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From: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
To: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: WGLC for draft-ietf-tcpm-dctcp-04
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Date: Wed, 1 Mar 2017 10:18:12 +0000
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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I have a comment on the following text:

   It would be useful to implement DCTCP as additional actions on top of
   an existing congestion control algorithm like NewReno.  The DCTCP
   implementation MAY also allow configuration of resetting the value of
   DCTCP.Alpha as part of processing any loss episodes.

Setting alpha to its max value (btw resetting might imply setting to zero) =
has many=20
complications and in the current Linux implementation turned out to not fun=
ction.
The Linux patch I provided additionally halves the window (once per RTT) wh=
en a=20
drop is encountered, independent from alpha. This is also, as far as I unde=
rstood,=20
how the Windows version works. Both implementations are working now in
drop-based networks next to Classic TCP flows.

As far as I know there is no working implementation that is setting alpha t=
o max so=20
I propose to describe the working implementation instead of a theoretical o=
ne.

For the rest I support publication.

Koen.

> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael
> (Nokia - DE)
> Sent: dinsdag 14 februari 2017 17:32
> To: tcpm@ietf.org
> Cc: tcpm-chairs@ietf.org
> Subject: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
>=20
> Hi,
>=20
> TCPM chairs are not aware of outstanding issues in draft-ietf-tcpm-dctcp-=
04
> "Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters".
>=20
> Therefore, this e-mail starts a working group last call for draft-ietf-tc=
pm-
> dctcp-04.
>=20
> The WGLC runs until Friday March 3. Please send any comments to the TCPM
> mailing list by then. Statements supporting the publication of the docume=
nt
> are also helpful.
>=20
> We assume that TCPM working group is aware of the IPR disclosure related
> to draft-bensley-tcpm-dctcp-00 (https://datatracker.ietf.org/ipr/2319/). =
A
> separate note will be sent out to confirm that all IPR has already been
> disclosed, in conformance with BCPs 78 and 79.
>=20
> The draft is available at https://tools.ietf.org/html/draft-ietf-tcpm-dct=
cp-04
>=20
> Thanks
>=20
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Mar  3 15:56:20 2017
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Subject: [tcpm] tcpm - Requested session has been scheduled for IETF 98
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Dear Michael Scharf,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 

tcpm Session 1 (2:30:00)
    Wednesday, Morning Session I 0900-1130
    Room Name: Zurich E/F size: 200
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
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


People who must be present:
  Yoshifumi Nishida
  Michael Tuexen
  Michael Scharf
  Mirja Kuehlewind

Resources Requested:
  Meetecho support in room

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


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From: Praveen Balasubramanian <pravb@microsoft.com>
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From: Praveen Balasubramanian <pravb@microsoft.com>
To: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>, "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>, "tcpm@ietf.org" <tcpm@ietf.org>
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Cc: "tcpm-chairs@ietf.org" <tcpm-chairs@ietf.org>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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The text around allowing a reset of Alpha was added on suggestions from the=
 Linux implementation experience. The text already suggests that this is op=
tional. The Windows implementation does not reset the value of Alpha or the=
 other tracking variables. I think that we don't need to remove this text. =
If you still strongly object and I don't hear any other opinions, I will re=
move it in the next revision.=20

>> The Linux patch I provided additionally halves the window (once per RTT)=
 when a drop is encountered, independent from alpha
The handling of a timeout and dup acks should ideally be identical to NewRe=
no. The only change in response is in presence of ECN marks.

Thanks

-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of De Schepper, Koen (N=
okia - BE)
Sent: Wednesday, March 1, 2017 2:18 AM
To: Scharf, Michael (Nokia - DE) <michael.scharf@nokia.com>; tcpm@ietf.org
Cc: tcpm-chairs@ietf.org
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04

I have a comment on the following text:

   It would be useful to implement DCTCP as additional actions on top of
   an existing congestion control algorithm like NewReno.  The DCTCP
   implementation MAY also allow configuration of resetting the value of
   DCTCP.Alpha as part of processing any loss episodes.

Setting alpha to its max value (btw resetting might imply setting to zero) =
has many complications and in the current Linux implementation turned out t=
o not function.
The Linux patch I provided additionally halves the window (once per RTT) wh=
en a drop is encountered, independent from alpha. This is also, as far as I=
 understood, how the Windows version works. Both implementations are workin=
g now in drop-based networks next to Classic TCP flows.

As far as I know there is no working implementation that is setting alpha t=
o max so I propose to describe the working implementation instead of a theo=
retical one.

For the rest I support publication.

Koen.

> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael=20
> (Nokia - DE)
> Sent: dinsdag 14 februari 2017 17:32
> To: tcpm@ietf.org
> Cc: tcpm-chairs@ietf.org
> Subject: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
>=20
> Hi,
>=20
> TCPM chairs are not aware of outstanding issues in=20
> draft-ietf-tcpm-dctcp-04 "Datacenter TCP (DCTCP): TCP Congestion Control =
for Datacenters".
>=20
> Therefore, this e-mail starts a working group last call for=20
> draft-ietf-tcpm- dctcp-04.
>=20
> The WGLC runs until Friday March 3. Please send any comments to the=20
> TCPM mailing list by then. Statements supporting the publication of=20
> the document are also helpful.
>=20
> We assume that TCPM working group is aware of the IPR disclosure=20
> related to draft-bensley-tcpm-dctcp-00=20
> (https://datatracker.ietf.org/ipr/2319/). A separate note will be sent=20
> out to confirm that all IPR has already been disclosed, in conformance wi=
th BCPs 78 and 79.
>=20
> The draft is available at=20
> https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04
>=20
> Thanks
>=20
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

_______________________________________________
tcpm mailing list
tcpm@ietf.org
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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On Fri, Mar 3, 2017 at 4:41 PM, Praveen Balasubramanian
<pravb@microsoft.com> wrote:
>
> Yes DCTCP adjusts cwnd at most once every round trip.
>
>
>
> See step 4:
>
>
>
> If the acknowledgment number is less than or equal to
>
>        DCTCP.WindowEnd, stop processing.  Otherwise, the end of the
>
>        observation window has been reached, so proceed to update the
>
>        congestion estimate as follows:
>
>
>
> So steps 5 through 8 are conditional and only executed once per round. Wh=
at is a bit confusing is that the cwnd calculation is not indicated as step=
 9. I will fix this in an update to the draft.
>
>
>
> Other than this we also have the following clarifying sentence:
>
> Just as specified in [RFC3168], DCTCP does not react to congestion
>
>    indications more than once for every window of data.
Thanks for clarification. I notice the draft is also less clear about
the window reduction on packet losses. For example, if sender
experiences ECE and then losses in the same window of data. Does DCTCP
reduces the window twice (once on ECN, then again on Loss)?

The draft implies cwnd would be reduced twice:

A DCTCP sender MUST deal with loss episodes in the same way as
   conventional TCP.  In case of a timeout or fast retransmit or any
   change in delay (for delay based congestion control), the cwnd and
   other state variables like ssthresh must be changed in the same way
   that a conventional TCP would have changed them.

AFAICT Linux's implementation for both DCTCP and RFC3168 would only
reduce cwnd once on ECN + loss. Since this is a common scenario, it'd
be nice to be explicit in the DCTCP draft.

>
>
>
>
>
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Yuchung Cheng
> Sent: Tuesday, February 28, 2017 3:03 PM
> To: tcpm@ietf.org
> Cc: Abdul Kabbani <akabbani@google.com>
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
>
>
>
> The wordings on cwnd adjustment in section 3.3 can be more clear:
>
>
>
> """
>
> Rather than always halving the congestion window as described in
>
>    [RFC3168], when the sender receives an indication of congestion
>
>    (ECE), the sender MUST update cwnd as follows:
>
>
>
>       cwnd =3D cwnd * (1 - DCTCP.Alpha / 2)
>
>
>
>    Thus, when no sent byte experienced congestion, DCTCP.Alpha equals
>
>    zero, and cwnd is left unchanged.  When all sent bytes experienced
>
>    congestion, DCTCP.Alpha equals one, and cwnd is reduced by half.
>
>    Lower levels of congestion will result in correspondingly smaller
>
>    reductions to cwnd.
>
> """
>
>
>
> AFAICT DCTCP paper only adjusts cwnd at most every round trip (which is a=
lso how Linux implements it). But the text implies the cwnd reduction can h=
appen for every ECE mark?
>
>
>
>
>
> On Tue, Feb 14, 2017 at 8:31 AM, Scharf, Michael (Nokia - DE) <michael.sc=
harf@nokia.com> wrote:
>
> Hi,
>
> TCPM chairs are not aware of outstanding issues in draft-ietf-tcpm-dctcp-=
04 "Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters".
>
> Therefore, this e-mail starts a working group last call for draft-ietf-tc=
pm-dctcp-04.
>
> The WGLC runs until Friday March 3. Please send any comments to the TCPM =
mailing list by then. Statements supporting the publication of the document=
 are also helpful.
>
> We assume that TCPM working group is aware of the IPR disclosure related =
to draft-bensley-tcpm-dctcp-00 (https://datatracker.ietf.org/ipr/2319/). A =
separate note will be sent out to confirm that all IPR has already been dis=
closed, in conformance with BCPs 78 and 79.
>
> The draft is available at https://tools.ietf.org/html/draft-ietf-tcpm-dct=
cp-04
>
> Thanks
>
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>


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To: tcpm@ietf.org
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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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I support this document moving forward.

El 14/02/17 a las 17:31, Scharf, Michael (Nokia - DE) escribió:
> Hi,
>
> TCPM chairs are not aware of outstanding issues in draft-ietf-tcpm-dctcp-04 "Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters".
>
> Therefore, this e-mail starts a working group last call for draft-ietf-tcpm-dctcp-04.
>
> The WGLC runs until Friday March 3. Please send any comments to the TCPM mailing list by then. Statements supporting the publication of the document are also helpful.
>
> We assume that TCPM working group is aware of the IPR disclosure related to draft-bensley-tcpm-dctcp-00 (https://datatracker.ietf.org/ipr/2319/). A separate note will be sent out to confirm that all IPR has already been disclosed, in conformance with BCPs 78 and 79.
>
> The draft is available at https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04
>
> Thanks
>
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From nobody Mon Mar  6 08:01:14 2017
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tcpm] Agenda requests for IETF 98
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TCP Alternative Backoff with ECN (ABE)
draft-ietf-tcpm-alternativebackoff-ecn
Presenter G Fairhurst / M Welzl
Total time (including Q&A): 5 minutes

We'd be happy for a slide to be presented on ABE status, now that this 
has been adopted.

Gorry, Michael, Naeem.


> I will present updates on the draft as well as experimentation results
> that've been requested in previous meetings.
>
> On Mon, Feb 27, 2017 at 9:38 AM, Scharf, Michael (Nokia - DE)
> <michael.scharf@nokia.com <mailto:michael.scharf@nokia.com>> wrote:
>
>     Hi all,
>
>     For the Chicago meeting, tentatively (!) TCPM got the 2.5h slot on
>     Wednesday morning.
>
>     If you want to present something in the TCPM meeting, please let the
>     chairs know until Sunday March 12:
>
>     * Draft name / title
>     * Presenter
>     * Total time (including Q&A)
>
>     Thanks!
>
>     Michael
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>     <https://www.ietf.org/mailman/listinfo/tcpm>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
To: Praveen Balasubramanian <pravb@microsoft.com>, "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>, "tcpm@ietf.org" <tcpm@ietf.org>
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> -----Original Message-----
> From: Praveen Balasubramanian [mailto:pravb@microsoft.com]
> Sent: zaterdag 4 maart 2017 1:52
> To: De Schepper, Koen (Nokia - BE) <koen.de_schepper@nokia-bell-
> labs.com>; Scharf, Michael (Nokia - DE) <michael.scharf@nokia.com>;
> tcpm@ietf.org
> Cc: tcpm-chairs@ietf.org
> Subject: RE: WGLC for draft-ietf-tcpm-dctcp-04
>=20
> The text around allowing a reset of Alpha was added on suggestions from t=
he
> Linux implementation experience.

