTT.  Now consider interaction with timestamps,
DSACK, etc. where this potentially *can* be disambiguated.

My point isn't that this is the end of the world, or that it's the
only potential problem, or even that it's a problem worth worrying
about (it might just be a corner case), it's simply that someone needs
to sit down and carefully think about all of the scenarios, and I
don't think this rev of 3517 is the time to do it. It took more than
two years to get 3517 published, and (for which I take some blame)
poor Ilpo and Markku have been sitting on their sack-entry draft for
many months now already.  This change to 3517 is small, easily
understood, has been understood and "approved" by many involved
parties since the time that 3517 was originally published, and
represents a real benefit.

I would very much prefer to see you explore the options for this
end-of-window loss scenario, including situations where it fails and
where the retransmission should be stifled (if any).  I think you'll
find that it's *not* as easy as looking for a cumulative ACK for the
highest SACKed octet + 1.  (Or, rather, it may be that it *is* that
simple for the *first* retransmission, but possibly not later
retransmissions.)

Ethan

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Inline

On Fri, Oct 22, 2010 at 9:30 PM, Ethan Blanton <eblanton@cs.ohiou.edu> wrot=
e:
> Hi, Matt.
>
> Matt Mathis spake unto us the following wisdom:
>> A couple of comments: =A0There is an errata against 2018 about the
>> requirement that timeouts MUST clear the score board. =A0 From the
>> errata:
>>
>> "At least one OS (Linux) violates the MUST to good effect: Even when tim=
eout
>> driven, it keeps old SACK data so it can avoid retransmitting data alrea=
dy at
>> the receiver. Thus even under severe bandwidth exhaustion, 100% of the d=
ata
>> delivered to the receiver causes forward progress and the system is not =
subject
>> to classical congestion collapse (that is, congestion collapse from
>> unnecessarily-retransmitted packets)."
>
> We have discussed this, actually. =A0We prefer *not* to address it as
> part of 3517bis for a number of reasons (my coauthors will hopefully
> correct me if I misrepresent anything):

Fair enough, but then please don't strengthen the incorrect language
in 2018 by citing or repeating it, especially without mentioning the
errata.  There isn't a trivial fix here, because the real problem is
that section 5.1 conflates two separate issues: what to do on a
timeout, and what to do when clearing the scoreboard due to reneging
or other events (such as timeouts as incorrectly required by 2018).
I think these should be two separate sections.   I am not  aware of
any existing description of how to handle a timeout w/o clearing the
scoreboard.

The current text clashes badly with existing, well functioning
implementations.  Insisting on keeping it will hurt the credibility of
the entire document.

>> Also there is another issue that should be introduced in the security
>> considerations section:
>>
>> Traditional recovery mechanism serve to protect the Internet from
>> implementers who change congestion control to make it too aggressive,
>> because doing so tends to increase the probability of timeouts, and
>> often reduces overall throuthput. =A0This creates a strong disincentive
>> to deploying extra aggressive congestion control.
>>
>> As recovery algorithms improve, this disincentive is reduced. =A0In the
>> limit an ideal recovery mechanism might never suffer from a timeout
>> except after losing an entire window of data or ACKs. =A0Such a recovery
>> algorithm might create an incentive to deploy arbitrarily aggressive
>> congestion control. =A0The algorithms presented in this document are are
>> believed to reduce the negative consequence of making congestion
>> control more aggressive than permitted by RFC 5681.
>
> OK; I agree with your point above entirely. =A0We will have to think
> about this with respect to 3517bis. =A0It is absolutely the case that
> smarter loss recovery makes it more feasible to back off aggressively.
> I'm not completely sure this is 3517's problem, though.

It is in scope for 3517bis, because this issue is entirely about Fast
Recovery strengthening TCP's self clock during recovery, which I
believe is an explicitly stated goal of the algorithm.

For the record, I think more aggressive CC is a good thing. Part of my
motivation for wanting this language here is to set the stage for some
future more aggressive CC algorithms.  This language explicitly
connects the future evolution in congestion control to the current
evolution in recovery algorithms.   Although as I reread it, I may
have suggested text that is just a bit too strong.

Thanks,
--MM--

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Thread-Topic: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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References: <DE08556E79FD1649AEE892A8C353EF611348C9477C@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <A1FD783C-6EDB-41D4-A0DE-9C77A486395D@windriver.com> <5254FC29-AD81-4B83-8AC9-7C03A7F4E74C@comsys.rwth-aachen.de> <49576D00-CF67-4A91-A00A-9B6CD37EEA46@windriver.com> <5FDC413D5FA246468C200652D63E627A0AC76DED@LDCMVEXC1-PRD.hq.netapp.com> <AANLkTincZy08rQq-fCP5PrLQn5JeooZFkq=mLjJ4zibg@mail.gmail.com> <5FDC413D5FA246468C200652D63E627A0B02D269@LDCMVEXC1-PRD.hq.netapp.com> <AANLkTi=5gRLuyqeFiN-es8oK4K0=cg=uHNpRCLdz+cLB@mail.gmail.com> <20101023014200.GD12426@colt> <5FDC413D5FA246468C200652D63E627A0B02D296@LDCMVEXC1-PRD.hq.netapp.com> <20101023132527.GE12426@colt>
From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Ethan Blanton" <eblanton@cs.ohiou.edu>
X-OriginalArrivalTime: 23 Oct 2010 18:31:31.0826 (UTC) FILETIME=[83B74D20:01CB72E0]
Cc: tcpm@ietf.org
Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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Thank you Ethan!



