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Subject: [tcpm] I-D Action: draft-ietf-tcpm-cubic-03.txt
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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 of the IETF.

        Title           : CUBIC for Fast Long-Distance Networks
        Authors         : Injong Rhee
                          Lisong Xu
                          Sangtae Ha
                          Alexander Zimmermann
                          Lars Eggert
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-cubic-03.txt
	Pages           : 15
	Date            : 2016-12-02

Abstract:
   CUBIC is an extension to the current TCP standards.  The protocol
   differs from the current TCP standards only in the congestion window
   adjustment function in the sender side.  In particular, it uses a
   cubic function instead of a linear window increase of the current TCP
   standards to improve scalability and stability under fast and long
   distance networks.  BIC-TCP, a predecessor of CUBIC, has been a
   default TCP adopted by Linux since year 2005 and has already been
   deployed globally and in use for several years by the Internet
   community at large.  CUBIC is using a similar window growth function
   as BIC-TCP and is designed to be less aggressive and fairer to TCP in
   bandwidth usage than BIC-TCP while maintaining the strengths of BIC-
   TCP such as stability, window scalability and RTT fairness.  Through
   extensive testing in various Internet scenarios, we believe that
   CUBIC is safe for deployment and testing in the global Internet.  The
   intent of this document is to provide the protocol specification of
   CUBIC for a third party implementation and solicit the community
   feedback through experimentation on the performance of CUBIC.


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

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

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-cubic-03


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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Subject: [tcpm] A new Internet draft for TCP Cubic
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Greetings,

We just submitted a new Internet draft for TCP Cubic available at the 
following link.

https://tools.ietf.org/html/draft-ietf-tcpm-cubic-03

In this new version, we made the following changes based on the comments 
received on tcpm-dubic@ietf.org in September and October 2016.

* Updated the contact information of two authors

* Section 1: Briefly explain that the throughput of a TCP is equal to 
"approximately, window/RTT". As there was some confusion about the 
window size and throughput of TCP.

* Section 3.1:
    + Add explanation of W_cubic(0)=W_max*beta_cubic
    + Add explanation about RTT "where RTT is the weighed average RTT 
calculated by the standard TCP"

* Section 3.2: Add a new paragraph to explain intuitively about the TCP 
friendly region. "Standard TCP performs well in certain types of 
networks, for example, under short RTT and small bandwidth (or small 
BDP) networks. In these networks, we use the TCP-friendly region to 
ensure that CUBIC achieves at least the same throughput as the standard 
TCP."

* Sections 3.3 and 3.4: Add "where W_cubic(t+RTT) is calculated using 
Eq. 1".

* Section 3.5: add slow start threshold calculation "ssthresh = cwnd * 
beta_cubic; // new slow start threshold"

* Section 4.3: Remove sentence "It is not designed for wireless 
networks" to avoid some confusion about the types of wireless networks, 
types of problems in wireless networks, and the potential problems of 
CUBIC in wireless networks.

* Section 4.5: changes
   from "In case that there is congestion collapse, CUBIC behaves likely 
standard TCP"
    to "With regard to the potential of causing congestion collapse, 
CUBIC behaves like standard TCP"

* Section 4.8: changes
   from "In cases of idle periods, t in Eq. 1 should not include the 
idle time"
   to "In cases of idle periods, t in Eq. 1 MUST NOT include the idle time"

* Revised the whole draft to clearly distinguish between the loss events 
detected by dup ack and the timeouts.
    + Section 1: from "after a loss event" to "during congestion avoidance"
    + Section 1: from "following a loss event" to "following a loss 
event detected by duplicate ACKs"
    + Section 1: from "since the last loss event" to "since the 
beginning of congestion avoidance"
    + Section 3.1: from "when a packet loss occurs" to "when a packet 
loss (detected by duplicate ACKs) occurs"
    + Section 3.5: from "when a packet loss occurs" to "when a packet 
loss (detected by duplicate ACKs) occurs"
    + Add a new section 3.7 to describe the behavior of CUBIC in 
timeout: "In case of timeout, CUBIC follows the standard TCP to reduce 
cwnd, but sets ssthresh using beta_cubic (same as in Section 3.5)."

Thanks
Lisong


From nobody Fri Dec  2 09:26:30 2016
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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 2 Dec 2016 09:25:43 -0800
Message-ID: <CAK6E8=fYwRsRodeUGya21hwxvaKB+ws3Y5nRjyNrSepXxeBDcw@mail.gmail.com>
To: Lisong Xu <xu@unl.edu>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
Subject: Re: [tcpm] A new Internet draft for TCP Cubic
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Hi Lisong,

The new draft looks much better. For clarity, I suggest the draft
explicitly mentions the interactions with (classic) ECN, even RFC3168
already mandates treating ECN as a loss. If Cubic only
reset the elapsed time "t" only on loss but not (classic) ECN, then even if it
reduces window by beta on ECE ACKs, it'll still grow the window too
rapidly afterwards.

Current draft says:

  The window growth function of CUBIC uses the following function:

       W_cubic(t) = C*(t-K)^3 + W_max (Eq. 1)

   where C is a constant fixed to determine the aggressiveness of window
   growth in high BDP networks, t is the elapsed time from the last
   window reduction (measured right after the fast recovery)
                                ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^


Fortunately, Linux does implement this tricky part correctly. Not sure
about other implementations tho.


On Fri, Dec 2, 2016 at 8:06 AM, Lisong Xu <xu@unl.edu> wrote:
> Greetings,
>
> We just submitted a new Internet draft for TCP Cubic available at the
> following link.
>
> https://tools.ietf.org/html/draft-ietf-tcpm-cubic-03
>
> In this new version, we made the following changes based on the comments
> received on tcpm-dubic@ietf.org in September and October 2016.
>
> * Updated the contact information of two authors
>
> * Section 1: Briefly explain that the throughput of a TCP is equal to
> "approximately, window/RTT". As there was some confusion about the window
> size and throughput of TCP.
>
> * Section 3.1:
>    + Add explanation of W_cubic(0)=W_max*beta_cubic
>    + Add explanation about RTT "where RTT is the weighed average RTT
> calculated by the standard TCP"
>
> * Section 3.2: Add a new paragraph to explain intuitively about the TCP
> friendly region. "Standard TCP performs well in certain types of networks,
> for example, under short RTT and small bandwidth (or small BDP) networks. In
> these networks, we use the TCP-friendly region to ensure that CUBIC achieves
> at least the same throughput as the standard TCP."
>
> * Sections 3.3 and 3.4: Add "where W_cubic(t+RTT) is calculated using Eq.
> 1".
>
> * Section 3.5: add slow start threshold calculation "ssthresh = cwnd *
> beta_cubic; // new slow start threshold"
>
> * Section 4.3: Remove sentence "It is not designed for wireless networks" to
> avoid some confusion about the types of wireless networks, types of problems
> in wireless networks, and the potential problems of CUBIC in wireless
> networks.
>
> * Section 4.5: changes
>   from "In case that there is congestion collapse, CUBIC behaves likely
> standard TCP"
>    to "With regard to the potential of causing congestion collapse, CUBIC
> behaves like standard TCP"
>
> * Section 4.8: changes
>   from "In cases of idle periods, t in Eq. 1 should not include the idle
> time"
>   to "In cases of idle periods, t in Eq. 1 MUST NOT include the idle time"
>
> * Revised the whole draft to clearly distinguish between the loss events
> detected by dup ack and the timeouts.
>    + Section 1: from "after a loss event" to "during congestion avoidance"
>    + Section 1: from "following a loss event" to "following a loss event
> detected by duplicate ACKs"
>    + Section 1: from "since the last loss event" to "since the beginning of
> congestion avoidance"
>    + Section 3.1: from "when a packet loss occurs" to "when a packet loss
> (detected by duplicate ACKs) occurs"
>    + Section 3.5: from "when a packet loss occurs" to "when a packet loss
> (detected by duplicate ACKs) occurs"
>    + Add a new section 3.7 to describe the behavior of CUBIC in timeout: "In
> case of timeout, CUBIC follows the standard TCP to reduce cwnd, but sets
> ssthresh using beta_cubic (same as in Section 3.5)."
>
> Thanks
> Lisong
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Dec  2 11:21:45 2016
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To: Yuchung Cheng <ycheng@google.com>
References: <7910_1470581922_u77Ewfxw007346_85d0d695-d8f1-e404-d0e5-23b13963bfd9@unl.edu> <e10515b6-d0cb-9fb4-f6cd-86c6bedc8679@unl.edu> <CAK6E8=fYwRsRodeUGya21hwxvaKB+ws3Y5nRjyNrSepXxeBDcw@mail.gmail.com>
From: Lisong Xu <xu@unl.edu>
Organization: University of Nebraska-Lincoln
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Date: Fri, 2 Dec 2016 13:21:37 -0600
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
Subject: Re: [tcpm] A new Internet draft for TCP Cubic
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Thanks, Yuchung. Will change it in the next version.

Lisong


On 12/2/2016 11:25 AM, Yuchung Cheng wrote:
> Hi Lisong,
>
> The new draft looks much better. For clarity, I suggest the draft
> explicitly mentions the interactions with (classic) ECN, even RFC3168
> already mandates treating ECN as a loss. If Cubic only
> reset the elapsed time "t" only on loss but not (classic) ECN, then even if it
> reduces window by beta on ECE ACKs, it'll still grow the window too
> rapidly afterwards.
>
> Current draft says:
>
>    The window growth function of CUBIC uses the following function:
>
>         W_cubic(t) = C*(t-K)^3 + W_max (Eq. 1)
>
>     where C is a constant fixed to determine the aggressiveness of window
>     growth in high BDP networks, t is the elapsed time from the last
>     window reduction (measured right after the fast recovery)
>                                  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>
>
> Fortunately, Linux does implement this tricky part correctly. Not sure
> about other implementations tho.
>
>
> On Fri, Dec 2, 2016 at 8:06 AM, Lisong Xu <xu@unl.edu> wrote:
>> Greetings,
>>
>> We just submitted a new Internet draft for TCP Cubic available at the
>> following link.
>>
>> https://tools.ietf.org/html/draft-ietf-tcpm-cubic-03
>>
>> In this new version, we made the following changes based on the comments
>> received on tcpm-dubic@ietf.org in September and October 2016.
>>
>> * Updated the contact information of two authors
>>
>> * Section 1: Briefly explain that the throughput of a TCP is equal to
>> "approximately, window/RTT". As there was some confusion about the window
>> size and throughput of TCP.
>>
>> * Section 3.1:
>>     + Add explanation of W_cubic(0)=W_max*beta_cubic
>>     + Add explanation about RTT "where RTT is the weighed average RTT
>> calculated by the standard TCP"
>>
>> * Section 3.2: Add a new paragraph to explain intuitively about the TCP
>> friendly region. "Standard TCP performs well in certain types of networks,
>> for example, under short RTT and small bandwidth (or small BDP) networks. In
>> these networks, we use the TCP-friendly region to ensure that CUBIC achieves
>> at least the same throughput as the standard TCP."
>>
>> * Sections 3.3 and 3.4: Add "where W_cubic(t+RTT) is calculated using Eq.
>> 1".
>>
>> * Section 3.5: add slow start threshold calculation "ssthresh = cwnd *
>> beta_cubic; // new slow start threshold"
>>
>> * Section 4.3: Remove sentence "It is not designed for wireless networks" to
>> avoid some confusion about the types of wireless networks, types of problems
>> in wireless networks, and the potential problems of CUBIC in wireless
>> networks.
>>
>> * Section 4.5: changes
>>    from "In case that there is congestion collapse, CUBIC behaves likely
>> standard TCP"
>>     to "With regard to the potential of causing congestion collapse, CUBIC
>> behaves like standard TCP"
>>
>> * Section 4.8: changes
>>    from "In cases of idle periods, t in Eq. 1 should not include the idle
>> time"
>>    to "In cases of idle periods, t in Eq. 1 MUST NOT include the idle time"
>>
>> * Revised the whole draft to clearly distinguish between the loss events
>> detected by dup ack and the timeouts.
>>     + Section 1: from "after a loss event" to "during congestion avoidance"
>>     + Section 1: from "following a loss event" to "following a loss event
>> detected by duplicate ACKs"
>>     + Section 1: from "since the last loss event" to "since the beginning of
>> congestion avoidance"
>>     + Section 3.1: from "when a packet loss occurs" to "when a packet loss
>> (detected by duplicate ACKs) occurs"
>>     + Section 3.5: from "when a packet loss occurs" to "when a packet loss
>> (detected by duplicate ACKs) occurs"
>>     + Add a new section 3.7 to describe the behavior of CUBIC in timeout: "In
>> case of timeout, CUBIC follows the standard TCP to reduce cwnd, but sets
>> ssthresh using beta_cubic (same as in Section 3.5)."
>>
>> Thanks
>> Lisong
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
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To: tsvwg@ietf.org, draft-black-tsvwg-ecn-experimentation@ietf.org,  "Black, David" <David.Black@dell.com>
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Subject: [tcpm] DCTCP and draft-black-tsvwg-ecn-experimentation ?
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Hi,

This very likely has been discussed length and I just missed it (Sorry).

Why is dctcp, https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-03, not
mentioned in this draft ?

Even if it does not fall within the 4ls experiments (not sure about the
answer to this) then it might be useful to add a few lines explaining
the status of DCTCP in this respect ?

Something else entirely then are mechanisms like DCQCN, not specified by
IETF, but relying generally on ECN markings in UDP transport,
though not following neither the classical nor the l4s approach.
Such mechanisms as is, whether they use ECT(0) or ECT(1), remain to be
treated as alians (or plainly as not compliant with standards) at this time
in stage. Correct ?


BR, Karen


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To: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "draft-black-tsvwg-ecn-experimentation@ietf.org" <draft-black-tsvwg-ecn-experimentation@ietf.org>
Thread-Topic: DCTCP and draft-black-tsvwg-ecn-experimentation ?
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Date: Tue, 6 Dec 2016 16:39:02 +0000
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, "Black, David" <David.Black@dell.com>
Subject: Re: [tcpm] DCTCP and draft-black-tsvwg-ecn-experimentation ?
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From nobody Wed Dec  7 06:46:21 2016
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To: taps WG <taps@ietf.org>, tcpm IETF list <tcpm@ietf.org>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/5Yx8rOeijFSUvkSa6oqHdxFPRZs>
Subject: [tcpm] Experimental TCP RFCs affecting the API?
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Hi,

As I'm working on an update of draft-ietf-taps-transports-usage to also =
incorporate Experimental TCP RFCs, I wonder which of them had an impact =
on the API - I mean any form of text describing how applications use the =
protocol.
Clearly, TFO is such a case (RFC 7413). Any others?

Thanks in advance!

Cheers,
Michael


From nobody Thu Dec  8 20:02:43 2016
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-04.txt
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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 of the IETF.

        Title           : Transmission Control Protocol Specification
        Author          : Wesley M. Eddy
	Filename        : draft-ietf-tcpm-rfc793bis-04.txt
	Pages           : 94
	Date            : 2016-12-08

Abstract:
   This document specifies the Internet's Transmission Control Protocol
   (TCP).  TCP is an important transport layer protocol in the Internet
   stack, and has continuously evolved over decades of use and growth of
   the Internet.  Over this time, a number of changes have been made to
   TCP as it was specified in RFC 793, though these have only been
   documented in a piecemeal fashion.  This document collects and brings
   those changes together with the protocol specification from RFC 793.
   This document obsoletes RFC 793 and several other RFCs (TODO: list
   all actual RFCs when finished).

   RFC EDITOR NOTE: If approved for publication as an RFC, this should
   be marked additionally as "STD: 7" and replace RFC 793 in that role.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rfc793bis/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-04

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Thu Dec  8 20:07:37 2016
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From: Wesley Eddy <wes@mti-systems.com>
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Date: Thu, 8 Dec 2016 23:07:26 -0500
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-04.txt
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The main progress in this revision is that I think all of the necessary 
RFC 1122 material is incorporated.  There are a couple of items that the 
WG should provide feedback on, which I'll open separate threads for.


On 12/8/2016 11:02 PM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions of the IETF.
>
>          Title           : Transmission Control Protocol Specification
>          Author          : Wesley M. Eddy
> 	Filename        : draft-ietf-tcpm-rfc793bis-04.txt
> 	Pages           : 94
> 	Date            : 2016-12-08
>
> Abstract:
>     This document specifies the Internet's Transmission Control Protocol
>     (TCP).  TCP is an important transport layer protocol in the Internet
>     stack, and has continuously evolved over decades of use and growth of
>     the Internet.  Over this time, a number of changes have been made to
>     TCP as it was specified in RFC 793, though these have only been
>     documented in a piecemeal fashion.  This document collects and brings
>     those changes together with the protocol specification from RFC 793.
>     This document obsoletes RFC 793 and several other RFCs (TODO: list
>     all actual RFCs when finished).
>
>     RFC EDITOR NOTE: If approved for publication as an RFC, this should
>     be marked additionally as "STD: 7" and replace RFC 793 in that role.
>
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rfc793bis/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-04
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-04
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Dec  9 18:36:38 2016
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To: =?utf-8?B?TWlyamEgS8O8aGxld2luZA==?= <mirja.kuehlewind@tik.ee.ethz.ch>, "tsvwg@ietf.org" <tsvwg@ietf.org>, marcelo bagnulo braun <marcelo@it.uc3m.es>, Bob Briscoe <ietf@bobbriscoe.net>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tsvwg] Review of draft-bagnulo-tsvwg-generalized-ecn-01
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Subject: Re: [tcpm] [tsvwg] Review of draft-bagnulo-tsvwg-generalized-ecn-01
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From nobody Sat Dec 10 01:39:33 2016
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From: =?utf-8?Q?Mirja_K=C3=BChlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
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To: "Black, David" <David.Black@dell.com>
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Subject: Re: [tcpm] [tsvwg] Review of draft-bagnulo-tsvwg-generalized-ecn-01
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Hi David,

yes, I already realized that I reviewed the wrong doc. The right review =
on the tcpm list will follow soon. I set the =E2=80=9Areplaces=E2=80=98 =
relation yesterday after noticing.=20

@all: Please make sure you set this correctly if you upload a draft with =
a new name!

Mirja


> Am 10.12.2016 um 03:36 schrieb Black, David <David.Black@dell.com>:
>=20
> Hi Mirja
> [+tcpm WG]
>=20
> The datatracker records draft-bagnulo-tsvwg-generalized-ecn as =
replaced
> by draft-bagnulo-tcpm-generalized-ecn - the latter draft is expected =
to move
> this work forward in the tcpm WG.
>=20
> Thanks, --David
>=20
>> -----Original Message-----
>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Mirja =
K=C3=BChlewind
>> Sent: Friday, December 09, 2016 11:56 AM
>> To: tsvwg@ietf.org; marcelo bagnulo braun; Bob Briscoe
>> Subject: [tsvwg] Review of draft-bagnulo-tsvwg-generalized-ecn-01
>>=20
>> Hi all,
>>=20
>> find below review comments for draft-bagnulo-tsvwg-generalized-ecn-01 =
as an
>> individual contributor (classified in different categories):
>>=20
>> Comments on Argumentation:
>> Please note that I do support the document and marking of control =
packets
>> given the otherwise negative impact on performance especially if we =
would aim
>> for an all-ECN Internet in future. These additional arguments are =
provided to
>> be added for discussion in the doc for completeness.
>>=20
>> Section 2 (the reliability argument)
>> -----
>> Even though this is not spelled out like this in RFC3168, I would =
like to
>> extend the argument here a little:
>>=20
>> 1) I'm okay to say that if loss is not detected reliably for this =
kind of
>> packet, you may as well not require to detect CE marks. However, =
receiving a
>> CE mark, detecting it, and just no reacting to it, is a different =
thing. I
>> clearly see the trade-off here for TCP but I don't want this to be a
>> statement where other protocols say it's okay to not react to CE =
because TCP
>> is doing this as well for ACKs. Note sure yet how to solve this =
problem and
>> I'm not comfortable with recommending to mark control packets =
whiteout at the
>> same time defining standard mechanisms for TCP how to react to it.
>>=20
>> 2) If a packet is dropped, it's gone and cannot further congest the =
link.
>> This is the mechanism of the network to protect itself. However, if =
you mark
>> a packet as ECN capable, it keeps sitting in the queue and another =
non-ECT
>> packet might be dropped instead. This is a well-known general problem =
of ECN,
>> however, if the ECT-marked is then also unresponsive to CE marking, =
this
>> might make the situation worse.
>>=20
>> Section 3 (TCP SYNs): Argument 3 (DoS attacks)
>> ---
>> Having ECT marked attack traffic makes it worse because this might =
leads to a
>> even higher loss rate for non-ECT traffic while ECT packets keep =
sitting in
>> the queue.
>>=20
>> Section 4 (Pure ACKs): second argument
>> ---
>> The difference it that an CE marking on a ACK can be detected by the =
receiver
>> (even though it not fully clear how to react to it or just ignore
>> it...hopefully not) but you can't detect ACK loss because you don't =
know how
>> many ACKs have been send initially. I don't think this is an issue =
but you say:
>> "If the pure ACK carrying the
>>    ECT and the CE bits set is later dropped by the network, it will =
be
>>    essentially falling back to the use of drop as congestion signal."
>> which seems wrong.
>>=20
>> However, this is on the other had a pro argument because while loss =
cannot be
>> detected at all, there is at least a chance to detect CE marked ACKs.
>>=20
>>=20
>> Comments on the more editorial side:
>> 1) One more argument to make in the intro, especially as you mention =
l4s, is
>> that in this case, having the packets not marked as ECT could even =
lead to
>> differential network treatment, which might influence performance =
even more.
>> 2) I would recommend to move the discussion on data centers in =
section 3 to a
>> separate section at the end of the doc because that's the only time =
you talk
>> about data centers.
>> 3) In section 3 regarding [ecn-pam], you are assuming that the =
packets were
>> dropped the the endpoint/receiver, however, there could also be =
middleboxes
>> doing this. I don't think we looked at RST but that would be =
interesting.
>> 3) section 3 "The responder may drop the SYN (either silently or by =
sending a
>> RST) or may reply with a non ECT marked SYN/ACK.  If it is the =
latter,
>>    then this is a non-issue" -> I wouldn't call it a non-issue =
because at
>> least the congestion signal got lost.
>> 4) I think you need security considerations.
>>=20
>> Nit:
>> section 3, 1. sentence: s/We next describe he arguments exhibited/We =
next
>> describe the arguments exhibited/
>>=20
>> And the important bit at the end:
>>=20
>> One high-level comment:
>> This document is a good read, however, it is not clear what an =
implementor
>> should actually do if he/she want to enable ECN on control packets. =
Instead
>> of only have an informational discussion, I would rather see an =
experiment
>> where you proposed concrete things/machanims that should/must be done
>> when
>> ECN is used on control packets.
>>=20
>> Further, at the end of section 3, you not only discuss potential
>> fallbacks/safety guards on how to use ECN on control packets but you =
actually
>> propose changes to the ECN protocol as specified in RFC3168. This =
part really
>> doesn't seem to be appropriate for an informational doc and also =
would need
>> further discussion as an experiment. As you have to rely for these =
mechanism
>> on changes on both the sender and receiver side, I would simply just =
use
>> AccECN instead. This would mean there should maybe be a =
recommendation that
>> if the initiator wants to set ECT on control packets incl. the SYN it
>> should/must also try to negotiate AccECN in the handshake (which =
anyway as an
>> implict fallback to calssic ECN if AccECN is not supported by the =
receiver).
>>=20
>> Thanks for writing the doc. I think this is a very good starting =
point!
>>=20
>> Mirja
>=20


From nobody Mon Dec 12 12:04:13 2016
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This is a thread to check group consensus on how to discuss RTT 
estimation and RTO computation in RFC 793bis.

RFC 793 originally had some material on this, described as "An Example 
Retransmission Timeout Procedure", but that is replaced by several later 
documents.
- RFC 1122 adds mention of Karn's algorithm plus exponential backoff of 
the RTO, and more strongly adds requirements ("MUST") for these, rather 
than just being an "example".
-  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298 
currently updates 1122 in this regard.

I think 6298 is "not too long" and has clear requirements.  I think 
793bis should simply reference 6298 as the requirement for computation 
of the RTO, rather than trying to actually discuss this as a 
self-contained topic in 793bis.  I think that would be duplicative, 
unnecessary, and confusing.

Do people agree with this?


For reference, the current 793bis text is below.  The first paragraph is 
direct from 793, the fourth from 1122, and the text in-between is 
attempting to say what I described above:

3.8.1.  Retransmission Timeout

    Because of the variability of the networks that compose an
    internetwork system and the wide range of uses of TCP connections the
    retransmission timeout (RTO) must be dynamically determined.

    The RTO MUST be computed according to the algorithm in [7], including
    Karn's algorithm for taking RTT samples.

    RFC 793 contains an early example procedure for computing the RTO.
    This was then replaced by the algorithm described in RFC 1122, and
    subsequently updated in RFC 2988, and then again in RFC 6298.

    If a retransmitted packet is identical to the original packet (which
    implies not only that the data boundaries have not changed, but also
    that the window and acknowledgment fields of the header have not
    changed), then the same IP Identification field MAY be used (see
    Section 3.2.1.5 of RFC 1122).






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From: Jana Iyengar <jri@google.com>
Date: Mon, 12 Dec 2016 12:14:29 -0800
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To: Wesley Eddy <wes@mti-systems.com>
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Cc: tcpm@ietf.org
Subject: Re: [tcpm] 793bis: requirement on RTO computation
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--f403045dd8f04cac8605437bc04f
Content-Type: text/plain; charset=UTF-8

On Mon, Dec 12, 2016 at 12:04 PM, Wesley Eddy <wes@mti-systems.com> wrote:

> This is a thread to check group consensus on how to discuss RTT estimation
> and RTO computation in RFC 793bis.
>
> RFC 793 originally had some material on this, described as "An Example
> Retransmission Timeout Procedure", but that is replaced by several later
> documents.
> - RFC 1122 adds mention of Karn's algorithm plus exponential backoff of
> the RTO, and more strongly adds requirements ("MUST") for these, rather
> than just being an "example".
> -  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298
> currently updates 1122 in this regard.
>
> I think 6298 is "not too long" and has clear requirements.  I think 793bis
> should simply reference 6298 as the requirement for computation of the RTO,
> rather than trying to actually discuss this as a self-contained topic in
> 793bis.  I think that would be duplicative, unnecessary, and confusing.
>
> Do people agree with this?
>

+1. Nobody refers to 793 for RTT/RTO computation anyways.

- jana


> For reference, the current 793bis text is below.  The first paragraph is
> direct from 793, the fourth from 1122, and the text in-between is
> attempting to say what I described above:
>
> 3.8.1.  Retransmission Timeout
>
>    Because of the variability of the networks that compose an
>    internetwork system and the wide range of uses of TCP connections the
>    retransmission timeout (RTO) must be dynamically determined.
>
>    The RTO MUST be computed according to the algorithm in [7], including
>    Karn's algorithm for taking RTT samples.
>
>    RFC 793 contains an early example procedure for computing the RTO.
>    This was then replaced by the algorithm described in RFC 1122, and
>    subsequently updated in RFC 2988, and then again in RFC 6298.
>
>    If a retransmitted packet is identical to the original packet (which
>    implies not only that the data boundaries have not changed, but also
>    that the window and acknowledgment fields of the header have not
>    changed), then the same IP Identification field MAY be used (see
>    Section 3.2.1.5 of RFC 1122).
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--f403045dd8f04cac8605437bc04f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Dec 12, 2016 at 12:04 PM, Wesley Eddy <span dir=3D"ltr">&lt;<a =
href=3D"mailto:wes@mti-systems.com" target=3D"_blank">wes@mti-systems.com</=
a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">This is a thread to =
check group consensus on how to discuss RTT estimation and RTO computation =
in RFC 793bis.<br>
<br>
RFC 793 originally had some material on this, described as &quot;An Example=
 Retransmission Timeout Procedure&quot;, but that is replaced by several la=
ter documents.<br>
- RFC 1122 adds mention of Karn&#39;s algorithm plus exponential backoff of=
 the RTO, and more strongly adds requirements (&quot;MUST&quot;) for these,=
 rather than just being an &quot;example&quot;.<br>
-=C2=A0 RFC 2988 and then 6298 expanded on RTT and RTO computation, and 629=
8 currently updates 1122 in this regard.<br>
<br>
I think 6298 is &quot;not too long&quot; and has clear requirements.=C2=A0 =
I think 793bis should simply reference 6298 as the requirement for computat=
ion of the RTO, rather than trying to actually discuss this as a self-conta=
ined topic in 793bis.=C2=A0 I think that would be duplicative, unnecessary,=
 and confusing.<br>
<br>
Do people agree with this?<br></blockquote><div><br></div><div>+1. Nobody r=
efers to 793 for RTT/RTO computation anyways.</div><div><br></div><div>- ja=
na</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
For reference, the current 793bis text is below.=C2=A0 The first paragraph =
is direct from 793, the fourth from 1122, and the text in-between is attemp=
ting to say what I described above:<br>
<br>
3.8.1.=C2=A0 Retransmission Timeout<br>
<br>
=C2=A0 =C2=A0Because of the variability of the networks that compose an<br>
=C2=A0 =C2=A0internetwork system and the wide range of uses of TCP connecti=
ons the<br>
=C2=A0 =C2=A0retransmission timeout (RTO) must be dynamically determined.<b=
r>
<br>
=C2=A0 =C2=A0The RTO MUST be computed according to the algorithm in [7], in=
cluding<br>
=C2=A0 =C2=A0Karn&#39;s algorithm for taking RTT samples.<br>
<br>
=C2=A0 =C2=A0RFC 793 contains an early example procedure for computing the =
RTO.<br>
=C2=A0 =C2=A0This was then replaced by the algorithm described in RFC 1122,=
 and<br>
=C2=A0 =C2=A0subsequently updated in RFC 2988, and then again in RFC 6298.<=
br>
<br>
=C2=A0 =C2=A0If a retransmitted packet is identical to the original packet =
(which<br>
=C2=A0 =C2=A0implies not only that the data boundaries have not changed, bu=
t also<br>
=C2=A0 =C2=A0that the window and acknowledgment fields of the header have n=
ot<br>
=C2=A0 =C2=A0changed), then the same IP Identification field MAY be used (s=
ee<br>
=C2=A0 =C2=A0Section 3.2.1.5 of RFC 1122).<br>
<br>
<br>
<br>
<br>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
</blockquote></div><br></div></div>

--f403045dd8f04cac8605437bc04f--


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Subject: Re: [tcpm] 793bis: requirement on RTO computation
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On Mon, Dec 12, 2016 at 12:04 PM, Wesley Eddy <wes@mti-systems.com> wrote:
>
> This is a thread to check group consensus on how to discuss RTT estimatio=
n and RTO computation in RFC 793bis.
>
> RFC 793 originally had some material on this, described as "An Example Re=
transmission Timeout Procedure", but that is replaced by several later docu=
ments.
> - RFC 1122 adds mention of Karn's algorithm plus exponential backoff of t=
he RTO, and more strongly adds requirements ("MUST") for these, rather than=
 just being an "example".
> -  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298 c=
urrently updates 1122 in this regard.
>
> I think 6298 is "not too long" and has clear requirements.  I think 793bi=
s should simply reference 6298 as the requirement for computation of the RT=
O, rather than trying to actually discuss this as a self-contained topic in=
 793bis.  I think that would be duplicative, unnecessary, and confusing.
>
> Do people agree with this?
>
>
> For reference, the current 793bis text is below.  The first paragraph is =
direct from 793, the fourth from 1122, and the text in-between is attemptin=
g to say what I described above:
>
> 3.8.1.  Retransmission Timeout
>
>    Because of the variability of the networks that compose an
>    internetwork system and the wide range of uses of TCP connections the
>    retransmission timeout (RTO) must be dynamically determined.
>
>    The RTO MUST be computed according to the algorithm in [7], including
>    Karn's algorithm for taking RTT samples.
>
>    RFC 793 contains an early example procedure for computing the RTO.
>    This was then replaced by the algorithm described in RFC 1122, and
>    subsequently updated in RFC 2988, and then again in RFC 6298.
>
>    If a retransmitted packet is identical to the original packet (which
>    implies not only that the data boundaries have not changed, but also
>    that the window and acknowledgment fields of the header have not
>    changed), then the same IP Identification field MAY be used (see
>    Section 3.2.1.5 of RFC 1122).
What will the new text looks like in the bis?