[KDS] I see it was added from the Linux implementation, but clearly not bas=
ed on
the experience ;-). This implementation is not working.
I don't know if the FreeBSD implementation has drop compatibility, and how =
it works.
If that has a working implementation by setting alpha to max, we can keep t=
he text.

> The text already suggests that this is
> optional.

[KDS] That is OK, as in the Linux patch it is also optional (default on).=20
The new parameter is called dctcp_drop_compatible as it does not have anyth=
ing
to do with alpha clamping.

Koen.

> The Windows implementation does not reset the value of Alpha or
> the other tracking variables. I think that we don't need to remove this t=
ext. If
> you still strongly object and I don't hear any other opinions, I will rem=
ove it in
> the next revision.
>=20
> >> The Linux patch I provided additionally halves the window (once per RT=
T)
> when a drop is encountered, independent from alpha
> The handling of a timeout and dup acks should ideally be identical to
> NewReno. The only change in response is in presence of ECN marks.
>=20
> Thanks
>=20
> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of De Schepper,
> Koen (Nokia - BE)
> Sent: Wednesday, March 1, 2017 2:18 AM
> To: Scharf, Michael (Nokia - DE) <michael.scharf@nokia.com>; tcpm@ietf.or=
g
> Cc: tcpm-chairs@ietf.org
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
>=20
> I have a comment on the following text:
>=20
>    It would be useful to implement DCTCP as additional actions on top of
>    an existing congestion control algorithm like NewReno.  The DCTCP
>    implementation MAY also allow configuration of resetting the value of
>    DCTCP.Alpha as part of processing any loss episodes.
>=20
> Setting alpha to its max value (btw resetting might imply setting to zero=
) has
> many complications and in the current Linux implementation turned out to
> not function.
> The Linux patch I provided additionally halves the window (once per RTT)
> when a drop is encountered, independent from alpha. This is also, as far =
as I
> understood, how the Windows version works. Both implementations are
> working now in drop-based networks next to Classic TCP flows.
>=20
> As far as I know there is no working implementation that is setting alpha=
 to
> max so I propose to describe the working implementation instead of a
> theoretical one.
>=20
> For the rest I support publication.
>=20
> Koen.
>=20
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael
> > (Nokia - DE)
> > Sent: dinsdag 14 februari 2017 17:32
> > To: tcpm@ietf.org
> > Cc: tcpm-chairs@ietf.org
> > Subject: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
> >
> > Hi,
> >
> > TCPM chairs are not aware of outstanding issues in
> > draft-ietf-tcpm-dctcp-04 "Datacenter TCP (DCTCP): TCP Congestion Contro=
l
> for Datacenters".
> >
> > Therefore, this e-mail starts a working group last call for
> > draft-ietf-tcpm- dctcp-04.
> >
> > The WGLC runs until Friday March 3. Please send any comments to the
> > TCPM mailing list by then. Statements supporting the publication of
> > the document are also helpful.
> >
> > We assume that TCPM working group is aware of the IPR disclosure
> > related to draft-bensley-tcpm-dctcp-00
> > (https://datatracker.ietf.org/ipr/2319/). A separate note will be sent
> > out to confirm that all IPR has already been disclosed, in conformance =
with
> BCPs 78 and 79.
> >
> > The draft is available at
> > https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04
> >
> > Thanks
> >
> > Michael
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Mar 10 12:33:23 2017
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rto-consider-05.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           : Retransmission Timeout Requirements
        Author          : Mark Allman
	Filename        : draft-ietf-tcpm-rto-consider-05.txt
	Pages           : 9
	Date            : 2017-03-10

Abstract:
    Ensuring reliable communication often manifests in a timeout and
    retry mechanism.  Each implementation of a retransmission timeout
    mechanism represents a balance between correctness and timeliness
    and therefore no implementation suits all situations.  This document
    provides high-level requirements for retransmission timeout schemes
    appropriate for general use in the Internet.  Within the
    requirements, implementations have latitude to define particulars
    that best address each situation.

Terminology

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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rto-consider-05


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:
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From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
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Subject: [tcpm] [Fwd: New Version Notification for draft-gomez-lwig-tcp-constrained-node-networks-02.txt]
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Dear all,

We have submitted version -02 of the "TCP over Constrained-Node Networks"
draft. Please find pointers to the updated document below.

Cheers,

Carles

---------------------------- Original Message ----------------------------
Subject: New Version Notification for
draft-gomez-lwig-tcp-constrained-node-networks-02.txt
From:    internet-drafts@ietf.org
Date:    Sat, March 11, 2017 8:33 am
To:      "Jon Crowcroft" <jon.crowcroft@cl.cam.ac.uk>
         "Carles Gomez" <carlesgo@entel.upc.edu>
--------------------------------------------------------------------------


A new version of I-D, draft-gomez-lwig-tcp-constrained-node-networks-02.txt
has been successfully submitted by Carles Gomez and posted to the
IETF repository.

Name:		draft-gomez-lwig-tcp-constrained-node-networks
Revision:	02
Title:		TCP over Constrained-Node Networks
Document date:	2017-03-10
Group:		Individual Submission
Pages:		15
URL:           
https://www.ietf.org/internet-drafts/draft-gomez-lwig-tcp-constrained-node-networks-02.txt
Status:        
https://datatracker.ietf.org/doc/draft-gomez-lwig-tcp-constrained-node-networks/
Htmlized:      
https://tools.ietf.org/html/draft-gomez-lwig-tcp-constrained-node-networks-02
Diff:          
https://www.ietf.org/rfcdiff?url2=draft-gomez-lwig-tcp-constrained-node-networks-02

Abstract:
   This document provides a profile for the Transmission Control
   Protocol (TCP) over Constrained-Node Networks (CNNs).  The
   overarching goal is to offer simple measures to allow for lightweight
   TCP implementation and suitable operation in such environments.




Please note that it may take a couple of minutes from the time of submission
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The IETF Secretariat




From nobody Sat Mar 11 00:31:53 2017
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Hi Michael,

First of all, sorry for the delay in this response, and thanks a lot for
your comments, suggestions and pointers.

We have updated the draft with the aim to address your comments. Latest
version is -02.

Please find below inline responses:

> I like this version better than the former one. Yet, I still have quite a
> number of observations:
>
> 1/ Target of the document (Best Current Practice)
>
> The survey in the Appendix is very useful and it would be good to expand
> it. It states:
>
> 7.2.  lwIP
>
>    lwIP is a TCP/IP stack, targetted for 8- and 16-bit microcontrollers.
>    lwIP has a total code size of ~14 kB to ~22 kB (which comprises
>    memory management, checksumming, network interfaces, IP, ICMP and
>    TCP), and a TCP code size of ~9 kB to ~14 kB [Dunk].
>
>    In contrast with uIP, lwIP decouples applications from the network
>    stack. lwIP supports a TCP transmission window greater than a single
>    segment, as well as buffering of incoming and outcoming data.  Other
>    implemented mechanisms comprise slow start, congestion avoidance,
>    fast retransmit and fast recovery.  SACK and Window Scale are not
>    implemented.
>
> So, it seems to me that the guidance written in the document (only a
> single window etc.) is actually *not* the best current practice actually
> used in lightweight implementations. So, to me the document needs
> significant changes to cover the lwIP implementation correctly, which I
> think would be appropriate.
>
> Or does the document target only environments for which this lwIP stack
> does not apply?

The correct intended status for the draft is Informational. The BCP
previously shown in the draft should have already been replaced by
"Informational" in -01. Solved in -02.

> 2/ Section 2
>
> I think you should better define the what exactly you mean by
> Constrained-Node Networks. For instance, is this limited to 6LoWPAN? What
> about some of the emerging 4G/5G radio interfaces that focus on energy
> constrained devices? Does the same guidance really apply to all (radio)
> networks?

CNNs are defined in RFC 7228 as networks whose characteristics are
influenced by being composed of a significant portion of constrained
nodes.

CNNs are not limited to 6LoWPAN/6Lo networks. Guidance may apply to other
networks as long as devices (and the links/networks they use) suffer the
type of constraints  described for constrained nodes. Please note that
the actual definition of CNNs [RFC 7228] is not tied to 6LoWPAN or any
particular type of network/technology.


> 2/ Section 4.2 Maximum Segment Size (MSS)
>
> I think this section would first benefit from a wider survey of link
> technologies before coming to conclusions.

-02 refers also to other technologies used in the CNN space (such as
MS/TP, IEEE 802.11ah and NB-IoT) which do not suffer the same degree of
frame (payload) size limitation as the ones already mentioned.

> This sentence needs rewording regarding the "in any case", which could be
> read in different ways in particular in isolation: "In any case, the TCP
> MSS must not be set to a value leading to an IPv6 datagram size exceeding
> 1280 bytes."

Agreed and done.

> The next sentence also needs rewording: "If a link layer technology used
> by a constrained device offers a link layer MTU greater than 1280 bytes,
> it is still useful to set the MSS so that IPv6 datagram size does not
> exceed 1280 bytes, in order to avoid issues due to Internet links that do
> not support greater MTUs." The Internet as a whole has means to handle
> MTUs larger than 1280 bytes, so this guidance seems not in general
> "useful".

Agreed. This text has been removed.

> 3/ Section 4.3.  Window Size
>
> Why does this document include this recommendation even if there are
> relevant lightweight stacks that can do much better?
>
> I think what is more appropriate would be a wording such as: "A TCP stack
> can reduce the implementation complexity by advertising a TCP window size
> of one MSS, and also transmit at most one MSS of unacknowledged data, at
> the cost of decreased performance."

Agreed and applied to the document.

> And the following sentence should probably replace "use" by "allow": "For
> devices that can afford greater TCP window size, it may be useful to use
> window sizes of at least five MSSs, in order to allow Fast Retransmit and
> Fast Recovery [RFC5681]." If a sender has larger window sizes, it has to
> implement congestion control, and as a result the window size will be
> variable. After an RTO, it can be as low as one MSS and is thus less than
> 5 MSS. I could read this sentence as suggesting a fixed minimum window
> size, which is not appropriate.

Agreed and applied to the document.

> 4/ Section 4.4.  RTO estimation
>
> This section must be entirely rewritten. First of all, this section should
> cite draft-ietf-tcpm-rto-consider and consider the guidance therein (in
> particular Section 3). One of the key insights of
> draft-ietf-tcpm-rto-consider is that there is typically no single best
> algorithm, as there are always tradeoffs. If this conclusion was wrong,
> this document would have to explicitly explain why.
> draft-ietf-tsvwg-rfc5405bis could also be referenced.
>
> To me, the key aspect that this section should probably discuss is that if
> a sender uses only very small window sizes and can hardly use fast
> retransmit (or even SACK), the RTO algorithm has a larger impact on the
> performance than for a more powerful TCP stack. This may open room for
> some algorithm tuning. But then the downsides of doing that should also be
> discussed (e.g., spurious timeouts etc.).

Modified the section accordingly. However, CoCoA RTO work is still
mentioned (now as a note on a related area, just for informational
purposes).

> 5/ Section 4.5.  TCP connection lifetime
>
> I think in this section you should better separate what a TCP stack should
> support, or not, and how the TCP stack should be used by applications,
> e.g., whether to close connections or not.

Agreed. This will be done in future revisions.

> 6/ Section 4.6.  Explicit congestion notification
>
> As motivation for ECN the text notes "As transactional data size
> decreases, the probability of detecting congestion by the presence of
> three duplicate ACKs decreases." But three DUPACKs are quite impossible if
> the sender follows Section 4.3, right?

After the changes in section 4.3, this comment is hopefully addressed.

> 7/ Section TCP options
>
> The sentence
>
>   "A TCP implementation for a constrained device that uses a single-MSS
> TCP receive or transmit window size will benefit from not supporting,
> and ignoring if received, the following TCP options: Window scale
> [RFC1323], TCP Timestamps [RFC1323], Selective Acknowledgements (SACK)
> and SACK-Permitted [RFC2018]."
>
> should at least be reworded to
>
>   "A TCP implementation for a constrained device that uses a single-MSS
> TCP receive or transmit window size may not benefit supporting from the
> following TCP options: Window scale [RFC1323], TCP Timestamps [RFC1323],
> Selective Acknowledgements (SACK) and SACK-Permitted [RFC2018]."