I'll skip the discussion about whether it should be in 3517bis or a
seperate document.



Obviously, if S6 + S7 are lost, the sender should retransmit, right?
(But the sender and receiver lack the information to be 100% sure about
that).=20

If S6 (or S7 for that matter) are delayed by more than 1 RTT, RTO would
also trigger the resending of those frames - potentially prematurely
[assuming RTT > RTO(min)].=20
This scheme would only cause the sending of one of the segments earlier,
than RTO may have done.

If either one is delayed by less than RTT, an ACK with a SACK entry
would move SND.FACK forward, preventing undue retransmissions in time.

Some goes if we are not at the end of stream, and new (S8) segments are
injected that are not delayed.


Regarding the timing/ordering of the retransmissions: A cumulative ACK
(with no SACK entries) should trigger the retransmission of only a
single further packet presumed lost at the end of stream. In your
example, the sender could fill the pipe by immedately sending S6 + S7. A
conservate sender should however still wait a little more for S7 to
arrive, if it was heavy delayed by the network. Thank you for bringing
up that point.

That way, the sender is allowing for 1 RTT reordering delay for S6, but
2 RTT reordering delay for S7 (and so forth). Assuming, of course, RTT <
RTO(min). If RTT is larger than RTO(min), and reordering delay larger
than 1 RTT, RTO will trigger retransmission of segments sooner than the
cumulative ACK for S6 could cause sack loss recovery to retransmit under
that scheme. Since this would also happen currently, we end up no worse
than without the modification.


And yes, I'm quite sure that there will be corner cases, where simply
using cumulative ACK for the highest SACKed octet + 1 may cause
superfluous retransmits. I use superfluous here in order to discreminate
from spurious retransmits, which are preceeded by an undue entering of
loss recovery. Here we are always in loss recovery.

(If I may, being interested primarily in low-latency, high bandwidth
environment where boundary synchronized traffic is encountered, the
timely recovery (few RTT) is of more criticality than absolute
conservation of packets. There are other means in TCP to address
environments, where heavy reordering is dominant, and I would argue that
once loss recovery started, this is out of scope.)


I think the above arguments can be interpreted such that

o) If RTT > RTO(min), and reordering delay >=3D RTT:
	No change in the way the sender behaves (all these delayed
segments may end up retransmitted unnecessarily, under both schemes, due
to RTO)

o) If RTT < RTO(min), and reordering delay between RTT and RTO(min):
	Some few superfluous retransmits can happen. This is bounded,
however, by (cwnd-4)*7/12 segments, if my calculations yesterday were
correct; with delayed ACK enabled, the upper bound can be as low as
(cwnd-4)*3/12, even under pathological worst case scenarios (such as
LIFO reordering and exact timing).

o) If reordering delay < RTT:
	The sender will not see a cumulative ACK without SACK entries,
thus not prematurely retransmit delayed segments.

If there is indeed loss, and RTT < RTO(min), full recovery will takes
n*RTT with n the number of segments lost prior to end-of-stream. Even
that "lengthy" recovery time will typically beat RTO by orders of
magnitude in local networks. If more then (cwnd-3)/2 is lost, the ACK
clock will eventually stall, and only a RTO + slow start can get the
flow going again.=20

Obviously, the above statements are not as thorough as you probably
would like them to be. I tried to flesh out a scenario yesterday, which
would result in the *most* superfluous retransmissions possible -
excluding additional information such as DSACK, delayed ACKs,
Timestamps, but taking the reduction of the congestion window into
account (and the variants of rate halving, that seem to be running in
the lastest BSD and Linux stacks).