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


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From: Neal Cardwell <ncardwell@google.com>
Date: Mon, 12 Dec 2016 15:34:45 -0500
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Subject: Re: [tcpm] 793bis: requirement on RTO computation
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On Mon, Dec 12, 2016 at 3:14 PM, Jana Iyengar <jri@google.com> wrote:

>
>
> On Mon, Dec 12, 2016 at 12:04 PM, Wesley Eddy <wes@mti-systems.com> wrote:
>
>> This is a thread to check group consensus on how to discuss RTT
>> estimation and RTO computation in RFC 793bis.
>>
>> RFC 793 originally had some material on this, described as "An Example
>> Retransmission Timeout Procedure", but that is replaced by several later
>> documents.
>> - RFC 1122 adds mention of Karn's algorithm plus exponential backoff of
>> the RTO, and more strongly adds requirements ("MUST") for these, rather
>> than just being an "example".
>> -  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298
>> currently updates 1122 in this regard.
>>
>> I think 6298 is "not too long" and has clear requirements.  I think
>> 793bis should simply reference 6298 as the requirement for computation of
>> the RTO, rather than trying to actually discuss this as a self-contained
>> topic in 793bis.  I think that would be duplicative, unnecessary, and
>> confusing.
>>
>> Do people agree with this?
>>
>
> +1. Nobody refers to 793 for RTT/RTO computation anyways.
>

+1 for just citing 6298, with text along the proposed lines.

neal


>
> - jana
>
>
>> For reference, the current 793bis text is below.  The first paragraph is
>> direct from 793, the fourth from 1122, and the text in-between is
>> attempting to say what I described above:
>>
>> 3.8.1.  Retransmission Timeout
>>
>>    Because of the variability of the networks that compose an
>>    internetwork system and the wide range of uses of TCP connections the
>>    retransmission timeout (RTO) must be dynamically determined.
>>
>>    The RTO MUST be computed according to the algorithm in [7], including
>>    Karn's algorithm for taking RTT samples.
>>
>>    RFC 793 contains an early example procedure for computing the RTO.
>>    This was then replaced by the algorithm described in RFC 1122, and
>>    subsequently updated in RFC 2988, and then again in RFC 6298.
>>
>>    If a retransmitted packet is identical to the original packet (which
>>    implies not only that the data boundaries have not changed, but also
>>    that the window and acknowledgment fields of the header have not
>>    changed), then the same IP Identification field MAY be used (see
>>    Section 3.2.1.5 of RFC 1122).
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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
>
>

--94eb2c04432c8b8f7d05437c0ae8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Dec 12, 2016 at 3:14 PM, Jana Iyengar <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jri@google.com" target=3D"_blank">jri@google.com</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div clas=
s=3D"gmail_extra"><br><div class=3D"gmail_quote"><span class=3D"">On Mon, D=
ec 12, 2016 at 12:04 PM, Wesley Eddy <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:wes@mti-systems.com" target=3D"_blank">wes@mti-systems.com</a>&gt;</span>=
 wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex">This is a thread to check group c=
onsensus on how to discuss RTT estimation and RTO computation in RFC 793bis=
.<br>
<br>
RFC 793 originally had some material on this, described as &quot;An Example=
 Retransmission Timeout Procedure&quot;, but that is replaced by several la=
ter documents.<br>
- RFC 1122 adds mention of Karn&#39;s algorithm plus exponential backoff of=
 the RTO, and more strongly adds requirements (&quot;MUST&quot;) for these,=
 rather than just being an &quot;example&quot;.<br>
-=C2=A0 RFC 2988 and then 6298 expanded on RTT and RTO computation, and 629=
8 currently updates 1122 in this regard.<br>
<br>
I think 6298 is &quot;not too long&quot; and has clear requirements.=C2=A0 =
I think 793bis should simply reference 6298 as the requirement for computat=
ion of the RTO, rather than trying to actually discuss this as a self-conta=
ined topic in 793bis.=C2=A0 I think that would be duplicative, unnecessary,=
 and confusing.<br>
<br>
Do people agree with this?<br></blockquote><div><br></div></span><div>+1. N=
obody refers to 793 for RTT/RTO computation anyways.</div></div></div></div=
></blockquote><div><br></div><div>+1 for just citing 6298, with text along =
the proposed lines.</div><div><br></div><div>neal</div><div>=C2=A0</div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><di=
v class=3D"gmail_quote"><span class=3D"HOEnZb"><font color=3D"#888888"><div=
><br></div><div>- jana</div></font></span><span class=3D""><div>=C2=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">
For reference, the current 793bis text is below.=C2=A0 The first paragraph =
is direct from 793, the fourth from 1122, and the text in-between is attemp=
ting to say what I described above:<br>
<br>
3.8.1.=C2=A0 Retransmission Timeout<br>
<br>
=C2=A0 =C2=A0Because of the variability of the networks that compose an<br>
=C2=A0 =C2=A0internetwork system and the wide range of uses of TCP connecti=
ons the<br>
=C2=A0 =C2=A0retransmission timeout (RTO) must be dynamically determined.<b=
r>
<br>
=C2=A0 =C2=A0The RTO MUST be computed according to the algorithm in [7], in=
cluding<br>
=C2=A0 =C2=A0Karn&#39;s algorithm for taking RTT samples.<br>
<br>
=C2=A0 =C2=A0RFC 793 contains an early example procedure for computing the =
RTO.<br>
=C2=A0 =C2=A0This was then replaced by the algorithm described in RFC 1122,=
 and<br>
=C2=A0 =C2=A0subsequently updated in RFC 2988, and then again in RFC 6298.<=
br>
<br>
=C2=A0 =C2=A0If a retransmitted packet is identical to the original packet =
(which<br>
=C2=A0 =C2=A0implies not only that the data boundaries have not changed, bu=
t also<br>
=C2=A0 =C2=A0that the window and acknowledgment fields of the header have n=
ot<br>
=C2=A0 =C2=A0changed), then the same IP Identification field MAY be used (s=
ee<br>
=C2=A0 =C2=A0Section 3.2.1.5 of RFC 1122).<br>
<br>
<br>
<br>
<br>
<br>
______________________________<wbr>_________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
</blockquote></span></div><br></div></div>
<br>______________________________<wbr>_________________<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" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a><br>
<br></blockquote></div><br></div></div>

--94eb2c04432c8b8f7d05437c0ae8--


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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: Wesley Eddy <wes@mti-systems.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] 793bis: requirement on RTO computation
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Subject: Re: [tcpm] 793bis: requirement on RTO computation
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Hi all,

May I rephrase the question a bit: Should the reference to RFC 6298 be a MU=
ST?

For what it is worth, I can hardly think of any reason why we would want to=
 duplicate text RTO in 793bis, so IMHO the reference wording seems the key =
question.

Michael


> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Wesley Eddy
> Sent: Monday, December 12, 2016 9:04 PM
> To: tcpm@ietf.org
> Subject: [tcpm] 793bis: requirement on RTO computation
>=20
> This is a thread to check group consensus on how to discuss RTT
> estimation and RTO computation in RFC 793bis.
>=20
> RFC 793 originally had some material on this, described as "An Example
> Retransmission Timeout Procedure", but that is replaced by several later
> documents.
> - RFC 1122 adds mention of Karn's algorithm plus exponential backoff of
> the RTO, and more strongly adds requirements ("MUST") for these, rather
> than just being an "example".
> -  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298
> currently updates 1122 in this regard.
>=20
> I think 6298 is "not too long" and has clear requirements.  I think
> 793bis should simply reference 6298 as the requirement for computation
> of the RTO, rather than trying to actually discuss this as a
> self-contained topic in 793bis.  I think that would be duplicative,
> unnecessary, and confusing.
>=20
> Do people agree with this?
>=20
>=20
> For reference, the current 793bis text is below.  The first paragraph is
> direct from 793, the fourth from 1122, and the text in-between is
> attempting to say what I described above:
>=20
> 3.8.1.  Retransmission Timeout
>=20
>     Because of the variability of the networks that compose an
>     internetwork system and the wide range of uses of TCP connections the
>     retransmission timeout (RTO) must be dynamically determined.
>=20
>     The RTO MUST be computed according to the algorithm in [7], including
>     Karn's algorithm for taking RTT samples.
>=20
>     RFC 793 contains an early example procedure for computing the RTO.
>     This was then replaced by the algorithm described in RFC 1122, and
>     subsequently updated in RFC 2988, and then again in RFC 6298.
>=20
>     If a retransmitted packet is identical to the original packet (which
>     implies not only that the data boundaries have not changed, but also
>     that the window and acknowledgment fields of the header have not
>     changed), then the same IP Identification field MAY be used (see
>     Section 3.2.1.5 of RFC 1122).
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] 793bis: requirement on RTO computation
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On Mon, Dec 12, 2016 at 12:41 PM, Scharf, Michael (Nokia - DE)
<michael.scharf@nokia.com> wrote:
> Hi all,
>
> May I rephrase the question a bit: Should the reference to RFC 6298 be a MUST?
>
> For what it is worth, I can hardly think of any reason why we would want to duplicate text RTO in 793bis, so IMHO the reference wording seems the key question.
>
> Michael
>
>
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Wesley Eddy
>> Sent: Monday, December 12, 2016 9:04 PM
>> To: tcpm@ietf.org
>> Subject: [tcpm] 793bis: requirement on RTO computation
>>
>> This is a thread to check group consensus on how to discuss RTT
>> estimation and RTO computation in RFC 793bis.
>>
>> RFC 793 originally had some material on this, described as "An Example
>> Retransmission Timeout Procedure", but that is replaced by several later
>> documents.
>> - RFC 1122 adds mention of Karn's algorithm plus exponential backoff of
>> the RTO, and more strongly adds requirements ("MUST") for these, rather
>> than just being an "example".
>> -  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298
>> currently updates 1122 in this regard.
>>
>> I think 6298 is "not too long" and has clear requirements.  I think
>> 793bis should simply reference 6298 as the requirement for computation
>> of the RTO, rather than trying to actually discuss this as a
>> self-contained topic in 793bis.  I think that would be duplicative,
>> unnecessary, and confusing.
>>
>> Do people agree with this?
+1 but some small suggestions below
>>
>>
>> For reference, the current 793bis text is below.  The first paragraph is
>> direct from 793, the fourth from 1122, and the text in-between is
>> attempting to say what I described above:
>>
>> 3.8.1.  Retransmission Timeout
>>
>>     Because of the variability of the networks that compose an
>>     internetwork system and the wide range of uses of TCP connections the
>>     retransmission timeout (RTO) must be dynamically determined.
>>
>>     The RTO MUST be computed according to the algorithm in [7], including
>>     Karn's algorithm for taking RTT samples.
why not just s/[7]/RFC6298  to save reader a round trip b/c he almost
has to know what that is.

>>
>>     RFC 793 contains an early example procedure for computing the RTO.
>>     This was then replaced by the algorithm described in RFC 1122, and
>>     subsequently updated in RFC 2988, and then again in RFC 6298.
>>
>>     If a retransmitted packet is identical to the original packet (which
>>     implies not only that the data boundaries have not changed, but also
>>     that the window and acknowledgment fields of the header have not
>>     changed), then the same IP Identification field MAY be used (see
>>     Section 3.2.1.5 of RFC 1122).
is this ipid text still useful. The text in RFC1122 seems to suggest
the RFC authors
didn't even believe that re-assembly benefit.

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


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Subject: Re: [tcpm] 793bis: requirement on RTO computation
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--94eb2c190cf0b6285705437e4df9
Content-Type: text/plain; charset=UTF-8

On Mon, Dec 12, 2016 at 12:41 PM, Scharf, Michael (Nokia - DE) <
michael.scharf@nokia.com> wrote:

> Hi all,
>
> May I rephrase the question a bit: Should the reference to RFC 6298 be a
> MUST?
>

Good question. MUST seems too strong for the reference to 6298, and there
are no interoperability issues by doing something different. I wonder what
folks think about saying "SHOULD implement 6298, and MUST adhere to
guidelines in draft-ietf-tcpm-rto-consider".



> For what it is worth, I can hardly think of any reason why we would want
> to duplicate text RTO in 793bis, so IMHO the reference wording seems the
> key question.
>
> Michael
>
>
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Wesley Eddy
> > Sent: Monday, December 12, 2016 9:04 PM
> > To: tcpm@ietf.org
> > Subject: [tcpm] 793bis: requirement on RTO computation
> >
> > This is a thread to check group consensus on how to discuss RTT
> > estimation and RTO computation in RFC 793bis.
> >
> > RFC 793 originally had some material on this, described as "An Example
> > Retransmission Timeout Procedure", but that is replaced by several later
> > documents.
> > - RFC 1122 adds mention of Karn's algorithm plus exponential backoff of
> > the RTO, and more strongly adds requirements ("MUST") for these, rather
> > than just being an "example".
> > -  RFC 2988 and then 6298 expanded on RTT and RTO computation, and 6298
> > currently updates 1122 in this regard.
> >
> > I think 6298 is "not too long" and has clear requirements.  I think
> > 793bis should simply reference 6298 as the requirement for computation
> > of the RTO, rather than trying to actually discuss this as a
> > self-contained topic in 793bis.  I think that would be duplicative,
> > unnecessary, and confusing.
> >
> > Do people agree with this?
> >
> >
> > For reference, the current 793bis text is below.  The first paragraph is
> > direct from 793, the fourth from 1122, and the text in-between is
> > attempting to say what I described above:
> >
> > 3.8.1.  Retransmission Timeout
> >
> >     Because of the variability of the networks that compose an
> >     internetwork system and the wide range of uses of TCP connections the
> >     retransmission timeout (RTO) must be dynamically determined.
> >
> >     The RTO MUST be computed according to the algorithm in [7], including
> >     Karn's algorithm for taking RTT samples.
> >
> >     RFC 793 contains an early example procedure for computing the RTO.
> >     This was then replaced by the algorithm described in RFC 1122, and
> >     subsequently updated in RFC 2988, and then again in RFC 6298.
> >
> >     If a retransmitted packet is identical to the original packet (which
> >     implies not only that the data boundaries have not changed, but also
> >     that the window and acknowledgment fields of the header have not
> >     changed), then the same IP Identification field MAY be used (see
> >     Section 3.2.1.5 of RFC 1122).
> >
> >
> >
> >
> >
> > _______________________________________________
> > 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
>

--94eb2c190cf0b6285705437e4df9
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Dec 12, 2016 at 12:41 PM, Scharf, Michael (Nokia - DE) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:michael.scharf@nokia.com" target=3D"_blank">michael=
.scharf@nokia.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">H=
i all,<br>
<br>
May I rephrase the question a bit: Should the reference to RFC 6298 be a MU=
ST?<br></blockquote><div><br></div><div>Good question. MUST seems too stron=
g for the reference to 6298, and there are no interoperability issues by do=
ing something different. I wonder what folks think about saying &quot;SHOUL=
D implement 6298, and MUST adhere to guidelines in draft-ietf-tcpm-rto-cons=
ider&quot;.</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">
For what it is worth, I can hardly think of any reason why we would want to=
 duplicate text RTO in 793bis, so IMHO the reference wording seems the key =
question.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Michael<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: tcpm [mailto:<a href=3D"mailto:tcpm-bounces@ietf.org">tcpm-bounc=
es@ietf.org</a>] On Behalf Of Wesley Eddy<br>
&gt; Sent: Monday, December 12, 2016 9:04 PM<br>
&gt; To: <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; Subject: [tcpm] 793bis: requirement on RTO computation<br>
&gt;<br>
&gt; This is a thread to check group consensus on how to discuss RTT<br>
&gt; estimation and RTO computation in RFC 793bis.<br>
&gt;<br>
&gt; RFC 793 originally had some material on this, described as &quot;An Ex=
ample<br>
&gt; Retransmission Timeout Procedure&quot;, but that is replaced by severa=
l later<br>
&gt; documents.<br>
&gt; - RFC 1122 adds mention of Karn&#39;s algorithm plus exponential backo=
ff of<br>
&gt; the RTO, and more strongly adds requirements (&quot;MUST&quot;) for th=
ese, rather<br>
&gt; than just being an &quot;example&quot;.<br>
&gt; -=C2=A0 RFC 2988 and then 6298 expanded on RTT and RTO computation, an=
d 6298<br>
&gt; currently updates 1122 in this regard.<br>
&gt;<br>
&gt; I think 6298 is &quot;not too long&quot; and has clear requirements.=
=C2=A0 I think<br>
&gt; 793bis should simply reference 6298 as the requirement for computation=
<br>
&gt; of the RTO, rather than trying to actually discuss this as a<br>
&gt; self-contained topic in 793bis.=C2=A0 I think that would be duplicativ=
e,<br>
&gt; unnecessary, and confusing.<br>
&gt;<br>
&gt; Do people agree with this?<br>
&gt;<br>
&gt;<br>
&gt; For reference, the current 793bis text is below.=C2=A0 The first parag=
raph is<br>
&gt; direct from 793, the fourth from 1122, and the text in-between is<br>
&gt; attempting to say what I described above:<br>
&gt;<br>
&gt; 3.8.1.=C2=A0 Retransmission Timeout<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Because of the variability of the networks that com=
pose an<br>
&gt;=C2=A0 =C2=A0 =C2=A0internetwork system and the wide range of uses of T=
CP connections the<br>
&gt;=C2=A0 =C2=A0 =C2=A0retransmission timeout (RTO) must be dynamically de=
termined.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0The RTO MUST be computed according to the algorithm=
 in [7], including<br>
&gt;=C2=A0 =C2=A0 =C2=A0Karn&#39;s algorithm for taking RTT samples.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0RFC 793 contains an early example procedure for com=
puting the RTO.<br>
&gt;=C2=A0 =C2=A0 =C2=A0This was then replaced by the algorithm described i=
n RFC 1122, and<br>
&gt;=C2=A0 =C2=A0 =C2=A0subsequently updated in RFC 2988, and then again in=
 RFC 6298.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0If a retransmitted packet is identical to the origi=
nal packet (which<br>
&gt;=C2=A0 =C2=A0 =C2=A0implies not only that the data boundaries have not =
changed, but also<br>
&gt;=C2=A0 =C2=A0 =C2=A0that the window and acknowledgment fields of the he=
ader have not<br>
&gt;=C2=A0 =C2=A0 =C2=A0changed), then the same IP Identification field MAY=
 be used (see<br>
&gt;=C2=A0 =C2=A0 =C2=A0Section 3.2.1.5 of RFC 1122).<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; ______________________________<wbr>_________________<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" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a><b=
r>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div>

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From: "zhangyali (D)" <zhangyali369@huawei.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: A question about Delayed ACK and RTO
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Subject: [tcpm] A question about Delayed ACK and RTO
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Hi all,

Recent days, I am doing some simulation about TCP performance in NS3. I fou=
nd a phenomenon is a default setting of TCP's delayed ACK is two packets or=
 200ms. I am wondering if this setting is accord with  some RFCs in IETF. B=
ut After I referred to some RFCs, I just found some restrictions, such as, =
the delay must be less than 0.5ms (RFC1122).

I think the delayed ACK has a close relationship with RTO. Take an extreme =
scenario, if the delay is longer than RTO, many packets will be retransmitt=
ed, which will waste many network resources.

So I want to know do we have some RFCs have given the exact value of both? =
And if we permit any TCP stack to set them freely, what is the mechanism to=
 balance the mismatch between RTO and delayed ACK?


Best Regards,
Yali

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<p class=3D"MsoNormal">Hi all,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP&#8217;s =
delayed ACK is two packets or 200ms. I am wondering if this setting is acco=
rd with &nbsp;some RFCs in IETF. But After I
 referred to some RFCs, I just found some restrictions, such as, the delay =
must be less than 0.5ms (RFC1122).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network resources.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between RTO and delayed ACK?<o=
:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Best Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">Yali<o:p></o:p></p>
</div>
</body>
</html>

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Cc: tcpm@ietf.org
Subject: Re: [tcpm] Ref to timestamp negotiation draft (expired)
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Hi Neal,

did you have time to glance over the timestamp negotiation draft?=20

Here is a brief abstract of some of the properties that the proposed =
negotiation scheme offers. However, at the time, there was too little =
interest in this (and no other vendor to support development efforts) =
thus it expired.

a) no additional TCP Option space useage, particular in the initial SYN

b) Up to 24 bits for arbitrary signalling - the draft uses 16 =
(https://tools.ietf.org/html/draft-scheffenegger-tcpm-timestamp-negotiati=
on-05) to signal the timestamp interval; the value is encoded =
differently from a IEEE floating point representation to allow further =
freedom to indicate relative precision of the injected timestamp (ie. =
envision a NIC capable of hw clocking each segment in a TSO transfer, =
vs. using a coarse grained software clock in a virtual host)

c) generic mechanism to make (parts of) the  timestamp useful for the =
receiver, allowing the sender to weakly authenticate returned timestamps =
(e.g. avoiding the cubic epoch exploit to game a sender by manipulating =
timestamps)

d) In that basic version, 8 bits would still be available to provide the =
signals envisioned by your proposal for indicating Max ACK Delay - =
perhaps using a shift value offset by a constant biad as for the =
timestamp interval, more granualar signaling would be possible than in =
your proposal.


Drawback of the scheme: duing 3WHS, the sender needs to keep state of =
the original TSval (which should be a non-issue for Linux stacks, and is =
manageable for byte-oriented stacks).

Also, a small probability of falsely activating the TS negotiation. =
However, against servers on the internet, this probability is smaller =
than expected as TSval still relates linearly to system uptime, in most =
implementations; I've investigated this during writing intensively (not =
what you could do, though) - see section 5.3=20

Re-reading, I see that my descriptive language  is perhaps not clear =
enough, the signals spoken about in the draft are always exchanged in a =
TSecr field of a TS Option, with the TSecr of the SYN/ACK being the XOR =
sum of the TSval from the SYN, plus the capabilites signal of the =
receiver...

Best regards,
  Richard




Great. Thanks, Bob. We'll take a look at these...

neal

On Wed, Nov 16, 2016 at 9:59 AM, Bob Briscoe <ietf@bobbriscoe.net> =
wrote:

> Neil, [re-sending from correct address, so it is acceptable to the =
tcpm
> list]
>
> As promised in the discussion just now about your timer negotiation =
talk
> in tcpm, here's the previous draft on a similar subject:
> Additional negotiation in the TCP Timestamp Option field during the =
TCP
> handshake
> =
<https://tools.ietf.org/html/draft-scheffenegger-tcpm-timestamp-negotiati=
on-05>
>
> This was originally prompted by the research here (see section 3.1 for =
the
> timestamp discussion), which you might also be interested in:
> Chirping for Congestion Control - Implementation Feasibility
> <http://www.bobbriscoe.net/pubs.html#chirp_impl>
>
>
>
> Bob
>
> PS. I can recommend a good search engine to find stuff like this :)
>
>
> --
> ________________________________________________________________
> Bob Briscoe                               http://bobbriscoe.net/
>
>
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<DIV><FONT size=3D2 face=3DArial>Hi Neal,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>did you have time to glance over the =
timestamp=20
negotiation draft? </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>Here is a brief abstract of some of the =
properties=20
that the proposed negotiation scheme offers. However, at the time, there =
was too=20
little interest in this (and no other vendor to support development =
efforts)=20
thus it expired.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>a) no additional TCP Option space =
useage,=20
particular in the initial SYN</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>b) Up to 24 bits for arbitrary =
signalling - the=20
draft uses 16 (<A=20
href=3D"https://tools.ietf.org/html/draft-scheffenegger-tcpm-timestamp-ne=
gotiation-05">https://tools.ietf.org/html/draft-scheffenegger-tcpm-timest=
amp-negotiation-05</A>)=20
to signal the timestamp interval; the value is encoded differently from =
a IEEE=20
floating point representation to allow further freedom to indicate =
relative=20
precision of the injected timestamp (ie. envision a NIC capable of hw =
clocking=20
each segment in a TSO transfer, vs. using a coarse grained software =
clock in a=20
virtual host)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>c) generic mechanism to make (parts of) =
the&nbsp;=20
timestamp useful for the receiver, allowing the sender to weakly =
authenticate=20
returned timestamps (e.g. avoiding the cubic epoch exploit to game a =
sender by=20
manipulating timestamps)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>d) In that basic version, 8 bits would =
still be=20
available to provide the signals envisioned by your proposal for =
indicating Max=20
ACK Delay - perhaps using a shift value offset by a constant biad as for =
the=20
timestamp interval, more granualar signaling would be possible than in =
your=20
proposal.</FONT></DIV>
<DIV>&nbsp;</DIV><FONT size=3D2 face=3DArial>
<DIV><BR>Drawback of the scheme: duing 3WHS, the sender needs to keep =
state of=20
the original TSval (which should be a non-issue for Linux stacks, and is =

manageable for byte-oriented stacks).</DIV>
<DIV>&nbsp;</DIV>
<DIV>Also, a small probability of falsely activating the TS negotiation. =

However, against servers on the internet, this probability is smaller =
than=20
expected as TSval still relates linearly to system uptime, in most=20
implementations; I've investigated this during writing intensively (not =
what you=20
could do, though) - see section 5.3 </DIV>
<DIV>&nbsp;</DIV>
<DIV>Re-reading, I see that my descriptive language&nbsp;&nbsp;is =
perhaps not=20
clear enough, the signals spoken about in the draft are always exchanged =
in a=20
TSecr field of a TS Option, with the TSecr of the SYN/ACK being the XOR =
sum of=20
the TSval from the SYN, plus the capabilites signal of the =
receiver...</DIV>
<DIV>&nbsp;</DIV>
<DIV>Best regards,<BR>&nbsp; Richard</FONT></DIV>
<DIV><FONT size=3D2 face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>Great. Thanks, Bob. We'll take a look =
at=20
these...</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>neal</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>On Wed, Nov 16, 2016 at 9:59 AM, Bob =
Briscoe &lt;<A=20
href=3D"mailto:ietf@bobbriscoe.net">ietf@bobbriscoe.net</A>&gt;=20
wrote:</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>&gt; Neil, [re-sending from correct =
address, so it=20
is acceptable to the tcpm<BR>&gt; list]<BR>&gt;<BR>&gt; As promised in =
the=20
discussion just now about your timer negotiation talk<BR>&gt; in tcpm, =
here's=20
the previous draft on a similar subject:<BR>&gt; Additional negotiation =
in the=20
TCP Timestamp Option field during the TCP<BR>&gt; handshake<BR>&gt; =
&lt;<A=20
href=3D"https://tools.ietf.org/html/draft-scheffenegger-tcpm-timestamp-ne=
gotiation-05">https://tools.ietf.org/html/draft-scheffenegger-tcpm-timest=
amp-negotiation-05</A>&gt;<BR>&gt;<BR>&gt;=20
This was originally prompted by the research here (see section 3.1 for=20
the<BR>&gt; timestamp discussion), which you might also be interested=20
in:<BR>&gt; Chirping for Congestion Control - Implementation =
Feasibility<BR>&gt;=20
&lt;<A=20
href=3D"http://www.bobbriscoe.net/pubs.html#chirp_impl">http://www.bobbri=
scoe.net/pubs.html#chirp_impl</A>&gt;<BR>&gt;<BR>&gt;<BR>&gt;<BR>&gt;=20
Bob<BR>&gt;<BR>&gt; PS. I can recommend a good search engine to find =
stuff like=20
this :)<BR>&gt;<BR>&gt;<BR>&gt; --<BR>&gt;=20
________________________________________________________________<BR>&gt; =
Bob=20
Briscoe&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<A=20
href=3D"http://bobbriscoe.net/">http://bobbriscoe.net/</A><BR>&gt;<BR>&gt=
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From: Neal Cardwell <ncardwell@google.com>
Date: Tue, 13 Dec 2016 10:17:51 -0500
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Subject: Re: [tcpm] A question about Delayed ACK and RTO
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On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com>
wrote:

> Hi all,
>
>
>
> Recent days, I am doing some simulation about TCP performance in NS3. I
> found a phenomenon is a default setting of TCP=E2=80=99s delayed ACK is t=
wo packets
> or 200ms. I am wondering if this setting is accord with  some RFCs in IET=
F.
> But After I referred to some RFCs, I just found some restrictions, such a=
s,
> the delay must be less than 0.5ms (RFC1122).
>
>
I believe the 200ms figure is due to historical reasons. AFAIK it's due to
the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume 2",
section 25.4 and figure 25.7, which describes the 200ms delayed ACK timer).
Then this value was inherited by other widely-deployed OSes.



> I think the delayed ACK has a close relationship with RTO. Take an extrem=
e
> scenario, if the delay is longer than RTO, many packets will be
> retransmitted, which will waste many network resources.
>

Yes, exactly. In theory, the RTO tries to be adaptive enough to measure any
RTT variations caused by delayed ACKs, and increase the RTO in response to
this. However, it's quite easy for a bulk transfer to never have any
delayed ACKs for most of its lifetime, during which the RTO gradually
converges toward the raw RTT value. Then when there is suddenly a delayed
ACK, there can be a spurious RTO.

To avoid that, many TCP stacks (at least the major open source OSes) have a
hard-coded 200ms "slop factor" or "fudge factor", to try to never let their
estimate of RTT variation fall below that 200ms value, to avoid this
effect.


> So I want to know do we have some RFCs have given the exact value of both=
?
> And if we permit any TCP stack to set them freely, what is the mechanism =
to
> balance the mismatch between RTO and delayed ACK?
>

I'm not aware of RFC specifications for exact values of both (delayed ACK
and RTO). However, the historical precedent is very strong, and the 200ms
delayed ACK value was very pronounced in Internet traces at least as
recently as 2011, when I last looked at the effect. (Probably others have
more recent data points for the prevalence of 200ms delayed ACKs.)

At IETF 97 our team at Google presented some features we use for internal
TCP traffic at Google, where the endpoints can negotiate the specific
constant to use for the maximum delayed ACK from the receiver and the
corresponding minimum RTT delay variation for budgeting in the RTO at the
sender:


https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options-for-l=
ow-latency-00.pdf

As this slide deck notes, within Google we negotiate 5ms for delayed ACKs.

cheers,
neal

--001a113d2a781f720705438bbb8e
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <span dir=3D"ltr">&lt;<a href=3D=
"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@huawei.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">





<div lang=3D"EN-US">
<div class=3D"gmail-m_5004409303414270934WordSection1">
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP=E2=80=99=
s delayed ACK is two packets or 200ms. I am wondering if this setting is ac=
cord with =C2=A0some RFCs in IETF. But After I
 referred to some RFCs, I just found some restrictions, such as, the delay =
must be less than 0.5ms (RFC1122).<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u></p></div></div></blockquote><div><br></div><=
div>I believe the 200ms figure is due to historical reasons. AFAIK it&#39;s=
 due to the BSD delayed ACK behavior (see Stevens &quot;TCP/IP Illustrated =
Volume 2&quot;, section 25.4 and figure 25.7, which describes the 200ms del=
ayed ACK timer). Then this value was inherited by other widely-deployed OSe=
s.</div><div><br></div><div>=C2=A0=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,20=
4);padding-left:1ex"><div lang=3D"EN-US"><div class=3D"gmail-m_500440930341=
4270934WordSection1"><p class=3D"MsoNormal"><u></u></p>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network resources.</p></di=
v></div></blockquote><div><br></div><div>Yes, exactly. In theory, the RTO t=
ries to be adaptive enough to measure any RTT variations caused by delayed =
ACKs, and increase the RTO in response to this. However, it&#39;s quite eas=
y for a bulk transfer to never have any delayed ACKs for most of its lifeti=
me, during which the RTO gradually converges toward the raw RTT value. Then=
 when there is suddenly a delayed ACK, there can be a spurious RTO.</div><d=
iv><br></div><div>To avoid that, many TCP stacks (at least the major open s=
ource OSes) have a hard-coded 200ms &quot;slop factor&quot; or &quot;fudge =
factor&quot;, to try to never let their estimate of RTT variation fall belo=
w that 200ms value, to avoid this effect.=C2=A0</div><div>=C2=A0<br></div><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft:1px solid rgb(204,204,204);padding-left:1ex"><div lang=3D"EN-US"><div c=
lass=3D"gmail-m_5004409303414270934WordSection1"><p class=3D"MsoNormal"><u>=
</u></p>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between RTO and delayed ACK?</=
p></div></div></blockquote><div><br></div><div>I&#39;m not aware of RFC spe=
cifications for exact values of both (delayed ACK and RTO). However, the hi=
storical precedent is very strong, and the 200ms delayed ACK value was very=
 pronounced in Internet traces at least as recently as 2011, when I last lo=
oked at the effect. (Probably others have more recent data points for the p=
revalence of 200ms delayed ACKs.)</div><div><br></div><div>At IETF 97 our t=
eam at Google presented some features we use for internal TCP traffic at Go=
ogle, where the endpoints can negotiate the specific constant to use for th=
e maximum delayed ACK from the receiver and the corresponding minimum RTT d=
elay variation for budgeting in the RTO at the sender:</div><div><br></div>=
<div>=C2=A0 <a href=3D"https://www.ietf.org/proceedings/97/slides/slides-97=
-tcpm-tcp-options-for-low-latency-00.pdf">https://www.ietf.org/proceedings/=
97/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf</a><br></div><d=
iv><br></div><div>As this slide deck notes, within Google we negotiate 5ms =
for delayed ACKs.</div><div><br></div><div>cheers,=C2=A0</div><div>neal</di=
v><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div lan=
g=3D"EN-US"><div class=3D"gmail-m_5004409303414270934WordSection1"><p class=
=3D"MsoNormal"><u></u></p></div></div></blockquote></div></div></div>

--001a113d2a781f720705438bbb8e--


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I don't disagree with anything Neal said, but want to emphasize
something.=20

>     I think the delayed ACK has a close relationship with RTO. Take
>     an extreme scenario, if the delay is longer than RTO, many
>     packets will be retransmitted, which will waste many network
>     resources.
>=20
> Yes, exactly. In theory, the RTO tries to be adaptive enough to
> measure any RTT variations caused by delayed ACKs, and increase the
> RTO in response to this.