Agreed. Applied to the document.

> I do not believe that RFC 2119 language of the form "Other TCP options
> SHOULD NOT be used, in keeping with the principle of lightweight
> operation." is appropriate, in particular since it may also exclude future
> new TCP options that specifically target lightweight operation.

Sorry, we missed that one in -01. Removed this 2119 form now.

> Also, this section may have to consider TFO options as this is discussed
> elsewhere in the draft.

TFO is now mentioned in the section for completeness, although the main
description of TFO comes already in section 4.5.

> 8/ Section 4.8.  Delayed Acknowledgments
>
> I think the last paragraph in this section should be moved earlier in the
> document, i.e., the document should probably explain early the different
> types of workloads in a CCN as well as their possibly different needs
> regarding TCP stack configuration. The assumption that most communication
> sizes is smaller than one MSS is very fundamental and if it is violated, a
> lot of the reasoning in the document does not apply.

Updated text in -02 just describes when delayed ACKs are recommended and
when they are not, without stating whether some scenarios are more likely
than others. Even if my current *impression* is the one formerly expressed
in the document, a more solid support for it would be needed, and also the
way CNNs are used may vary over time (and it definitely varies across
scenarios). Implementers or administrators may choose the most appropriate
setting for each particular scenario.

> 9/ Section 5.  Security Considerations
>
> I am not a security expert but here is a question: If an attacker knows
> that a TCP stack only uses a window of one MSS (because it sits in a CCN),
> and if some packet can be passively intercepted, isn't the likelihood of
> successfully injecting segments into an established TCP connection larger?

Do you mean something like a replay attack? Generating a spoofed duplicate?

Why is the likelihood of successfully injecting segments greater? Is it
e.g. because it makes it easier for the attacker to appear as a legitimate
sender when the actual legitimate sender is silent?

> Another aspect to be mentioned is that certain TCP options can improve TCP
> security (e.g., MD5, TCP-AO) but come along with complexity.

Agreed. A brief description of these has been added.

Cheers,

Carles

> Thanks
>
> Michael


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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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I can confirm that all my earlier concerns have been addressed.
So I support this draft going forward, assuming the proposed 
improvements around loss handling currently being discussed will be sorted.


Bob

On 14/02/17 16:31, Scharf, Michael (Nokia - DE) wrote:
> Hi,
>
> TCPM chairs are not aware of outstanding issues in draft-ietf-tcpm-dctcp-04 "Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters".
>
> Therefore, this e-mail starts a working group last call for draft-ietf-tcpm-dctcp-04.
>
> The WGLC runs until Friday March 3. Please send any comments to the TCPM mailing list by then. Statements supporting the publication of the document are also helpful.
>
> We assume that TCPM working group is aware of the IPR disclosure related to draft-bensley-tcpm-dctcp-00 (https://datatracker.ietf.org/ipr/2319/). A separate note will be sent out to confirm that all IPR has already been disclosed, in conformance with BCPs 78 and 79.
>
> The draft is available at https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04
>
> Thanks
>
> Michael
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

-- 
________________________________________________________________
Bob Briscoe                               http://bobbriscoe.net/


From nobody Mon Mar 13 02:52:21 2017
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Date: Mon, 13 Mar 2017 09:51:47 +0000
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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I think this document looks in good shape. I have two sets of WG 
comments relating to the language used in the current draft:

(1) I think the language surrounding loss treatment needs to be still 
strengthened.
(2) I would like to see the contradition with RFC3168 reworked to avoid 
this INFO docuemnt attempting to over-ride that PS.

Best wishes,

Gorry

---
In section 1:

" This protocol is not meant for uncontrolled deployment in the global 
Internet. "

- To me this statement seems far too opn, and the draft has alreday 
found a clearer recommendations that I would agree to, as cited in the 
abstract:

"DCTCP as described in this draft
    is applicable to deployments in controlled environments like
    datacenters but it MUST NOT be deployed over the public Internet
    without additional measures"

- Is it possible to re-use this statement in place of the one quoted 
above at the end of the conclusion?

---

I'd also really like to see a statement about loss-treatment in the 
introduction. (see my comemt on section 5, but calling-out from the 
start that loss triggers standard TCP congestion control measures is I 
think a very important thing to make the reader aware about. Especially 
when the introduction (quite correctly) points to the merits of ECN 
reacting faster than loss-based methods.

---
In section 3:

"A DCTCP sender MUST deal with loss episodes in the same way as
    conventional TCP."

- I agree. Could we replace "deal with" perhaps with something like 
"react to"

---

3.4.  Handling of SYN, SYN-ACK, RST Packets

   " The switching fabric can drop TCP packets that do not have the ECT
    set in the IP header.  If SYN and SYN-ACK packets for DCTCP
    connections do not have ECT set, they will be dropped with high
    probability.  For DCTCP connections, the sender SHOULD set ECT for
    SYN, SYN-ACK and RST packets."

- I'd take the position that the fabric can and will drop any packets
under overload, as per RFC7567. I'd prefer to explicitly state that
to avoid a misconception that ECT eliminates all drop (rather than
nearly all drops).

---

Section 4:

   "It is RECOMMENDED
    that an implementation provide a configuration knob that will cause
    ECT to be set on such control packes"
- please correct typo /packes"

- I'm not a fan of an informational document over-riding a PS. I do not 
think
this is appropriate.

- I suggest once way to help, could be to say that
   "future versions of this protocol provide a configuration knob that 
will cause
    ECT to be set on such control packets [xx]."

and then  cite "draft-ietf-tsvwg-ecn-experimentation".

- The present text is clearly at odds with the RFC series.

---

In section 5:

The following statement is correct, in as much as SCTP needs to go back 
loss-based CC, but I'd prefer personally to see this expressed as 
"reverts to loss-based congestion control" - to make this a little more 
clear, and I'd much rather it said when it encounters loss, rather than 
perhaps suggesting this is only drop-tail:

"DCTCP will degrade to loss-based congestion control when transiting a 
congested drop-tail link."

- e.g. something like this:

"DCTCP will revert to loss-based congestion control when packet loss is 
experienced (e.g. when
transiting a congested drop-tail link, or a link with an AQM drop 
behaviour).

---

In section 6:

This statement could be better written:

"If the estimation gain is small relative to the packet loss rate, the 
estimate may not be too inaccurate."

- First, I think the loss rate noted, may be the loss rate for the 
return path (i.e., the opposite
direction to the packets bing CE-marked?)
- Second, may not be too inacurrate sounds odd - do you think the 
accuracy is acceptable for normal use?

If this relates to ACK loss, the following statement isn't necessarily 
true. And especially wculd be far from true for an internet with 
asymmetric routing:

"o  If packet loss mostly occurs under heavy congestion, most drops
       will occur during an unbroken string of CE packets, and the
       estimate will be unaffected.
"


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From: Praveen Balasubramanian <pravb@microsoft.com>
To: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>, "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: WGLC for draft-ietf-tcpm-dctcp-04
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Cc: "tcpm-chairs@ietf.org" <tcpm-chairs@ietf.org>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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> The DCTCP
>    implementation MAY also allow configuration of resetting the value of
>    DCTCP.Alpha as part of processing any loss episodes.

Changing this to:

If experimental results show a benefit, a DCTCP implementation MAY reset th=
e value of DCTCP.Alpha to 1 value as part of processing any loss episodes.=
=20

Thanks

-----Original Message-----
From: De Schepper, Koen (Nokia - BE) [mailto:koen.de_schepper@nokia-bell-la=
bs.com]=20
Sent: Wednesday, March 8, 2017 8:24 AM
To: Praveen Balasubramanian <pravb@microsoft.com>; Scharf, Michael (Nokia -=
 DE) <michael.scharf@nokia.com>; tcpm@ietf.org
Cc: tcpm-chairs@ietf.org
Subject: RE: WGLC for draft-ietf-tcpm-dctcp-04


> -----Original Message-----
> From: Praveen Balasubramanian [mailto:pravb@microsoft.com]
> Sent: zaterdag 4 maart 2017 1:52
> To: De Schepper, Koen (Nokia - BE) <koen.de_schepper@nokia-bell-=20
> labs.com>; Scharf, Michael (Nokia - DE) <michael.scharf@nokia.com>;=20
> tcpm@ietf.org
> Cc: tcpm-chairs@ietf.org
> Subject: RE: WGLC for draft-ietf-tcpm-dctcp-04
>=20
> The text around allowing a reset of Alpha was added on suggestions=20
> from the Linux implementation experience.

[KDS] I see it was added from the Linux implementation, but clearly not bas=
ed on the experience ;-). This implementation is not working.
I don't know if the FreeBSD implementation has drop compatibility, and how =
it works.
If that has a working implementation by setting alpha to max, we can keep t=
he text.

> The text already suggests that this is optional.

[KDS] That is OK, as in the Linux patch it is also optional (default on).=20
The new parameter is called dctcp_drop_compatible as it does not have anyth=
ing to do with alpha clamping.

Koen.

> The Windows implementation does not reset the value of Alpha or the=20
> other tracking variables. I think that we don't need to remove this=20
> text. If you still strongly object and I don't hear any other=20
> opinions, I will remove it in the next revision.
>=20
> >> The Linux patch I provided additionally halves the window (once per=20
> >> RTT)
> when a drop is encountered, independent from alpha The handling of a=20
> timeout and dup acks should ideally be identical to NewReno. The only=20
> change in response is in presence of ECN marks.
>=20
> Thanks
>=20
> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of De Schepper,=20
> Koen (Nokia - BE)
> Sent: Wednesday, March 1, 2017 2:18 AM
> To: Scharf, Michael (Nokia - DE) <michael.scharf@nokia.com>;=20
> tcpm@ietf.org
> Cc: tcpm-chairs@ietf.org
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
>=20
> I have a comment on the following text:
>=20
>    It would be useful to implement DCTCP as additional actions on top of
>    an existing congestion control algorithm like NewReno.  The DCTCP
>    implementation MAY also allow configuration of resetting the value of
>    DCTCP.Alpha as part of processing any loss episodes.
>=20
> Setting alpha to its max value (btw resetting might imply setting to=20
> zero) has many complications and in the current Linux implementation=20
> turned out to not function.
> The Linux patch I provided additionally halves the window (once per=20
> RTT) when a drop is encountered, independent from alpha. This is also,=20
> as far as I understood, how the Windows version works. Both=20
> implementations are working now in drop-based networks next to Classic TC=
P flows.
>=20
> As far as I know there is no working implementation that is setting=20
> alpha to max so I propose to describe the working implementation=20
> instead of a theoretical one.
>=20
> For the rest I support publication.
>=20
> Koen.
>=20
> > -----Original Message-----
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf,=20
> > Michael (Nokia - DE)
> > Sent: dinsdag 14 februari 2017 17:32
> > To: tcpm@ietf.org
> > Cc: tcpm-chairs@ietf.org
> > Subject: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
> >
> > Hi,
> >
> > TCPM chairs are not aware of outstanding issues in
> > draft-ietf-tcpm-dctcp-04 "Datacenter TCP (DCTCP): TCP Congestion=20
> > Control
> for Datacenters".
> >
> > Therefore, this e-mail starts a working group last call for
> > draft-ietf-tcpm- dctcp-04.
> >
> > The WGLC runs until Friday March 3. Please send any comments to the=20
> > TCPM mailing list by then. Statements supporting the publication of=20
> > the document are also helpful.
> >
> > We assume that TCPM working group is aware of the IPR disclosure=20
> > related to draft-bensley-tcpm-dctcp-00=20
> > (https://datatracker.ietf.org/ipr/2319/). A separate note will be=20
> > sent out to confirm that all IPR has already been disclosed, in=20
> > conformance with
> BCPs 78 and 79.
> >
> > The draft is available at
> > https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04
> >
> > Thanks
> >
> > Michael
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon Mar 13 13:28:54 2017
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rack-02.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           : RACK: a time-based fast loss detection algorithm for TCP
        Authors         : Yuchung Cheng
                          Neal Cardwell
                          Nandita Dukkipati
	Filename        : draft-ietf-tcpm-rack-02.txt
	Pages           : 22
	Date            : 2017-03-13