Best regards,




Richard Scheffenegger
=20

> -----Original Message-----
> From: Ethan Blanton [mailto:eblanton@cs.ohiou.edu]=20
> Sent: Samstag, 23. Oktober 2010 09:25
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
>=20
> Richard, I'm pulling this back to the list, because I suspect=20
> you're not the only reader who didn't follow ... my original=20
> example was too terse.
>=20
> Scheffenegger, Richard spake unto us the following wisdom:
> > Unfortunately, I'm afraid I am not following what you wanted to=20
> > describe
> > here:
>=20
> It requires three losses, and your change of rules.
>=20
> > > I largely agree with this characterization.  There may be some=20
> > > potential for confusion if the segment immediately preceding the=20
> > > incoming cumulative ACK was retransmitted, where loss recovery is=20
> > > indeed complete and the next segment *should not* be=20
> retransmitted.
> > > Due to this and other possible issues, I'd much rather see this=20
> > > hammered out carefully and integrated later, while=20
> 3517bis is pushed=20
> > > through and finished in the meantime.
>=20
> [snip your example]
>=20
> S1 ... S7 are sent.  S1, S6, and S7 are lost.  The sender=20
> retransmits S1, gets a cumulative ACK for S6, retransmits S6,=20
> then gets a cumulative ACK for S7.  Should it retransmit S7? =20
> Did the original transmission or the retransmission trigger=20
> the cum ACK for S7?
> Consider the case were S6 was not lost in the first place,=20
> but delayed in the network for ~1RTT.  Now consider=20
> interaction with timestamps, DSACK, etc. where this=20
> potentially *can* be disambiguated.
>=20
> My point isn't that this is the end of the world, or that=20
> it's the only potential problem, or even that it's a problem=20
> worth worrying about (it might just be a corner case), it's=20
> simply that someone needs to sit down and carefully think=20
> about all of the scenarios, and I don't think this rev of=20
> 3517 is the time to do it. It took more than two years to get=20
> 3517 published, and (for which I take some blame) poor Ilpo=20
> and Markku have been sitting on their sack-entry draft for=20
> many months now already.  This change to 3517 is small,=20
> easily understood, has been understood and "approved" by many=20
> involved parties since the time that 3517 was originally=20
> published, and represents a real benefit.
>=20
> I would very much prefer to see you explore the options for=20
> this end-of-window loss scenario, including situations where=20
> it fails and where the retransmission should be stifled (if=20
> any).  I think you'll find that it's *not* as easy as looking=20
> for a cumulative ACK for the highest SACKed octet + 1.  (Or,=20
> rather, it may be that it *is* that simple for the *first*=20
> retransmission, but possibly not later
> retransmissions.)
>=20
> Ethan
>=20

From ananth@cisco.com  Sat Oct 23 19:41:54 2010
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Thread-Topic: Readiness of "draft-ietf-tcpm-persist-00.txt"  for last call.
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From: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
To: <tcpm@ietf.org>
X-OriginalArrivalTime: 24 Oct 2010 02:43:33.0939 (UTC) FILETIME=[40444030:01CB7325]
Subject: [tcpm] Readiness of "draft-ietf-tcpm-persist-00.txt" for last call.
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Hi,
  =20
  Following is the summary (from the authors perspective) as to where we
stand w.r.t this draft. We believe, this short and harmless draft is
ready for the last call.
   =20
 During the last review of the persist draft, some folks were not happy
about the API section (section 7) of the document.  The authors
discussed this concern among themselves and also have received some
other reviews (both internal and external) and are kind of confused as
to why the presence of an API guideline (for illustration purpose
mainly) should be of any concern. Following reasons as to why we think
why some illustration (like API) would be nicer to have :-

- This is an informational draft. The purpose of the API section is to
illustrate what how an application can do something about connections
which are stuck in persist state. The objective is NOT to recommend any
sockets API syntax or format (again we are not standardizing anything),
it is purely there to illustrate the essence of the draft.

- There was some talk about SO_LINGER API etc.,. I think we have
clarified earlier that SO_LINGER comes into play only when the
application tries to close the socket. SO_LINGER with 0 value is like
causing an abort of the TCP connection. Here we acknowledge that there
is a way (possibly more ways) that exists which can be used to abort the
TCP connection. This is orthogonal to the current API discussion which
we are having which simply illustrates how an application can abort the
TCP connection without violating any RFC (793 or 1122)

In a nutshell, we don't see any reason why mentioning an API example is
violation of the RFC. We believe adding an API example would not deter
that simplicity in any way, but that is just our take.

Maybe there is some scope to rewrite that section if folks think
otherwise.

We would like to move forward on getting closure on this aspect as this
seems to be the last hurdle in getting this draft to the last call
process.=20

Sorry about the delay in getting back.

Thanks,
-Anantha

From yoshifumi.nishida@gmail.com  Sat Oct 23 23:39:21 2010
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Subject: Re: [tcpm] end-of-stream loss recovery (RFC2018 SACK errata?)
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2010/10/23 Ethan Blanton <eblanton@cs.ohiou.edu>:

> S1 ... S7 are sent. =A0S1, S6, and S7 are lost. =A0The sender retransmits
> S1, gets a cumulative ACK for S6, retransmits S6, then gets a
> cumulative ACK for S7. =A0Should it retransmit S7? =A0Did the original
> transmission or the retransmission trigger the cum ACK for S7?
> Consider the case were S6 was not lost in the first place, but delayed
> in the network for ~1RTT. =A0Now consider interaction with timestamps,
> DSACK, etc. where this potentially *can* be disambiguated.

In my feeling, this depends on the pipe size.
if cwnd - pipe >=3D 1 SMSS, we can send at least 1 packet.
In this example, S8 will be sent. ( rule (2) in NextSeg() will be applied)
If S8 can arrive at the receiver successfully, the sender can get good
estimation for lost packets by the ACK for S8. Then, it can retransmit
proper packets.

However, as Richard already pointed out, if this is the end of
transmission, there is no S8.
In this case, I think retransmitting S7 will not be harmful so much.

If this is not the end of transmission (e.g. application stall), I'm
not very sure whether we should retransmit a packet.

Thanks,
--
Yoshifumi Nishida
nishida@sfc.wide.ad.jp