In other words, the delayed ACK timer and the RTO are not
competing.  The delayed ACK timer should be a subset of the RTO.

That said, ...

> However, it's quite easy for a bulk transfer to never have any
> delayed ACKs for most of its lifetime, during which the RTO
> gradually converges toward the raw RTT value.  Then when there is
> suddenly a delayed ACK, there can be a spurious RTO.

... this is certainly possible.  And, especially so as folks have
moved towards trying to make the RTO converge to the RTT (e.g., by
not reseting the RTO on ACK reception or not using a minRTO or the
like).  The less variance the RTO accounts for the more a spike in
delay (from various places) will fool the RTO.  Many moons ago we
showed there is a fundamental tradeoff between RTO timeliness and
RTO accuracy.  You can optimize for one of those, but it will come
at the expense of the other.

allman




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From: "zhangyali (D)" <zhangyali369@huawei.com>
To: Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] A question about Delayed ACK and RTO
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Y2xhc3M9Ik1zb05vcm1hbCI+PG86cD4mbmJzcDs8L286cD48L3A+DQo8L2Rpdj4NCjwvZGl2Pg0K
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--_000_A747A0713F56294D8FBE33E5C6B8F5815F545E28DGGEMA502MBSchi_--


From nobody Wed Dec 14 05:24:18 2016
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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com>
From: Neal Cardwell <ncardwell@google.com>
Date: Wed, 14 Dec 2016 08:23:41 -0500
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiAgQSBxdWVzdGlvbiBhYm91dCBEZWxheWVkIEFD?= =?utf-8?q?K_and_RTO?=
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On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <zhangyali369@huawei.com>
wrote:

> Hi Neal,
>
>
>
> Thanks for your providing request information.
>
>
>
> About the delay of delayed ACK, I found another clue in a SIGCOMM paper i=
n
> 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps senders were tal=
king to
> 4.3BSD receivers which would delay an ack until 35% of the window was
> filled or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on
> average).=E2=80=9D There is no reference about the 200 ms, so I guess thi=
s is the
> particular delay appears for the first time.
>
> I try to calculate the value based on delayed packet numbers, packet size
> and sending rate. In this case, the delayed packet number is 7, the packe=
t
> size is 576Byte (refer to RFC879 in 1983), and sundering rate is 4.5Bps.
> The value is 896 ms! Longer than 200 ms.
>
>
>
> >. However, it's quite easy for a bulk transfer to never have any delayed
> ACKs for most of its lifetime, during which the RTO gradually converges
> toward the raw RTT value. Then when there is suddenly a delayed ACK, ther=
e
> can be a spurious RTO.
>
> I am not very sure if I see your point. I think the key point is the RTT
> variation will be weakened when packets are sent in a burst (the
> consequence of delayed ACK).   And the spurious  could only happen for th=
e
> last few packets whose number is smaller than delayed packets, right?
>

Yes, there should only be a delayed ACK at the end of an application chunk
if there is an odd number of packets. But we would expect roughly half of
application chunks to have an odd number of packets. Though the proportion
is probably higher than that, since many application chunks are just one
packet (e.g. an HTTP or RPC request or response).



> About your proposal about negotiation of delayed ACK, a potential problem
> is that RTO will be stretched than before because you add extra delay. As
> you have said, most of the packets will not exceed RTO even host enable
> delayed ACK for most large flows, but the retransmission will be delayed
> (i.e., 5ms)also once one packet is lost.
>

The basic idea of the proposal is to tweak the RTO calculation and turn a
previously existing, historically motivated 200ms fixed "slop factor" into
a dynamically negotiated 5ms "slop factor". In our experience that is
almost always a win.

neal


>
>
> Best,
>
> Yali
>
> *=E5=8F=91=E4=BB=B6=E4=BA=BA**:* Neal Cardwell [mailto:ncardwell@google.c=
om]
> *=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4:* 2016=E5=B9=B412=E6=9C=8813=E6=97=
=A5 23:18
> *=E6=94=B6=E4=BB=B6=E4=BA=BA:* zhangyali (D) <zhangyali369@huawei.com>
> *=E6=8A=84=E9=80=81:* tcpm@ietf.org
> *=E4=B8=BB=E9=A2=98:* Re: [tcpm] A question about Delayed ACK and RTO
>
>
>
> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com>
> wrote:
>
> Hi all,
>
>
>
> Recent days, I am doing some simulation about TCP performance in NS3. I
> found a phenomenon is a default setting of TCP=E2=80=99s delayed ACK is t=
wo packets
> or 200ms. I am wondering if this setting is accord with  some RFCs in IET=
F.
> But After I referred to some RFCs, I just found some restrictions, such a=
s,
> the delay must be less than 0.5ms (RFC1122).
>
>
>
> I believe the 200ms figure is due to historical reasons. AFAIK it's due t=
o
> the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume 2",
> section 25.4 and figure 25.7, which describes the 200ms delayed ACK timer=
).
> Then this value was inherited by other widely-deployed OSes.
>
>
>
>
>
> I think the delayed ACK has a close relationship with RTO. Take an extrem=
e
> scenario, if the delay is longer than RTO, many packets will be
> retransmitted, which will waste many network resources.
>
>
>
> Yes, exactly. In theory, the RTO tries to be adaptive enough to measure
> any RTT variations caused by delayed ACKs, and increase the RTO in respon=
se
> to this. However, it's quite easy for a bulk transfer to never have any
> delayed ACKs for most of its lifetime, during which the RTO gradually
> converges toward the raw RTT value. Then when there is suddenly a delayed
> ACK, there can be a spurious RTO.
>
>
>
> To avoid that, many TCP stacks (at least the major open source OSes) have
> a hard-coded 200ms "slop factor" or "fudge factor", to try to never let
> their estimate of RTT variation fall below that 200ms value, to avoid thi=
s
> effect.
>
>
>
> So I want to know do we have some RFCs have given the exact value of both=
?
> And if we permit any TCP stack to set them freely, what is the mechanism =
to
> balance the mismatch between RTO and delayed ACK?
>
>
>
> I'm not aware of RFC specifications for exact values of both (delayed ACK
> and RTO). However, the historical precedent is very strong, and the 200ms
> delayed ACK value was very pronounced in Internet traces at least as
> recently as 2011, when I last looked at the effect. (Probably others have
> more recent data points for the prevalence of 200ms delayed ACKs.)
>
>
>
> At IETF 97 our team at Google presented some features we use for internal
> TCP traffic at Google, where the endpoints can negotiate the specific
> constant to use for the maximum delayed ACK from the receiver and the
> corresponding minimum RTT delay variation for budgeting in the RTO at the
> sender:
>
>
>
>   https://www.ietf.org/proceedings/97/slides/slides-
> 97-tcpm-tcp-options-for-low-latency-00.pdf
>
>
>
> As this slide deck notes, within Google we negotiate 5ms for delayed ACKs=
.
>
>
>
> cheers,
>
> neal
>
>
>

--001a1135cefca612ca05439e4032
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <span dir=3D"ltr">&lt;<a href=3D=
"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@huawei.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_-5765726088193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">Hi Neal,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">Thanks for your providing request information.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">About the delay of delayed ACK, I found another clue i=
n a SIGCOMM paper in 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KB=
ps senders were talking to 4.3BSD receivers which would
 delay an ack until 35% of the window was filled or 200 ms had passed (i.e.=
, an ack was delayed for 5-7 packets on average).=E2=80=9D There is no refe=
rence about the 200 ms, so I guess this is the particular delay appears for=
 the first time.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">I try to calculate the value based on delayed packet n=
umbers, packet size and sending rate. In this case, the delayed packet numb=
er is 7, the packet size is 576Byte (refer to
 RFC879 in 1983), and sundering rate is 4.5Bps. The value is 896 ms! Longer=
 than 200 ms.
<u></u><u></u></span></p><span class=3D"">
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">&gt;</span>. However, it&#39;s quite easy for a bulk t=
ransfer to never have any delayed ACKs for most of its lifetime, during whi=
ch the RTO gradually converges toward the raw RTT value.
 Then when there is suddenly a delayed ACK, there can be a spurious RTO.<u>=
</u><u></u></p>
</span><p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot=
;,sans-serif;color:#1f497d">I am not very sure if I see your point. I think=
 the key point is the RTT variation will be weakened when packets are sent =
in a burst (the consequence of delayed ACK). =C2=A0=C2=A0And the
 spurious =C2=A0could only happen for the last few packets whose number is =
smaller than delayed packets, right?</span></p></div></div></blockquote><di=
v><br></div><div>Yes, there should only be a delayed ACK at the end of an a=
pplication chunk if there is an odd number of packets. But we would expect =
roughly half of application chunks to have an odd number of packets. Though=
 the proportion is probably higher than that, since many application chunks=
 are just one packet (e.g. an HTTP or RPC request or response).</div><div><=
br></div><div>=C2=A0<span style=3D"color:rgb(31,73,125);font-family:Calibri=
,sans-serif;font-size:11pt">=C2=A0</span></div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"m_-5765=
726088193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">About your proposal about negotiation=
 of delayed ACK, a potential problem is that RTO will be stretched than bef=
ore because you add extra delay. As you have said,
 most of the packets will not exceed RTO even host enable delayed ACK for m=
ost large flows, but the retransmission will be delayed (i.e., 5ms)also onc=
e one packet is lost.</span></p></div></div></blockquote><div><br></div><di=
v>The basic idea of the proposal is to tweak the RTO calculation and turn a=
 previously existing, historically motivated 200ms fixed &quot;slop factor&=
quot; into a dynamically negotiated 5ms &quot;slop factor&quot;. In our exp=
erience that is almost always a win.</div><div><br></div><div>neal</div><di=
v>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"bl=
ue" vlink=3D"purple"><div class=3D"m_-5765726088193305192WordSection1"><p c=
lass=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Calibr=
i&quot;,sans-serif;color:#1f497d">
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Best,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Yali<u></u><u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"ZH-CN" style=3D"font-size:11.0pt;fo=
nt-family:&quot;\005fae\008f6f\0096c5\009ed1&quot;,sans-serif">=E5=8F=91=E4=
=BB=B6=E4=BA=BA</span></b><b><span style=3D"font-size:11.0pt;font-family:&q=
uot;\005fae\008f6f\0096c5\009ed1&quot;,sans-serif">:</span></b><span style=
=3D"font-size:11.0pt;font-family:&quot;\005fae\008f6f\0096c5\009ed1&quot;,s=
ans-serif"> Neal Cardwell
 [mailto:<a href=3D"mailto:ncardwell@google.com" target=3D"_blank">ncardwel=
l@google.com</a>] <br>
<b><span lang=3D"ZH-CN">=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4</span>:</b> 20=
16<span lang=3D"ZH-CN">=E5=B9=B4</span>12<span lang=3D"ZH-CN">=E6=9C=88</sp=
an>13<span lang=3D"ZH-CN">=E6=97=A5</span> 23:18<br>
<b><span lang=3D"ZH-CN">=E6=94=B6=E4=BB=B6=E4=BA=BA</span>:</b> zhangyali (=
D) &lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangya=
li369@huawei.com</a>&gt;<br>
<b><span lang=3D"ZH-CN">=E6=8A=84=E9=80=81</span>:</b> <a href=3D"mailto:tc=
pm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<b><span lang=3D"ZH-CN">=E4=B8=BB=E9=A2=98</span>:</b> Re: [tcpm] A questio=
n about Delayed ACK and RTO<u></u><u></u></span></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) &lt;<=
a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@hu=
awei.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP=E2=80=99=
s delayed ACK is two packets or 200ms. I am wondering if this
 setting is accord with =C2=A0some RFCs in IETF. But After I referred to so=
me RFCs, I just found some restrictions, such as, the delay must be less th=
an 0.5ms (RFC1122).<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I believe the 200ms figure is due to historical reas=
ons. AFAIK it&#39;s due to the BSD delayed ACK behavior (see Stevens &quot;=
TCP/IP Illustrated Volume 2&quot;, section 25.4 and figure 25.7, which desc=
ribes the 200ms delayed ACK timer). Then this value
 was inherited by other widely-deployed OSes.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network
 resources.<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Yes, exactly. In theory, the RTO tries to be adaptiv=
e enough to measure any RTT variations caused by delayed ACKs, and increase=
 the RTO in response to this. However, it&#39;s quite easy for a bulk trans=
fer to never have any delayed ACKs for
 most of its lifetime, during which the RTO gradually converges toward the =
raw RTT value. Then when there is suddenly a delayed ACK, there can be a sp=
urious RTO.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To avoid that, many TCP stacks (at least the major o=
pen source OSes) have a hard-coded 200ms &quot;slop factor&quot; or &quot;f=
udge factor&quot;, to try to never let their estimate of RTT variation fall=
 below that 200ms value, to avoid this effect.=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between
 RTO and delayed ACK?<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I&#39;m not aware of RFC specifications for exact va=
lues of both (delayed ACK and RTO). However, the historical precedent is ve=
ry strong, and the 200ms delayed ACK value was very pronounced in Internet =
traces at least as recently as 2011, when
 I last looked at the effect. (Probably others have more recent data points=
 for the prevalence of 200ms delayed ACKs.)<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">At IETF 97 our team at Google presented some feature=
s we use for internal TCP traffic at Google, where the endpoints can negoti=
ate the specific constant to use for the maximum delayed ACK from the recei=
ver and the corresponding minimum
 RTT delay variation for budgeting in the RTO at the sender:<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0 <a href=3D"https://www.ietf.org/proceedings/9=
7/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf" target=3D"_blan=
k">
https://www.ietf.org/<wbr>proceedings/97/slides/slides-<wbr>97-tcpm-tcp-opt=
ions-for-low-<wbr>latency-00.pdf</a><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">As this slide deck notes, within Google we negotiate=
 5ms for delayed ACKs.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">cheers,=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">neal<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</div>
</div></div></div>
</div>

</blockquote></div><br></div></div>

--001a1135cefca612ca05439e4032--


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From: Neal Cardwell <ncardwell@google.com>
Date: Wed, 14 Dec 2016 11:42:02 -0500
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To: Richard Scheffenegger <rs.ietf@gmx.at>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Eric Dumazet <edumazet@google.com>
Subject: Re: [tcpm] Ref to timestamp negotiation draft (expired)
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--94eb2c04432c01e59f0543a10665
Content-Type: text/plain; charset=UTF-8

Hi Richard,

Thanks for your note. Unfortunately, I've been busy with other projects
recently (largely BBR) and haven't had time to read your draft on improving
timestamps (draft-scheffenegger-tcpm-timestamp-negotiation-05). It looks
like a very nice, thorough draft. So, rest assured that I will read it
before our team embarks on any draft related to the timestamp code we are
running within Google (which was written by Eric Dumazet, cc-ed).

thanks,
neal


On Tue, Dec 13, 2016 at 4:06 AM, Richard Scheffenegger <rs.ietf@gmx.at>
wrote:

> Hi Neal,
>
> did you have time to glance over the timestamp negotiation draft?
>
> Here is a brief abstract of some of the properties that the proposed
> negotiation scheme offers. However, at the time, there was too little
> interest in this (and no other vendor to support development efforts) thus
> it expired.
>
> a) no additional TCP Option space useage, particular in the initial SYN
>
> b) Up to 24 bits for arbitrary signalling - the draft uses 16 (
> https://tools.ietf.org/html/draft-scheffenegger-tcpm-
> timestamp-negotiation-05) to signal the timestamp interval; the value is
> encoded differently from a IEEE floating point representation to allow
> further freedom to indicate relative precision of the injected timestamp
> (ie. envision a NIC capable of hw clocking each segment in a TSO transfer,
> vs. using a coarse grained software clock in a virtual host)
>
> c) generic mechanism to make (parts of) the  timestamp useful for the
> receiver, allowing the sender to weakly authenticate returned timestamps
> (e.g. avoiding the cubic epoch exploit to game a sender by manipulating
> timestamps)
>
> d) In that basic version, 8 bits would still be available to provide the
> signals envisioned by your proposal for indicating Max ACK Delay - perhaps
> using a shift value offset by a constant biad as for the timestamp
> interval, more granualar signaling would be possible than in your proposal.
>
>
> Drawback of the scheme: duing 3WHS, the sender needs to keep state of the
> original TSval (which should be a non-issue for Linux stacks, and is
> manageable for byte-oriented stacks).
>
> Also, a small probability of falsely activating the TS negotiation.
> However, against servers on the internet, this probability is smaller than
> expected as TSval still relates linearly to system uptime, in most
> implementations; I've investigated this during writing intensively (not
> what you could do, though) - see section 5.3
>
> Re-reading, I see that my descriptive language  is perhaps not clear
> enough, the signals spoken about in the draft are always exchanged in a
> TSecr field of a TS Option, with the TSecr of the SYN/ACK being the XOR sum
> of the TSval from the SYN, plus the capabilites signal of the receiver...
>
> Best regards,
>   Richard
>
>
>
>
> Great. Thanks, Bob. We'll take a look at these...
>
> neal
>
> On Wed, Nov 16, 2016 at 9:59 AM, Bob Briscoe <ietf@bobbriscoe.net> wrote:
>
> > Neil, [re-sending from correct address, so it is acceptable to the tcpm
> > list]
> >
> > As promised in the discussion just now about your timer negotiation talk
> > in tcpm, here's the previous draft on a similar subject:
> > Additional negotiation in the TCP Timestamp Option field during the TCP
> > handshake
> > <https://tools.ietf.org/html/draft-scheffenegger-tcpm-
> timestamp-negotiation-05>
> >
> > This was originally prompted by the research here (see section 3.1 for
> the
> > timestamp discussion), which you might also be interested in:
> > Chirping for Congestion Control - Implementation Feasibility
> > <http://www.bobbriscoe.net/pubs.html#chirp_impl>
> >
> >
> >
> > Bob
> >
> > PS. I can recommend a good search engine to find stuff like this :)
> >
> >
> > --
> > ________________________________________________________________
> > Bob Briscoe                               http://bobbriscoe.net/
> >
> >
>

--94eb2c04432c01e59f0543a10665
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Richard,<div><br>Thanks for your note. Unfortunately, I=
&#39;ve been busy with other projects recently (largely BBR) and haven&#39;=
t had time to read your draft on improving timestamps (draft-scheffenegger-=
tcpm-timestamp-negotiation-05). It looks like a very nice, thorough draft. =
So, rest assured that I will read it before our team embarks on any draft r=
elated to the timestamp code we are running within Google (which was writte=
n by Eric Dumazet, cc-ed).</div><div><br></div><div>thanks,</div><div>neal<=
/div><div><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote">On Tue, Dec 13, 2016 at 4:06 AM, Richard Scheffenegger <span dir=
=3D"ltr">&lt;<a href=3D"mailto:rs.ietf@gmx.at" target=3D"_blank">rs.ietf@gm=
x.at</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>





<div bgcolor=3D"#ffffff">
<div><font size=3D"2" face=3D"Arial">Hi Neal,</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">did you have time to glance over the t=
imestamp=20
negotiation draft? </font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">Here is a brief abstract of some of th=
e properties=20
that the proposed negotiation scheme offers. However, at the time, there wa=
s too=20
little interest in this (and no other vendor to support development efforts=
)=20
thus it expired.</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">a) no additional TCP Option space usea=
ge,=20
particular in the initial SYN</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">b) Up to 24 bits for arbitrary signall=
ing - the=20
draft uses 16 (<a href=3D"https://tools.ietf.org/html/draft-scheffenegger-t=
cpm-timestamp-negotiation-05" target=3D"_blank">https://tools.ietf.org/html=
/<wbr>draft-scheffenegger-tcpm-<wbr>timestamp-negotiation-05</a>)=20
to signal the timestamp interval; the value is encoded differently from a I=
EEE=20
floating point representation to allow further freedom to indicate relative=
=20
precision of the injected timestamp (ie. envision a NIC capable of hw clock=
ing=20
each segment in a TSO transfer, vs. using a coarse grained software clock i=
n a=20
virtual host)</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">c) generic mechanism to make (parts of=
) the=C2=A0=20
timestamp useful for the receiver, allowing the sender to weakly authentica=
te=20
returned timestamps (e.g. avoiding the cubic epoch exploit to game a sender=
 by=20
manipulating timestamps)</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">d) In that basic version, 8 bits would=
 still be=20
available to provide the signals envisioned by your proposal for indicating=
 Max=20
ACK Delay - perhaps using a shift value offset by a constant biad as for th=
e=20
timestamp interval, more granualar signaling would be possible than in your=
=20
proposal.</font></div>
<div>=C2=A0</div><font size=3D"2" face=3D"Arial">
<div><br>Drawback of the scheme: duing 3WHS, the sender needs to keep state=
 of=20
the original TSval (which should be a non-issue for Linux stacks, and is=20
manageable for byte-oriented stacks).</div>
<div>=C2=A0</div>
<div>Also, a small probability of falsely activating the TS negotiation.=20
However, against servers on the internet, this probability is smaller than=
=20
expected as TSval still relates linearly to system uptime, in most=20
implementations; I&#39;ve investigated this during writing intensively (not=
 what you=20
could do, though) - see section 5.3 </div>
<div>=C2=A0</div>
<div>Re-reading, I see that my descriptive language=C2=A0=C2=A0is perhaps n=
ot=20
clear enough, the signals spoken about in the draft are always exchanged in=
 a=20
TSecr field of a TS Option, with the TSecr of the SYN/ACK being the XOR sum=
 of=20
the TSval from the SYN, plus the capabilites signal of the receiver...</div=
>
<div>=C2=A0</div>
<div>Best regards,<br>=C2=A0 Richard</div></font></div>
<div><font size=3D"2" face=3D"Arial"></font>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial"></font>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial"></font>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial"></font>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">Great. Thanks, Bob. We&#39;ll take a l=
ook at=20
these...</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">neal</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">On Wed, Nov 16, 2016 at 9:59 AM, Bob B=
riscoe &lt;<a href=3D"mailto:ietf@bobbriscoe.net" target=3D"_blank">ietf@bo=
bbriscoe.net</a>&gt;=20
wrote:</font></div>
<div>=C2=A0</div>
<div><font size=3D"2" face=3D"Arial">&gt; Neil, [re-sending from correct ad=
dress, so it=20
is acceptable to the tcpm<br>&gt; list]<br>&gt;<br>&gt; As promised in the=
=20
discussion just now about your timer negotiation talk<br>&gt; in tcpm, here=
&#39;s=20
the previous draft on a similar subject:<br>&gt; Additional negotiation in =
the=20
TCP Timestamp Option field during the TCP<br>&gt; handshake<br>&gt; &lt;<a =
href=3D"https://tools.ietf.org/html/draft-scheffenegger-tcpm-timestamp-nego=
tiation-05" target=3D"_blank">https://tools.ietf.org/html/<wbr>draft-scheff=
enegger-tcpm-<wbr>timestamp-negotiation-05</a>&gt;<br>&gt;<br>&gt;=20
This was originally prompted by the research here (see section 3.1 for=20
the<br>&gt; timestamp discussion), which you might also be interested=20
in:<br>&gt; Chirping for Congestion Control - Implementation Feasibility<br=
>&gt;=20
&lt;<a href=3D"http://www.bobbriscoe.net/pubs.html#chirp_impl" target=3D"_b=
lank">http://www.bobbriscoe.net/<wbr>pubs.html#chirp_impl</a>&gt;<br>&gt;<b=
r>&gt;<br>&gt;<br>&gt;=20
Bob<br>&gt;<br>&gt; PS. I can recommend a good search engine to find stuff =
like=20
this :)<br>&gt;<span class=3D"HOEnZb"><font color=3D"#888888"><br>&gt;<br>&=
gt; --<br>&gt;=20
______________________________<wbr>______________________________<wbr>____<=
br>&gt; Bob=20
Briscoe=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0<wb=
r>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=20
<a href=3D"http://bobbriscoe.net/" target=3D"_blank">http://bobbriscoe.net/=
</a><br>&gt;<br>&gt;</font></span></font></div>
</blockquote></div><br></div>

--94eb2c04432c01e59f0543a10665--


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From: "Jakob Heitz (jheitz)" <jheitz@cisco.com>
To: Neal Cardwell <ncardwell@google.com>
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<div>Historically, the minimum RTO is 1 second and actual RTT is very rarel=
y more than 1 second, so all this RTT calculation hardly ever matters anywa=
y.<br>
<br>
Thanks,<br>
<div>Jakob.</div>
<div><br>
</div>
</div>
<div><br>
On Dec 14, 2016, at 5:24 AM, Neal Cardwell &lt;<a href=3D"mailto:ncardwell@=
google.com">ncardwell@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali3=
69@huawei.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">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_-5765726088193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">Hi Neal,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">Thanks for your providing request information.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">About the delay of delayed ACK, I found another clue i=
n a SIGCOMM paper in 1988. In the page 14, a sentence is =A1=A7The 4.5KBps =
senders were talking to 4.3BSD receivers which would
 delay an ack until 35% of the window was filled or 200 ms had passed (i.e.=
, an ack was delayed for 5-7 packets on average).=A1=A8 There is no referen=
ce about the 200 ms, so I guess this is the particular delay appears for th=
e first time.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">I try to calculate the value based on delayed packet n=
umbers, packet size and sending rate. In this case, the delayed packet numb=
er is 7, the packet size is 576Byte (refer to
 RFC879 in 1983), and sundering rate is 4.5Bps. The value is 896 ms! Longer=
 than 200 ms.
<u></u><u></u></span></p>
<span class=3D"">
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">&gt;</span>. However, it's quite easy for a bulk trans=
fer to never have any delayed ACKs for most of its lifetime, during which t=
he RTO gradually converges toward the raw RTT value.
 Then when there is suddenly a delayed ACK, there can be a spurious RTO.<u>=
</u><u></u></p>
</span>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">I am not very sure if I see your point. I think the ke=
y point is the RTT variation will be weakened when packets are sent in a bu=
rst (the consequence of delayed ACK). &nbsp;&nbsp;And the
 spurious &nbsp;could only happen for the last few packets whose number is =
smaller than delayed packets, right?</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Yes, there should only be a delayed ACK at the end of an application c=
hunk if there is an odd number of packets. But we would expect roughly half=
 of application chunks to have an odd number of packets. Though the proport=
ion is probably higher than that,
 since many application chunks are just one packet (e.g. an HTTP or RPC req=
uest or response).</div>
<div><br>
</div>
<div>&nbsp;<span style=3D"color:rgb(31,73,125);font-family:Calibri,sans-ser=
if;font-size:11pt">&nbsp;</span></div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_-5765726088193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">About your proposal about negotiation=
 of delayed ACK, a potential problem is that RTO will be stretched than bef=
ore because you add extra delay. As you have said,
 most of the packets will not exceed RTO even host enable delayed ACK for m=
ost large flows, but the retransmission will be delayed (i.e., 5ms)also onc=
e one packet is lost.</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>The basic idea of the proposal is to tweak the RTO calculation and tur=
n a previously existing, historically motivated 200ms fixed &quot;slop fact=
or&quot; into a dynamically negotiated 5ms &quot;slop factor&quot;. In our =
experience that is almost always a win.</div>
<div><br>
</div>
<div>neal</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_-5765726088193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Best,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Yali<u></u><u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"ZH-CN" style=3D"font-size:11.0pt;fo=
nt-family:&quot;\005fae\008f6f\0096c5\009ed1&quot;,sans-serif">&#21457;=A5=
=F3=A4H</span></b><b><span style=3D"font-size:11.0pt;font-family:&quot;\005=
fae\008f6f\0096c5\009ed1&quot;,sans-serif">:</span></b><span style=3D"font-=
size:11.0pt;font-family:&quot;\005fae\008f6f\0096c5\009ed1&quot;,sans-serif=
">
 Neal Cardwell [mailto:<a href=3D"mailto:ncardwell@google.com" target=3D"_b=
lank">ncardwell@google.com</a>]
<br>
<b><span lang=3D"ZH-CN">&#21457;=B0e&#26102;&#38388;</span>:</b> 2016<span =
lang=3D"ZH-CN">=A6~</span>12<span lang=3D"ZH-CN">=A4=EB</span>13<span lang=
=3D"ZH-CN">=A4=E9</span> 23:18<br>
<b><span lang=3D"ZH-CN">=A6=AC=A5=F3=A4H</span>:</b> zhangyali (D) &lt;<a h=
ref=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@huawe=
i.com</a>&gt;<br>
<b><span lang=3D"ZH-CN">=A7=DB=B0e</span>:</b> <a href=3D"mailto:tcpm@ietf.=
org" target=3D"_blank">
tcpm@ietf.org</a><br>
<b><span lang=3D"ZH-CN">=A5D&#39064;</span>:</b> Re: [tcpm] A question abou=
t Delayed ACK and RTO<u></u><u></u></span></p>
<div>
<div class=3D"h5">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) &lt;<=
a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@hu=
awei.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP=A1=A6s d=
elayed ACK is two packets or 200ms. I am wondering if this setting is accor=
d with &nbsp;some RFCs in IETF. But After I
 referred to some RFCs, I just found some restrictions, such as, the delay =
must be less than 0.5ms (RFC1122).<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I believe the 200ms figure is due to historical reas=
ons. AFAIK it's due to the BSD delayed ACK behavior (see Stevens &quot;TCP/=
IP Illustrated Volume 2&quot;, section 25.4 and figure 25.7, which describe=
s the 200ms delayed ACK timer). Then this value
 was inherited by other widely-deployed OSes.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;&nbsp;<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network resources.<u></u><=
u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Yes, exactly. In theory, the RTO tries to be adaptiv=
e enough to measure any RTT variations caused by delayed ACKs, and increase=
 the RTO in response to this. However, it's quite easy for a bulk transfer =
to never have any delayed ACKs for
 most of its lifetime, during which the RTO gradually converges toward the =
raw RTT value. Then when there is suddenly a delayed ACK, there can be a sp=
urious RTO.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To avoid that, many TCP stacks (at least the major o=
pen source OSes) have a hard-coded 200ms &quot;slop factor&quot; or &quot;f=
udge factor&quot;, to try to never let their estimate of RTT variation fall=
 below that 200ms value, to avoid this effect.&nbsp;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between RTO and delayed ACK?<u=
></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I'm not aware of RFC specifications for exact values=
 of both (delayed ACK and RTO). However, the historical precedent is very s=
trong, and the 200ms delayed ACK value was very pronounced in Internet trac=
es at least as recently as 2011, when
 I last looked at the effect. (Probably others have more recent data points=
 for the prevalence of 200ms delayed ACKs.)<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">At IETF 97 our team at Google presented some feature=
s we use for internal TCP traffic at Google, where the endpoints can negoti=
ate the specific constant to use for the maximum delayed ACK from the recei=
ver and the corresponding minimum
 RTT delay variation for budgeting in the RTO at the sender:<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; <a href=3D"https://www.ietf.org/proceedings/9=
7/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf" target=3D"_blan=
k">
https://www.ietf.org/<wbr>proceedings/97/slides/slides-<wbr>97-tcpm-tcp-opt=
ions-for-low-<wbr>latency-00.pdf</a><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">As this slide deck notes, within Google we negotiate=
 5ms for delayed ACKs.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">cheers,&nbsp;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">neal<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>tcpm mailing list</span><br>
<span><a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/tcpm">https://www.ie=
tf.org/mailman/listinfo/tcpm</a></span><br>
</div>
</blockquote>
</body>
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From: Neal Cardwell <ncardwell@google.com>
Date: Wed, 14 Dec 2016 12:04:06 -0500
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On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (jheitz) <jheitz@cisco.com>
wrote:

> Historically, the minimum RTO is 1 second and actual RTT is very rarely
> more than 1 second, so all this RTT calculation hardly ever matters anywa=
y.
>

That may be true historically, but for many years major TCP implementations
(including Linux and FreeBSD) have used a minimum RTO closer to 200ms. And
in datacenter environments even 200ms can be infeasibly high.

neal


>
> Thanks,
> Jakob.
>
>
> On Dec 14, 2016, at 5:24 AM, Neal Cardwell <ncardwell@google.com> wrote:
>
> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <zhangyali369@huawei.com>
> wrote:
>
>> Hi Neal,
>>
>>
>>
>> Thanks for your providing request information.
>>
>>
>>
>> About the delay of delayed ACK, I found another clue in a SIGCOMM paper
>> in 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps senders were=
 talking to
>> 4.3BSD receivers which would delay an ack until 35% of the window was
>> filled or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on
>> average).=E2=80=9D There is no reference about the 200 ms, so I guess th=
is is the
>> particular delay appears for the first time.
>>
>> I try to calculate the value based on delayed packet numbers, packet siz=
e
>> and sending rate. In this case, the delayed packet number is 7, the pack=
et
>> size is 576Byte (refer to RFC879 in 1983), and sundering rate is 4.5Bps.
>> The value is 896 ms! Longer than 200 ms.
>>
>>
>>
>> >. However, it's quite easy for a bulk transfer to never have any
>> delayed ACKs for most of its lifetime, during which the RTO gradually
>> converges toward the raw RTT value. Then when there is suddenly a delaye=
d
>> ACK, there can be a spurious RTO.
>>
>> I am not very sure if I see your point. I think the key point is the RTT
>> variation will be weakened when packets are sent in a burst (the
>> consequence of delayed ACK).   And the spurious  could only happen for t=
he
>> last few packets whose number is smaller than delayed packets, right?
>>
>
> Yes, there should only be a delayed ACK at the end of an application chun=
k
> if there is an odd number of packets. But we would expect roughly half of
> application chunks to have an odd number of packets. Though the proportio=
n
> is probably higher than that, since many application chunks are just one
> packet (e.g. an HTTP or RPC request or response).
>
>
>
>> About your proposal about negotiation of delayed ACK, a potential proble=
m
>> is that RTO will be stretched than before because you add extra delay. A=
s
>> you have said, most of the packets will not exceed RTO even host enable
>> delayed ACK for most large flows, but the retransmission will be delayed
>> (i.e., 5ms)also once one packet is lost.
>>
>
> The basic idea of the proposal is to tweak the RTO calculation and turn a
> previously existing, historically motivated 200ms fixed "slop factor" int=
o
> a dynamically negotiated 5ms "slop factor". In our experience that is
> almost always a win.
>
> neal
>
>
>>
>>
>> Best,
>>
>> Yali
>>
>> *=E5=8F=91=E4=BB=B6=E4=BA=BA**:* Neal Cardwell [mailto:ncardwell@google.=
com]
>> *=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4:* 2016=E5=B9=B412=E6=9C=8813=E6=97=
=A5 23:18
>> *=E6=94=B6=E4=BB=B6=E4=BA=BA:* zhangyali (D) <zhangyali369@huawei.com>
>> *=E6=8A=84=E9=80=81:* tcpm@ietf.org
>> *=E4=B8=BB=E9=A2=98:* Re: [tcpm] A question about Delayed ACK and RTO
>>
>>
>>
>> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com>
>> wrote:
>>
>> Hi all,
>>
>>
>>
>> Recent days, I am doing some simulation about TCP performance in NS3. I
>> found a phenomenon is a default setting of TCP=E2=80=99s delayed ACK is =
two packets
>> or 200ms. I am wondering if this setting is accord with  some RFCs in IE=
TF.
>> But After I referred to some RFCs, I just found some restrictions, such =
as,
>> the delay must be less than 0.5ms (RFC1122).
>>
>>
>>
>> I believe the 200ms figure is due to historical reasons. AFAIK it's due
>> to the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume =
2",
>> section 25.4 and figure 25.7, which describes the 200ms delayed ACK time=
r).
>> Then this value was inherited by other widely-deployed OSes.
>>
>>
>>
>>
>>
>> I think the delayed ACK has a close relationship with RTO. Take an
>> extreme scenario, if the delay is longer than RTO, many packets will be
>> retransmitted, which will waste many network resources.
>>
>>
>>
>> Yes, exactly. In theory, the RTO tries to be adaptive enough to measure
>> any RTT variations caused by delayed ACKs, and increase the RTO in respo=
nse
>> to this. However, it's quite easy for a bulk transfer to never have any
>> delayed ACKs for most of its lifetime, during which the RTO gradually
>> converges toward the raw RTT value. Then when there is suddenly a delaye=
d
>> ACK, there can be a spurious RTO.
>>
>>
>>
>> To avoid that, many TCP stacks (at least the major open source OSes) hav=
e
>> a hard-coded 200ms "slop factor" or "fudge factor", to try to never let
>> their estimate of RTT variation fall below that 200ms value, to avoid th=
is
>> effect.
>>
>>
>>
>> So I want to know do we have some RFCs have given the exact value of
>> both? And if we permit any TCP stack to set them freely, what is the
>> mechanism to balance the mismatch between RTO and delayed ACK?
>>
>>
>>
>> I'm not aware of RFC specifications for exact values of both (delayed AC=
K
>> and RTO). However, the historical precedent is very strong, and the 200m=
s
>> delayed ACK value was very pronounced in Internet traces at least as
>> recently as 2011, when I last looked at the effect. (Probably others hav=
e
>> more recent data points for the prevalence of 200ms delayed ACKs.)
>>
>>
>>
>> At IETF 97 our team at Google presented some features we use for interna=
l
>> TCP traffic at Google, where the endpoints can negotiate the specific
>> constant to use for the maximum delayed ACK from the receiver and the
>> corresponding minimum RTT delay variation for budgeting in the RTO at th=
e
>> sender:
>>
>>
>>
>>   https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-
>> tcp-options-for-low-latency-00.pdf
>>
>>
>>
>> As this slide deck notes, within Google we negotiate 5ms for delayed ACK=
s.
>>
>>
>>
>> cheers,
>>
>> neal
>>
>>
>>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>

--001a1134fbcef152090543a15431
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (jheitz) <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jheitz@cisco.com" target=3D"_blank">jheitz@cisco.com</a>&gt=
;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">



<div dir=3D"auto">
<div>Historically, the minimum RTO is 1 second and actual RTT is very rarel=
y more than 1 second, so all this RTT calculation hardly ever matters anywa=
y.<br></div></div></blockquote><div><br></div>That may be true historically=
, but for many years major TCP implementations (including Linux and FreeBSD=
) have used a minimum RTO closer to 200ms. And in datacenter environments e=
ven 200ms can be infeasibly high.<div><br></div><div>neal</div><div>=C2=A0<=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"auto"><d=
iv>
<br>
Thanks,<br>
<div>Jakob.</div>
<div><br>
</div>
</div><div><div class=3D"gmail-h5">
<div><br>
On Dec 14, 2016, at 5:24 AM, Neal Cardwell &lt;<a href=3D"mailto:ncardwell@=
google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali3=
69@huawei.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"gmail-m_-954577182682849028m_-5765726088193305192WordSection1=
">
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">Hi Neal,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">Thanks for your providing request information.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">About the delay of delayed ACK, I found another clue in a S=
IGCOMM paper in 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps se=
nders were talking to 4.3BSD receivers which would
 delay an ack until 35% of the window was filled or 200 ms had passed (i.e.=
, an ack was delayed for 5-7 packets on average).=E2=80=9D There is no refe=
rence about the 200 ms, so I guess this is the particular delay appears for=
 the first time.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">I try to calculate the value based on delayed packet number=
s, packet size and sending rate. In this case, the delayed packet number is=
 7, the packet size is 576Byte (refer to
 RFC879 in 1983), and sundering rate is 4.5Bps. The value is 896 ms! Longer=
 than 200 ms.
<u></u><u></u></span></p>
<span>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">&gt;</span>. However, it&#39;s quite easy for a bulk transf=
er to never have any delayed ACKs for most of its lifetime, during which th=
e RTO gradually converges toward the raw RTT value.
 Then when there is suddenly a delayed ACK, there can be a spurious RTO.<u>=
</u><u></u></p>
</span>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">I am not very sure if I see your point. I think the key poi=
nt is the RTT variation will be weakened when packets are sent in a burst (=
the consequence of delayed ACK). =C2=A0=C2=A0And the
 spurious =C2=A0could only happen for the last few packets whose number is =
smaller than delayed packets, right?</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Yes, there should only be a delayed ACK at the end of an application c=
hunk if there is an odd number of packets. But we would expect roughly half=
 of application chunks to have an odd number of packets. Though the proport=
ion is probably higher than that,
 since many application chunks are just one packet (e.g. an HTTP or RPC req=
uest or response).</div>
<div><br>
</div>
<div>=C2=A0<span style=3D"color:rgb(31,73,125);font-family:calibri,sans-ser=
if;font-size:11pt">=C2=A0</span></div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"gmail-m_-954577182682849028m_-5765726088193305192WordSection1=
">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">About your proposal about negotiation of del=
ayed ACK, a potential problem is that RTO will be stretched than before bec=
ause you add extra delay. As you have said,
 most of the packets will not exceed RTO even host enable delayed ACK for m=
ost large flows, but the retransmission will be delayed (i.e., 5ms)also onc=
e one packet is lost.</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>The basic idea of the proposal is to tweak the RTO calculation and tur=
n a previously existing, historically motivated 200ms fixed &quot;slop fact=
or&quot; into a dynamically negotiated 5ms &quot;slop factor&quot;. In our =
experience that is almost always a win.</div>
<div><br>
</div>
<div>neal</div>
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"gmail-m_-954577182682849028m_-5765726088193305192WordSection1=
">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Best,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Yali<u></u><u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"ZH-CN" style=3D"font-size:11pt;font=
-family:=E5=BE=AE=E8=BD=AF=E9=9B=85=E9=BB=91,sans-serif">=E5=8F=91=E4=BB=B6=
=E4=BA=BA</span></b><b><span style=3D"font-size:11pt;font-family:=E5=BE=AE=
=E8=BD=AF=E9=9B=85=E9=BB=91,sans-serif">:</span></b><span style=3D"font-siz=
e:11pt;font-family:=E5=BE=AE=E8=BD=AF=E9=9B=85=E9=BB=91,sans-serif">
 Neal Cardwell [mailto:<a href=3D"mailto:ncardwell@google.com" target=3D"_b=
lank">ncardwell@google.com</a>]
<br>
<b><span lang=3D"ZH-CN">=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4</span>:</b> 20=
16<span lang=3D"ZH-CN">=E5=B9=B4</span>12<span lang=3D"ZH-CN">=E6=9C=88</sp=
an>13<span lang=3D"ZH-CN">=E6=97=A5</span> 23:18<br>
<b><span lang=3D"ZH-CN">=E6=94=B6=E4=BB=B6=E4=BA=BA</span>:</b> zhangyali (=
D) &lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangya=
li369@huawei.com</a>&gt;<br>
<b><span lang=3D"ZH-CN">=E6=8A=84=E9=80=81</span>:</b> <a href=3D"mailto:tc=
pm@ietf.org" target=3D"_blank">
tcpm@ietf.org</a><br>
<b><span lang=3D"ZH-CN">=E4=B8=BB=E9=A2=98</span>:</b> Re: [tcpm] A questio=
n about Delayed ACK and RTO<u></u><u></u></span></p>
<div>
<div class=3D"gmail-m_-954577182682849028h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) &lt;<=
a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@hu=
awei.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP=E2=80=99=
s delayed ACK is two packets or 200ms. I am wondering if this setting is ac=
cord with =C2=A0some RFCs in IETF. But After I
 referred to some RFCs, I just found some restrictions, such as, the delay =
must be less than 0.5ms (RFC1122).<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I believe the 200ms figure is due to historical reas=
ons. AFAIK it&#39;s due to the BSD delayed ACK behavior (see Stevens &quot;=
TCP/IP Illustrated Volume 2&quot;, section 25.4 and figure 25.7, which desc=
ribes the 200ms delayed ACK timer). Then this value
 was inherited by other widely-deployed OSes.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network resources.<u></u><=
u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Yes, exactly. In theory, the RTO tries to be adaptiv=
e enough to measure any RTT variations caused by delayed ACKs, and increase=
 the RTO in response to this. However, it&#39;s quite easy for a bulk trans=
fer to never have any delayed ACKs for
 most of its lifetime, during which the RTO gradually converges toward the =
raw RTT value. Then when there is suddenly a delayed ACK, there can be a sp=
urious RTO.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To avoid that, many TCP stacks (at least the major o=
pen source OSes) have a hard-coded 200ms &quot;slop factor&quot; or &quot;f=
udge factor&quot;, to try to never let their estimate of RTT variation fall=
 below that 200ms value, to avoid this effect.=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between RTO and delayed ACK?<u=
></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I&#39;m not aware of RFC specifications for exact va=
lues of both (delayed ACK and RTO). However, the historical precedent is ve=
ry strong, and the 200ms delayed ACK value was very pronounced in Internet =
traces at least as recently as 2011, when
 I last looked at the effect. (Probably others have more recent data points=
 for the prevalence of 200ms delayed ACKs.)<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">At IETF 97 our team at Google presented some feature=
s we use for internal TCP traffic at Google, where the endpoints can negoti=
ate the specific constant to use for the maximum delayed ACK from the recei=
ver and the corresponding minimum
 RTT delay variation for budgeting in the RTO at the sender:<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0 <a href=3D"https://www.ietf.org/proceedings/9=
7/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf" target=3D"_blan=
k">
https://www.ietf.org/proceedin<wbr>gs/97/slides/slides-97-tcpm-<wbr>tcp-opt=
ions-for-low-latency-<wbr>00.pdf</a><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">As this slide deck notes, within Google we negotiate=
 5ms for delayed ACKs.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">cheers,=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">neal<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div></div><blockquote type=3D"cite">
<div><span>______________________________<wbr>_________________</span><br>
<span>tcpm mailing list</span><br>
<span><a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><=
/span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_bla=
nk">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a></span><br>
</div>
</blockquote>
</div>

</blockquote></div><br></div></div>

--001a1134fbcef152090543a15431--


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From: "Jakob Heitz (jheitz)" <jheitz@cisco.com>
To: Neal Cardwell <ncardwell@google.com>
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Date: Wed, 14 Dec 2016 17:21:32 +0000
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cmc8bWFpbHRvOnRjcG1AaWV0Zi5vcmc+DQpodHRwczovL3d3dy5pZXRmLm9yZy9tYWlsbWFuL2xp
c3RpbmZvL3RjcG0NCg0K

--_000_3C746F8573B9428FAD1882E6BB99BE52ciscocom_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dgb2312">
</head>
<body dir=3D"auto">
<div>The problem in a virtualized environment is that RTT is highly variabl=
e. Mostly sub millisecond with occasional spikes at several 100 ms. And not=
 just because of delayed ack, but due to effects of VM scheduling and load.=
<br>
<br>
Thanks,<br>
<div>Jakob.</div>
<div><br>
</div>
</div>
<div><br>
On Dec 14, 2016, at 9:06 AM, Neal Cardwell &lt;<a href=3D"mailto:ncardwell@=
google.com">ncardwell@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (j=
heitz) <span dir=3D"ltr">
&lt;<a href=3D"mailto:jheitz@cisco.com" target=3D"_blank">jheitz@cisco.com<=
/a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir=3D"auto">
<div>Historically, the minimum RTO is 1 second and actual RTT is very rarel=
y more than 1 second, so all this RTT calculation hardly ever matters anywa=
y.<br>
</div>
</div>
</blockquote>
<div><br>
</div>
That may be true historically, but for many years major TCP implementations=
 (including Linux and FreeBSD) have used a minimum RTO closer to 200ms. And=
 in datacenter environments even 200ms can be infeasibly high.
<div><br>
</div>
<div>neal</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir=3D"auto">
<div><br>
Thanks,<br>
<div>Jakob.</div>
<div><br>
</div>
</div>
<div>
<div class=3D"gmail-h5">
<div><br>
On Dec 14, 2016, at 5:24 AM, Neal Cardwell &lt;<a href=3D"mailto:ncardwell@=
google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali3=
69@huawei.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"gmail-m_-954577182682849028m_-5765726088193305192WordSection1=
">
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">Hi Neal,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">Thanks for your providing request information.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">About the delay of delayed ACK, I found another clue in a S=
IGCOMM paper in 1988. In the page 14, a sentence is =A1=B0The 4.5KBps sende=
rs were talking to 4.3BSD receivers which
 would delay an ack until 35% of the window was filled or 200 ms had passed=
 (i.e., an ack was delayed for 5-7 packets on average).=A1=B1 There is no r=
eference about the 200 ms, so I guess this is the particular delay appears =
for the first time.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">I try to calculate the value based on delayed packet number=
s, packet size and sending rate. In this case, the delayed packet number is=
 7, the packet size is 576Byte (refer
 to RFC879 in 1983), and sundering rate is 4.5Bps. The value is 896 ms! Lon=
ger than 200 ms.
<u></u><u></u></span></p>
<span>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">&gt;</span>. However, it's quite easy for a bulk transfer t=
o never have any delayed ACKs for most of its lifetime, during which the RT=
O gradually converges toward the raw RTT
 value. Then when there is suddenly a delayed ACK, there can be a spurious =
RTO.<u></u><u></u></p>
</span>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">I am not very sure if I see your point. I think the key poi=
nt is the RTT variation will be weakened when packets are sent in a burst (=
the consequence of delayed ACK). &nbsp;&nbsp;And
 the spurious &nbsp;could only happen for the last few packets whose number=
 is smaller than delayed packets, right?</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Yes, there should only be a delayed ACK at the end of an application c=
hunk if there is an odd number of packets. But we would expect roughly half=
 of application chunks to have an odd number of packets. Though the proport=
ion is probably higher than that,
 since many application chunks are just one packet (e.g. an HTTP or RPC req=
uest or response).</div>
<div><br>
</div>
<div>&nbsp;<span style=3D"color:rgb(31,73,125);font-family:calibri,sans-ser=
if;font-size:11pt">&nbsp;</span></div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"gmail-m_-954577182682849028m_-5765726088193305192WordSection1=
">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">About your proposal about negotiation of del=
ayed ACK, a potential problem is that RTO will be stretched than before bec=
ause you add extra delay. As you have
 said, most of the packets will not exceed RTO even host enable delayed ACK=
 for most large flows, but the retransmission will be delayed (i.e., 5ms)al=
so once one packet is lost.</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>The basic idea of the proposal is to tweak the RTO calculation and tur=
n a previously existing, historically motivated 200ms fixed &quot;slop fact=
or&quot; into a dynamically negotiated 5ms &quot;slop factor&quot;. In our =
experience that is almost always a win.</div>
<div><br>
</div>
<div>neal</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"gmail-m_-954577182682849028m_-5765726088193305192WordSection1=
">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Best,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Yali<u></u><u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"ZH-CN" style=3D"font-size:11pt;font=
-family:=CE=A2=C8=ED=D1=C5=BA=DA,sans-serif">=B7=A2=BC=FE=C8=CB</span></b><=
b><span style=3D"font-size:11pt;font-family:=CE=A2=C8=ED=D1=C5=BA=DA,sans-s=
erif">:</span></b><span style=3D"font-size:11pt;font-family:=CE=A2=C8=ED=D1=
=C5=BA=DA,sans-serif"> Neal Cardwell [mailto:<a href=3D"mailto:ncardwell@go=
ogle.com" target=3D"_blank">ncardwell@google.com</a>]
<br>
<b><span lang=3D"ZH-CN">=B7=A2=CB=CD=CA=B1=BC=E4</span>:</b> 2016<span lang=
=3D"ZH-CN">=C4=EA</span>12<span lang=3D"ZH-CN">=D4=C2</span>13<span lang=3D=
"ZH-CN">=C8=D5</span> 23:18<br>
<b><span lang=3D"ZH-CN">=CA=D5=BC=FE=C8=CB</span>:</b> zhangyali (D) &lt;<a=
 href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@hua=
wei.com</a>&gt;<br>
<b><span lang=3D"ZH-CN">=B3=AD=CB=CD</span>:</b> <a href=3D"mailto:tcpm@iet=
f.org" target=3D"_blank">
tcpm@ietf.org</a><br>
<b><span lang=3D"ZH-CN">=D6=F7=CC=E2</span>:</b> Re: [tcpm] A question abou=
t Delayed ACK and RTO<u></u><u></u></span></p>
<div>
<div class=3D"gmail-m_-954577182682849028h5">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) &lt;<=
a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@hu=
awei.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP=A1=AFs d=
elayed ACK is two packets or 200ms. I am wondering if this setting is accor=
d with &nbsp;some RFCs in IETF. But After I
 referred to some RFCs, I just found some restrictions, such as, the delay =
must be less than 0.5ms (RFC1122).<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I believe the 200ms figure is due to historical reas=
ons. AFAIK it's due to the BSD delayed ACK behavior (see Stevens &quot;TCP/=
IP Illustrated Volume 2&quot;, section 25.4 and figure 25.7, which describe=
s the 200ms delayed ACK timer). Then this value
 was inherited by other widely-deployed OSes.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;&nbsp;<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network resources.<u></u><=
u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Yes, exactly. In theory, the RTO tries to be adaptiv=
e enough to measure any RTT variations caused by delayed ACKs, and increase=
 the RTO in response to this. However, it's quite easy for a bulk transfer =
to never have any delayed ACKs for
 most of its lifetime, during which the RTO gradually converges toward the =
raw RTT value. Then when there is suddenly a delayed ACK, there can be a sp=
urious RTO.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To avoid that, many TCP stacks (at least the major o=
pen source OSes) have a hard-coded 200ms &quot;slop factor&quot; or &quot;f=
udge factor&quot;, to try to never let their estimate of RTT variation fall=
 below that 200ms value, to avoid this effect.&nbsp;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between RTO and delayed ACK?<u=
></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I'm not aware of RFC specifications for exact values=
 of both (delayed ACK and RTO). However, the historical precedent is very s=
trong, and the 200ms delayed ACK value was very pronounced in Internet trac=
es at least as recently as 2011, when
 I last looked at the effect. (Probably others have more recent data points=
 for the prevalence of 200ms delayed ACKs.)<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">At IETF 97 our team at Google presented some feature=
s we use for internal TCP traffic at Google, where the endpoints can negoti=
ate the specific constant to use for the maximum delayed ACK from the recei=
ver and the corresponding minimum
 RTT delay variation for budgeting in the RTO at the sender:<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; <a href=3D"https://www.ietf.org/proceedings/9=
7/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf" target=3D"_blan=
k">
https://www.ietf.org/proceedin<wbr>gs/97/slides/slides-97-tcpm-<wbr>tcp-opt=
ions-for-low-latency-<wbr>00.pdf</a><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">As this slide deck notes, within Google we negotiate=
 5ms for delayed ACKs.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">cheers,&nbsp;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">neal<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
</div>
<blockquote type=3D"cite">
<div><span>______________________________<wbr>_________________</span><br>
<span>tcpm mailing list</span><br>
<span><a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><=
/span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_bla=
nk">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a></span><br>
</div>
</blockquote>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</body>
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From: Neal Cardwell <ncardwell@google.com>
Date: Wed, 14 Dec 2016 14:13:15 -0500
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To: "Jakob Heitz (jheitz)" <jheitz@cisco.com>
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiBBIHF1ZXN0aW9uIGFib3V0IERlbGF5ZWQgQUNL?= =?utf-8?q?_and_RTO?=
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On Wed, Dec 14, 2016 at 12:21 PM, Jakob Heitz (jheitz) <jheitz@cisco.com>
wrote:

> The problem in a virtualized environment is that RTT is highly variable.
> Mostly sub millisecond with occasional spikes at several 100 ms. And not
> just because of delayed ack, but due to effects of VM scheduling and load=
.
>

True enough. But the RTO mechanism need not try to drive the spurious RTO
rate down to zero. There is some trade-off between faster loss recovery and
an increase in spurious retransmits, and some associated cost-benefit
analysis. These days IMHO the cost of a spurious timer-based loss repair
attempt is relatively low, now that we have TLP and a number of nice undo
mechanisms (FRTO, Eifel, DSACKs). By contrast, in datacenter apps w/
sub-millisecond RTTs, there are prohibitive costs for delaying every
timer-driven loss repair for 200ms.

cheers,
neal



>
> Thanks,
> Jakob.
>
>
> On Dec 14, 2016, at 9:06 AM, Neal Cardwell <ncardwell@google.com> wrote:
>
> On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (jheitz) <jheitz@cisco.com>
> wrote:
>
>> Historically, the minimum RTO is 1 second and actual RTT is very rarely
>> more than 1 second, so all this RTT calculation hardly ever matters anyw=
ay.
>>
>
> That may be true historically, but for many years major TCP
> implementations (including Linux and FreeBSD) have used a minimum RTO
> closer to 200ms. And in datacenter environments even 200ms can be
> infeasibly high.
>
> neal
>
>
>>
>> Thanks,
>> Jakob.
>>
>>
>> On Dec 14, 2016, at 5:24 AM, Neal Cardwell <ncardwell@google.com> wrote:
>>
>> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <zhangyali369@huawei.com>
>> wrote:
>>
>>> Hi Neal,
>>>
>>>
>>>
>>> Thanks for your providing request information.
>>>
>>>
>>>
>>> About the delay of delayed ACK, I found another clue in a SIGCOMM paper
>>> in 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps senders wer=
e talking to
>>> 4.3BSD receivers which would delay an ack until 35% of the window was
>>> filled or 200 ms had passed (i.e., an ack was delayed for 5-7 packets o=
n
>>> average).=E2=80=9D There is no reference about the 200 ms, so I guess t=
his is the
>>> particular delay appears for the first time.
>>>
>>> I try to calculate the value based on delayed packet numbers, packet
>>> size and sending rate. In this case, the delayed packet number is 7, th=
e
>>> packet size is 576Byte (refer to RFC879 in 1983), and sundering rate is
>>> 4.5Bps. The value is 896 ms! Longer than 200 ms.
>>>
>>>
>>>
>>> >. However, it's quite easy for a bulk transfer to never have any
>>> delayed ACKs for most of its lifetime, during which the RTO gradually
>>> converges toward the raw RTT value. Then when there is suddenly a delay=
ed
>>> ACK, there can be a spurious RTO.
>>>
>>> I am not very sure if I see your point. I think the key point is the RT=
T
>>> variation will be weakened when packets are sent in a burst (the
>>> consequence of delayed ACK).   And the spurious  could only happen for =
the
>>> last few packets whose number is smaller than delayed packets, right?
>>>
>>
>> Yes, there should only be a delayed ACK at the end of an application
>> chunk if there is an odd number of packets. But we would expect roughly
>> half of application chunks to have an odd number of packets. Though the
>> proportion is probably higher than that, since many application chunks a=
re
>> just one packet (e.g. an HTTP or RPC request or response).
>>
>>
>>
>>> About your proposal about negotiation of delayed ACK, a potential
>>> problem is that RTO will be stretched than before because you add extra
>>> delay. As you have said, most of the packets will not exceed RTO even h=
ost
>>> enable delayed ACK for most large flows, but the retransmission will be
>>> delayed (i.e., 5ms)also once one packet is lost.
>>>
>>
>> The basic idea of the proposal is to tweak the RTO calculation and turn =
a
>> previously existing, historically motivated 200ms fixed "slop factor" in=
to
>> a dynamically negotiated 5ms "slop factor". In our experience that is
>> almost always a win.
>>
>> neal
>>
>>
>>>
>>>
>>> Best,
>>>
>>> Yali
>>>
>>> *=E5=8F=91=E4=BB=B6=E4=BA=BA**:* Neal Cardwell [mailto:ncardwell@google=
.com]
>>> *=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4:* 2016=E5=B9=B412=E6=9C=8813=E6=
=97=A5 23:18
>>> *=E6=94=B6=E4=BB=B6=E4=BA=BA:* zhangyali (D) <zhangyali369@huawei.com>
>>> *=E6=8A=84=E9=80=81:* tcpm@ietf.org
>>> *=E4=B8=BB=E9=A2=98:* Re: [tcpm] A question about Delayed ACK and RTO
>>>
>>>
>>>
>>> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com=
>
>>> wrote:
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Recent days, I am doing some simulation about TCP performance in NS3. I
>>> found a phenomenon is a default setting of TCP=E2=80=99s delayed ACK is=
 two packets
>>> or 200ms. I am wondering if this setting is accord with  some RFCs in I=
ETF.
>>> But After I referred to some RFCs, I just found some restrictions, such=
 as,
>>> the delay must be less than 0.5ms (RFC1122).
>>>
>>>
>>>
>>> I believe the 200ms figure is due to historical reasons. AFAIK it's due
>>> to the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume=
 2",
>>> section 25.4 and figure 25.7, which describes the 200ms delayed ACK tim=
er).
>>> Then this value was inherited by other widely-deployed OSes.
>>>
>>>
>>>
>>>
>>>
>>> I think the delayed ACK has a close relationship with RTO. Take an
>>> extreme scenario, if the delay is longer than RTO, many packets will be
>>> retransmitted, which will waste many network resources.
>>>
>>>
>>>
>>> Yes, exactly. In theory, the RTO tries to be adaptive enough to measure
>>> any RTT variations caused by delayed ACKs, and increase the RTO in resp=
onse
>>> to this. However, it's quite easy for a bulk transfer to never have any
>>> delayed ACKs for most of its lifetime, during which the RTO gradually
>>> converges toward the raw RTT value. Then when there is suddenly a delay=
ed
>>> ACK, there can be a spurious RTO.
>>>
>>>
>>>
>>> To avoid that, many TCP stacks (at least the major open source OSes)
>>> have a hard-coded 200ms "slop factor" or "fudge factor", to try to neve=
r
>>> let their estimate of RTT variation fall below that 200ms value, to avo=
id
>>> this effect.
>>>
>>>
>>>
>>> So I want to know do we have some RFCs have given the exact value of
>>> both? And if we permit any TCP stack to set them freely, what is the
>>> mechanism to balance the mismatch between RTO and delayed ACK?
>>>
>>>
>>>
>>> I'm not aware of RFC specifications for exact values of both (delayed
>>> ACK and RTO). However, the historical precedent is very strong, and the
>>> 200ms delayed ACK value was very pronounced in Internet traces at least=
 as
>>> recently as 2011, when I last looked at the effect. (Probably others ha=
ve
>>> more recent data points for the prevalence of 200ms delayed ACKs.)
>>>
>>>
>>>
>>> At IETF 97 our team at Google presented some features we use for
>>> internal TCP traffic at Google, where the endpoints can negotiate the
>>> specific constant to use for the maximum delayed ACK from the receiver =
and
>>> the corresponding minimum RTT delay variation for budgeting in the RTO =
at
>>> the sender:
>>>
>>>
>>>
>>>   https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tc
>>> p-options-for-low-latency-00.pdf
>>>
>>>
>>>
>>> As this slide deck notes, within Google we negotiate 5ms for delayed
>>> ACKs.
>>>
>>>
>>>
>>> cheers,
>>>
>>> neal
>>>
>>>
>>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>>
>