Abstract:
   This document presents a new TCP loss detection algorithm called RACK
   ("Recent ACKnowledgment").  RACK uses the notion of time, instead of
   packet or sequence counts, to detect losses, for modern TCP
   implementations that can support per-packet timestamps and the
   selective acknowledgment (SACK) option.  It is intended to replace
   the conventional DUPACK threshold approach and its variants, as well
   as other nonstandard approaches.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rack-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rack-02


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 Mon Mar 13 15:12:36 2017
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 13 Mar 2017 15:11:52 -0700
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Cc: Abdul Kabbani <akabbani@google.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
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Content-Type: text/plain; charset=UTF-8

On Mon, Mar 13, 2017 at 9:38 AM, Praveen Balasubramanian <
pravb@microsoft.com> wrote:

> > Thanks for clarification. I notice the draft is also less clear about
> the window reduction on packet losses. For example, if sender experiences
> ECE and then losses in the same window of > data. Does DCTCP reduces the
> window twice (once on ECN, then again on Loss)?
>
> DCTCP will reduce the window only once per window of data for the loss
> signals of ECN and duplicate ACKs. On a timeout cwnd will be
> unconditionally reduced to 1 MSS even if there was a preceding cwnd
> reduction due to a loss signal in a given window of data.
>
> > The draft implies cwnd would be reduced twice:
>
> > A DCTCP sender MUST deal with loss episodes in the same way as
> >    conventional TCP.  In case of a timeout or fast retransmit or any
> >    change in delay (for delay based congestion control), the cwnd and
> >    other state variables like ssthresh must be changed in the same way
> >    that a conventional TCP would have changed them.
>
> This wording comes from the fact that in Windows at least DCTCP is
> implemented as an extension to NewReno, in the sense that only the ECN
> handler behaves differently. I will change this wording to explicitly
> mention the above behavior of handling other loss events.
>
Thanks for the clarification. I assume you mean Reno not NewReno. NewReno
is not a congestion control... okay I really need to start a thread to
correct this.



>
> > AFAICT Linux's implementation for both DCTCP and RFC3168 would only
> reduce cwnd once on ECN + loss. Since this is a common scenario, it'd be
> nice to be explicit in the DCTCP draft.
>
> Yes agreed. We will publish an update soon.
>
> Thanks
>
>
> -----Original Message-----
> From: Yuchung Cheng [mailto:ycheng@google.com]
> Sent: Sunday, March 5, 2017 8:34 AM
> To: Praveen Balasubramanian <pravb@microsoft.com>
> Cc: tcpm@ietf.org; Abdul Kabbani <akabbani@google.com>
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
>
> On Fri, Mar 3, 2017 at 4:41 PM, Praveen Balasubramanian <
> pravb@microsoft.com> wrote:
> >
> > Yes DCTCP adjusts cwnd at most once every round trip.
> >
> >
> >
> > See step 4:
> >
> >
> >
> > If the acknowledgment number is less than or equal to
> >
> >        DCTCP.WindowEnd, stop processing.  Otherwise, the end of the
> >
> >        observation window has been reached, so proceed to update the
> >
> >        congestion estimate as follows:
> >
> >
> >
> > So steps 5 through 8 are conditional and only executed once per round.
> What is a bit confusing is that the cwnd calculation is not indicated as
> step 9. I will fix this in an update to the draft.
> >
> >
> >
> > Other than this we also have the following clarifying sentence:
> >
> > Just as specified in [RFC3168], DCTCP does not react to congestion
> >
> >    indications more than once for every window of data.
> Thanks for clarification. I notice the draft is also less clear about the
> window reduction on packet losses. For example, if sender experiences ECE
> and then losses in the same window of data. Does DCTCP reduces the window
> twice (once on ECN, then again on Loss)?
>
> The draft implies cwnd would be reduced twice:
>
> A DCTCP sender MUST deal with loss episodes in the same way as
>    conventional TCP.  In case of a timeout or fast retransmit or any
>    change in delay (for delay based congestion control), the cwnd and
>    other state variables like ssthresh must be changed in the same way
>    that a conventional TCP would have changed them.
>
> AFAICT Linux's implementation for both DCTCP and RFC3168 would only reduce
> cwnd once on ECN + loss. Since this is a common scenario, it'd be nice to
> be explicit in the DCTCP draft.
>
> >
> >
> >
> >
> >
> > From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Yuchung Cheng
> > Sent: Tuesday, February 28, 2017 3:03 PM
> > To: tcpm@ietf.org
> > Cc: Abdul Kabbani <akabbani@google.com>
> > Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04
> >
> >
> >
> > The wordings on cwnd adjustment in section 3.3 can be more clear:
> >
> >
> >
> > """
> >
> > Rather than always halving the congestion window as described in
> >
> >    [RFC3168], when the sender receives an indication of congestion
> >
> >    (ECE), the sender MUST update cwnd as follows:
> >
> >
> >
> >       cwnd = cwnd * (1 - DCTCP.Alpha / 2)
> >
> >
> >
> >    Thus, when no sent byte experienced congestion, DCTCP.Alpha equals
> >
> >    zero, and cwnd is left unchanged.  When all sent bytes experienced
> >
> >    congestion, DCTCP.Alpha equals one, and cwnd is reduced by half.
> >
> >    Lower levels of congestion will result in correspondingly smaller
> >
> >    reductions to cwnd.
> >
> > """
> >
> >
> >
> > AFAICT DCTCP paper only adjusts cwnd at most every round trip (which is
> also how Linux implements it). But the text implies the cwnd reduction can
> happen for every ECE mark?
> >
> >
> >
> >
> >
> > On Tue, Feb 14, 2017 at 8:31 AM, Scharf, Michael (Nokia - DE) <
> michael.scharf@nokia.com> wrote:
> >
> > Hi,
> >
> > TCPM chairs are not aware of outstanding issues in
> draft-ietf-tcpm-dctcp-04 "Datacenter TCP (DCTCP): TCP Congestion Control
> for Datacenters".
> >
> > Therefore, this e-mail starts a working group last call for
> draft-ietf-tcpm-dctcp-04.
> >
> > The WGLC runs until Friday March 3. Please send any comments to the TCPM
> mailing list by then. Statements supporting the publication of the document
> are also helpful.
> >
> > We assume that TCPM working group is aware of the IPR disclosure related
> to draft-bensley-tcpm-dctcp-00 (https://datatracker.ietf.org/ipr/2319/).
> A separate note will be sent out to confirm that all IPR has already been
> disclosed, in conformance with BCPs 78 and 79.
> >
> > The draft is available at
> > https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04
> >
> > Thanks
> >
> > Michael
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
> >
>

--001a1145f2fa097df3054aa4026f
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, Mar 13, 2017 at 9:38 AM, Praveen Balasubramanian <span dir=3D"l=
tr">&lt;<a href=3D"mailto:pravb@microsoft.com" target=3D"_blank">pravb@micr=
osoft.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"><span cl=
ass=3D"">&gt; Thanks for clarification. I notice the draft is also less cle=
ar about the window reduction on packet losses. For example, if sender expe=
riences ECE and then losses in the same window of &gt; data. Does DCTCP red=
uces the window twice (once on ECN, then again on Loss)?<br>
<br>
</span>DCTCP will reduce the window only once per window of data for the lo=
ss signals of ECN and duplicate ACKs. On a timeout cwnd will be uncondition=
ally reduced to 1 MSS even if there was a preceding cwnd reduction due to a=
 loss signal in a given window of data.<br>
<span class=3D""><br>
&gt; The draft implies cwnd would be reduced twice:<br>
<br>
&gt; A DCTCP sender MUST deal with loss episodes in the same way as<br>
&gt;=C2=A0 =C2=A0 conventional TCP.=C2=A0 In case of a timeout or fast retr=
ansmit or any<br>
&gt;=C2=A0 =C2=A0 change in delay (for delay based congestion control), the=
 cwnd and<br>
&gt;=C2=A0 =C2=A0 other state variables like ssthresh must be changed in th=
e same way<br>
&gt;=C2=A0 =C2=A0 that a conventional TCP would have changed them.<br>
<br>
</span>This wording comes from the fact that in Windows at least DCTCP is i=
mplemented as an extension to NewReno, in the sense that only the ECN handl=
er behaves differently. I will change this wording to explicitly mention th=
e above behavior of handling other loss events.<br></blockquote><div>Thanks=
 for the clarification. I assume you mean Reno not NewReno. NewReno is not =
a congestion control... okay I really need to start a thread to correct thi=
s.</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<span class=3D""><br>
&gt; AFAICT Linux&#39;s implementation for both DCTCP and RFC3168 would onl=
y reduce cwnd once on ECN + loss. Since this is a common scenario, it&#39;d=
 be nice to be explicit in the DCTCP draft.<br>
<br>
</span>Yes agreed. We will publish an update soon.<br>
<br>
Thanks<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
-----Original Message-----<br>
From: Yuchung Cheng [mailto:<a href=3D"mailto:ycheng@google.com">ycheng@goo=
gle.com</a>]<br>
Sent: Sunday, March 5, 2017 8:34 AM<br>
To: Praveen Balasubramanian &lt;<a href=3D"mailto:pravb@microsoft.com">prav=
b@microsoft.com</a>&gt;<br>
Cc: <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a>; Abdul Kabbani &lt;<=
a href=3D"mailto:akabbani@google.com">akabbani@google.com</a>&gt;<br>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04<br>
<br>
On Fri, Mar 3, 2017 at 4:41 PM, Praveen Balasubramanian &lt;<a href=3D"mail=
to:pravb@microsoft.com">pravb@microsoft.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Yes DCTCP adjusts cwnd at most once every round trip.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; See step 4:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; If the acknowledgment number is less than or equal to<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 DCTCP.WindowEnd, stop processing.=C2=A0 Oth=
erwise, the end of the<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 observation window has been reached, so pro=
ceed to update the<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 congestion estimate as follows:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; So steps 5 through 8 are conditional and only executed once per round.=
 What is a bit confusing is that the cwnd calculation is not indicated as s=
tep 9. I will fix this in an update to the draft.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Other than this we also have the following clarifying sentence:<br>
&gt;<br>
&gt; Just as specified in [RFC3168], DCTCP does not react to congestion<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 indications more than once for every window of data.<br>
Thanks for clarification. I notice the draft is also less clear about the w=
indow reduction on packet losses. For example, if sender experiences ECE an=
d then losses in the same window of data. Does DCTCP reduces the window twi=
ce (once on ECN, then again on Loss)?<br>
<br>
The draft implies cwnd would be reduced twice:<br>
<br>
A DCTCP sender MUST deal with loss episodes in the same way as<br>
=C2=A0 =C2=A0conventional TCP.=C2=A0 In case of a timeout or fast retransmi=
t or any<br>
=C2=A0 =C2=A0change in delay (for delay based congestion control), the cwnd=
 and<br>
=C2=A0 =C2=A0other state variables like ssthresh must be changed in the sam=
e way<br>
=C2=A0 =C2=A0that a conventional TCP would have changed them.<br>
<br>
AFAICT Linux&#39;s implementation for both DCTCP and RFC3168 would only red=
uce cwnd once on ECN + loss. Since this is a common scenario, it&#39;d be n=
ice to be explicit in the DCTCP draft.<br>
<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; From: tcpm [mailto:<a href=3D"mailto:tcpm-bounces@ietf.org">tcpm-bounc=
es@ietf.org</a>] On Behalf Of Yuchung Cheng<br>
&gt; Sent: Tuesday, February 28, 2017 3:03 PM<br>
&gt; To: <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; Cc: Abdul Kabbani &lt;<a href=3D"mailto:akabbani@google.com">akabbani@=
google.com</a>&gt;<br>
&gt; Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-dctcp-04<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The wordings on cwnd adjustment in section 3.3 can be more clear:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; &quot;&quot;&quot;<br>
&gt;<br>
&gt; Rather than always halving the congestion window as described in<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 [RFC3168], when the sender receives an indication of cong=
estion<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 (ECE), the sender MUST update cwnd as follows:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0cwnd =3D cwnd * (1 - DCTCP.Alpha / 2)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 Thus, when no sent byte experienced congestion, DCTCP.Alp=
ha equals<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 zero, and cwnd is left unchanged.=C2=A0 When all sent byt=
es experienced<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 congestion, DCTCP.Alpha equals one, and cwnd is reduced b=
y half.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 Lower levels of congestion will result in correspondingly=
 smaller<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 reductions to cwnd.<br>
&gt;<br>
&gt; &quot;&quot;&quot;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; AFAICT DCTCP paper only adjusts cwnd at most every round trip (which i=
s also how Linux implements it). But the text implies the cwnd reduction ca=
n happen for every ECE mark?<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Tue, Feb 14, 2017 at 8:31 AM, Scharf, Michael (Nokia - DE) &lt;<a h=
ref=3D"mailto:michael.scharf@nokia.com">michael.scharf@nokia.com</a>&gt; wr=
ote:<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; TCPM chairs are not aware of outstanding issues in draft-ietf-tcpm-dct=
cp-04 &quot;Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters&=
quot;.<br>
&gt;<br>
&gt; Therefore, this e-mail starts a working group last call for draft-ietf=
-tcpm-dctcp-04.<br>
&gt;<br>
&gt; The WGLC runs until Friday March 3. Please send any comments to the TC=
PM mailing list by then. Statements supporting the publication of the docum=
ent are also helpful.<br>
&gt;<br>
&gt; We assume that TCPM working group is aware of the IPR disclosure relat=
ed to draft-bensley-tcpm-dctcp-00 (<a href=3D"https://datatracker.ietf.org/=
ipr/2319/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.or=
g/<wbr>ipr/2319/</a>). A separate note will be sent out to confirm that all=
 IPR has already been disclosed, in conformance with BCPs 78 and 79.<br>
&gt;<br>
&gt; The draft is available at<br>
&gt; <a href=3D"https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-04" rel=
=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/<wbr>draft-ie=
tf-tcpm-dctcp-04</a><br>
&gt;<br>
&gt; Thanks<br>
&gt;<br>
&gt; Michael<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>
&gt;<br>
&gt;<br>
</div></div></blockquote></div><br></div></div>

--001a1145f2fa097df3054aa4026f--


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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 13 Mar 2017 15:24:48 -0700
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Subject: [tcpm] Clarification on NewReno vs Reno
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Over the past years, I have seem many people in both academia and
industry citing NewReno as a congestion control which really confuses
me. I would like to attempt to ask a clarification from list.

AFAIK, NewReno is NOT a congestion control. It is an improvement to
loss recovery in the absence of SACK support, per the abstract in
RFC6582

"""    ... in the absence of SACK. This document describes a specific
   algorithm for responding to partial acknowledgments, referred to as
   "NewReno".  This response to partial acknowledgments was first
   proposed by Janey Hoe.  This document obsoletes RFC 3782.

"""

In a nutshell, NewReno simply retransmits the next unacked packet when
SND.UNA advances. Using NewReno impacts the congestion control which
can be either Reno, Cubic, etc. But it is not a congestion control by
itself. In Linux NewReno is used only if SACK is not supported which
is rare.

Reno in contrast is the standard AIMD defined by RFC5681.


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Subject: Re: [tcpm] [iccrg] Clarification on NewReno vs Reno
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--001a114708b279bfd9054aa4dcf8
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Hi Yuchung,

This seems to be a rather philosophical question to me. So, I am not very
sure if we can gain something from this..

In my personal view, congestion control in TCP is "infer network status
based on feedback info, adjust amount of packet transmissions  based on it"
I think what newreno does fit this. It uses partial acks as feedback to do
some controls.
Hence, it looks congestion control to me. But, it's just my personal view
and can be different from yours.

Thanks,
--
Yoshi


On Mon, Mar 13, 2017 at 3:24 PM, Yuchung Cheng <ycheng@google.com> wrote:

> Over the past years, I have seem many people in both academia and
> industry citing NewReno as a congestion control which really confuses
> me. I would like to attempt to ask a clarification from list.
>
> AFAIK, NewReno is NOT a congestion control. It is an improvement to
> loss recovery in the absence of SACK support, per the abstract in
> RFC6582
>
> """    ... in the absence of SACK. This document describes a specific
>    algorithm for responding to partial acknowledgments, referred to as
>    "NewReno".  This response to partial acknowledgments was first
>    proposed by Janey Hoe.  This document obsoletes RFC 3782.
>
> """
>
> In a nutshell, NewReno simply retransmits the next unacked packet when
> SND.UNA advances. Using NewReno impacts the congestion control which
> can be either Reno, Cubic, etc. But it is not a congestion control by
> itself. In Linux NewReno is used only if SACK is not supported which
> is rare.
>
> Reno in contrast is the standard AIMD defined by RFC5681.
>
> _______________________________________________
> iccrg mailing list
> iccrg@irtf.org
> https://www.irtf.org/mailman/listinfo/iccrg
>

--001a114708b279bfd9054aa4dcf8
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<div dir=3D"ltr">Hi Yuchung,<div><br></div><div>This seems to be a rather p=
hilosophical question to me. So, I am not very sure if we can gain somethin=
g from this..</div><div><br></div><div>In my personal view, congestion cont=
rol in TCP is &quot;infer network status based on feedback info, adjust amo=
unt of packet transmissions =C2=A0based on it&quot;=C2=A0</div><div>I think=
 what newreno does fit this. It uses partial acks as feedback to do some co=
ntrols.</div><div>Hence, it looks congestion control to me. But, it&#39;s j=
ust my personal view and can be different from yours.=C2=A0</div><div><br><=
/div><div>Thanks,</div><div>--<br></div><div>Yoshi</div><div><br></div><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Mar 13, 2017 =
at 3:24 PM, Yuchung Cheng <span dir=3D"ltr">&lt;<a href=3D"mailto:ycheng@go=
ogle.com" target=3D"_blank">ycheng@google.com</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">Over the past years, I have seem many people in =
both academia and<br>
industry citing NewReno as a congestion control which really confuses<br>
me. I would like to attempt to ask a clarification from list.<br>
<br>
AFAIK, NewReno is NOT a congestion control. It is an improvement to<br>
loss recovery in the absence of SACK support, per the abstract in<br>
RFC6582<br>
<br>
&quot;&quot;&quot;=C2=A0 =C2=A0 ... in the absence of SACK. This document d=
escribes a specific<br>
=C2=A0 =C2=A0algorithm for responding to partial acknowledgments, referred =
to as<br>
=C2=A0 =C2=A0&quot;NewReno&quot;.=C2=A0 This response to partial acknowledg=
ments was first<br>
=C2=A0 =C2=A0proposed by Janey Hoe.=C2=A0 This document obsoletes RFC 3782.=
<br>
<br>
&quot;&quot;&quot;<br>
<br>
In a nutshell, NewReno simply retransmits the next unacked packet when<br>
SND.UNA advances. Using NewReno impacts the congestion control which<br>
can be either Reno, Cubic, etc. But it is not a congestion control by<br>
itself. In Linux NewReno is used only if SACK is not supported which<br>
is rare.<br>
<br>
Reno in contrast is the standard AIMD defined by RFC5681.<br>
<br>
______________________________<wbr>_________________<br>
iccrg mailing list<br>
<a href=3D"mailto:iccrg@irtf.org" target=3D"_blank">iccrg@irtf.org</a><br>
<a href=3D"https://www.irtf.org/mailman/listinfo/iccrg" rel=3D"noreferrer" =
target=3D"_blank">https://www.irtf.org/mailman/l<wbr>istinfo/iccrg</a><br>
</blockquote></div><br></div></div>

--001a114708b279bfd9054aa4dcf8--


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Subject: Re: [tcpm] [iccrg] Clarification on NewReno vs Reno
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Hi Yuchung,

I'm completely with you - I noticed the same mis-use and found it =
annoying. However, I have meanwhile given in and also use "NewReno" in =
papers. There are two reasons:

1) I once had a reviewer point out that we worked with the old Reno =
instead of NewReno, because I used the name "Reno" in a paper. I know =
that's a horrible reason, but pragmatically I don't want to risk paper =
failure because of a reviewer who makes that mistake.
Of course I wouldn't write truly wrong things into a paper on that =
basis, but saying "NewReno" instead of "Reno" strikes me as harmless... =
misleading, indeed, and somewhat annoying, but what can we do when =
people don't get it. Blame the name! It should have been "TCP with =
partial ACK support", or something like that.

2) When I write something like "We did tests on FreeBSD, choosing =
newreno congestion control", then that's actually correct because that's =
the name given to the default Reno-type CC. in FreeBSD's pluggable =
congestion control. The name isn't really wrong either, it really does =
implement NewReno, but it's a bit awkward as a naming choice (could have =
just as well been "TCP Reno with SACK support and Limited Transmit and =
ABC and the following spurious timeout detection / recovery algorithms =
and ... and ..." ). Oh well.

My suggestion is to accept that Reno is now called NewReno, because Reno =
is old and NewReno is new.

Cheers,
Michael



> On 13 Mar 2017, at 23:24, Yuchung Cheng <ycheng@google.com> wrote:
>=20
> Over the past years, I have seem many people in both academia and
> industry citing NewReno as a congestion control which really confuses
> me. I would like to attempt to ask a clarification from list.
>=20
> AFAIK, NewReno is NOT a congestion control. It is an improvement to
> loss recovery in the absence of SACK support, per the abstract in
> RFC6582
>=20
> """    ... in the absence of SACK. This document describes a specific
>   algorithm for responding to partial acknowledgments, referred to as
>   "NewReno".  This response to partial acknowledgments was first
>   proposed by Janey Hoe.  This document obsoletes RFC 3782.
>=20
> """
>=20
> In a nutshell, NewReno simply retransmits the next unacked packet when
> SND.UNA advances. Using NewReno impacts the congestion control which
> can be either Reno, Cubic, etc. But it is not a congestion control by
> itself. In Linux NewReno is used only if SACK is not supported which
> is rare.
>=20
> Reno in contrast is the standard AIMD defined by RFC5681.
>=20
> _______________________________________________
> iccrg mailing list
> iccrg@irtf.org
> https://www.irtf.org/mailman/listinfo/iccrg


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From: Praveen Balasubramanian <pravb@microsoft.com>
To: Michael Welzl <michawe@ifi.uio.no>, Yuchung Cheng <ycheng@google.com>
CC: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "iccrg@irtf.org" <iccrg@irtf.org>
Thread-Topic: [tcpm] [iccrg] Clarification on NewReno vs Reno
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Date: Tue, 14 Mar 2017 15:57:51 +0000
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Subject: Re: [tcpm] [iccrg] Clarification on NewReno vs Reno
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>> When I write something like "We did tests on FreeBSD, choosing newreno c=
ongestion control", then that's actually correct because that's the name gi=
ven to the default Reno-type CC
This is true of Windows as well - code as well as the config surface.

-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Michael Welzl
Sent: Tuesday, March 14, 2017 1:04 AM
To: Yuchung Cheng <ycheng@google.com>
Cc: tcpm@ietf.org Extensions <tcpm@ietf.org>; iccrg@irtf.org
Subject: Re: [tcpm] [iccrg] Clarification on NewReno vs Reno

Hi Yuchung,

I'm completely with you - I noticed the same mis-use and found it annoying.=
 However, I have meanwhile given in and also use "NewReno" in papers. There=
 are two reasons:

1) I once had a reviewer point out that we worked with the old Reno instead=
 of NewReno, because I used the name "Reno" in a paper. I know that's a hor=
rible reason, but pragmatically I don't want to risk paper failure because =
of a reviewer who makes that mistake.
Of course I wouldn't write truly wrong things into a paper on that basis, b=
ut saying "NewReno" instead of "Reno" strikes me as harmless... misleading,=
 indeed, and somewhat annoying, but what can we do when people don't get it=
. Blame the name! It should have been "TCP with partial ACK support", or so=
mething like that.