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Dec 14, 2016 at 12:21 PM, Jakob Heitz (jheitz) <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jheitz@cisco.com" target=3D"_blank">jheitz@cisco.com</a>&gt=
;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">



<div dir=3D"auto">
<div>The problem in a virtualized environment is that RTT is highly variabl=
e. Mostly sub millisecond with occasional spikes at several 100 ms. And not=
 just because of delayed ack, but due to effects of VM scheduling and load.=
<br></div></div></blockquote><div><br></div><div>True enough. But the RTO m=
echanism need not try to drive the spurious RTO rate down to zero. There is=
 some trade-off between faster loss recovery and an increase in spurious re=
transmits, and some associated cost-benefit analysis. These days IMHO the c=
ost of a spurious timer-based loss repair attempt is relatively low, now th=
at we have TLP and a number of nice undo mechanisms (FRTO, Eifel, DSACKs). =
By contrast, in datacenter apps w/ sub-millisecond RTTs, there are prohibit=
ive costs for delaying every timer-driven loss repair for 200ms.</div><div>=
<br></div><div>cheers,</div><div>neal</div><div><br></div><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"auto"><div>
<br>
Thanks,<br>
<div>Jakob.</div>
<div><br>
</div>
</div><div><div class=3D"h5">
<div><br>
On Dec 14, 2016, at 9:06 AM, Neal Cardwell &lt;<a href=3D"mailto:ncardwell@=
google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (j=
heitz) <span dir=3D"ltr">
&lt;<a href=3D"mailto:jheitz@cisco.com" target=3D"_blank">jheitz@cisco.com<=
/a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir=3D"auto">
<div>Historically, the minimum RTO is 1 second and actual RTT is very rarel=
y more than 1 second, so all this RTT calculation hardly ever matters anywa=
y.<br>
</div>
</div>
</blockquote>
<div><br>
</div>
That may be true historically, but for many years major TCP implementations=
 (including Linux and FreeBSD) have used a minimum RTO closer to 200ms. And=
 in datacenter environments even 200ms can be infeasibly high.
<div><br>
</div>
<div>neal</div>
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir=3D"auto">
<div><br>
Thanks,<br>
<div>Jakob.</div>
<div><br>
</div>
</div>
<div>
<div class=3D"m_-4879702542152661443gmail-h5">
<div><br>
On Dec 14, 2016, at 5:24 AM, Neal Cardwell &lt;<a href=3D"mailto:ncardwell@=
google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali3=
69@huawei.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"m_-4879702542152661443gmail-m_-954577182682849028m_-576572608=
8193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">Hi Neal,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">Thanks for your providing request information.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">About the delay of delayed ACK, I found another clue in a S=
IGCOMM paper in 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps se=
nders were talking to 4.3BSD receivers which
 would delay an ack until 35% of the window was filled or 200 ms had passed=
 (i.e., an ack was delayed for 5-7 packets on average).=E2=80=9D There is n=
o reference about the 200 ms, so I guess this is the particular delay appea=
rs for the first time.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">I try to calculate the value based on delayed packet number=
s, packet size and sending rate. In this case, the delayed packet number is=
 7, the packet size is 576Byte (refer
 to RFC879 in 1983), and sundering rate is 4.5Bps. The value is 896 ms! Lon=
ger than 200 ms.
<u></u><u></u></span></p>
<span>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">&gt;</span>. However, it&#39;s quite easy for a bulk transf=
er to never have any delayed ACKs for most of its lifetime, during which th=
e RTO gradually converges toward the raw RTT
 value. Then when there is suddenly a delayed ACK, there can be a spurious =
RTO.<u></u><u></u></p>
</span>
<p class=3D"MsoNormal"><span style=3D"font-family:calibri,sans-serif;color:=
rgb(31,73,125)">I am not very sure if I see your point. I think the key poi=
nt is the RTT variation will be weakened when packets are sent in a burst (=
the consequence of delayed ACK). =C2=A0=C2=A0And
 the spurious =C2=A0could only happen for the last few packets whose number=
 is smaller than delayed packets, right?</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Yes, there should only be a delayed ACK at the end of an application c=
hunk if there is an odd number of packets. But we would expect roughly half=
 of application chunks to have an odd number of packets. Though the proport=
ion is probably higher than that,
 since many application chunks are just one packet (e.g. an HTTP or RPC req=
uest or response).</div>
<div><br>
</div>
<div>=C2=A0<span style=3D"color:rgb(31,73,125);font-family:calibri,sans-ser=
if;font-size:11pt">=C2=A0</span></div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"m_-4879702542152661443gmail-m_-954577182682849028m_-576572608=
8193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">About your proposal about negotiation of del=
ayed ACK, a potential problem is that RTO will be stretched than before bec=
ause you add extra delay. As you have
 said, most of the packets will not exceed RTO even host enable delayed ACK=
 for most large flows, but the retransmission will be delayed (i.e., 5ms)al=
so once one packet is lost.</span></p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>The basic idea of the proposal is to tweak the RTO calculation and tur=
n a previously existing, historically motivated 200ms fixed &quot;slop fact=
or&quot; into a dynamically negotiated 5ms &quot;slop factor&quot;. In our =
experience that is almost always a win.</div>
<div><br>
</div>
<div>neal</div>
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-US">
<div class=3D"m_-4879702542152661443gmail-m_-954577182682849028m_-576572608=
8193305192WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Best,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Yali<u></u><u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"ZH-CN" style=3D"font-size:11pt;font=
-family:=E5=BE=AE=E8=BD=AF=E9=9B=85=E9=BB=91,sans-serif">=E5=8F=91=E4=BB=B6=
=E4=BA=BA</span></b><b><span style=3D"font-size:11pt;font-family:=E5=BE=AE=
=E8=BD=AF=E9=9B=85=E9=BB=91,sans-serif">:</span></b><span style=3D"font-siz=
e:11pt;font-family:=E5=BE=AE=E8=BD=AF=E9=9B=85=E9=BB=91,sans-serif"> Neal C=
ardwell [mailto:<a href=3D"mailto:ncardwell@google.com" target=3D"_blank">n=
cardwell@google.com</a>]
<br>
<b><span lang=3D"ZH-CN">=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4</span>:</b> 20=
16<span lang=3D"ZH-CN">=E5=B9=B4</span>12<span lang=3D"ZH-CN">=E6=9C=88</sp=
an>13<span lang=3D"ZH-CN">=E6=97=A5</span> 23:18<br>
<b><span lang=3D"ZH-CN">=E6=94=B6=E4=BB=B6=E4=BA=BA</span>:</b> zhangyali (=
D) &lt;<a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangya=
li369@huawei.com</a>&gt;<br>
<b><span lang=3D"ZH-CN">=E6=8A=84=E9=80=81</span>:</b> <a href=3D"mailto:tc=
pm@ietf.org" target=3D"_blank">
tcpm@ietf.org</a><br>
<b><span lang=3D"ZH-CN">=E4=B8=BB=E9=A2=98</span>:</b> Re: [tcpm] A questio=
n about Delayed ACK and RTO<u></u><u></u></span></p>
<div>
<div class=3D"m_-4879702542152661443gmail-m_-954577182682849028h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) &lt;<=
a href=3D"mailto:zhangyali369@huawei.com" target=3D"_blank">zhangyali369@hu=
awei.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Recent days, I am doing some simulation about TCP pe=
rformance in NS3. I found a phenomenon is a default setting of TCP=E2=80=99=
s delayed ACK is two packets or 200ms. I am wondering if this setting is ac=
cord with =C2=A0some RFCs in IETF. But After I
 referred to some RFCs, I just found some restrictions, such as, the delay =
must be less than 0.5ms (RFC1122).<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I believe the 200ms figure is due to historical reas=
ons. AFAIK it&#39;s due to the BSD delayed ACK behavior (see Stevens &quot;=
TCP/IP Illustrated Volume 2&quot;, section 25.4 and figure 25.7, which desc=
ribes the 200ms delayed ACK timer). Then this value
 was inherited by other widely-deployed OSes.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">I think the delayed ACK has a close relationship wit=
h RTO. Take an extreme scenario, if the delay is longer than RTO, many pack=
ets will be retransmitted, which will waste many network resources.<u></u><=
u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Yes, exactly. In theory, the RTO tries to be adaptiv=
e enough to measure any RTT variations caused by delayed ACKs, and increase=
 the RTO in response to this. However, it&#39;s quite easy for a bulk trans=
fer to never have any delayed ACKs for
 most of its lifetime, during which the RTO gradually converges toward the =
raw RTT value. Then when there is suddenly a delayed ACK, there can be a sp=
urious RTO.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To avoid that, many TCP stacks (at least the major o=
pen source OSes) have a hard-coded 200ms &quot;slop factor&quot; or &quot;f=
udge factor&quot;, to try to never let their estimate of RTT variation fall=
 below that 200ms value, to avoid this effect.=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4=
.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">So I want to know do we have some RFCs have given th=
e exact value of both? And if we permit any TCP stack to set them freely, w=
hat is the mechanism to balance the mismatch between RTO and delayed ACK?<u=
></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I&#39;m not aware of RFC specifications for exact va=
lues of both (delayed ACK and RTO). However, the historical precedent is ve=
ry strong, and the 200ms delayed ACK value was very pronounced in Internet =
traces at least as recently as 2011, when
 I last looked at the effect. (Probably others have more recent data points=
 for the prevalence of 200ms delayed ACKs.)<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">At IETF 97 our team at Google presented some feature=
s we use for internal TCP traffic at Google, where the endpoints can negoti=
ate the specific constant to use for the maximum delayed ACK from the recei=
ver and the corresponding minimum
 RTT delay variation for budgeting in the RTO at the sender:<u></u><u></u><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0 <a href=3D"https://www.ietf.org/proceedings/9=
7/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf" target=3D"_blan=
k">
https://www.ietf.org/proceedin<wbr>gs/97/slides/slides-97-tcpm-tc<wbr>p-opt=
ions-for-low-latency-00.<wbr>pdf</a><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">As this slide deck notes, within Google we negotiate=
 5ms for delayed ACKs.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">cheers,=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">neal<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
</div>
<blockquote type=3D"cite">
<div><span>______________________________<wbr>_________________</span><br>
<span>tcpm mailing list</span><br>
<span><a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><=
/span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_bla=
nk">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a></span><br>
</div>
</blockquote>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div></div></div>

</blockquote></div><br></div></div>

--001a1142d920c627cc0543a3220c--


From nobody Wed Dec 14 11:20:35 2016
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From: Joe Touch <touch@isi.edu>
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We should not be optimizing TCP for specific environments; that's for
operators to override.

Joe


On 12/14/2016 9:04 AM, Neal Cardwell wrote:
> On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (jheitz)
> <jheitz@cisco.com <mailto:jheitz@cisco.com>> wrote:
>
>     Historically, the minimum RTO is 1 second and actual RTT is very
>     rarely more than 1 second, so all this RTT calculation hardly ever
>     matters anyway.
>
>
> That may be true historically, but for many years major TCP
> implementations (including Linux and FreeBSD) have used a minimum RTO
> closer to 200ms. And in datacenter environments even 200ms can be
> infeasibly high.
>
> neal
>  


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    <p>We should not be optimizing TCP for specific environments; that's
      for operators to override.</p>
    <p>Joe<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 12/14/2016 9:04 AM, Neal Cardwell
      wrote:<br>
    </div>
    <blockquote
cite="mid:CADVnQy=2vuxwybswBgzEEjinp92DRKFZgphaVdBGvguJBirbJw@mail.gmail.com"
      type="cite">On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (jheitz)
      <span dir="ltr">&lt;<a moz-do-not-send="true"
          href="mailto:jheitz@cisco.com" target="_blank">jheitz@cisco.com</a>&gt;</span>
      wrote:<br>
      <blockquote class="gmail_quote" style="margin:0px 0px 0px
        0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
        <div dir="auto">
          <div>Historically, the minimum RTO is 1 second and actual RTT
            is very rarely more than 1 second, so all this RTT
            calculation hardly ever matters anyway.<br>
          </div>
        </div>
      </blockquote>
      <div><br>
      </div>
      That may be true historically, but for many years major TCP
      implementations (including Linux and FreeBSD) have used a minimum
      RTO closer to 200ms. And in datacenter environments even 200ms can
      be infeasibly high.
      <div><br>
      </div>
      <div>neal</div>
      <div> </div>
    </blockquote>
    <br>
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--------------210A99D607FAAD82E01B539E--


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Hi Joe,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">I really agree with you about TCP mec=
hanism should general for all areas, including wan and data center.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">But every scenarios have their own ch=
aracteristics. Data center has some distinctive features, such as, ultralow=
 RTT, because of centralized location, less hops,
 and so on. These features determines different requirement for TCP default=
 configuration, compared with wan scenario. It is a solution by having the =
aids of operator, but maybe we could have a more automatic and general mech=
anism to solve discrepancy. Just
 a simple idea. </span><span style=3D"font-size:11.0pt;font-family:Wingding=
s;color:#1F497D">J</span><span style=3D"font-size:11.0pt;font-family:&quot;=
Calibri&quot;,sans-serif;color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Best Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Yali
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"ZH-CN" style=3D"font-size:11.0pt;fo=
nt-family:&quot;=CE=A2=C8=ED=D1=C5=BA=DA&quot;,sans-serif;color:windowtext"=
>=B7=A2=BC=FE=C8=CB</span></b><b><span style=3D"font-size:11.0pt;font-famil=
y:&quot;=CE=A2=C8=ED=D1=C5=BA=DA&quot;,sans-serif;color:windowtext">:</span=
></b><span style=3D"font-size:11.0pt;font-family:&quot;=CE=A2=C8=ED=D1=C5=
=BA=DA&quot;,sans-serif;color:windowtext">
 tcpm [mailto:tcpm-bounces@ietf.org] <b><span lang=3D"ZH-CN">=B4=FA=B1=ED <=
/span></b>Joe Touch<br>
<b><span lang=3D"ZH-CN">=B7=A2=CB=CD=CA=B1=BC=E4</span>:</b> 2016<span lang=
=3D"ZH-CN">=C4=EA</span>12<span lang=3D"ZH-CN">=D4=C2</span>15<span lang=3D=
"ZH-CN">=C8=D5</span> 3:20<br>
<b><span lang=3D"ZH-CN">=CA=D5=BC=FE=C8=CB</span>:</b> Neal Cardwell &lt;nc=
ardwell@google.com&gt;; Jakob Heitz (jheitz) &lt;jheitz@cisco.com&gt;<br>
<b><span lang=3D"ZH-CN">=B3=AD=CB=CD</span>:</b> tcpm@ietf.org<br>
<b><span lang=3D"ZH-CN">=D6=F7=CC=E2</span>:</b> Re: [tcpm] <span lang=3D"Z=
H-CN">=B4=F0=B8=B4</span>: A question about Delayed ACK and RTO<o:p></o:p><=
/span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p>We should not be optimizing TCP for specific environments; that's for op=
erators to override.<o:p></o:p></p>
<p>Joe<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 12/14/2016 9:04 AM, Neal Cardwell wrote:<o:p></o:=
p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal">On Wed, Dec 14, 2016 at 11:56 AM, Jakob Heitz (jheit=
z) &lt;<a href=3D"mailto:jheitz@cisco.com" target=3D"_blank">jheitz@cisco.c=
om</a>&gt; wrote:<o:p></o:p></p>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">Historically, the minimum RTO is 1 second and actual=
 RTT is very rarely more than 1 second, so all this RTT calculation hardly =
ever matters anyway.<o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">That may be true historically, but for many years ma=
jor TCP implementations (including Linux and FreeBSD) have used a minimum R=
TO closer to 200ms. And in datacenter environments even 200ms can be infeas=
ibly high.
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">neal<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</blockquote>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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From: "zhangyali (D)" <zhangyali369@huawei.com>
To: "zhangyali (D)" <zhangyali369@huawei.com>, Neal Cardwell <ncardwell@google.com>
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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com>
From: Sepherosa Ziehau <sepherosa@gmail.com>
Date: Thu, 15 Dec 2016 11:21:13 +0800
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To: "zhangyali (D)" <zhangyali369@huawei.com>
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiDnrZTlpI06IOetlOWkjTogQSBxdWVzdGlvbiBh?= =?utf-8?q?bout_Delayed_ACK_and_RTO?=
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I believe delack timeout has been 100ms on FreeBSD since 1999, while
RTO min is >=3D 200ms by default on FreeBSD.

On Thu, Dec 15, 2016 at 11:08 AM, zhangyali (D) <zhangyali369@huawei.com> w=
rote:
> Please allow me to add one more point. If the probability of odd number o=
f
> packets occurring  spurious RTO is so outstanding, why nobody try to solv=
e
> this problem?
>
>
>
> =E5=8F=91=E4=BB=B6=E4=BA=BA: tcpm [mailto:tcpm-bounces@ietf.org] =E4=BB=
=A3=E8=A1=A8 zhangyali (D)
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2016=E5=B9=B412=E6=9C=8815=E6=97=A5=
 9:45
> =E6=94=B6=E4=BB=B6=E4=BA=BA: Neal Cardwell <ncardwell@google.com>
> =E6=8A=84=E9=80=81: tcpm@ietf.org
> =E4=B8=BB=E9=A2=98: [tcpm] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: A ques=
tion about Delayed ACK and RTO
>
>
>
>> Yes, there should only be a delayed ACK at the end of an application chu=
nk
>> if there is an odd number of packets. But we would expect roughly half o=
f
>> application chunks to have an odd number of packets. Though the proporti=
on
>> is probably higher than that, since many application chunks are just one
>> packet (e.g. an HTTP or RPC request or response).
>
>
>
> I agree with you that odd number of packets may occur spurious RTO more
> easily, but I think for the flows owning just one packet should not be
> affected by delayed ACK. AFAIK, slow-start stage will begin with one pack=
et,
> and sender will send two packets after it receives an ACK. If the first
> packet is obstructed by the delayed ACK, the =E2=80=98clock algorithm=E2=
=80=99 will be
> broken down. So receiver should judge if this packet is the first one in =
the
> slow-start stage, if yes, send the ack immediately once receiving the
> packet.
>
>
>
> Yali
>
>
>
> =E5=8F=91=E4=BB=B6=E4=BA=BA: Neal Cardwell [mailto:ncardwell@google.com]
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2016=E5=B9=B412=E6=9C=8814=E6=97=A5=
 21:24
> =E6=94=B6=E4=BB=B6=E4=BA=BA: zhangyali (D) <zhangyali369@huawei.com>
> =E6=8A=84=E9=80=81: tcpm@ietf.org
> =E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [tcpm] A question about Delay=
ed ACK and RTO
>
>
>
> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <zhangyali369@huawei.com>
> wrote:
>
> Hi Neal,
>
>
>
> Thanks for your providing request information.
>
>
>
> About the delay of delayed ACK, I found another clue in a SIGCOMM paper i=
n
> 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps senders were tal=
king to
> 4.3BSD receivers which would delay an ack until 35% of the window was fil=
led
> or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on average=
).=E2=80=9D
> There is no reference about the 200 ms, so I guess this is the particular
> delay appears for the first time.
>
> I try to calculate the value based on delayed packet numbers, packet size
> and sending rate. In this case, the delayed packet number is 7, the packe=
t
> size is 576Byte (refer to RFC879 in 1983), and sundering rate is 4.5Bps. =
The
> value is 896 ms! Longer than 200 ms.
>
>
>
>>. However, it's quite easy for a bulk transfer to never have any delayed
>> ACKs for most of its lifetime, during which the RTO gradually converges
>> toward the raw RTT value. Then when there is suddenly a delayed ACK, the=
re
>> can be a spurious RTO.
>
> I am not very sure if I see your point. I think the key point is the RTT
> variation will be weakened when packets are sent in a burst (the conseque=
nce
> of delayed ACK).   And the spurious  could only happen for the last few
> packets whose number is smaller than delayed packets, right?
>
>
>
> Yes, there should only be a delayed ACK at the end of an application chun=
k
> if there is an odd number of packets. But we would expect roughly half of
> application chunks to have an odd number of packets. Though the proportio=
n
> is probably higher than that, since many application chunks are just one
> packet (e.g. an HTTP or RPC request or response).
>
>
>
>
>
> About your proposal about negotiation of delayed ACK, a potential problem=
 is
> that RTO will be stretched than before because you add extra delay. As yo=
u
> have said, most of the packets will not exceed RTO even host enable delay=
ed
> ACK for most large flows, but the retransmission will be delayed (i.e.,
> 5ms)also once one packet is lost.
>
>
>
> The basic idea of the proposal is to tweak the RTO calculation and turn a
> previously existing, historically motivated 200ms fixed "slop factor" int=
o a
> dynamically negotiated 5ms "slop factor". In our experience that is almos=
t
> always a win.
>
>
>
> neal
>
>
>
>
>
> Best,
>
> Yali
>
> =E5=8F=91=E4=BB=B6=E4=BA=BA: Neal Cardwell [mailto:ncardwell@google.com]
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2016=E5=B9=B412=E6=9C=8813=E6=97=A5=
 23:18
> =E6=94=B6=E4=BB=B6=E4=BA=BA: zhangyali (D) <zhangyali369@huawei.com>
> =E6=8A=84=E9=80=81: tcpm@ietf.org
> =E4=B8=BB=E9=A2=98: Re: [tcpm] A question about Delayed ACK and RTO
>
>
>
> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com>
> wrote:
>
> Hi all,
>
>
>
> Recent days, I am doing some simulation about TCP performance in NS3. I
> found a phenomenon is a default setting of TCP=E2=80=99s delayed ACK is t=
wo packets
> or 200ms. I am wondering if this setting is accord with  some RFCs in IET=
F.
> But After I referred to some RFCs, I just found some restrictions, such a=
s,
> the delay must be less than 0.5ms (RFC1122).
>
>
>
> I believe the 200ms figure is due to historical reasons. AFAIK it's due t=
o
> the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume 2",
> section 25.4 and figure 25.7, which describes the 200ms delayed ACK timer=
).
> Then this value was inherited by other widely-deployed OSes.
>
>
>
>
>
> I think the delayed ACK has a close relationship with RTO. Take an extrem=
e
> scenario, if the delay is longer than RTO, many packets will be
> retransmitted, which will waste many network resources.
>
>
>
> Yes, exactly. In theory, the RTO tries to be adaptive enough to measure a=
ny
> RTT variations caused by delayed ACKs, and increase the RTO in response t=
o
> this. However, it's quite easy for a bulk transfer to never have any dela=
yed
> ACKs for most of its lifetime, during which the RTO gradually converges
> toward the raw RTT value. Then when there is suddenly a delayed ACK, ther=
e
> can be a spurious RTO.
>
>
>
> To avoid that, many TCP stacks (at least the major open source OSes) have=
 a
> hard-coded 200ms "slop factor" or "fudge factor", to try to never let the=
ir
> estimate of RTT variation fall below that 200ms value, to avoid this effe=
ct.
>
>
>
> So I want to know do we have some RFCs have given the exact value of both=
?
> And if we permit any TCP stack to set them freely, what is the mechanism =
to
> balance the mismatch between RTO and delayed ACK?
>
>
>
> I'm not aware of RFC specifications for exact values of both (delayed ACK
> and RTO). However, the historical precedent is very strong, and the 200ms
> delayed ACK value was very pronounced in Internet traces at least as
> recently as 2011, when I last looked at the effect. (Probably others have
> more recent data points for the prevalence of 200ms delayed ACKs.)
>
>
>
> At IETF 97 our team at Google presented some features we use for internal
> TCP traffic at Google, where the endpoints can negotiate the specific
> constant to use for the maximum delayed ACK from the receiver and the
> corresponding minimum RTT delay variation for budgeting in the RTO at the
> sender:
>
>
>
>
> https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options-for=
-low-latency-00.pdf
>
>
>
> As this slide deck notes, within Google we negotiate 5ms for delayed ACKs=
.
>
>
>
> cheers,
>
> neal
>
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



--=20
Tomorrow Will Never Die


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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com>
From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Thu, 15 Dec 2016 00:24:24 -0800
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To: "zhangyali (D)" <zhangyali369@huawei.com>
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiDnrZTlpI06IOetlOWkjTogQSBxdWVzdGlvbiBh?= =?utf-8?q?bout_Delayed_ACK_and_RTO?=
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A question would be when TCP sent odd number of packets, how it can
know whether another packets will come from upper layer very soon or
not. Also, I'm not very sure yet if the probability is so obvious. I
can agree that RTO can get close to raw rtt value under bulk transfer.
But, I am guessing not so many apps change the behavior from bulk
transfer to spontaneous small burst transfer.
--
Yoshi


On Wed, Dec 14, 2016 at 7:08 PM, zhangyali (D) <zhangyali369@huawei.com> wr=
ote:
> Please allow me to add one more point. If the probability of odd number o=
f
> packets occurring  spurious RTO is so outstanding, why nobody try to solv=
e
> this problem?
>
>
>
> =E5=8F=91=E4=BB=B6=E4=BA=BA: tcpm [mailto:tcpm-bounces@ietf.org] =E4=BB=
=A3=E8=A1=A8 zhangyali (D)
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2016=E5=B9=B412=E6=9C=8815=E6=97=A5=
 9:45
> =E6=94=B6=E4=BB=B6=E4=BA=BA: Neal Cardwell <ncardwell@google.com>
> =E6=8A=84=E9=80=81: tcpm@ietf.org
> =E4=B8=BB=E9=A2=98: [tcpm] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: A ques=
tion about Delayed ACK and RTO
>
>
>
>> Yes, there should only be a delayed ACK at the end of an application chu=
nk
>> if there is an odd number of packets. But we would expect roughly half o=
f
>> application chunks to have an odd number of packets. Though the proporti=
on
>> is probably higher than that, since many application chunks are just one
>> packet (e.g. an HTTP or RPC request or response).
>
>
>
> I agree with you that odd number of packets may occur spurious RTO more
> easily, but I think for the flows owning just one packet should not be
> affected by delayed ACK. AFAIK, slow-start stage will begin with one pack=
et,
> and sender will send two packets after it receives an ACK. If the first
> packet is obstructed by the delayed ACK, the =E2=80=98clock algorithm=E2=
=80=99 will be
> broken down. So receiver should judge if this packet is the first one in =
the
> slow-start stage, if yes, send the ack immediately once receiving the
> packet.
>
>
>
> Yali
>
>
>
> =E5=8F=91=E4=BB=B6=E4=BA=BA: Neal Cardwell [mailto:ncardwell@google.com]
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2016=E5=B9=B412=E6=9C=8814=E6=97=A5=
 21:24
> =E6=94=B6=E4=BB=B6=E4=BA=BA: zhangyali (D) <zhangyali369@huawei.com>
> =E6=8A=84=E9=80=81: tcpm@ietf.org
> =E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [tcpm] A question about Delay=
ed ACK and RTO
>
>
>
> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <zhangyali369@huawei.com>
> wrote:
>
> Hi Neal,
>
>
>
> Thanks for your providing request information.
>
>
>
> About the delay of delayed ACK, I found another clue in a SIGCOMM paper i=
n
> 1988. In the page 14, a sentence is =E2=80=9CThe 4.5KBps senders were tal=
king to
> 4.3BSD receivers which would delay an ack until 35% of the window was fil=
led
> or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on average=
).=E2=80=9D
> There is no reference about the 200 ms, so I guess this is the particular
> delay appears for the first time.
>
> I try to calculate the value based on delayed packet numbers, packet size
> and sending rate. In this case, the delayed packet number is 7, the packe=
t
> size is 576Byte (refer to RFC879 in 1983), and sundering rate is 4.5Bps. =
The
> value is 896 ms! Longer than 200 ms.
>
>
>
>>. However, it's quite easy for a bulk transfer to never have any delayed
>> ACKs for most of its lifetime, during which the RTO gradually converges
>> toward the raw RTT value. Then when there is suddenly a delayed ACK, the=
re
>> can be a spurious RTO.
>
> I am not very sure if I see your point. I think the key point is the RTT
> variation will be weakened when packets are sent in a burst (the conseque=
nce
> of delayed ACK).   And the spurious  could only happen for the last few
> packets whose number is smaller than delayed packets, right?
>
>
>
> Yes, there should only be a delayed ACK at the end of an application chun=
k
> if there is an odd number of packets. But we would expect roughly half of
> application chunks to have an odd number of packets. Though the proportio=
n
> is probably higher than that, since many application chunks are just one
> packet (e.g. an HTTP or RPC request or response).
>
>
>
>
>
> About your proposal about negotiation of delayed ACK, a potential problem=
 is
> that RTO will be stretched than before because you add extra delay. As yo=
u
> have said, most of the packets will not exceed RTO even host enable delay=
ed
> ACK for most large flows, but the retransmission will be delayed (i.e.,
> 5ms)also once one packet is lost.
>
>
>
> The basic idea of the proposal is to tweak the RTO calculation and turn a
> previously existing, historically motivated 200ms fixed "slop factor" int=
o a
> dynamically negotiated 5ms "slop factor". In our experience that is almos=
t
> always a win.
>
>
>
> neal
>
>
>
>
>
> Best,
>
> Yali
>
> =E5=8F=91=E4=BB=B6=E4=BA=BA: Neal Cardwell [mailto:ncardwell@google.com]
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2016=E5=B9=B412=E6=9C=8813=E6=97=A5=
 23:18
> =E6=94=B6=E4=BB=B6=E4=BA=BA: zhangyali (D) <zhangyali369@huawei.com>
> =E6=8A=84=E9=80=81: tcpm@ietf.org
> =E4=B8=BB=E9=A2=98: Re: [tcpm] A question about Delayed ACK and RTO
>
>
>
> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com>
> wrote:
>
> Hi all,
>
>
>
> Recent days, I am doing some simulation about TCP performance in NS3. I
> found a phenomenon is a default setting of TCP=E2=80=99s delayed ACK is t=
wo packets
> or 200ms. I am wondering if this setting is accord with  some RFCs in IET=
F.
> But After I referred to some RFCs, I just found some restrictions, such a=
s,
> the delay must be less than 0.5ms (RFC1122).
>
>
>
> I believe the 200ms figure is due to historical reasons. AFAIK it's due t=
o
> the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume 2",
> section 25.4 and figure 25.7, which describes the 200ms delayed ACK timer=
).
> Then this value was inherited by other widely-deployed OSes.
>
>
>
>
>
> I think the delayed ACK has a close relationship with RTO. Take an extrem=
e
> scenario, if the delay is longer than RTO, many packets will be
> retransmitted, which will waste many network resources.
>
>
>
> Yes, exactly. In theory, the RTO tries to be adaptive enough to measure a=
ny
> RTT variations caused by delayed ACKs, and increase the RTO in response t=
o
> this. However, it's quite easy for a bulk transfer to never have any dela=
yed
> ACKs for most of its lifetime, during which the RTO gradually converges
> toward the raw RTT value. Then when there is suddenly a delayed ACK, ther=
e
> can be a spurious RTO.
>
>
>
> To avoid that, many TCP stacks (at least the major open source OSes) have=
 a
> hard-coded 200ms "slop factor" or "fudge factor", to try to never let the=
ir
> estimate of RTT variation fall below that 200ms value, to avoid this effe=
ct.
>
>
>
> So I want to know do we have some RFCs have given the exact value of both=
?
> And if we permit any TCP stack to set them freely, what is the mechanism =
to
> balance the mismatch between RTO and delayed ACK?
>
>
>
> I'm not aware of RFC specifications for exact values of both (delayed ACK
> and RTO). However, the historical precedent is very strong, and the 200ms
> delayed ACK value was very pronounced in Internet traces at least as
> recently as 2011, when I last looked at the effect. (Probably others have
> more recent data points for the prevalence of 200ms delayed ACKs.)
>
>
>
> At IETF 97 our team at Google presented some features we use for internal
> TCP traffic at Google, where the endpoints can negotiate the specific
> constant to use for the maximum delayed ACK from the receiver and the
> corresponding minimum RTT delay variation for budgeting in the RTO at the
> sender:
>
>
>
>
> https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options-for=
-low-latency-00.pdf
>
>
>
> As this slide deck notes, within Google we negotiate 5ms for delayed ACKs=
.
>
>
>
> cheers,
>
> neal
>
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From nobody Thu Dec 15 02:21:11 2016
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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com>
To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>, "zhangyali (D)" <zhangyali369@huawei.com>
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
Organization: The University of Aberdeen is a charity registered in Scotland,  No SC013683.
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiDnrZTlpI06IOetlOWkjTogQSBxdWVzdGlvbiBh?= =?utf-8?q?bout_Delayed_ACK_and_RTO?=
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Just adding a few more comments here, because there has been much talk 
about how desirable it is for the RTO to converge to the RTT, and I 
think we need to be clear.