2) When I write something like "We did tests on FreeBSD, choosing newreno c=
ongestion control", then that's actually correct because that's the name gi=
ven to the default Reno-type CC. in FreeBSD's pluggable congestion control.=
 The name isn't really wrong either, it really does implement NewReno, but =
it's a bit awkward as a naming choice (could have just as well been "TCP Re=
no with SACK support and Limited Transmit and ABC and the following spuriou=
s timeout detection / recovery algorithms and ... and ..." ). Oh well.

My suggestion is to accept that Reno is now called NewReno, because Reno is=
 old and NewReno is new.

Cheers,
Michael



> On 13 Mar 2017, at 23:24, Yuchung Cheng <ycheng@google.com> wrote:
>=20
> Over the past years, I have seem many people in both academia and=20
> industry citing NewReno as a congestion control which really confuses=20
> me. I would like to attempt to ask a clarification from list.
>=20
> AFAIK, NewReno is NOT a congestion control. It is an improvement to=20
> loss recovery in the absence of SACK support, per the abstract in
> RFC6582
>=20
> """    ... in the absence of SACK. This document describes a specific
>   algorithm for responding to partial acknowledgments, referred to as
>   "NewReno".  This response to partial acknowledgments was first
>   proposed by Janey Hoe.  This document obsoletes RFC 3782.
>=20
> """
>=20
> In a nutshell, NewReno simply retransmits the next unacked packet when=20
> SND.UNA advances. Using NewReno impacts the congestion control which=20
> can be either Reno, Cubic, etc. But it is not a congestion control by=20
> itself. In Linux NewReno is used only if SACK is not supported which=20
> is rare.
>=20
> Reno in contrast is the standard AIMD defined by RFC5681.
>=20
> _______________________________________________
> iccrg mailing list
> iccrg@irtf.org
> https://www.irtf.org/mailman/listinfo/iccrg

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


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From: Yuchung Cheng <ycheng@google.com>
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To: Praveen Balasubramanian <pravb@microsoft.com>
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Subject: Re: [tcpm] [iccrg] Clarification on NewReno vs Reno
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Thanks for the replies: now I know why people use NewReno it's b/c of
implementation misnomer.

On Tue, Mar 14, 2017 at 8:57 AM, Praveen Balasubramanian
<pravb@microsoft.com> wrote:
>>> When I write something like "We did tests on FreeBSD, choosing newreno =
congestion control", then that's actually correct because that's the name g=
iven to the default Reno-type CC
> This is true of Windows as well - code as well as the config surface.
>
> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Michael Welzl
> Sent: Tuesday, March 14, 2017 1:04 AM
> To: Yuchung Cheng <ycheng@google.com>
> Cc: tcpm@ietf.org Extensions <tcpm@ietf.org>; iccrg@irtf.org
> Subject: Re: [tcpm] [iccrg] Clarification on NewReno vs Reno
>
> Hi Yuchung,
>
> I'm completely with you - I noticed the same mis-use and found it annoyin=
g. However, I have meanwhile given in and also use "NewReno" in papers. The=
re are two reasons:
>
> 1) I once had a reviewer point out that we worked with the old Reno inste=
ad of NewReno, because I used the name "Reno" in a paper. I know that's a h=
orrible reason, but pragmatically I don't want to risk paper failure becaus=
e of a reviewer who makes that mistake.
> Of course I wouldn't write truly wrong things into a paper on that basis,=
 but saying "NewReno" instead of "Reno" strikes me as harmless... misleadin=
g, indeed, and somewhat annoying, but what can we do when people don't get =
it. Blame the name! It should have been "TCP with partial ACK support", or =
something like that.
>
> 2) When I write something like "We did tests on FreeBSD, choosing newreno=
 congestion control", then that's actually correct because that's the name =
given to the default Reno-type CC. in FreeBSD's pluggable congestion contro=
l. The name isn't really wrong either, it really does implement NewReno, bu=
t it's a bit awkward as a naming choice (could have just as well been "TCP =
Reno with SACK support and Limited Transmit and ABC and the following spuri=
ous timeout detection / recovery algorithms and ... and ..." ). Oh well.
>
> My suggestion is to accept that Reno is now called NewReno, because Reno =
is old and NewReno is new.
>
> Cheers,
> Michael
>
>
>
>> On 13 Mar 2017, at 23:24, Yuchung Cheng <ycheng@google.com> wrote:
>>
>> Over the past years, I have seem many people in both academia and
>> industry citing NewReno as a congestion control which really confuses
>> me. I would like to attempt to ask a clarification from list.
>>
>> AFAIK, NewReno is NOT a congestion control. It is an improvement to
>> loss recovery in the absence of SACK support, per the abstract in
>> RFC6582
>>
>> """    ... in the absence of SACK. This document describes a specific
>>   algorithm for responding to partial acknowledgments, referred to as
>>   "NewReno".  This response to partial acknowledgments was first
>>   proposed by Janey Hoe.  This document obsoletes RFC 3782.
>>
>> """
>>
>> In a nutshell, NewReno simply retransmits the next unacked packet when
>> SND.UNA advances. Using NewReno impacts the congestion control which
>> can be either Reno, Cubic, etc. But it is not a congestion control by
>> itself. In Linux NewReno is used only if SACK is not supported which
>> is rare.
>>
>> Reno in contrast is the standard AIMD defined by RFC5681.
>>
>> _______________________________________________
>> iccrg mailing list
>> iccrg@irtf.org
>> https://www.irtf.org/mailman/listinfo/iccrg
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: Carles Gomez Montenegro <carlesgo@entel.upc.edu>
CC: "lwip@ietf.org" <lwip@ietf.org>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "jon.crowcroft@cl.cam.ac.uk" <jon.crowcroft@cl.cam.ac.uk>
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Subject: Re: [tcpm] [Lwip] [Fwd: New Version Notification for draft-gomez-lwig-tcp-constrained-node-networks-01.txt]
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Hi Carles,

Thanks a lot for the update. The version -02 indeed reads better from my po=
int of view. Also making this document informational is IMHO appropriate.

I still have one comment on Section 4.7.  "TCP options":

   A TCP implementation needs to support options 0, 1 and 2 [RFC793].  A
   TCP implementation for a constrained device that uses a single-MSS
   TCP receive or transmit window size may not benefit from supporting
   the following TCP options: Window scale [RFC1323], TCP Timestamps
   [RFC1323], Selective Acknowledgements (SACK) and SACK-Permitted
   [RFC2018].  Other TCP options should not be used, in keeping with the
   principle of lightweight operation.

   Other TCP options should not be supported by a constrained device, in
   keeping with the principle of lightweight implementation and
   operation.

I think the last two sentences are still not perfect even if the RFC 2119 l=
anguage is now removed. This sort of statement can easily get outdated, sin=
ce there continues to be innovation on TCP extensions and they typically us=
e options. I think a more future-proof wording would be something of the fo=
rm

  "At the time of publication, lightweight TCP implementations on constrain=
ed devices do not have to use other TCP options."

As a side note, the inconsistency regarding the TFO option still exist in t=
his wording. And the last two sentences also have quite a bit of repetition=
. In addition, it is not clear to me whether the different wording ("use" v=
s. "support") in the last two sentences is intentional. For some TCP option=
s, it is up to the active opener to decide whether to send the option, and =
it could refrain from doing so by default even if the option is actually "s=
upported" (i.e., implemented in the stack). This may be a reasonable TCP st=
ack configuration in some deployment scenarios, e.g., if window scale is no=
t needed for sensor data but would help for firmware updates.

Thanks

Michael=


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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
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Hi all,

We have uploaded a first draft agenda to https://datatracker.ietf.org/meeti=
ng/98/agenda/tcpm/

It is a draft and not final, so feel free to bash ;-)

Michael


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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">We have uploaded a first draft agenda to <a href=3D"=
https://datatracker.ietf.org/meeting/98/agenda/tcpm/">
https://datatracker.ietf.org/meeting/98/agenda/tcpm/</a><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It is a draft and not final, so feel free to bash ;-=
)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Michael<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From: Jonathan Looney <jtl.ietf@gmail.com>
Date: Wed, 15 Mar 2017 09:17:08 -0400
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Subject: [tcpm] 64-bit sequence numbers
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Hi all,

At the Seoul meeting, we saw two proposals for making TCP perform better on
high-speed networks: increasing the maximum window size to 2**31 bits and
increasing the precision of timestamps to the microsecond or nanosecond
level. As I said in the meeting at Seoul, the combination of these two
proposals led me to believe that we need to begin planning for 64-bit
sequence numbers.

Given that it takes years to rollout new TCP features (especially something
as fundamental as changing sequence numbers), it seems like it is time to
begin thinking about how we will do this.

I submitted the below I-D with a proposal for 64-bit sequence numbers. I'm
sure the proposal isn't perfect. The draft itself highlights some
challenges. I asked the chairs to allocate a few minutes in Chicago to
discuss this topic. Even if we decide to go in a different direction than
this draft, I think it is important that we begin to think about 64-bit
sequence numbers.

Jonathan

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, Mar 13, 2017 at 7:32 PM
Subject: New Version Notification for draft-looney-tcpm-64-bit-seqnos-00.txt
To: Jonathan Looney <jtl.ietf@gmail.com>



A new version of I-D, draft-looney-tcpm-64-bit-seqnos-00.txt
has been successfully submitted by Jonathan Looney and posted to the
IETF repository.

Name:           draft-looney-tcpm-64-bit-seqnos
Revision:       00
Title:          64-bit Sequence Numbers for TCP
Document date:  2017-03-13
Group:          Individual Submission
Pages:          17
URL:            https://www.ietf.org/internet-drafts/draft-looney-tcpm-64-
bit-seqnos-00.txt
Status:         https://datatracker.ietf.org/doc/draft-looney-tcpm-64-bit-
seqnos/
Htmlized:       https://tools.ietf.org/html/draft-looney-tcpm-64-bit-
seqnos-00


Abstract:
   This draft updates RFC 793 to allow the optional use of 64-bit
   sequence numbers.  It also updates other standards to support the
   extended sequence number space.




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

The IETF Secretariat

--94eb2c092b64f5cda6054ac4c23c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi all,<div><br></div><div>At the Seoul meeting, we saw tw=
o proposals for making TCP perform better on high-speed networks: increasin=
g the maximum window size to 2**31 bits and increasing the precision of tim=
estamps to the microsecond or nanosecond level. As I said in the meeting at=
 Seoul, the combination of these two proposals led me to believe that we ne=
ed to begin planning for 64-bit sequence numbers.<br></div><div><br></div><=
div>Given that it takes years to rollout new TCP features (especially somet=
hing as fundamental as changing sequence numbers), it seems like it is time=
 to begin thinking about how we will do this.</div><div><br></div><div>I su=
bmitted the below I-D with a proposal for 64-bit sequence numbers. I&#39;m =
sure the proposal isn&#39;t perfect. The draft itself highlights some chall=
enges. I asked the chairs to allocate a few minutes in Chicago to discuss t=
his topic. Even if we decide to go in a different direction than this draft=
, I think it is important that we begin to think about 64-bit sequence numb=
ers.</div><div><br></div><div>Jonathan</div><div><br><div class=3D"gmail_qu=
ote">---------- Forwarded message ----------<br>From: <b class=3D"gmail_sen=
dername"></b> <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.=
org">internet-drafts@ietf.org</a>&gt;</span><br>Date: Mon, Mar 13, 2017 at =
7:32 PM<br>Subject: New Version Notification for draft-looney-tcpm-64-bit-s=
eqnos-00.txt<br>To: Jonathan Looney &lt;<a href=3D"mailto:jtl.ietf@gmail.co=
m">jtl.ietf@gmail.com</a>&gt;<br><br><br><br>
A new version of I-D, draft-looney-tcpm-64-bit-<wbr>seqnos-00.txt<br>
has been successfully submitted by Jonathan Looney and posted to the<br>
IETF repository.<br>
<br>
Name:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-looney-tcpm-64-bit-<wbr=
>seqnos<br>
Revision:=C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 64-bit Sequence Numbers for TCP<br=
>
Document date:=C2=A0 2017-03-13<br>
Group:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Pages:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 17<br>
URL:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"https://www.ietf.o=
rg/internet-drafts/draft-looney-tcpm-64-bit-seqnos-00.txt" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/internet-<wbr>drafts/draft-looney=
-tcpm-64-<wbr>bit-seqnos-00.txt</a><br>
Status:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://datatracker.iet=
f.org/doc/draft-looney-tcpm-64-bit-seqnos/" rel=3D"noreferrer" target=3D"_b=
lank">https://datatracker.ietf.org/<wbr>doc/draft-looney-tcpm-64-bit-<wbr>s=
eqnos/</a><br>
Htmlized:=C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://tools.ietf.org/html/=
draft-looney-tcpm-64-bit-seqnos-00" rel=3D"noreferrer" target=3D"_blank">ht=
tps://tools.ietf.org/html/<wbr>draft-looney-tcpm-64-bit-<wbr>seqnos-00</a><=
br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This draft updates RFC 793 to allow the optional use of 64-bit=
<br>
=C2=A0 =C2=A0sequence numbers.=C2=A0 It also updates other standards to sup=
port the<br>
=C2=A0 =C2=A0extended sequence number space.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div></div>