The TCP RTO has two distinct functions:

(1)  the RTO is of course the method of last resort for the recovery of 
a lost segment (the final segment of a burst, following persistent loss, 
etc).

(2) Second the RTO is  method to detect a path failure - where something 
has changed significantly. As such, triggering a conservative (1 or 3 
seconds) RTO implies that state about the path needs to be reset (RTT, 
perhaps invoking PMTUD, congestion state variables, etc, possibly in the 
future falling back from using ECN, etc)

For the first function, there are many ways to detect lost segments 
(Dupacks, probe packets, recovery timers, etc). All of these act faster 
than a conservatively set timeout. They also do not need to erase 
collected path state, they can simply re-transmit the lost segments.

An RTO close to the RTT will be disastrous for some paths. This is one 
of the reasons why people have deployed PEPs to support radio technology 
- we should not be encouraging this. An RTO set close to the RTT can 
produce complex interactions when the path characteristics change (which 
does happen with propagation impairments and radio resource management, 
on wireless, microwave satellite and other paths, and also with 
middleboxes that attempt to control capacity/volume usage). This type of 
path is not going away, talk of wireless links with top speeds of Gbps 
for 5G will likely be accompanied by very large variation in path 
characteristics - at the other extreme people in large parts of the 
world are still with kbps technology. If the timer only recovers 
packets, it would simply result in retransmission, if it triggers other 
probing that would be unfortunate. I think the IETF needs to ensure our 
specs work for all people.

What I am saying is that the second function does not need a timeout 
period near the RTT. Robust protocols have been designed to have a 
conservative RTO timer that is set to much more than a RTT, designed to 
detect an unresponsive endpoint or path problem. The IETF has argued - 
and to my knowledge still argues that Min_RTO should be one second, 
which I think reflects this position. If the loss recovery is efficient, 
this will trigger only infrequently (e.g., persistent loss of one 
packet). I think this is the correct way to design TCP. It motivates 
that we should actually strive for a one second (or so) RTO.

Gorry


On 15/12/2016 08:24, Yoshifumi Nishida wrote:
> A question would be when TCP sent odd number of packets, how it can
> know whether another packets will come from upper layer very soon or
> not. Also, I'm not very sure yet if the probability is so obvious. I
> can agree that RTO can get close to raw rtt value under bulk transfer.
> But, I am guessing not so many apps change the behavior from bulk
> transfer to spontaneous small burst transfer.
> --
> Yoshi
>
>
> On Wed, Dec 14, 2016 at 7:08 PM, zhangyali (D) <zhangyali369@huawei.com> wrote:
>> Please allow me to add one more point. If the probability of odd number of
>> packets occurring  spurious RTO is so outstanding, why nobody try to solve
>> this problem?
>>
>>
>>
>> 发件人: tcpm [mailto:tcpm-bounces@ietf.org] 代表 zhangyali (D)
>> 发送时间: 2016年12月15日 9:45
>> 收件人: Neal Cardwell <ncardwell@google.com>
>> 抄送: tcpm@ietf.org
>> 主题: [tcpm] 答复: 答复: A question about Delayed ACK and RTO
>>
>>
>>
>>> Yes, there should only be a delayed ACK at the end of an application chunk
>>> if there is an odd number of packets. But we would expect roughly half of
>>> application chunks to have an odd number of packets. Though the proportion
>>> is probably higher than that, since many application chunks are just one
>>> packet (e.g. an HTTP or RPC request or response).
>>
>>
>>
>> I agree with you that odd number of packets may occur spurious RTO more
>> easily, but I think for the flows owning just one packet should not be
>> affected by delayed ACK. AFAIK, slow-start stage will begin with one packet,
>> and sender will send two packets after it receives an ACK. If the first
>> packet is obstructed by the delayed ACK, the ‘clock algorithm’ will be
>> broken down. So receiver should judge if this packet is the first one in the
>> slow-start stage, if yes, send the ack immediately once receiving the
>> packet.
>>
>>
>>
>> Yali
>>
>>
>>
>> 发件人: Neal Cardwell [mailto:ncardwell@google.com]
>> 发送时间: 2016年12月14日 21:24
>> 收件人: zhangyali (D) <zhangyali369@huawei.com>
>> 抄送: tcpm@ietf.org
>> 主题: Re: 答复: [tcpm] A question about Delayed ACK and RTO
>>
>>
>>
>> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) <zhangyali369@huawei.com>
>> wrote:
>>
>> Hi Neal,
>>
>>
>>
>> Thanks for your providing request information.
>>
>>
>>
>> About the delay of delayed ACK, I found another clue in a SIGCOMM paper in
>> 1988. In the page 14, a sentence is “The 4.5KBps senders were talking to
>> 4.3BSD receivers which would delay an ack until 35% of the window was filled
>> or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on average).”
>> There is no reference about the 200 ms, so I guess this is the particular
>> delay appears for the first time.
>>
>> I try to calculate the value based on delayed packet numbers, packet size
>> and sending rate. In this case, the delayed packet number is 7, the packet
>> size is 576Byte (refer to RFC879 in 1983), and sundering rate is 4.5Bps. The
>> value is 896 ms! Longer than 200 ms.
>>
>>
>>
>>> . However, it's quite easy for a bulk transfer to never have any delayed
>>> ACKs for most of its lifetime, during which the RTO gradually converges
>>> toward the raw RTT value. Then when there is suddenly a delayed ACK, there
>>> can be a spurious RTO.
>>
>> I am not very sure if I see your point. I think the key point is the RTT
>> variation will be weakened when packets are sent in a burst (the consequence
>> of delayed ACK).   And the spurious  could only happen for the last few
>> packets whose number is smaller than delayed packets, right?
>>
>>
>>
>> Yes, there should only be a delayed ACK at the end of an application chunk
>> if there is an odd number of packets. But we would expect roughly half of
>> application chunks to have an odd number of packets. Though the proportion
>> is probably higher than that, since many application chunks are just one
>> packet (e.g. an HTTP or RPC request or response).
>>
>>
>>
>>
>>
>> About your proposal about negotiation of delayed ACK, a potential problem is
>> that RTO will be stretched than before because you add extra delay. As you
>> have said, most of the packets will not exceed RTO even host enable delayed
>> ACK for most large flows, but the retransmission will be delayed (i.e.,
>> 5ms)also once one packet is lost.
>>
>>
>>
>> The basic idea of the proposal is to tweak the RTO calculation and turn a
>> previously existing, historically motivated 200ms fixed "slop factor" into a
>> dynamically negotiated 5ms "slop factor". In our experience that is almost
>> always a win.
>>
>>
>>
>> neal
>>
>>
>>
>>
>>
>> Best,
>>
>> Yali
>>
>> 发件人: Neal Cardwell [mailto:ncardwell@google.com]
>> 发送时间: 2016年12月13日 23:18
>> 收件人: zhangyali (D) <zhangyali369@huawei.com>
>> 抄送: tcpm@ietf.org
>> 主题: Re: [tcpm] A question about Delayed ACK and RTO
>>
>>
>>
>> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) <zhangyali369@huawei.com>
>> wrote:
>>
>> Hi all,
>>
>>
>>
>> Recent days, I am doing some simulation about TCP performance in NS3. I
>> found a phenomenon is a default setting of TCP’s delayed ACK is two packets
>> or 200ms. I am wondering if this setting is accord with  some RFCs in IETF.
>> But After I referred to some RFCs, I just found some restrictions, such as,
>> the delay must be less than 0.5ms (RFC1122).
>>
>>
>>
>> I believe the 200ms figure is due to historical reasons. AFAIK it's due to
>> the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume 2",
>> section 25.4 and figure 25.7, which describes the 200ms delayed ACK timer).
>> Then this value was inherited by other widely-deployed OSes.
>>
>>
>>
>>
>>
>> I think the delayed ACK has a close relationship with RTO. Take an extreme
>> scenario, if the delay is longer than RTO, many packets will be
>> retransmitted, which will waste many network resources.
>>
>>
>>
>> Yes, exactly. In theory, the RTO tries to be adaptive enough to measure any
>> RTT variations caused by delayed ACKs, and increase the RTO in response to
>> this. However, it's quite easy for a bulk transfer to never have any delayed
>> ACKs for most of its lifetime, during which the RTO gradually converges
>> toward the raw RTT value. Then when there is suddenly a delayed ACK, there
>> can be a spurious RTO.
>>
>>
>>
>> To avoid that, many TCP stacks (at least the major open source OSes) have a
>> hard-coded 200ms "slop factor" or "fudge factor", to try to never let their
>> estimate of RTT variation fall below that 200ms value, to avoid this effect.
>>
>>
>>
>> So I want to know do we have some RFCs have given the exact value of both?
>> And if we permit any TCP stack to set them freely, what is the mechanism to
>> balance the mismatch between RTO and delayed ACK?
>>
>>
>>
>> I'm not aware of RFC specifications for exact values of both (delayed ACK
>> and RTO). However, the historical precedent is very strong, and the 200ms
>> delayed ACK value was very pronounced in Internet traces at least as
>> recently as 2011, when I last looked at the effect. (Probably others have
>> more recent data points for the prevalence of 200ms delayed ACKs.)
>>
>>
>>
>> At IETF 97 our team at Google presented some features we use for internal
>> TCP traffic at Google, where the endpoints can negotiate the specific
>> constant to use for the maximum delayed ACK from the receiver and the
>> corresponding minimum RTT delay variation for budgeting in the RTO at the
>> sender:
>>
>>
>>
>>
>> https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf
>>
>>
>>
>> As this slide deck notes, within Google we negotiate 5ms for delayed ACKs.
>>
>>
>>
>> cheers,
>>
>> neal
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From nobody Thu Dec 15 03:25:26 2016
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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com> <0a234a82-5333-e3f7-cfa9-812d93144c54@erg.abdn.ac.uk>
From: Alejandro Popovsky <apopov@palermo.edu>
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Tcp would benefit much from and 'end of app write call' signal or may be 
'end of sender buffer data' signal.

TLP, adaptive RTO at the end of bursts, end of burst reordering, and may 
be some other mechanisms
were introduced to help with this issue.

May be the addition of an option for this signal may help, or may be the 
reinterpretation of the push flag.
This would need the modification of both senders and receivers, so the 
effect of not having a modified
receiver getting the flag should be contemplated.

Alejandro.


On 15/12/2016 7:20 a. m., Gorry Fairhurst wrote:
>
> Just adding a few more comments here, because there has been much talk 
> about how desirable it is for the RTO to converge to the RTT, and I 
> think we need to be clear.
>
> The TCP RTO has two distinct functions:
>
> (1)  the RTO is of course the method of last resort for the recovery 
> of a lost segment (the final segment of a burst, following persistent 
> loss, etc).
>
> (2) Second the RTO is  method to detect a path failure - where 
> something has changed significantly. As such, triggering a 
> conservative (1 or 3 seconds) RTO implies that state about the path 
> needs to be reset (RTT, perhaps invoking PMTUD, congestion state 
> variables, etc, possibly in the future falling back from using ECN, etc)
>
> For the first function, there are many ways to detect lost segments 
> (Dupacks, probe packets, recovery timers, etc). All of these act 
> faster than a conservatively set timeout. They also do not need to 
> erase collected path state, they can simply re-transmit the lost 
> segments.
>
> An RTO close to the RTT will be disastrous for some paths. This is one 
> of the reasons why people have deployed PEPs to support radio 
> technology - we should not be encouraging this. An RTO set close to 
> the RTT can produce complex interactions when the path characteristics 
> change (which does happen with propagation impairments and radio 
> resource management, on wireless, microwave satellite and other paths, 
> and also with middleboxes that attempt to control capacity/volume 
> usage). This type of path is not going away, talk of wireless links 
> with top speeds of Gbps for 5G will likely be accompanied by very 
> large variation in path characteristics - at the other extreme people 
> in large parts of the world are still with kbps technology. If the 
> timer only recovers packets, it would simply result in retransmission, 
> if it triggers other probing that would be unfortunate. I think the 
> IETF needs to ensure our specs work for all people.
>
> What I am saying is that the second function does not need a timeout 
> period near the RTT. Robust protocols have been designed to have a 
> conservative RTO timer that is set to much more than a RTT, designed 
> to detect an unresponsive endpoint or path problem. The IETF has 
> argued - and to my knowledge still argues that Min_RTO should be one 
> second, which I think reflects this position. If the loss recovery is 
> efficient, this will trigger only infrequently (e.g., persistent loss 
> of one packet). I think this is the correct way to design TCP. It 
> motivates that we should actually strive for a one second (or so) RTO.
>
> Gorry
>
>
> On 15/12/2016 08:24, Yoshifumi Nishida wrote:
>> A question would be when TCP sent odd number of packets, how it can
>> know whether another packets will come from upper layer very soon or
>> not. Also, I'm not very sure yet if the probability is so obvious. I
>> can agree that RTO can get close to raw rtt value under bulk transfer.
>> But, I am guessing not so many apps change the behavior from bulk
>> transfer to spontaneous small burst transfer.
>> -- 
>> Yoshi
>>
>>
>> On Wed, Dec 14, 2016 at 7:08 PM, zhangyali (D) 
>> <zhangyali369@huawei.com> wrote:
>>> Please allow me to add one more point. If the probability of odd 
>>> number of
>>> packets occurring  spurious RTO is so outstanding, why nobody try to 
>>> solve
>>> this problem?
>>>
>>>
>>>
>>> 发件人: tcpm [mailto:tcpm-bounces@ietf.org] 代表 zhangyali (D)
>>> 发送时间: 2016年12月15日 9:45
>>> 收件人: Neal Cardwell <ncardwell@google.com>
>>> 抄送: tcpm@ietf.org
>>> 主题: [tcpm] 答复: 答复: A question about Delayed ACK and RTO
>>>
>>>
>>>
>>>> Yes, there should only be a delayed ACK at the end of an 
>>>> application chunk
>>>> if there is an odd number of packets. But we would expect roughly 
>>>> half of
>>>> application chunks to have an odd number of packets. Though the 
>>>> proportion
>>>> is probably higher than that, since many application chunks are 
>>>> just one
>>>> packet (e.g. an HTTP or RPC request or response).
>>>
>>>
>>>
>>> I agree with you that odd number of packets may occur spurious RTO more
>>> easily, but I think for the flows owning just one packet should not be
>>> affected by delayed ACK. AFAIK, slow-start stage will begin with one 
>>> packet,
>>> and sender will send two packets after it receives an ACK. If the first
>>> packet is obstructed by the delayed ACK, the ‘clock algorithm’ will be
>>> broken down. So receiver should judge if this packet is the first 
>>> one in the
>>> slow-start stage, if yes, send the ack immediately once receiving the
>>> packet.
>>>
>>>
>>>
>>> Yali
>>>
>>>
>>>
>>> 发件人: Neal Cardwell [mailto:ncardwell@google.com]
>>> 发送时间: 2016年12月14日 21:24
>>> 收件人: zhangyali (D) <zhangyali369@huawei.com>
>>> 抄送: tcpm@ietf.org
>>> 主题: Re: 答复: [tcpm] A question about Delayed ACK and RTO
>>>
>>>
>>>
>>> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D) 
>>> <zhangyali369@huawei.com>
>>> wrote:
>>>
>>> Hi Neal,
>>>
>>>
>>>
>>> Thanks for your providing request information.
>>>
>>>
>>>
>>> About the delay of delayed ACK, I found another clue in a SIGCOMM 
>>> paper in
>>> 1988. In the page 14, a sentence is “The 4.5KBps senders were 
>>> talking to
>>> 4.3BSD receivers which would delay an ack until 35% of the window 
>>> was filled
>>> or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on 
>>> average).”
>>> There is no reference about the 200 ms, so I guess this is the 
>>> particular
>>> delay appears for the first time.
>>>
>>> I try to calculate the value based on delayed packet numbers, packet 
>>> size
>>> and sending rate. In this case, the delayed packet number is 7, the 
>>> packet
>>> size is 576Byte (refer to RFC879 in 1983), and sundering rate is 
>>> 4.5Bps. The
>>> value is 896 ms! Longer than 200 ms.
>>>
>>>
>>>
>>>> . However, it's quite easy for a bulk transfer to never have any 
>>>> delayed
>>>> ACKs for most of its lifetime, during which the RTO gradually 
>>>> converges
>>>> toward the raw RTT value. Then when there is suddenly a delayed 
>>>> ACK, there
>>>> can be a spurious RTO.
>>>
>>> I am not very sure if I see your point. I think the key point is the 
>>> RTT
>>> variation will be weakened when packets are sent in a burst (the 
>>> consequence
>>> of delayed ACK).   And the spurious  could only happen for the last few
>>> packets whose number is smaller than delayed packets, right?
>>>
>>>
>>>
>>> Yes, there should only be a delayed ACK at the end of an application 
>>> chunk
>>> if there is an odd number of packets. But we would expect roughly 
>>> half of
>>> application chunks to have an odd number of packets. Though the 
>>> proportion
>>> is probably higher than that, since many application chunks are just 
>>> one
>>> packet (e.g. an HTTP or RPC request or response).
>>>
>>>
>>>
>>>
>>>
>>> About your proposal about negotiation of delayed ACK, a potential 
>>> problem is
>>> that RTO will be stretched than before because you add extra delay. 
>>> As you
>>> have said, most of the packets will not exceed RTO even host enable 
>>> delayed
>>> ACK for most large flows, but the retransmission will be delayed (i.e.,
>>> 5ms)also once one packet is lost.
>>>
>>>
>>>
>>> The basic idea of the proposal is to tweak the RTO calculation and 
>>> turn a
>>> previously existing, historically motivated 200ms fixed "slop 
>>> factor" into a
>>> dynamically negotiated 5ms "slop factor". In our experience that is 
>>> almost
>>> always a win.
>>>
>>>
>>>
>>> neal
>>>
>>>
>>>
>>>
>>>
>>> Best,
>>>
>>> Yali
>>>
>>> 发件人: Neal Cardwell [mailto:ncardwell@google.com]
>>> 发送时间: 2016年12月13日 23:18
>>> 收件人: zhangyali (D) <zhangyali369@huawei.com>
>>> 抄送: tcpm@ietf.org
>>> 主题: Re: [tcpm] A question about Delayed ACK and RTO
>>>
>>>
>>>
>>> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D) 
>>> <zhangyali369@huawei.com>
>>> wrote:
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Recent days, I am doing some simulation about TCP performance in NS3. I
>>> found a phenomenon is a default setting of TCP’s delayed ACK is two 
>>> packets
>>> or 200ms. I am wondering if this setting is accord with  some RFCs 
>>> in IETF.
>>> But After I referred to some RFCs, I just found some restrictions, 
>>> such as,
>>> the delay must be less than 0.5ms (RFC1122).
>>>
>>>
>>>
>>> I believe the 200ms figure is due to historical reasons. AFAIK it's 
>>> due to
>>> the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated Volume 
>>> 2",
>>> section 25.4 and figure 25.7, which describes the 200ms delayed ACK 
>>> timer).
>>> Then this value was inherited by other widely-deployed OSes.
>>>
>>>
>>>
>>>
>>>
>>> I think the delayed ACK has a close relationship with RTO. Take an 
>>> extreme
>>> scenario, if the delay is longer than RTO, many packets will be
>>> retransmitted, which will waste many network resources.
>>>
>>>
>>>
>>> Yes, exactly. In theory, the RTO tries to be adaptive enough to 
>>> measure any
>>> RTT variations caused by delayed ACKs, and increase the RTO in 
>>> response to
>>> this. However, it's quite easy for a bulk transfer to never have any 
>>> delayed
>>> ACKs for most of its lifetime, during which the RTO gradually converges
>>> toward the raw RTT value. Then when there is suddenly a delayed ACK, 
>>> there
>>> can be a spurious RTO.
>>>
>>>
>>>
>>> To avoid that, many TCP stacks (at least the major open source OSes) 
>>> have a
>>> hard-coded 200ms "slop factor" or "fudge factor", to try to never 
>>> let their
>>> estimate of RTT variation fall below that 200ms value, to avoid this 
>>> effect.
>>>
>>>
>>>
>>> So I want to know do we have some RFCs have given the exact value of 
>>> both?
>>> And if we permit any TCP stack to set them freely, what is the 
>>> mechanism to
>>> balance the mismatch between RTO and delayed ACK?
>>>
>>>
>>>
>>> I'm not aware of RFC specifications for exact values of both 
>>> (delayed ACK
>>> and RTO). However, the historical precedent is very strong, and the 
>>> 200ms
>>> delayed ACK value was very pronounced in Internet traces at least as
>>> recently as 2011, when I last looked at the effect. (Probably others 
>>> have
>>> more recent data points for the prevalence of 200ms delayed ACKs.)
>>>
>>>
>>>
>>> At IETF 97 our team at Google presented some features we use for 
>>> internal
>>> TCP traffic at Google, where the endpoints can negotiate the specific
>>> constant to use for the maximum delayed ACK from the receiver and the
>>> corresponding minimum RTT delay variation for budgeting in the RTO 
>>> at the
>>> sender:
>>>
>>>
>>>
>>>
>>> https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf 
>>>
>>>
>>>
>>>
>>> As this slide deck notes, within Google we negotiate 5ms for delayed 
>>> ACKs.
>>>
>>>
>>>
>>> cheers,
>>>
>>> neal
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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I agree with Gorry's take on variable path characteristics and all.=20

> The TCP RTO has two distinct functions:
>=20
> (1)  the RTO is of course the method of last resort for the recovery
> of a lost segment (the final segment of a burst, following
> persistent loss, etc).
>=20
> (2) Second the RTO is  method to detect a path failure - where
> something has changed significantly.

I think this is an accident.  And, the way we should handle it is to
not conflate such disparate functions into one mechanism.

The RTO is a ***retransmission*** timeout.  It's original function
was to decide when to trigger resending data that has not been ACKed
because after this time period chances are the packet was lost
(i.e., case (1)).

Independently we do need to understand path failure.  And, part of
detecting path failure may well be detecting that packets are not
being delivered.  However, this is a different issue than figuring
out when to resend a segment.  The fact that we use the RTO for this
function boils down to laziness in re-using something that was
laying around instead of doing it right.

By trying to shoehorn both functions into one mechanism we are
***guaranteeing*** that the mechanism will be lousy (for at least
one of the tasks and probably both!).  This thread perfectly
illustrates this...  If we were to use the RTO for both functions
then it would be impossible to reconcile Gorry's notion that path
failure happens on long timescales and so RTOs on the order of a
second are needed with Neal's world where he wants to trigger
retransmissions three orders of magnitude sooner.  Path failure may
*leverage* loss detection, but it needn't be the exact same such
that it *hampers* loss detection.

IMHO. :-)

allman


=2D-
http://www.icir.org/mallman/
@mallman_icsi




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From nobody Thu Dec 15 07:30:10 2016
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To: Alejandro Popovsky <apopov@palermo.edu>, tcpm@ietf.org
References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com> <0a234a82-5333-e3f7-cfa9-812d93144c54@erg.abdn.ac.uk> <a5af440e-28da-f193-14f4-794cb04804d3@palermo.edu>
From: Joe Touch <touch@isi.edu>
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On 12/15/2016 3:25 AM, Alejandro Popovsky wrote:
> Tcp would benefit much from and 'end of app write call' signal or may
> be 'end of sender buffer data' signal.

 In the current TCP API, there is absolutely no correlation between app
SEND calls and any notion TCP might have of boundaries - TCP presents a
bytestream service only. There would be many consequences to changing
that API, but it might be useful to note that this was once considered.
Just two days ago, Bob Braden and I were discussing TCP v3 (before the
current one  - as documented in IEN  21), which did have this sort of
signal, but it was carried into the packet as begin/end (of "letter") flags.

Note that his behavior is a consequence of the interaction between
delayed ACKs and a lack of application data, and happens any time the
app layer "starves" TCP. It was well studied in the 90's in its
interaction with persistent web connections. It isn't really related to
RTO per se.

Joe


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From: "Jakob Heitz (jheitz)" <jheitz@cisco.com>
To: Joe Touch <touch@isi.edu>, Alejandro Popovsky <apopov@palermo.edu>, "tcpm@ietf.org" <tcpm@ietf.org>
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From nobody Thu Dec 15 15:17:25 2016
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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com> <0a234a82-5333-e3f7-cfa9-812d93144c54@erg.abdn.ac.uk> <a5af440e-28da-f193-14f4-794cb04804d3@palermo.edu> <45f0034f-70a2-2aa3-d87b-c7288a51ad77@isi.edu>
From: Alejandro Popovsky <apopov@palermo.edu>
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On the second option (no more data to send) there would be no need to
change the API. The tcp sender tcp would generate the signal automatically
with the last segment. And the receiver could just use the info to 
adjust the delay
ack timer for that segment.

And in order for the extra ack not to produce an extra cwnd growth
the sender should follow the recommendation of rfc5681, page 6.

Alejandro.

On 15/12/2016 12:29 p. m., Joe Touch wrote:
>
> On 12/15/2016 3:25 AM, Alejandro Popovsky wrote:
>> Tcp would benefit much from and 'end of app write call' signal or may
>> be 'end of sender buffer data' signal.
>   In the current TCP API, there is absolutely no correlation between app
> SEND calls and any notion TCP might have of boundaries - TCP presents a
> bytestream service only. There would be many consequences to changing
> that API, but it might be useful to note that this was once considered.
> Just two days ago, Bob Braden and I were discussing TCP v3 (before the
> current one  - as documented in IEN  21), which did have this sort of
> signal, but it was carried into the packet as begin/end (of "letter") flags.
>
> Note that his behavior is a consequence of the interaction between
> delayed ACKs and a lack of application data, and happens any time the
> app layer "starves" TCP. It was well studied in the 90's in its
> interaction with persistent web connections. It isn't really related to
> RTO per se.
>
> Joe
>


From nobody Thu Dec 15 16:24:36 2016
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Date: Thu, 15 Dec 2016 16:24:33 -0800
From: hiren panchasara <hiren@strugglingcoder.info>
To: Mark Allman <mallman@icir.org>
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Subject: Re: [tcpm] ??: ??: ??: A question about Delayed ACK and RTO
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On 12/15/16 at 09:52P, Mark Allman wrote:
>=20
> I agree with Gorry's take on variable path characteristics and all.=20
>=20
> > The TCP RTO has two distinct functions:
> >=20
> > (1)  the RTO is of course the method of last resort for the recovery
> > of a lost segment (the final segment of a burst, following
> > persistent loss, etc).
> >=20
> > (2) Second the RTO is  method to detect a path failure - where
> > something has changed significantly.
>=20
> I think this is an accident.  And, the way we should handle it is to
> not conflate such disparate functions into one mechanism.
>=20
> The RTO is a ***retransmission*** timeout.  It's original function
> was to decide when to trigger resending data that has not been ACKed
> because after this time period chances are the packet was lost
> (i.e., case (1)).
>=20
> Independently we do need to understand path failure.  And, part of
> detecting path failure may well be detecting that packets are not
> being delivered.  However, this is a different issue than figuring
> out when to resend a segment.  The fact that we use the RTO for this
> function boils down to laziness in re-using something that was
> laying around instead of doing it right.
>=20
> By trying to shoehorn both functions into one mechanism we are
> ***guaranteeing*** that the mechanism will be lousy (for at least
> one of the tasks and probably both!).  This thread perfectly
> illustrates this...  If we were to use the RTO for both functions
> then it would be impossible to reconcile Gorry's notion that path
> failure happens on long timescales and so RTOs on the order of a
> second are needed with Neal's world where he wants to trigger
> retransmissions three orders of magnitude sooner.  Path failure may
> *leverage* loss detection, but it needn't be the exact same such
> that it *hampers* loss detection.

Interesting.
How can these 2 be differentiated? Has that been discussed/researched in
IETF or elsewhere? OR can tcp borrow such a scheme from somewhere else?

Cheers,
Hiren

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> How can these 2 be differentiated?

Easy.

Retransmissions are triggered by the scheme outlined in RFC 6298.

Path failure is defined as not getting any packets through in 10sec.

There.  Differentiated.

I am not suggesting that is smart.  Or, the best way.  But, making
these two things triggered by their own criteria that is tuned to
the each specific task is trivial.

allman




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To: "zhangyali (D)" <zhangyali369@huawei.com>, Joe Touch <touch@isi.edu>, "Neal Cardwell" <ncardwell@google.com>, "Jakob Heitz (jheitz)" <jheitz@cisco.com>
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On 12/15/2016 3:13 PM, Alejandro Popovsky wrote:
> On the second option (no more data to send) there would be no need to
> change the API. The tcp sender tcp would generate the signal
> automatically
> with the last segment. And the receiver could just use the info to
> adjust the delay
> ack timer for that segment.

The two problems:
    - the sending app can't signal "last segment" of a data chunk to
TCP. All it can say is "close", after which it can't send data anymore.
    - the sending-side TCP can't signal "last segment" to the receiver
either. The only flag there is FIN, which similarly inhibits later data
on the connection in that direction.

Joe

>
> And in order for the extra ack not to produce an extra cwnd growth
> the sender should follow the recommendation of rfc5681, page 6.
>
> Alejandro.
>
> On 15/12/2016 12:29 p. m., Joe Touch wrote:
>>
>> On 12/15/2016 3:25 AM, Alejandro Popovsky wrote:
>>> Tcp would benefit much from and 'end of app write call' signal or may
>>> be 'end of sender buffer data' signal.
>>   In the current TCP API, there is absolutely no correlation between app
>> SEND calls and any notion TCP might have of boundaries - TCP presents a
>> bytestream service only. There would be many consequences to changing
>> that API, but it might be useful to note that this was once considered.
>> Just two days ago, Bob Braden and I were discussing TCP v3 (before the
>> current one  - as documented in IEN  21), which did have this sort of
>> signal, but it was carried into the packet as begin/end (of "letter")
>> flags.
>>
>> Note that his behavior is a consequence of the interaction between
>> delayed ACKs and a lack of application data, and happens any time the
>> app layer "starves" TCP. It was well studied in the 90's in its
>> interaction with persistent web connections. It isn't really related to
>> RTO per se.
>>
>> Joe
>>


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To: Praveen Balasubramanian <pravb@microsoft.com>, Alejandro Popovsky <apopov@palermo.edu>, "tcpm@ietf.org" <tcpm@ietf.org>
References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com> <0a234a82-5333-e3f7-cfa9-812d93144c54@erg.abdn.ac.uk> <a5af440e-28da-f193-14f4-794cb04804d3@palermo.edu> <45f0034f-70a2-2aa3-d87b-c7288a51ad77@isi.edu> <48aa99d0-113e-edda-94f7-48d81bea59fd@palermo.edu> <BN1PR03MB0085597E6BAF5F2F878A71FB69C0@BN1PR03MB008.namprd03.prod.outlook.com>
From: Joe Touch <touch@isi.edu>
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On 12/15/2016 5:13 PM, Praveen Balasubramanian wrote:
> FWIW the windows TCP implementation preserves the notion of application send() message boundary by setting the PSH bit only on the last segment generated from the posted buffer.
TCP isn't required to allow the app to decide when to set the PSH bit
(it's a MAY in the SEND call). It can coalesce multiple PSH bits to send
a larger segment.

However, it's also explicitly not a record boundary. PUSH is really just
to avoid deadlock between the app and TCP on the send side.

>  Unfortunately other stacks seem to set PSH on all segments making any receiver optimization of immediate ACK on PSH impossible, because that would effectively turn off delayed ACKs. That said, with ACK stretching, LRO etc. being very common maybe it is time for TCP implementations to turn off delayed ACKs across the board except for request-response applications.

There are many ways this could be fixed but they all require going
outside the current spec.

Joe

>
> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Alejandro Popovsky
> Sent: Thursday, December 15, 2016 3:13 PM
> To: Joe Touch <touch@isi.edu>; tcpm@ietf.org
> Subject: Re: [tcpm] µªÎ`: µªÎ`: µªÎ`: A question about Delayed ACK and RTO
>
> On the second option (no more data to send) there would be no need to change the API. The tcp sender tcp would generate the signal automatically with the last segment. And the receiver could just use the info to adjust the delay ack timer for that segment.
>
> And in order for the extra ack not to produce an extra cwnd growth the sender should follow the recommendation of rfc5681, page 6.
>
> Alejandro.
>
> On 15/12/2016 12:29 p. m., Joe Touch wrote:
>> On 12/15/2016 3:25 AM, Alejandro Popovsky wrote:
>>> Tcp would benefit much from and 'end of app write call' signal or may 
>>> be 'end of sender buffer data' signal.
>>   In the current TCP API, there is absolutely no correlation between 
>> app SEND calls and any notion TCP might have of boundaries - TCP 
>> presents a bytestream service only. There would be many consequences 
>> to changing that API, but it might be useful to note that this was once considered.
>> Just two days ago, Bob Braden and I were discussing TCP v3 (before the 
>> current one  - as documented in IEN  21), which did have this sort of 
>> signal, but it was carried into the packet as begin/end (of "letter") flags.
>>
>> Note that his behavior is a consequence of the interaction between 
>> delayed ACKs and a lack of application data, and happens any time the 
>> app layer "starves" TCP. It was well studied in the 90's in its 
>> interaction with persistent web connections. It isn't really related 
>> to RTO per se.
>>
>> Joe
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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To: Praveen Balasubramanian <pravb@microsoft.com>, "zhangyali (D)" <zhangyali369@huawei.com>, Neal Cardwell <ncardwell@google.com>, "Jakob Heitz (jheitz)" <jheitz@cisco.com>
References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <F9D5899E-435E-417A-B5D0-561183AFFC92@cisco.com> <CADVnQy=2vuxwybswBgzEEjinp92DRKFZgphaVdBGvguJBirbJw@mail.gmail.com> <37069250-13c5-7692-f137-0f441d57a90c@isi.edu> <A747A0713F56294D8FBE33E5C6B8F5815F5460C4@DGGEMA502-MBS.china.huawei.com> <BN1PR03MB0080450F8ED7E868BC5466AB69C0@BN1PR03MB008.namprd03.prod.outlook.com>
From: Joe Touch <touch@isi.edu>
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On 12/15/2016 5:21 PM, Praveen Balasubramanian wrote:
>
> While operators could tweak the settings, having TCP perform well out
> of box in low latency environments is very useful.
>

IMO, the only rule "out of the box" is that TCP ought to *work*.

Every other rule ends up compromising "works" for "works well", which is
a mistake IMO. It just makes the whole Internet brittle in the future,
WHEN (not if) your assumptions change.

Joe

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    <div class="moz-cite-prefix">On 12/15/2016 5:21 PM, Praveen
      Balasubramanian wrote:<br>
    </div>
    <blockquote
cite="mid:BN1PR03MB0080450F8ED7E868BC5466AB69C0@BN1PR03MB008.namprd03.prod.outlook.com"
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style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext;mso-fareast-language:EN-US">While
            operators could tweak the settings, having TCP perform well
            out of box in low latency environments is very useful. </span></p>
      </div>
    </blockquote>
    <br>
    IMO, the only rule "out of the box" is that TCP ought to *work*.<br>
    <br>
    Every other rule ends up compromising "works" for "works well",
    which is a mistake IMO. It just makes the whole Internet brittle in
    the future, WHEN (not if) your assumptions change.<br>
    <br>
    Joe<br>
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From nobody Thu Dec 15 18:07:57 2016
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To: Mark Allman <mallman@icir.org>
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On 12/15/16 at 07:48P, Mark Allman wrote:
>=20
> > How can these 2 be differentiated?
>=20
> Easy.
>=20
> Retransmissions are triggered by the scheme outlined in RFC 6298.
>=20
> Path failure is defined as not getting any packets through in 10sec.
>=20
> There.  Differentiated.
>=20
> I am not suggesting that is smart.  Or, the best way.  But, making
> these two things triggered by their own criteria that is tuned to
> the each specific task is trivial.

Thank you. I think I am getting the point you are trying to make and I
agree that we need to separate them but I thought coming up wiht that
'10sec' was the trickiest part. May be I am missing something or
over-thinking this. :-)

RTO, at least, gets derived from something realistic. I guess 'path
failure' needs to come up with something like that?

Cheers,
Hiren

PS: PMTUD implementation (at least in FreeBSD) comes to mind when I
think about detecting anomaly beyond just 'packets are getting lost'.
But thats again based on RTO in FreeBSD. Though I think we are talking
about more fatal failures here.

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From: "zhangyali (D)" <zhangyali369@huawei.com>
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiAg562U5aSNOiDnrZTlpI06IOetlOWkjTogQSBx?= =?utf-8?q?uestion_about_Delayed_ACK_and_RTO?=
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On 16/12/2016 07:47, zhangyali (D) wrote:
> I identify with your distinction function of RTO. I really agree that we should differentiate the effect caused by network congestion or path failure, and take different actions, such as, more aggressive action to the second cases. But now, we just have RTO as the last method to retransmission the segment that is lost or just be delayed if we do not have extra segments or window to trigger dupack. So another retansmission detection method faster than timeout and more effective should be thinked about.
>
> Yali
On this I also agree -  Tail Loss has become a problem for 
latency-sensitive shorter traffic bursts, amd we need widely deployed 
mechanisms to deal with this. There have been proposals in TCPM to add 
mechanisms that would help and this seems worth looking at.

Gorry
> -----邮件原件-----
> 发件人: Gorry Fairhurst [mailto:gorry@erg.abdn.ac.uk]
> 发送时间: 2016年12月15日 18:21
> 收件人: Yoshifumi Nishida<nishida@sfc.wide.ad.jp>; zhangyali (D)<zhangyali369@huawei.com>
> 抄送: tcpm@ietf.org
> 主题: Re: [tcpm] 答复: 答复: 答复: A question about Delayed ACK and RTO
>
>
> Just adding a few more comments here, because there has been much talk about how desirable it is for the RTO to converge to the RTT, and I think we need to be clear.
>
> The TCP RTO has two distinct functions:
>
> (1)  the RTO is of course the method of last resort for the recovery of a lost segment (the final segment of a burst, following persistent loss, etc).
>
> (2) Second the RTO is  method to detect a path failure - where something has changed significantly. As such, triggering a conservative (1 or 3
> seconds) RTO implies that state about the path needs to be reset (RTT, perhaps invoking PMTUD, congestion state variables, etc, possibly in the future falling back from using ECN, etc)
>
> For the first function, there are many ways to detect lost segments (Dupacks, probe packets, recovery timers, etc). All of these act faster than a conservatively set timeout. They also do not need to erase collected path state, they can simply re-transmit the lost segments.
>
> An RTO close to the RTT will be disastrous for some paths. This is one of the reasons why people have deployed PEPs to support radio technology
> - we should not be encouraging this. An RTO set close to the RTT can produce complex interactions when the path characteristics change (which does happen with propagation impairments and radio resource management, on wireless, microwave satellite and other paths, and also with middleboxes that attempt to control capacity/volume usage). This type of path is not going away, talk of wireless links with top speeds of Gbps for 5G will likely be accompanied by very large variation in path characteristics - at the other extreme people in large parts of the world are still with kbps technology. If the timer only recovers packets, it would simply result in retransmission, if it triggers other probing that would be unfortunate. I think the IETF needs to ensure our specs work for all people.
>
> What I am saying is that the second function does not need a timeout period near the RTT. Robust protocols have been designed to have a conservative RTO timer that is set to much more than a RTT, designed to detect an unresponsive endpoint or path problem. The IETF has argued - and to my knowledge still argues that Min_RTO should be one second, which I think reflects this position. If the loss recovery is efficient, this will trigger only infrequently (e.g., persistent loss of one packet). I think this is the correct way to design TCP. It motivates that we should actually strive for a one second (or so) RTO.
>
> Gorry
>
>
> On 15/12/2016 08:24, Yoshifumi Nishida wrote:
>> A question would be when TCP sent odd number of packets, how it can
>> know whether another packets will come from upper layer very soon or
>> not. Also, I'm not very sure yet if the probability is so obvious. I
>> can agree that RTO can get close to raw rtt value under bulk transfer.
>> But, I am guessing not so many apps change the behavior from bulk
>> transfer to spontaneous small burst transfer.
>> --
>> Yoshi
>>
>>
>> On Wed, Dec 14, 2016 at 7:08 PM, zhangyali (D)<zhangyali369@huawei.com>  wrote:
>>> Please allow me to add one more point. If the probability of odd
>>> number of packets occurring  spurious RTO is so outstanding, why
>>> nobody try to solve this problem?
>>>
>>>
>>>
>>> 发件人: tcpm [mailto:tcpm-bounces@ietf.org] 代表 zhangyali (D)
>>> 发送时间: 2016年12月15日 9:45
>>> 收件人: Neal Cardwell<ncardwell@google.com>
>>> 抄送: tcpm@ietf.org
>>> 主题: [tcpm] 答复: 答复: A question about Delayed ACK and RTO
>>>
>>>
>>>
>>>> Yes, there should only be a delayed ACK at the end of an application
>>>> chunk if there is an odd number of packets. But we would expect
>>>> roughly half of application chunks to have an odd number of packets.
>>>> Though the proportion is probably higher than that, since many
>>>> application chunks are just one packet (e.g. an HTTP or RPC request or response).
>>>
>>>
>>> I agree with you that odd number of packets may occur spurious RTO
>>> more easily, but I think for the flows owning just one packet should
>>> not be affected by delayed ACK. AFAIK, slow-start stage will begin
>>> with one packet, and sender will send two packets after it receives
>>> an ACK. If the first packet is obstructed by the delayed ACK, the
>>> ‘clock algorithm’ will be broken down. So receiver should judge if
>>> this packet is the first one in the slow-start stage, if yes, send
>>> the ack immediately once receiving the packet.
>>>
>>>
>>>
>>> Yali
>>>
>>>
>>>
>>> 发件人: Neal Cardwell [mailto:ncardwell@google.com]
>>> 发送时间: 2016年12月14日 21:24
>>> 收件人: zhangyali (D)<zhangyali369@huawei.com>
>>> 抄送: tcpm@ietf.org
>>> 主题: Re: 答复: [tcpm] A question about Delayed ACK and RTO
>>>
>>>
>>>
>>> On Wed, Dec 14, 2016 at 4:48 AM, zhangyali (D)
>>> <zhangyali369@huawei.com>
>>> wrote:
>>>
>>> Hi Neal,
>>>
>>>
>>>
>>> Thanks for your providing request information.
>>>
>>>
>>>
>>> About the delay of delayed ACK, I found another clue in a SIGCOMM
>>> paper in 1988. In the page 14, a sentence is “The 4.5KBps senders
>>> were talking to 4.3BSD receivers which would delay an ack until 35%
>>> of the window was filled or 200 ms had passed (i.e., an ack was delayed for 5-7 packets on average).”
>>> There is no reference about the 200 ms, so I guess this is the
>>> particular delay appears for the first time.
>>>
>>> I try to calculate the value based on delayed packet numbers, packet
>>> size and sending rate. In this case, the delayed packet number is 7,
>>> the packet size is 576Byte (refer to RFC879 in 1983), and sundering
>>> rate is 4.5Bps. The value is 896 ms! Longer than 200 ms.
>>>
>>>
>>>
>>>> . However, it's quite easy for a bulk transfer to never have any
>>>> delayed ACKs for most of its lifetime, during which the RTO
>>>> gradually converges toward the raw RTT value. Then when there is
>>>> suddenly a delayed ACK, there can be a spurious RTO.
>>> I am not very sure if I see your point. I think the key point is the
>>> RTT variation will be weakened when packets are sent in a burst (the consequence
>>> of delayed ACK).   And the spurious  could only happen for the last few
>>> packets whose number is smaller than delayed packets, right?
>>>
>>>
>>>
>>> Yes, there should only be a delayed ACK at the end of an application
>>> chunk if there is an odd number of packets. But we would expect
>>> roughly half of application chunks to have an odd number of packets.
>>> Though the proportion is probably higher than that, since many
>>> application chunks are just one packet (e.g. an HTTP or RPC request or response).
>>>
>>>
>>>
>>>
>>>
>>> About your proposal about negotiation of delayed ACK, a potential
>>> problem is that RTO will be stretched than before because you add
>>> extra delay. As you have said, most of the packets will not exceed
>>> RTO even host enable delayed ACK for most large flows, but the
>>> retransmission will be delayed (i.e., 5ms)also once one packet is lost.
>>>
>>>
>>>
>>> The basic idea of the proposal is to tweak the RTO calculation and
>>> turn a previously existing, historically motivated 200ms fixed "slop
>>> factor" into a dynamically negotiated 5ms "slop factor". In our
>>> experience that is almost always a win.
>>>
>>>
>>>
>>> neal
>>>
>>>
>>>
>>>
>>>
>>> Best,
>>>
>>> Yali
>>>
>>> 发件人: Neal Cardwell [mailto:ncardwell@google.com]
>>> 发送时间: 2016年12月13日 23:18
>>> 收件人: zhangyali (D)<zhangyali369@huawei.com>
>>> 抄送: tcpm@ietf.org
>>> 主题: Re: [tcpm] A question about Delayed ACK and RTO
>>>
>>>
>>>
>>> On Tue, Dec 13, 2016 at 3:58 AM, zhangyali (D)
>>> <zhangyali369@huawei.com>
>>> wrote:
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Recent days, I am doing some simulation about TCP performance in NS3.
>>> I found a phenomenon is a default setting of TCP’s delayed ACK is two
>>> packets or 200ms. I am wondering if this setting is accord with  some RFCs in IETF.
>>> But After I referred to some RFCs, I just found some restrictions,
>>> such as, the delay must be less than 0.5ms (RFC1122).
>>>
>>>
>>>
>>> I believe the 200ms figure is due to historical reasons. AFAIK it's
>>> due to the BSD delayed ACK behavior (see Stevens "TCP/IP Illustrated
>>> Volume 2", section 25.4 and figure 25.7, which describes the 200ms delayed ACK timer).
>>> Then this value was inherited by other widely-deployed OSes.
>>>
>>>
>>>
>>>
>>>
>>> I think the delayed ACK has a close relationship with RTO. Take an
>>> extreme scenario, if the delay is longer than RTO, many packets will
>>> be retransmitted, which will waste many network resources.
>>>
>>>
>>>
>>> Yes, exactly. In theory, the RTO tries to be adaptive enough to
>>> measure any RTT variations caused by delayed ACKs, and increase the
>>> RTO in response to this. However, it's quite easy for a bulk transfer
>>> to never have any delayed ACKs for most of its lifetime, during which
>>> the RTO gradually converges toward the raw RTT value. Then when there
>>> is suddenly a delayed ACK, there can be a spurious RTO.
>>>
>>>
>>>
>>> To avoid that, many TCP stacks (at least the major open source OSes)
>>> have a hard-coded 200ms "slop factor" or "fudge factor", to try to
>>> never let their estimate of RTT variation fall below that 200ms value, to avoid this effect.
>>>
>>>
>>>
>>> So I want to know do we have some RFCs have given the exact value of both?
>>> And if we permit any TCP stack to set them freely, what is the
>>> mechanism to balance the mismatch between RTO and delayed ACK?
>>>
>>>
>>>
>>> I'm not aware of RFC specifications for exact values of both (delayed
>>> ACK and RTO). However, the historical precedent is very strong, and
>>> the 200ms delayed ACK value was very pronounced in Internet traces at
>>> least as recently as 2011, when I last looked at the effect.
>>> (Probably others have more recent data points for the prevalence of
>>> 200ms delayed ACKs.)
>>>
>>>
>>>
>>> At IETF 97 our team at Google presented some features we use for
>>> internal TCP traffic at Google, where the endpoints can negotiate the
>>> specific constant to use for the maximum delayed ACK from the
>>> receiver and the corresponding minimum RTT delay variation for
>>> budgeting in the RTO at the
>>> sender:
>>>
>>>
>>>
>>>
>>> https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options
>>> -for-low-latency-00.pdf
>>>
>>>
>>>
>>> As this slide deck notes, within Google we negotiate 5ms for delayed ACKs.
>>>
>>>
>>>
>>> cheers,
>>>
>>> neal
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> 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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Changed milestone "Submit document on a time-based fast loss detection
algorithm for TCP to the IESG for publication as an Experimental RFC",
added draft-ietf-tcpm-rack to milestone.

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


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From: David Borman <dab@weston.borman.com>
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References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com> <0a234a82-5333-e3f7-cfa9-812d93144c54@erg.abdn.ac.uk> <a5af440e-28da-f193-14f4-794cb04804d3@palermo.edu> <45f0034f-70a2-2aa3-d87b-c7288a51ad77@isi.edu> <48aa99d0-113e-edda-94f7-48d81bea59fd@palermo.edu> <BN1PR03MB0085597E6BAF5F2F878A71FB69C0@BN1PR03MB008.namprd03.prod.outlook.com> <bf8ad6ba-38db-647a-18db-33557146adf9@isi.edu>
To: Joe Touch <touch@isi.edu>
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> On Dec 15, 2016, at 7:59 PM, Joe Touch <touch@isi.edu> wrote:
>=20
>=20
>=20
> On 12/15/2016 5:13 PM, Praveen Balasubramanian wrote:
>> FWIW the windows TCP implementation preserves the notion of =
application send() message boundary by setting the PSH bit only on the =
last segment generated from the posted buffer.

Other TCP implementations have done the same thing for decades.

> TCP isn't required to allow the app to decide when to set the PSH bit
> (it's a MAY in the SEND call). It can coalesce multiple PSH bits to =
send
> a larger segment.
>=20
> However, it's also explicitly not a record boundary. PUSH is really =
just
> to avoid deadlock between the app and TCP on the send side.

The PUSH bit is a way for the sending TCP to signal the receiving TCP =
that no more data is coming, and so if the receiving TCP is buffering =
data and has not yet made it available to the receiving application, it =
should now make all that data available to be read.  These days this =
doesn=E2=80=99t matter as much, because TCP stacks make all received =
data immediately available to the receiving application.  But senders =
should continue to set the PUSH bit.

			-David Borman

>> Unfortunately other stacks seem to set PSH on all segments making any =
receiver optimization of immediate ACK on PSH impossible, because that =
would effectively turn off delayed ACKs. That said, with ACK stretching, =
LRO etc. being very common maybe it is time for TCP implementations to =
turn off delayed ACKs across the board except for request-response =
applications.
>=20
> There are many ways this could be fixed but they all require going
> outside the current spec.
>=20
> Joe
>=20
>>=20
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Alejandro =
Popovsky
>> Sent: Thursday, December 15, 2016 3:13 PM
>> To: Joe Touch <touch@isi.edu>; tcpm@ietf.org
>> Subject: Re: [tcpm] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=
=A4=8D: A question about Delayed ACK and RTO
>>=20
>> On the second option (no more data to send) there would be no need to =
change the API. The tcp sender tcp would generate the signal =
automatically with the last segment. And the receiver could just use the =
info to adjust the delay ack timer for that segment.
>>=20
>> And in order for the extra ack not to produce an extra cwnd growth =
the sender should follow the recommendation of rfc5681, page 6.
>>=20
>> Alejandro.
>>=20
>> On 15/12/2016 12:29 p. m., Joe Touch wrote:
>>> On 12/15/2016 3:25 AM, Alejandro Popovsky wrote:
>>>> Tcp would benefit much from and 'end of app write call' signal or =
may=20
>>>> be 'end of sender buffer data' signal.
>>>  In the current TCP API, there is absolutely no correlation between=20=

>>> app SEND calls and any notion TCP might have of boundaries - TCP=20
>>> presents a bytestream service only. There would be many consequences=20=

>>> to changing that API, but it might be useful to note that this was =
once considered.
>>> Just two days ago, Bob Braden and I were discussing TCP v3 (before =
the=20
>>> current one  - as documented in IEN  21), which did have this sort =
of=20
>>> signal, but it was carried into the packet as begin/end (of =
"letter") flags.
>>>=20
>>> Note that his behavior is a consequence of the interaction between=20=

>>> delayed ACKs and a lack of application data, and happens any time =
the=20
>>> app layer "starves" TCP. It was well studied in the 90's in its=20
>>> interaction with persistent web connections. It isn't really related=20=

>>> to RTO per se.
>>>=20
>>> Joe
>>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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I appear to have run into either an edge case or ambiguity between RFC =
793 and 5681 SHOULDs and MUSTs where the sending TCP stack chooses to =
implement Zero Window Probing by sending 1 byte packets. Error scenario =
is as follows:

No. Time        Source      Destination   Info
  1 0.000000    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D1 Ack=3D1 =
Win=3D4644 Len=3D1460
  2 0.000853    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D1461 Ack=3D1 =
Win=3D4644 Len=3D1460
  3 0.000855    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D2921 Ack=3D1 =
Win=3D4644 Len=3D1460
  4 0.000855    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D4381 Ack=3D1 =
Win=3D4644 Len=3D264
  5 0.000856    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D1461 =
Win=3D3184 Len=3D0
  6 0.000856    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D2921 =
Win=3D1724 Len=3D0
  7 0.002107    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D4381 =
Win=3D264 Len=3D0
  8 0.002107    10.1.1.56   10.1.1.62     [TCP ZeroWindow] [ACK] Seq=3D1 =
Ack=3D4645 Win=3D0 Len=3D0

  9 4.966264    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
 10 4.966265    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP =
ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0

 11 5.465671    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
 12 5.466429    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP =
ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0

 13 6.465684    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
 14 6.466442    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP =
ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
 15 6.466443    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
 16 6.466443    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP =
ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
     ...

1-4. Sender Sends final data that fills RWND.
5-8. Receiver Sends ACK for SND.UNA. Receiver fails to respond in a =
timely manner with an increased window size.
9. Sender Sends Zero Window Probe with 1 byte, increasing SND.UNA by =
one.
10. Receiver responds with ACK for SND.UNA-1 and RWND of 0. This is the =
first Duplicate ACK.
11. After delaying, Sender Resends the ZWP.
12. Receiver Responds with ACK for SND.UNA-1 and RWND of 0. This is the =
second Duplicate ACK.
13. After delaying, Sender Resends the ZWD.
14. Receiver Responds with ACK for SND.UNA-1 and a RWND of 0. This is =
the third Duplicate ACK.
15. In response to three duplicate ACKs, the Sender now enters Fast =
Retransmission for the dropped packet, aka the ZWP.

15+. The Sender will respond to all ACKs from the Receiver with a =
retransmission of the Zero Window Probe due to Fast Retransmission.
16+. The Receiver will respond to all Probes with ACK for SND.UNA-1 and =
a RWND of 0, until it=E2=80=99s user application reads the data.

=E2=80=94=E2=80=94=E2=80=94=E2=80=94=E2=80=94

On a fast network (i.e. LAN), and a slow receiver (i.e. an embedded =
device), this becomes a Denial of Service as the receiver ends up =
spending all its resources ACKing the Probes instead of reading from the =
socket.

Suggested change, in RFC 5681:
Section 3.2, paragraph 2 says:
    The TCP sender SHOULD use the "fast retransmit" algorithm to detect
    and repair loss, based on incoming duplicate ACKs.  The fast
    retransmit algorithm uses the arrival of 3 duplicate ACKs (as =
defined
    in section 2, without any intervening ACKs which move SND.UNA) as an
    indication that a segment has been lost.  After receiving 3 =
duplicate
    ACKs, TCP performs a retransmission of what appears to be the =
missing
    segment, without waiting for the retransmission timer to expire.

It should say:
    The TCP sender SHOULD use the "fast retransmit" algorithm to detect
    and repair loss, based on incoming duplicate ACKs.  The fast
    retransmit algorithm uses the arrival of 3 duplicate ACKs (as =
defined
    in section 2, without any intervening ACKs which move SND.UNA) as an
    indication that a segment has been lost.  After receiving 3 =
duplicate
    ACKs, TCP performs a retransmission of what appears to be the =
missing
    segment, without waiting for the retransmission timer to expire.=20
    However, if all transmitted data from the sender has been =
acknowledged,
    the duplicate ACKs are in response to a Zero Window Probe, AND the
    advertised window is still zero, then the TCP sender MUST NOT =
consider
    these to be duplicate ACKs in relation to the fast retransmit =
algorithm.


-Dan=


From nobody Fri Dec 16 08:48:18 2016
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To: David Borman <dab@weston.borman.com>
References: <A747A0713F56294D8FBE33E5C6B8F5815F545A98@DGGEMA502-MBS.china.huawei.com> <CADVnQynhEt-bKeFz3bN=MHd4rx8csg-no9ifTp__rbmT_JDggw@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F545E28@DGGEMA502-MBS.china.huawei.com> <CADVnQykfYUOyrSsQLrz2EFL0_q5whjwdD-cEehowj+nL4zUMtQ@mail.gmail.com> <A747A0713F56294D8FBE33E5C6B8F5815F546099@DGGEMA502-MBS.china.huawei.com> <A747A0713F56294D8FBE33E5C6B8F5815F5460FF@DGGEMA502-MBS.china.huawei.com> <CAO249yck=og1TmyDLNb4GFATy4FPVVWT8gi5G=j6T8Ksj4Hh9w@mail.gmail.com> <0a234a82-5333-e3f7-cfa9-812d93144c54@erg.abdn.ac.uk> <a5af440e-28da-f193-14f4-794cb04804d3@palermo.edu> <45f0034f-70a2-2aa3-d87b-c7288a51ad77@isi.edu> <48aa99d0-113e-edda-94f7-48d81bea59fd@palermo.edu> <BN1PR03MB0085597E6BAF5F2F878A71FB69C0@BN1PR03MB008.namprd03.prod.outlook.com> <bf8ad6ba-38db-647a-18db-33557146adf9@isi.edu> <47FF18FE-34E6-479D-8B89-430390D15EC3@weston.borman.com>
From: Joe Touch <touch@isi.edu>
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Date: Fri, 16 Dec 2016 08:47:44 -0800
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Subject: Re: [tcpm] =?utf-8?b?562U5aSNOiDnrZTlpI06IOetlOWkjTogQSBxdWVzdGlvbiBh?= =?utf-8?q?bout_Delayed_ACK_and_RTO?=
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On 12/16/2016 6:06 AM, David Borman wrote:
>> On Dec 15, 2016, at 7:59 PM, Joe Touch <touch@isi.edu> wrote:
>>
>>
>>
>> On 12/15/2016 5:13 PM, Praveen Balasubramanian wrote:
>>> FWIW the windows TCP implementation preserves the notion of application send() message boundary by setting the PSH bit only on the last segment generated from the posted buffer.
> Other TCP implementations have done the same thing for decades.
RFC793 states that the PSH bit is independent of segment boundaries.
Both it and RFC1122 indicate that the send() call MAY support setting
that flag and give provisions for what to do when that flag is not
supported. However, we can't simply assume that this means "last segment
generated".

It means only "last segment in this group" as part of an interactive
transfer. There could be another group following it immediately.

>
>> TCP isn't required to allow the app to decide when to set the PSH bit
>> (it's a MAY in the SEND call). It can coalesce multiple PSH bits to send
>> a larger segment.
>>
>> However, it's also explicitly not a record boundary. PUSH is really just
>> to avoid deadlock between the app and TCP on the send side.
> The PUSH bit is a way for the sending TCP to signal the receiving TCP that no more data is coming, 

as part of this "group" only. There's no rule that a PSH means that the
sending app has paused at all.

> and so if the receiving TCP is buffering data and has not yet made it available to the receiving application, it should now make all that data available to be read. 

RIght - push this through as a group. This is about when to avoid delays
to the receiving app. It's not useful as a signal to override delayed
ACKs because it doesn't require the sender to pause.

>  These days this doesn’t matter as much, because TCP stacks make all received data immediately available to the receiving application.  But senders should continue to set the PUSH bit.

The conditions under which they need to do this are spec'd in RFC1122.
Applications can't assume access via the send() call, so they shouldn't
design their apps with this assumption.

Joe


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From: Neal Cardwell <ncardwell@google.com>
Date: Fri, 16 Dec 2016 12:47:46 -0500
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To: Dan Ciliske <dciliske@netburner.com>
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Subject: Re: [tcpm] Fast retransmit of Zero Window Probes with 1 byte
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Good point. The RFC does seem unclear on that.

The BSD code (tcp_input.c or Figure 29.3 in Stevens volume 2) is pretty
clear about simply not treating a window probe as "outstanding data":

  * If we have outstanding data (other than
  * a window probe), ...