--94eb2c092b64f5cda6054ac4c23c--


From nobody Wed Mar 15 06:32:03 2017
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I haven't read your document (although I'm putting it on my list of
things to look at).  But, I am going to leave this 11 year old piece
of tripe here ...

    https://www.icir.org/mallman/share/draft-allman-tcpx2-hack-00.txt

(At one point I had a student implement this and it isn't all that
difficult and can in fact co-exist with current TCP and largely use
the same code path for both "versions" of TCP.  Alas, it wasn't
properly finished and written up / code released / etc.---which is
clearly my fault.)

allman


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From nobody Wed Mar 15 13:26:20 2017
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Hi, Jonathan (et al.),

A few issues:

1) why do you want to use all 64 bits for the TCP-AO MAC, but only the
low 32 bits for the KDF? The latter won't help with backward
compatibility, so why do this? FWIW, one mode of TCP-AO ignores options;
I don't see how you can say "ignore options" (experimental - RFC6978).
This doc probably needs to explain what happens when "ignore options" is
set (for NAT traversal), i.e., the sequence number extensions will be
ignored completely for TCP-AO in that case, AFAICT.

2) the doc should address how to handle RSTs, esp. if the state goes
away. It seems like a connection negotiated with 64-bit sequence numbers
can't ever be reset if the connection state is lost (all the RSTs will
be treated as out of window

3) there are more things than just TCP-AO that use ISNs as input; it
seems like you need a blanket statement on how they're handled. This
also then seems to kill automatic fall-back, AFAICT.

4) the doc needs to explain the impact of using this feature on each
segment on the TCP option space

5) the doc should probably address the impact of odd middleboxes, e.g.,
that split or splice segments or change sequence numbers

That's just scratching the surface; I fear this is a Gordian knot of
issues for which there is no simple sword.

Joe




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Hi, Mark (et al.),

I was thinking about this too, but as I recall the challenge was legacy
fallback.

I'm not convinced the proposed approach has safe fallback either, though.

Joe


On 3/15/2017 6:32 AM, Mark Allman wrote:
> I haven't read your document (although I'm putting it on my list of
> things to look at).  But, I am going to leave this 11 year old piece
> of tripe here ...
>
>     https://www.icir.org/mallman/share/draft-allman-tcpx2-hack-00.txt
>
> (At one point I had a student implement this and it isn't all that
> difficult and can in fact co-exist with current TCP and largely use
> the same code path for both "versions" of TCP.  Alas, it wasn't
> properly finished and written up / code released / etc.---which is
> clearly my fault.)
>
> allman
>
>
> --
> https://www.icir.org/mallman/
> @mallman_icsi
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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<html>
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    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <p>Hi, Mark (et al.),</p>
    <p>I was thinking about this too, but as I recall the challenge was
      legacy fallback.</p>
    <p>I'm not convinced the proposed approach has safe fallback either,
      though.</p>
    <p>Joe<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 3/15/2017 6:32 AM, Mark Allman
      wrote:<br>
    </div>
    <blockquote cite="mid:4961.1489584720@lawyers.icir.org" type="cite">
      <pre wrap="">
I haven't read your document (although I'm putting it on my list of
things to look at).  But, I am going to leave this 11 year old piece
of tripe here ...

    <a class="moz-txt-link-freetext" href="https://www.icir.org/mallman/share/draft-allman-tcpx2-hack-00.txt">https://www.icir.org/mallman/share/draft-allman-tcpx2-hack-00.txt</a>

(At one point I had a student implement this and it isn't all that
difficult and can in fact co-exist with current TCP and largely use
the same code path for both "versions" of TCP.  Alas, it wasn't
properly finished and written up / code released / etc.---which is
clearly my fault.)

allman


--
<a class="moz-txt-link-freetext" href="https://www.icir.org/mallman/">https://www.icir.org/mallman/</a>
@mallman_icsi



</pre>
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      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
tcpm mailing list
<a class="moz-txt-link-abbreviated" href="mailto:tcpm@ietf.org">tcpm@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/tcpm">https://www.ietf.org/mailman/listinfo/tcpm</a>
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Thread-Topic: WGLC comments for draft-ietf-tcpm-dctcp-04
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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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Inline prefixed by >>

-----Original Message-----
From: Gorry Fairhurst [mailto:gorry@erg.abdn.ac.uk]=20
Sent: Monday, March 13, 2017 2:52 AM
To: tcpm@ietf.org
Cc: draft-ietf-tcpm-dctcp@ietf.org
Subject: WGLC comments for draft-ietf-tcpm-dctcp-04

I think this document looks in good shape. I have two sets of WG comments r=
elating to the language used in the current draft:

(1) I think the language surrounding loss treatment needs to be still stren=
gthened.
>> This is fixed in the next update per the other comments in the mailing l=
ist.

(2) I would like to see the contradition with RFC3168 reworked to avoid thi=
s INFO docuemnt attempting to over-ride that PS.
>> Sure

Best wishes,

Gorry

---
In section 1:

" This protocol is not meant for uncontrolled deployment in the global Inte=
rnet. "

- To me this statement seems far too opn, and the draft has alreday found a=
 clearer recommendations that I would agree to, as cited in the
abstract:

"DCTCP as described in this draft
    is applicable to deployments in controlled environments like
    datacenters but it MUST NOT be deployed over the public Internet
    without additional measures"

- Is it possible to re-use this statement in place of the one quoted above =
at the end of the conclusion?

>> Fixed


---

I'd also really like to see a statement about loss-treatment in the introdu=
ction. (see my comemt on section 5, but calling-out from the start that los=
s triggers standard TCP congestion control measures is I think a very impor=
tant thing to make the reader aware about. Especially when the introduction=
 (quite correctly) points to the merits of ECN reacting faster than loss-ba=
sed methods.


>> Fair enough. Added a statement: " DCTCP is a modification to the process=
ing of ECN by a conventional TCP and requires that standard TCP congestion =
control be used for handling packet loss."


---
In section 3:

"A DCTCP sender MUST deal with loss episodes in the same way as
    conventional TCP."

- I agree. Could we replace "deal with" perhaps with something like "react =
to"



>> Yes, replaced. Much better choice of words thanks.

---

3.4.  Handling of SYN, SYN-ACK, RST Packets

   " The switching fabric can drop TCP packets that do not have the ECT
    set in the IP header.  If SYN and SYN-ACK packets for DCTCP
    connections do not have ECT set, they will be dropped with high
    probability.  For DCTCP connections, the sender SHOULD set ECT for
    SYN, SYN-ACK and RST packets."

- I'd take the position that the fabric can and will drop any packets under=
 overload, as per RFC7567. I'd prefer to explicitly state that to avoid a m=
isconception that ECT eliminates all drop (rather than nearly all drops).


>> Change this to : " If SYN and SYN-ACK packets for DCTCP connections do n=
ot have ECT set in the IP header, they will likely be dropped by the switch=
ing fabric under load. For DCTCP connections, the sender SHOULD set ECT for=
 SYN, SYN-ACK and RST packets."

---

Section 4:

   "It is RECOMMENDED
    that an implementation provide a configuration knob that will cause
    ECT to be set on such control packes"
- please correct typo /packes"


>> Fixed

- I'm not a fan of an informational document over-riding a PS. I do not thi=
nk this is appropriate.
- I suggest once way to help, could be to say that
   "future versions of this protocol provide a configuration knob that will=
 cause
    ECT to be set on such control packets [xx]."
and then  cite "draft-ietf-tsvwg-ecn-experimentation".


>> Adding the citation to the "draft-ietf-tsvwg-ecn-experimentation" draft =
in the next update. Given that the document clearly limits the scope of dep=
loyment, I don't think this is at odds with the PS. Referencing the draft w=
ill provide enough context.


- The present text is clearly at odds with the RFC series.

---

In section 5:

The following statement is correct, in as much as SCTP needs to go back los=
s-based CC, but I'd prefer personally to see this expressed as "reverts to =
loss-based congestion control" - to make this a little more clear, and I'd =
much rather it said when it encounters loss, rather than perhaps suggesting=
 this is only drop-tail:

"DCTCP will degrade to loss-based congestion control when transiting a cong=
ested drop-tail link."

- e.g. something like this:

"DCTCP will revert to loss-based congestion control when packet loss is exp=
erienced (e.g. when transiting a congested drop-tail link, or a link with a=
n AQM drop behaviour).


>> Yes the intention is better captured with your suggested text. Fixed.

---

In section 6:

This statement could be better written:

"If the estimation gain is small relative to the packet loss rate, the esti=
mate may not be too inaccurate."

- First, I think the loss rate noted, may be the loss rate for the return p=
ath (i.e., the opposite direction to the packets bing CE-marked?)
- Second, may not be too inacurrate sounds odd - do you think the accuracy =
is acceptable for normal use?

If this relates to ACK loss, the following statement isn't necessarily true=
. And especially wculd be far from true for an internet with asymmetric rou=
ting:

"o  If packet loss mostly occurs under heavy congestion, most drops
       will occur during an unbroken string of CE packets, and the
       estimate will be unaffected.
"

>> Yes these considerations are for the case where ACKs are lost reducing t=
he accuracy of Alpha. The further sentence says that the impact of such los=
s has not been measured so its hard for me to say whether it will be "accep=
table for normal use". Also, none of this applies to the internet and AFAIK=
 routing is symmetric in the datacenter. I propose that we leave this text =
as is.


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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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Hi,

I haven't read the draft since long ago (one of the first versions), but =
I just read this dialogue and I have only one comment, so I'm removing =
everything else:


> ---
>=20
> 3.4.  Handling of SYN, SYN-ACK, RST Packets
>=20
>   " The switching fabric can drop TCP packets that do not have the ECT
>    set in the IP header.  If SYN and SYN-ACK packets for DCTCP
>    connections do not have ECT set, they will be dropped with high
>    probability.  For DCTCP connections, the sender SHOULD set ECT for
>    SYN, SYN-ACK and RST packets."
>=20
> - I'd take the position that the fabric can and will drop any packets =
under overload, as per RFC7567. I'd prefer to explicitly state that to =
avoid a misconception that ECT eliminates all drop (rather than nearly =
all drops).
>=20
>=20
>>> Change this to : " If SYN and SYN-ACK packets for DCTCP connections =
do not have ECT set in the IP header, they will likely be dropped by the =
switching fabric under load. For DCTCP connections, the sender SHOULD =
set ECT for SYN, SYN-ACK and RST packets."

To me, "will likely be dropped by the switching fabric under load" =
doesn't sound quite right - as if they would most probably be dropped =
whenever there is any form of load (no matter how much). I would suggest =
to change this to "...they are more likely to be dropped by the =
switching fabric under load."