So one option would be to go with that approach, and clarify in the
definition of a dupack that a window probe does not count as outstanding
data:

So the existing RFC 5681 text:

   DUPLICATE ACKNOWLEDGMENT: An acknowledgment is considered a
      "duplicate" in the following algorithms when (a) the receiver of
      the ACK has outstanding data,

could be:

   DUPLICATE ACKNOWLEDGMENT: An acknowledgment is considered a
      "duplicate" in the following algorithms when (a) the receiver of
      the ACK has outstanding data (other than a window probe),

Do you mind sharing which TCP implementation was running on the sender in
the scenario you describe? I haven't been able to reproduce that behavior
on Linux, and BSD doesn't seem like it would be susceptible. Just curious.

neal


On Thu, Dec 15, 2016 at 4:52 PM, Dan Ciliske <dciliske@netburner.com> wrote=
:

> I appear to have run into either an edge case or ambiguity between RFC 79=
3
> and 5681 SHOULDs and MUSTs where the sending TCP stack chooses to impleme=
nt
> Zero Window Probing by sending 1 byte packets. Error scenario is as follo=
ws:
>
> No. Time        Source      Destination   Info
>   1 0.000000    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D1 Ack=3D1 Win=
=3D4644
> Len=3D1460
>   2 0.000853    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D1461 Ack=3D1
> Win=3D4644 Len=3D1460
>   3 0.000855    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D2921 Ack=3D1
> Win=3D4644 Len=3D1460
>   4 0.000855    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D4381 Ack=3D1
> Win=3D4644 Len=3D264
>   5 0.000856    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D1461 Win=3D=
3184
> Len=3D0
>   6 0.000856    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D2921 Win=3D=
1724
> Len=3D0
>   7 0.002107    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D4381 Win=3D=
264
> Len=3D0
>   8 0.002107    10.1.1.56   10.1.1.62     [TCP ZeroWindow] [ACK] Seq=3D1
> Ack=3D4645 Win=3D0 Len=3D0
>
>   9 4.966264    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, ACK=
]
> Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  10 4.966265    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP
> ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>
>  11 5.465671    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, ACK=
]
> Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  12 5.466429    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP
> ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>
>  13 6.465684    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, ACK=
]
> Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  14 6.466442    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP
> ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>  15 6.466443    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, ACK=
]
> Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  16 6.466443    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] [TCP
> ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>      ...
>
> 1-4. Sender Sends final data that fills RWND.
> 5-8. Receiver Sends ACK for SND.UNA. Receiver fails to respond in a timel=
y
> manner with an increased window size.
> 9. Sender Sends Zero Window Probe with 1 byte, increasing SND.UNA by one.
> 10. Receiver responds with ACK for SND.UNA-1 and RWND of 0. This is the
> first Duplicate ACK.
> 11. After delaying, Sender Resends the ZWP.
> 12. Receiver Responds with ACK for SND.UNA-1 and RWND of 0. This is the
> second Duplicate ACK.
> 13. After delaying, Sender Resends the ZWD.
> 14. Receiver Responds with ACK for SND.UNA-1 and a RWND of 0. This is the
> third Duplicate ACK.
> 15. In response to three duplicate ACKs, the Sender now enters Fast
> Retransmission for the dropped packet, aka the ZWP.
>
> 15+. The Sender will respond to all ACKs from the Receiver with a
> retransmission of the Zero Window Probe due to Fast Retransmission.
> 16+. The Receiver will respond to all Probes with ACK for SND.UNA-1 and a
> RWND of 0, until it=E2=80=99s user application reads the data.
>
> =E2=80=94=E2=80=94=E2=80=94=E2=80=94=E2=80=94
>
> On a fast network (i.e. LAN), and a slow receiver (i.e. an embedded
> device), this becomes a Denial of Service as the receiver ends up spendin=
g
> all its resources ACKing the Probes instead of reading from the socket.
>
> Suggested change, in RFC 5681:
> Section 3.2, paragraph 2 says:
>     The TCP sender SHOULD use the "fast retransmit" algorithm to detect
>     and repair loss, based on incoming duplicate ACKs.  The fast
>     retransmit algorithm uses the arrival of 3 duplicate ACKs (as defined
>     in section 2, without any intervening ACKs which move SND.UNA) as an
>     indication that a segment has been lost.  After receiving 3 duplicate
>     ACKs, TCP performs a retransmission of what appears to be the missing
>     segment, without waiting for the retransmission timer to expire.
>
> It should say:
>     The TCP sender SHOULD use the "fast retransmit" algorithm to detect
>     and repair loss, based on incoming duplicate ACKs.  The fast
>     retransmit algorithm uses the arrival of 3 duplicate ACKs (as defined
>     in section 2, without any intervening ACKs which move SND.UNA) as an
>     indication that a segment has been lost.  After receiving 3 duplicate
>     ACKs, TCP performs a retransmission of what appears to be the missing
>     segment, without waiting for the retransmission timer to expire.
>     However, if all transmitted data from the sender has been acknowledge=
d,
>     the duplicate ACKs are in response to a Zero Window Probe, AND the
>     advertised window is still zero, then the TCP sender MUST NOT conside=
r
>     these to be duplicate ACKs in relation to the fast retransmit
> algorithm.
>
>
> -Dan
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--001a1137c88ec3ab5d0543ca2cf5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Good point. The RFC does seem unclear on that.</div><=
div><br></div><div>The BSD code (tcp_input.c or Figure 29.3 in Stevens volu=
me 2) is pretty clear about simply not treating a window probe as &quot;out=
standing data&quot;:</div><div><br></div><div>=C2=A0 * If we have outstandi=
ng data (other than</div><div>=C2=A0 * a window probe), ...</div><div><br><=
/div><div>So one option would be to go with that approach, and clarify in t=
he definition of a dupack that a window probe does not count as outstanding=
 data:</div><div><br></div><div>So the existing RFC 5681 text:</div><div><b=
r></div><div><div>=C2=A0 =C2=A0DUPLICATE ACKNOWLEDGMENT: An acknowledgment =
is considered a</div><div>=C2=A0 =C2=A0 =C2=A0 &quot;duplicate&quot; in the=
 following algorithms when (a) the receiver of</div><div>=C2=A0 =C2=A0 =C2=
=A0 the ACK has outstanding data,</div></div><div><br></div><div>could be:<=
/div><div><br></div><div><div>=C2=A0 =C2=A0DUPLICATE ACKNOWLEDGMENT: An ack=
nowledgment is considered a</div><div>=C2=A0 =C2=A0 =C2=A0 &quot;duplicate&=
quot; in the following algorithms when (a) the receiver of</div><div>=C2=A0=
 =C2=A0 =C2=A0 the ACK has outstanding data (other than a window probe),</d=
iv></div><div><br></div><div>Do you mind sharing which TCP implementation w=
as running on the sender in the scenario you describe? I haven&#39;t been a=
ble to reproduce that behavior on Linux, and BSD doesn&#39;t seem like it w=
ould be susceptible. Just curious.</div><div><br></div><div>neal</div><div>=
<br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Thu, Dec 15, 2016 at 4:52 PM, Dan Ciliske <span dir=3D"ltr">&lt;<a href=
=3D"mailto:dciliske@netburner.com" target=3D"_blank">dciliske@netburner.com=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I appear to have r=
un into either an edge case or ambiguity between RFC 793 and 5681 SHOULDs a=
nd MUSTs where the sending TCP stack chooses to implement Zero Window Probi=
ng by sending 1 byte packets. Error scenario is as follows:<br>
<br>
No. Time=C2=A0 =C2=A0 =C2=A0 =C2=A0 Source=C2=A0 =C2=A0 =C2=A0 Destination=
=C2=A0 =C2=A0Info<br>
=C2=A0 1 0.000000=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[PSH, ACK] Seq=3D1 Ack=3D1 Win=3D4644 Len=3D1460<br>
=C2=A0 2 0.000853=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[PSH, ACK] Seq=3D1461 Ack=3D1 Win=3D4644 Len=3D1460<br>
=C2=A0 3 0.000855=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[PSH, ACK] Seq=3D2921 Ack=3D1 Win=3D4644 Len=3D1460<br>
=C2=A0 4 0.000855=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[PSH, ACK] Seq=3D4381 Ack=3D1 Win=3D4644 Len=3D264<br>
=C2=A0 5 0.000856=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[ACK] Seq=3D1 Ack=3D1461 Win=3D3184 Len=3D0<br>
=C2=A0 6 0.000856=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[ACK] Seq=3D1 Ack=3D2921 Win=3D1724 Len=3D0<br>
=C2=A0 7 0.002107=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[ACK] Seq=3D1 Ack=3D4381 Win=3D264 Len=3D0<br>
=C2=A0 8 0.002107=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0<br>
<br>
=C2=A0 9 4.966264=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=
=3D1<br>
=C2=A010 4.966265=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 W=
in=3D0 Len=3D0<br>
<br>
=C2=A011 5.465671=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=
=3D1<br>
=C2=A012 5.466429=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 W=
in=3D0 Len=3D0<br>
<br>
=C2=A013 6.465684=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=
=3D1<br>
=C2=A014 6.466442=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 W=
in=3D0 Len=3D0<br>
=C2=A015 6.466443=C2=A0 =C2=A0 10.1.1.62=C2=A0 =C2=A010.1.1.56=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=
=3D1<br>
=C2=A016 6.466443=C2=A0 =C2=A0 10.1.1.56=C2=A0 =C2=A010.1.1.62=C2=A0 =C2=A0=
 =C2=A0[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 W=
in=3D0 Len=3D0<br>
=C2=A0 =C2=A0 =C2=A0...<br>
<br>
1-4. Sender Sends final data that fills RWND.<br>
5-8. Receiver Sends ACK for SND.UNA. Receiver fails to respond in a timely =
manner with an increased window size.<br>
9. Sender Sends Zero Window Probe with 1 byte, increasing SND.UNA by one.<b=
r>
10. Receiver responds with ACK for SND.UNA-1 and RWND of 0. This is the fir=
st Duplicate ACK.<br>
11. After delaying, Sender Resends the ZWP.<br>
12. Receiver Responds with ACK for SND.UNA-1 and RWND of 0. This is the sec=
ond Duplicate ACK.<br>
13. After delaying, Sender Resends the ZWD.<br>
14. Receiver Responds with ACK for SND.UNA-1 and a RWND of 0. This is the t=
hird Duplicate ACK.<br>
15. In response to three duplicate ACKs, the Sender now enters Fast Retrans=
mission for the dropped packet, aka the ZWP.<br>
<br>
15+. The Sender will respond to all ACKs from the Receiver with a retransmi=
ssion of the Zero Window Probe due to Fast Retransmission.<br>
16+. The Receiver will respond to all Probes with ACK for SND.UNA-1 and a R=
WND of 0, until it=E2=80=99s user application reads the data.<br>
<br>
=E2=80=94=E2=80=94=E2=80=94=E2=80=94=E2=80=94<br>
<br>
On a fast network (i.e. LAN), and a slow receiver (i.e. an embedded device)=
, this becomes a Denial of Service as the receiver ends up spending all its=
 resources ACKing the Probes instead of reading from the socket.<br>
<br>
Suggested change, in RFC 5681:<br>
Section 3.2, paragraph 2 says:<br>
=C2=A0 =C2=A0 The TCP sender SHOULD use the &quot;fast retransmit&quot; alg=
orithm to detect<br>
=C2=A0 =C2=A0 and repair loss, based on incoming duplicate ACKs.=C2=A0 The =
fast<br>
=C2=A0 =C2=A0 retransmit algorithm uses the arrival of 3 duplicate ACKs (as=
 defined<br>
=C2=A0 =C2=A0 in section 2, without any intervening ACKs which move SND.UNA=
) as an<br>
=C2=A0 =C2=A0 indication that a segment has been lost.=C2=A0 After receivin=
g 3 duplicate<br>
=C2=A0 =C2=A0 ACKs, TCP performs a retransmission of what appears to be the=
 missing<br>
=C2=A0 =C2=A0 segment, without waiting for the retransmission timer to expi=
re.<br>
<br>
It should say:<br>
=C2=A0 =C2=A0 The TCP sender SHOULD use the &quot;fast retransmit&quot; alg=
orithm to detect<br>
=C2=A0 =C2=A0 and repair loss, based on incoming duplicate ACKs.=C2=A0 The =
fast<br>
=C2=A0 =C2=A0 retransmit algorithm uses the arrival of 3 duplicate ACKs (as=
 defined<br>
=C2=A0 =C2=A0 in section 2, without any intervening ACKs which move SND.UNA=
) as an<br>
=C2=A0 =C2=A0 indication that a segment has been lost.=C2=A0 After receivin=
g 3 duplicate<br>
=C2=A0 =C2=A0 ACKs, TCP performs a retransmission of what appears to be the=
 missing<br>
=C2=A0 =C2=A0 segment, without waiting for the retransmission timer to expi=
re.<br>
=C2=A0 =C2=A0 However, if all transmitted data from the sender has been ack=
nowledged,<br>
=C2=A0 =C2=A0 the duplicate ACKs are in response to a Zero Window Probe, AN=
D the<br>
=C2=A0 =C2=A0 advertised window is still zero, then the TCP sender MUST NOT=
 consider<br>
=C2=A0 =C2=A0 these to be duplicate ACKs in relation to the fast retransmit=
 algorithm.<br>
<br>
<br>
-Dan<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/tcpm</a><br>
</blockquote></div><br></div>

--001a1137c88ec3ab5d0543ca2cf5--


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Subject: [tcpm] Adopting call for draft-khademi-tcpm-alternativebackoff-ecn
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Hello,

As we have discussed during Seoul meeting, the chairs decided to start
running a call for adoption of draft-khademi-tcpm-alternativebackoff-ecn.

We would like to confirm that the TCPM community agrees to:

  * adopt draft-khademi-tcpm-alternativebackoff-ecn as a TCPM working group
item to publish it as an experimental RFC.
  * add a corresponding milestone to the TCPM charter.

     Aug 2017  Submit document on alternative backoff with ECN algorithm to
the IESG for publication as an Experimental RFC


As the end of the year is approaching, we decided to run a longer call than
usual. Please let us know by * Friday January 6  2017 *, whether you
support or object adoption of the document.

Thanks,
--
tcpm co-chairs

--94eb2c18f7ee573c190543ceac33
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<div dir=3D"ltr">Hello,<br><br>As we have discussed during Seoul meeting, t=
he chairs decided to start running a call for adoption of draft-khademi-tcp=
m-alternativebackoff-ecn.=C2=A0<div><br>We would like to confirm that the T=
CPM community agrees to:<br><br>=C2=A0 * adopt draft-khademi-tcpm-alternati=
vebackoff-ecn as a TCPM working group item to publish it as an experimental=
 RFC.<br>=C2=A0 * add a corresponding milestone to the TCPM charter.<br><br=
>=C2=A0 =C2=A0 =C2=A0Aug 2017 =C2=A0Submit document on alternative backoff =
with ECN algorithm to the IESG for publication as an Experimental RFC<br><b=
r><br>As the end of the year is approaching, we decided to run a longer cal=
l than usual. Please let us know by * Friday January 6 =C2=A02017 *, whethe=
r you support or object adoption of the document.=C2=A0<div><br>Thanks,<br>=
--<br>tcpm co-chairs</div></div></div>

--94eb2c18f7ee573c190543ceac33--


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From: Dan Ciliske <dciliske@netburner.com>
In-Reply-To: <CADVnQyn8PnhBG0qPKWx0=uMGe-pi4xUKFZ=L+J5PWVNH74x0xg@mail.gmail.com>
Date: Fri, 16 Dec 2016 10:44:17 -0800
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To: Neal Cardwell <ncardwell@google.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] Fast retransmit of Zero Window Probes with 1 byte
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Hmm=E2=80=A6 looks like I may need to dredge up my copy. As to the =
stack: it=E2=80=99s our own custom embedded TCP/IP stack running on a =
derivative of uC/OS II. The code triggering this was written nearly 18 =
years ago=E2=80=A6 I have no idea why it=E2=80=99s just now showed up.

Also, your wording change seems to be more concise and clear.

-Dan

> On Dec 16, 2016, at 9:47 AM, Neal Cardwell <ncardwell@google.com> =
wrote:
>=20
> Good point. The RFC does seem unclear on that.
>=20
> The BSD code (tcp_input.c or Figure 29.3 in Stevens volume 2) is =
pretty clear about simply not treating a window probe as "outstanding =
data":
>=20
>   * If we have outstanding data (other than
>   * a window probe), ...
>=20
> So one option would be to go with that approach, and clarify in the =
definition of a dupack that a window probe does not count as outstanding =
data:
>=20
> So the existing RFC 5681 text:
>=20
>    DUPLICATE ACKNOWLEDGMENT: An acknowledgment is considered a
>       "duplicate" in the following algorithms when (a) the receiver of
>       the ACK has outstanding data,
>=20
> could be:
>=20
>    DUPLICATE ACKNOWLEDGMENT: An acknowledgment is considered a
>       "duplicate" in the following algorithms when (a) the receiver of
>       the ACK has outstanding data (other than a window probe),
>=20
> Do you mind sharing which TCP implementation was running on the sender =
in the scenario you describe? I haven't been able to reproduce that =
behavior on Linux, and BSD doesn't seem like it would be susceptible. =
Just curious.
>=20
> neal
>=20
>=20
> On Thu, Dec 15, 2016 at 4:52 PM, Dan Ciliske <dciliske@netburner.com =
<mailto:dciliske@netburner.com>> wrote:
> I appear to have run into either an edge case or ambiguity between RFC =
793 and 5681 SHOULDs and MUSTs where the sending TCP stack chooses to =
implement Zero Window Probing by sending 1 byte packets. Error scenario =
is as follows:
>=20
> No. Time        Source      Destination   Info
>   1 0.000000    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D1 Ack=3D1 =
Win=3D4644 Len=3D1460
>   2 0.000853    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D1461 Ack=3D1 =
Win=3D4644 Len=3D1460
>   3 0.000855    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D2921 Ack=3D1 =
Win=3D4644 Len=3D1460
>   4 0.000855    10.1.1.62   10.1.1.56     [PSH, ACK] Seq=3D4381 Ack=3D1 =
Win=3D4644 Len=3D264
>   5 0.000856    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D1461 =
Win=3D3184 Len=3D0
>   6 0.000856    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D2921 =
Win=3D1724 Len=3D0
>   7 0.002107    10.1.1.56   10.1.1.62     [ACK] Seq=3D1 Ack=3D4381 =
Win=3D264 Len=3D0
>   8 0.002107    10.1.1.56   10.1.1.62     [TCP ZeroWindow] [ACK] Seq=3D1=
 Ack=3D4645 Win=3D0 Len=3D0
>=20
>   9 4.966264    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  10 4.966265    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] =
[TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>=20
>  11 5.465671    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  12 5.466429    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] =
[TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>=20
>  13 6.465684    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  14 6.466442    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] =
[TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>  15 6.466443    10.1.1.62   10.1.1.56     [TCP ZeroWindowProbe] [PSH, =
ACK] Seq=3D4645 Ack=3D1 Win=3D4644 Len=3D1
>  16 6.466443    10.1.1.56   10.1.1.62     [TCP ZeroWindowProbeAck] =
[TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0
>      ...
>=20
> 1-4. Sender Sends final data that fills RWND.
> 5-8. Receiver Sends ACK for SND.UNA. Receiver fails to respond in a =
timely manner with an increased window size.
> 9. Sender Sends Zero Window Probe with 1 byte, increasing SND.UNA by =
one.
> 10. Receiver responds with ACK for SND.UNA-1 and RWND of 0. This is =
the first Duplicate ACK.
> 11. After delaying, Sender Resends the ZWP.
> 12. Receiver Responds with ACK for SND.UNA-1 and RWND of 0. This is =
the second Duplicate ACK.
> 13. After delaying, Sender Resends the ZWD.
> 14. Receiver Responds with ACK for SND.UNA-1 and a RWND of 0. This is =
the third Duplicate ACK.
> 15. In response to three duplicate ACKs, the Sender now enters Fast =
Retransmission for the dropped packet, aka the ZWP.
>=20
> 15+. The Sender will respond to all ACKs from the Receiver with a =
retransmission of the Zero Window Probe due to Fast Retransmission.
> 16+. The Receiver will respond to all Probes with ACK for SND.UNA-1 =
and a RWND of 0, until it=E2=80=99s user application reads the data.
>=20
> =E2=80=94=E2=80=94=E2=80=94=E2=80=94=E2=80=94
>=20
> On a fast network (i.e. LAN), and a slow receiver (i.e. an embedded =
device), this becomes a Denial of Service as the receiver ends up =
spending all its resources ACKing the Probes instead of reading from the =
socket.
>=20
> Suggested change, in RFC 5681:
> Section 3.2, paragraph 2 says:
>     The TCP sender SHOULD use the "fast retransmit" algorithm to =
detect
>     and repair loss, based on incoming duplicate ACKs.  The fast
>     retransmit algorithm uses the arrival of 3 duplicate ACKs (as =
defined
>     in section 2, without any intervening ACKs which move SND.UNA) as =
an
>     indication that a segment has been lost.  After receiving 3 =
duplicate
>     ACKs, TCP performs a retransmission of what appears to be the =
missing
>     segment, without waiting for the retransmission timer to expire.
>=20
> It should say:
>     The TCP sender SHOULD use the "fast retransmit" algorithm to =
detect
>     and repair loss, based on incoming duplicate ACKs.  The fast
>     retransmit algorithm uses the arrival of 3 duplicate ACKs (as =
defined
>     in section 2, without any intervening ACKs which move SND.UNA) as =
an
>     indication that a segment has been lost.  After receiving 3 =
duplicate
>     ACKs, TCP performs a retransmission of what appears to be the =
missing
>     segment, without waiting for the retransmission timer to expire.
>     However, if all transmitted data from the sender has been =
acknowledged,
>     the duplicate ACKs are in response to a Zero Window Probe, AND the
>     advertised window is still zero, then the TCP sender MUST NOT =
consider
>     these to be duplicate ACKs in relation to the fast retransmit =
algorithm.
>=20
>=20
> -Dan
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org <mailto:tcpm@ietf.org>
> https://www.ietf.org/mailman/listinfo/tcpm =
<https://www.ietf.org/mailman/listinfo/tcpm>
>=20


--Apple-Mail=_71553A00-E789-4DCC-B488-E1D9CF5E7751
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D"">Hmm=E2=80=A6 looks like I may need to dredge up my copy. As =
to the stack: it=E2=80=99s our own custom embedded TCP/IP stack running =
on a derivative of uC/OS II. The code triggering this was written nearly =
18 years ago=E2=80=A6 I have no idea why it=E2=80=99s just now showed =
up.<div class=3D""><br class=3D""></div><div class=3D"">Also, your =
wording change seems to be more concise and clear.</div><div =
class=3D""><br class=3D""></div><div class=3D"">-Dan</div><div =
class=3D""><br class=3D""></div><div class=3D""><div><blockquote =
type=3D"cite" class=3D""><div class=3D"">On Dec 16, 2016, at 9:47 AM, =
Neal Cardwell &lt;<a href=3D"mailto:ncardwell@google.com" =
class=3D"">ncardwell@google.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D""><div class=3D"">Good point. The RFC does seem unclear on =
that.</div><div class=3D""><br class=3D""></div><div class=3D"">The BSD =
code (tcp_input.c or Figure 29.3 in Stevens volume 2) is pretty clear =
about simply not treating a window probe as "outstanding =
data":</div><div class=3D""><br class=3D""></div><div class=3D"">&nbsp; =
* If we have outstanding data (other than</div><div class=3D"">&nbsp; * =
a window probe), ...</div><div class=3D""><br class=3D""></div><div =
class=3D"">So one option would be to go with that approach, and clarify =
in the definition of a dupack that a window probe does not count as =
outstanding data:</div><div class=3D""><br class=3D""></div><div =
class=3D"">So the existing RFC 5681 text:</div><div class=3D""><br =
class=3D""></div><div class=3D""><div class=3D"">&nbsp; &nbsp;DUPLICATE =
ACKNOWLEDGMENT: An acknowledgment is considered a</div><div =
class=3D"">&nbsp; &nbsp; &nbsp; "duplicate" in the following algorithms =
when (a) the receiver of</div><div class=3D"">&nbsp; &nbsp; &nbsp; the =
ACK has outstanding data,</div></div><div class=3D""><br =
class=3D""></div><div class=3D"">could be:</div><div class=3D""><br =
class=3D""></div><div class=3D""><div class=3D"">&nbsp; &nbsp;DUPLICATE =
ACKNOWLEDGMENT: An acknowledgment is considered a</div><div =
class=3D"">&nbsp; &nbsp; &nbsp; "duplicate" in the following algorithms =
when (a) the receiver of</div><div class=3D"">&nbsp; &nbsp; &nbsp; the =
ACK has outstanding data (other than a window probe),</div></div><div =
class=3D""><br class=3D""></div><div class=3D"">Do you mind sharing =
which TCP implementation was running on the sender in the scenario you =
describe? I haven't been able to reproduce that behavior on Linux, and =
BSD doesn't seem like it would be susceptible. Just curious.</div><div =
class=3D""><br class=3D""></div><div class=3D"">neal</div><div =
class=3D""><br class=3D""></div></div><div class=3D"gmail_extra"><br =
class=3D""><div class=3D"gmail_quote">On Thu, Dec 15, 2016 at 4:52 PM, =
Dan Ciliske <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:dciliske@netburner.com" target=3D"_blank" =
class=3D"">dciliske@netburner.com</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">I appear to have run =
into either an edge case or ambiguity between RFC 793 and 5681 SHOULDs =
and MUSTs where the sending TCP stack chooses to implement Zero Window =
Probing by sending 1 byte packets. Error scenario is as follows:<br =
class=3D"">
<br class=3D"">
No. Time&nbsp; &nbsp; &nbsp; &nbsp; Source&nbsp; &nbsp; &nbsp; =
Destination&nbsp; &nbsp;Info<br class=3D"">
&nbsp; 1 0.000000&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[PSH, ACK] Seq=3D1 Ack=3D1 Win=3D4644 Len=3D1460<br =
class=3D"">
&nbsp; 2 0.000853&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[PSH, ACK] Seq=3D1461 Ack=3D1 Win=3D4644 Len=3D1460<br =
class=3D"">
&nbsp; 3 0.000855&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[PSH, ACK] Seq=3D2921 Ack=3D1 Win=3D4644 Len=3D1460<br =
class=3D"">
&nbsp; 4 0.000855&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[PSH, ACK] Seq=3D4381 Ack=3D1 Win=3D4644 Len=3D264<br =
class=3D"">
&nbsp; 5 0.000856&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[ACK] Seq=3D1 Ack=3D1461 Win=3D3184 Len=3D0<br class=3D"">
&nbsp; 6 0.000856&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[ACK] Seq=3D1 Ack=3D2921 Win=3D1724 Len=3D0<br class=3D"">
&nbsp; 7 0.002107&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[ACK] Seq=3D1 Ack=3D4381 Win=3D264 Len=3D0<br class=3D"">
&nbsp; 8 0.002107&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindow] [ACK] Seq=3D1 Ack=3D4645 Win=3D0 Len=3D0<br =
class=3D"">
<br class=3D"">
&nbsp; 9 4.966264&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 =
Win=3D4644 Len=3D1<br class=3D"">
&nbsp;10 4.966265&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 =
Ack=3D4645 Win=3D0 Len=3D0<br class=3D"">
<br class=3D"">
&nbsp;11 5.465671&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 =
Win=3D4644 Len=3D1<br class=3D"">
&nbsp;12 5.466429&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 =
Ack=3D4645 Win=3D0 Len=3D0<br class=3D"">
<br class=3D"">
&nbsp;13 6.465684&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 =
Win=3D4644 Len=3D1<br class=3D"">
&nbsp;14 6.466442&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 =
Ack=3D4645 Win=3D0 Len=3D0<br class=3D"">
&nbsp;15 6.466443&nbsp; &nbsp; 10.1.1.62&nbsp; &nbsp;10.1.1.56&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbe] [PSH, ACK] Seq=3D4645 Ack=3D1 =
Win=3D4644 Len=3D1<br class=3D"">
&nbsp;16 6.466443&nbsp; &nbsp; 10.1.1.56&nbsp; &nbsp;10.1.1.62&nbsp; =
&nbsp; &nbsp;[TCP ZeroWindowProbeAck] [TCP ZeroWindow] [ACK] Seq=3D1 =
Ack=3D4645 Win=3D0 Len=3D0<br class=3D"">
&nbsp; &nbsp; &nbsp;...<br class=3D"">
<br class=3D"">
1-4. Sender Sends final data that fills RWND.<br class=3D"">
5-8. Receiver Sends ACK for SND.UNA. Receiver fails to respond in a =
timely manner with an increased window size.<br class=3D"">
9. Sender Sends Zero Window Probe with 1 byte, increasing SND.UNA by =
one.<br class=3D"">
10. Receiver responds with ACK for SND.UNA-1 and RWND of 0. This is the =
first Duplicate ACK.<br class=3D"">
11. After delaying, Sender Resends the ZWP.<br class=3D"">
12. Receiver Responds with ACK for SND.UNA-1 and RWND of 0. This is the =
second Duplicate ACK.<br class=3D"">
13. After delaying, Sender Resends the ZWD.<br class=3D"">
14. Receiver Responds with ACK for SND.UNA-1 and a RWND of 0. This is =
the third Duplicate ACK.<br class=3D"">
15. In response to three duplicate ACKs, the Sender now enters Fast =
Retransmission for the dropped packet, aka the ZWP.<br class=3D"">
<br class=3D"">
15+. The Sender will respond to all ACKs from the Receiver with a =
retransmission of the Zero Window Probe due to Fast Retransmission.<br =
class=3D"">
16+. The Receiver will respond to all Probes with ACK for SND.UNA-1 and =
a RWND of 0, until it=E2=80=99s user application reads the data.<br =
class=3D"">
<br class=3D"">
=E2=80=94=E2=80=94=E2=80=94=E2=80=94=E2=80=94<br class=3D"">
<br class=3D"">
On a fast network (i.e. LAN), and a slow receiver (i.e. an embedded =
device), this becomes a Denial of Service as the receiver ends up =
spending all its resources ACKing the Probes instead of reading from the =
socket.<br class=3D"">
<br class=3D"">
Suggested change, in RFC 5681:<br class=3D"">
Section 3.2, paragraph 2 says:<br class=3D"">
&nbsp; &nbsp; The TCP sender SHOULD use the "fast retransmit" algorithm =
to detect<br class=3D"">
&nbsp; &nbsp; and repair loss, based on incoming duplicate ACKs.&nbsp; =
The fast<br class=3D"">
&nbsp; &nbsp; retransmit algorithm uses the arrival of 3 duplicate ACKs =
(as defined<br class=3D"">
&nbsp; &nbsp; in section 2, without any intervening ACKs which move =
SND.UNA) as an<br class=3D"">
&nbsp; &nbsp; indication that a segment has been lost.&nbsp; After =
receiving 3 duplicate<br class=3D"">
&nbsp; &nbsp; ACKs, TCP performs a retransmission of what appears to be =
the missing<br class=3D"">
&nbsp; &nbsp; segment, without waiting for the retransmission timer to =
expire.<br class=3D"">
<br class=3D"">
It should say:<br class=3D"">
&nbsp; &nbsp; The TCP sender SHOULD use the "fast retransmit" algorithm =
to detect<br class=3D"">
&nbsp; &nbsp; and repair loss, based on incoming duplicate ACKs.&nbsp; =
The fast<br class=3D"">
&nbsp; &nbsp; retransmit algorithm uses the arrival of 3 duplicate ACKs =
(as defined<br class=3D"">
&nbsp; &nbsp; in section 2, without any intervening ACKs which move =
SND.UNA) as an<br class=3D"">
&nbsp; &nbsp; indication that a segment has been lost.&nbsp; After =
receiving 3 duplicate<br class=3D"">
&nbsp; &nbsp; ACKs, TCP performs a retransmission of what appears to be =
the missing<br class=3D"">
&nbsp; &nbsp; segment, without waiting for the retransmission timer to =
expire.<br class=3D"">
&nbsp; &nbsp; However, if all transmitted data from the sender has been =
acknowledged,<br class=3D"">
&nbsp; &nbsp; the duplicate ACKs are in response to a Zero Window Probe, =
AND the<br class=3D"">
&nbsp; &nbsp; advertised window is still zero, then the TCP sender MUST =
NOT consider<br class=3D"">
&nbsp; &nbsp; these to be duplicate ACKs in relation to the fast =
retransmit algorithm.<br class=3D"">
<br class=3D"">
<br class=3D"">
-Dan<br class=3D"">
______________________________<wbr class=3D"">_________________<br =
class=3D"">
tcpm mailing list<br class=3D"">
<a href=3D"mailto:tcpm@ietf.org" class=3D"">tcpm@ietf.org</a><br =
class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" =
target=3D"_blank" class=3D"">https://www.ietf.org/mailman/<wbr =
class=3D"">listinfo/tcpm</a><br class=3D"">
</blockquote></div><br class=3D""></div>
</div></blockquote></div><br class=3D""></div></body></html>=

--Apple-Mail=_71553A00-E789-4DCC-B488-E1D9CF5E7751--


From nobody Wed Dec 21 08:08:31 2016
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From: Jeremy Harris <jgh@wizmail.org>
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Subject: Re: [tcpm] Fwd: New Version Notification for draft-harris-estab-wscale-00.txt
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On 01/11/16 15:55, Jeremy Harris wrote:
> I have submitted a draft extending the use of the TCP Wscale option.

[Apologies for the threading break; I didn't receive Yoshifumi's
mail directly]

Yoshifumi Nishida <nishida@sfc.wide.ad.jp> wrote:


> 1: This might be beneficial for middle points that want to do some
> network analysis. However, I don't see benefits for sender and
> receiver. So, unless there are strong relationships between network
> analysis app and sender/receiver, sender/receiver won't have good
> motivation to implement and activate this feature. However, such
> relationship might not always exist.

Indeed.  The intent is to make it low-enough cost that it will be
at least commonly available, preferably commonly enabled as a default.

> 2: For the same reason, it seems to be difficult for senders to decide
> whether activate this feature or not.

If it is not enabled by default on a system, it should be possible for
operators to enable it when needed for a packet-capture debug step.

I'll add some words pointing this out.

>    Also, it's difficult to decide the interval for sending this option
> as it totally depends on network analysis apps.
>    It seems that the draft presumes to use one fixed value for
> interval, but I'm not sure if we can find a proper value as I guess
> the requirements can be varied for some reasons.

It's not intended that the option is required in the flow for proper
operation of anything, including any network analysis applications.
It is solely intended as a convenience datum which may be discovered
either manually or by an analysis application.

The draft does not require any specific interval, and I see no
reason for a fixed interval either.  A fixed number of packets is
simple to code, but that should not be part of the specification.

> 3: The requirements in Section 2.3 won't be needed as it's already
> stated in Section 2.2 in RFC7323.
>     So, I think this proposal will require sender-only modification.
> (which is better than updating both sides)

I'll change the wording, to clarify that RFC 7323 still controls this
aspect.

> 4: I think you'll need to refer RFC7323 instead of 1323 as 1323 is obsoleted.

Yes.

-- 
Cheers,
  Jeremy


From nobody Fri Dec 23 01:32:08 2016
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Subject: [tcpm] tcpm - New Meeting Session Request for IETF 98
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A new meeting session request has just been submitted by Michael Scharf, a Chair of the tcpm working group.


---------------------------------------------------------
Working Group Name: TCP Maintenance and Minor Extensions
Area Name: Transport Area
Session Requester: Michael Scharf

Number of Sessions: 1
Length of Session(s):  2.5 Hours
Number of Attendees: 80
Conflicts to Avoid: 
 First Priority: iccrg tcpinc mptcp taps tsvarea tsvwg quic
 Second Priority: httpbis lwig maprg rtcweb rmcat teas
 Third Priority: ippm


Special Requests:
  A plus BoF/WG would also be a first priority conflict
---------------------------------------------------------