Cheers,
Michael


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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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On 16/03/2017, 08:26, Michael Welzl wrote:
> Hi,
>
> I haven't read the draft since long ago (one of the first versions), but I just read this dialogue and I have only one comment, so I'm removing everything else:
>
>
>> ---
>>
>> 3.4.  Handling of SYN, SYN-ACK, RST Packets
>>
>>    " The switching fabric can drop TCP packets that do not have the ECT
>>     set in the IP header.  If SYN and SYN-ACK packets for DCTCP
>>     connections do not have ECT set, they will be dropped with high
>>     probability.  For DCTCP connections, the sender SHOULD set ECT for
>>     SYN, SYN-ACK and RST packets."
>>
>> - I'd take the position that the fabric can and will drop any packets under overload, as per RFC7567. I'd prefer to explicitly state that to avoid a misconception that ECT eliminates all drop (rather than nearly all drops).
>>
>>
>>>> Change this to : " If SYN and SYN-ACK packets for DCTCP connections do not have ECT set in the IP header, they will likely be dropped by the switching fabric under load. For DCTCP connections, the sender SHOULD set ECT for SYN, SYN-ACK and RST packets."
> To me, "will likely be dropped by the switching fabric under load" doesn't sound quite right - as if they would most probably be dropped whenever there is any form of load (no matter how much). I would suggest to change this to "...they are more likely to be dropped by the switching fabric under load."
>
> Cheers,
> Michael
I agree with Michael, but also I don't see it is correct for transport 
protocolsto need a very specific network behaviour. The choice of which 
packets are dropped is a design choice of the AQM: If we were to see 
FQ-Codel or FQ-PIE (for exmaple) deployed, then a common behaviour would 
be to separately queue new flows (even if non-ECT) - Switch design can 
change - I therefore think we should not be assuming a specific switch 
behaviour, but also I do think the ID does not need to go there ....  
because I suspect the clause could perhaps be written differently...

- I'd suggest to change to:
"SYN and SYN-ACK packets for DCTCP connections set the ECT(0) codepoint 
in the IP header, this ensures they receive the same treatment as other 
DCTP packets when forwarded by a switching fabric under load. For DCTCP 
connections, the sender SHOULD set ECT for SYN, SYN-ACK and RST packets."

Gorry




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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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> On 16 Mar 2017, at 09:52, Gorry Fairhurst <gorry@erg.abdn.ac.uk> =
wrote:
>=20
> On 16/03/2017, 08:26, Michael Welzl wrote:
>> Hi,
>>=20
>> I haven't read the draft since long ago (one of the first versions), =
but I just read this dialogue and I have only one comment, so I'm =
removing everything else:
>>=20
>>=20
>>> ---
>>>=20
>>> 3.4.  Handling of SYN, SYN-ACK, RST Packets
>>>=20
>>>   " The switching fabric can drop TCP packets that do not have the =
ECT
>>>    set in the IP header.  If SYN and SYN-ACK packets for DCTCP
>>>    connections do not have ECT set, they will be dropped with high
>>>    probability.  For DCTCP connections, the sender SHOULD set ECT =
for
>>>    SYN, SYN-ACK and RST packets."
>>>=20
>>> - I'd take the position that the fabric can and will drop any =
packets under overload, as per RFC7567. I'd prefer to explicitly state =
that to avoid a misconception that ECT eliminates all drop (rather than =
nearly all drops).
>>>=20
>>>=20
>>>>> Change this to : " If SYN and SYN-ACK packets for DCTCP =
connections do not have ECT set in the IP header, they will likely be =
dropped by the switching fabric under load. For DCTCP connections, the =
sender SHOULD set ECT for SYN, SYN-ACK and RST packets."
>> To me, "will likely be dropped by the switching fabric under load" =
doesn't sound quite right - as if they would most probably be dropped =
whenever there is any form of load (no matter how much). I would suggest =
to change this to "...they are more likely to be dropped by the =
switching fabric under load."
>>=20
>> Cheers,
>> Michael
> I agree with Michael, but also I don't see it is correct for transport =
protocolsto need a very specific network behaviour. The choice of which =
packets are dropped is a design choice of the AQM: If we were to see =
FQ-Codel or FQ-PIE (for exmaple) deployed, then a common behaviour would =
be to separately queue new flows (even if non-ECT) - Switch design can =
change - I therefore think we should not be assuming a specific switch =
behaviour, but also I do think the ID does not need to go there ....  =
because I suspect the clause could perhaps be written differently...
>=20
> - I'd suggest to change to:
> "SYN and SYN-ACK packets for DCTCP connections set the ECT(0) =
codepoint in the IP header, this ensures they receive the same treatment =
as other DCTP packets when forwarded by a switching fabric under load. =
For DCTCP connections, the sender SHOULD set ECT for SYN, SYN-ACK and =
RST packets."

This is just an ACK to say that I agree.

Cheers,
Michael


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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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I wonder if the following could be easily also resolved, see suggestion 
below.

On 16/03/2017, 02:52, Praveen Balasubramanian wrote:
> In section 6:
>
> This statement could be better written:
>
> "If the estimation gain is small relative to the packet loss rate, the estimate may not be too inaccurate."
>
> - First, I think the loss rate noted, may be the loss rate for the return path (i.e., the opposite direction to the packets bing CE-marked?)
> - Second, may not be too inacurrate sounds odd - do you think the accuracy is acceptable for normal use?
>
> If this relates to ACK loss, the following statement isn't necessarily true. And especially wculd be far from true for an internet with asymmetric routing:
>
> "o  If packet loss mostly occurs under heavy congestion, most drops
>         will occur during an unbroken string of CE packets, and the
>         estimate will be unaffected.
> "
>
>>> >>  Yes these considerations are for the case where ACKs are lost reducing the accuracy of Alpha. The further sentence says that the impact of such loss has not been measured so its hard for me to say whether it will be "acceptable for normal use". Also, none of this applies to the internet and AFAIK routing is symmetric in the datacenter. I propose that we leave this text as is.
- Is it worth inserting "ACK" in the following, to disambiguate packet 
loss from ACK loss.

    "o  If ACK packet loss mostly occurs under heavy congestion, most drops
        will occur during an unbroken string of CE packets, and the
        estimate will be unaffected.

Gorry


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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/xuLrohvZepfWt08IFgqFOPUnDF8>
Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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It is pretty clear that all the considerations are for ACK packet loss:

" However, if one or more ACK packets are dropped, it is possible that a su=
bsequent ACK will cumulatively acknowledge
      a mix of CE and non-CE segments. This will, of course, result in a le=
ss accurate congestion estimate. There are some potential considerations:"

The fact that CE is being set also implies this is a response packet.=20

"If packet loss mostly occurs under heavy congestion, most drops will occur=
 during an unbroken string of CE packets,
          and the estimate will be unaffected."

I wasn't sure this needed further qualification but I don't see the harm ad=
ding ACK to qualify the packet loss, so fixed.

-----Original Message-----
From: Gorry Fairhurst [mailto:gorry@erg.abdn.ac.uk]=20
Sent: Thursday, March 16, 2017 2:01 AM
To: Praveen Balasubramanian <pravb@microsoft.com>
Cc: tcpm@ietf.org; draft-ietf-tcpm-dctcp@ietf.org
Subject: Re: WGLC comments for draft-ietf-tcpm-dctcp-04

I wonder if the following could be easily also resolved, see suggestion bel=
ow.

On 16/03/2017, 02:52, Praveen Balasubramanian wrote:
> In section 6:
>
> This statement could be better written:
>
> "If the estimation gain is small relative to the packet loss rate, the es=
timate may not be too inaccurate."
>
> - First, I think the loss rate noted, may be the loss rate for the=20
> return path (i.e., the opposite direction to the packets bing=20
> CE-marked?)
> - Second, may not be too inacurrate sounds odd - do you think the accurac=
y is acceptable for normal use?
>
> If this relates to ACK loss, the following statement isn't necessarily tr=
ue. And especially wculd be far from true for an internet with asymmetric r=
outing:
>
> "o  If packet loss mostly occurs under heavy congestion, most drops
>         will occur during an unbroken string of CE packets, and the
>         estimate will be unaffected.
> "
>
>>> >>  Yes these considerations are for the case where ACKs are lost reduc=
ing the accuracy of Alpha. The further sentence says that the impact of suc=
h loss has not been measured so its hard for me to say whether it will be "=
acceptable for normal use". Also, none of this applies to the internet and =
AFAIK routing is symmetric in the datacenter. I propose that we leave this =
text as is.
- Is it worth inserting "ACK" in the following, to disambiguate packet loss=
 from ACK loss.

    "o  If ACK packet loss mostly occurs under heavy congestion, most drops
        will occur during an unbroken string of CE packets, and the
        estimate will be unaffected.

Gorry


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From: Praveen Balasubramanian <pravb@microsoft.com>
To: Michael Welzl <michawe@ifi.uio.no>, "<gorry@erg.abdn.ac.uk> Fairhurst" <gorry@erg.abdn.ac.uk>
CC: "tcpm@ietf.org" <tcpm@ietf.org>, "draft-ietf-tcpm-dctcp@ietf.org" <draft-ietf-tcpm-dctcp@ietf.org>
Thread-Topic: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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Changed to : " If SYN , SYN-ACK and RST packets for DCTCP connections have =
ECT set in the IP header, they will receive the same treatment as other DCT=
CP packets when forwarded by a switching fabric under load. Lack of ECT in =
these packets may result in a higher drop rate depending on the switching f=
abric configuration. Hence for DCTCP connections, the sender SHOULD set ECT=
 for SYN, SYN-ACK and RST packets."

-----Original Message-----
From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
Sent: Thursday, March 16, 2017 1:58 AM
To: <gorry@erg.abdn.ac.uk> Fairhurst <gorry@erg.abdn.ac.uk>
Cc: Praveen Balasubramanian <pravb@microsoft.com>; tcpm@ietf.org; draft-iet=
f-tcpm-dctcp@ietf.org
Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04


> On 16 Mar 2017, at 09:52, Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:
>=20
> On 16/03/2017, 08:26, Michael Welzl wrote:
>> Hi,
>>=20
>> I haven't read the draft since long ago (one of the first versions), but=
 I just read this dialogue and I have only one comment, so I'm removing eve=
rything else:
>>=20
>>=20
>>> ---
>>>=20
>>> 3.4.  Handling of SYN, SYN-ACK, RST Packets
>>>=20
>>>   " The switching fabric can drop TCP packets that do not have the ECT
>>>    set in the IP header.  If SYN and SYN-ACK packets for DCTCP
>>>    connections do not have ECT set, they will be dropped with high
>>>    probability.  For DCTCP connections, the sender SHOULD set ECT for
>>>    SYN, SYN-ACK and RST packets."
>>>=20
>>> - I'd take the position that the fabric can and will drop any packets u=
nder overload, as per RFC7567. I'd prefer to explicitly state that to avoid=
 a misconception that ECT eliminates all drop (rather than nearly all drops=
).
>>>=20
>>>=20
>>>>> Change this to : " If SYN and SYN-ACK packets for DCTCP connections d=
o not have ECT set in the IP header, they will likely be dropped by the swi=
tching fabric under load. For DCTCP connections, the sender SHOULD set ECT =
for SYN, SYN-ACK and RST packets."
>> To me, "will likely be dropped by the switching fabric under load" doesn=
't sound quite right - as if they would most probably be dropped whenever t=
here is any form of load (no matter how much). I would suggest to change th=
is to "...they are more likely to be dropped by the switching fabric under =
load."
>>=20
>> Cheers,
>> Michael
> I agree with Michael, but also I don't see it is correct for transport pr=
otocolsto need a very specific network behaviour. The choice of which packe=
ts are dropped is a design choice of the AQM: If we were to see FQ-Codel or=
 FQ-PIE (for exmaple) deployed, then a common behaviour would be to separat=
ely queue new flows (even if non-ECT) - Switch design can change - I theref=
ore think we should not be assuming a specific switch behaviour, but also I=
 do think the ID does not need to go there ....  because I suspect the clau=
se could perhaps be written differently...
>=20
> - I'd suggest to change to:
> "SYN and SYN-ACK packets for DCTCP connections set the ECT(0) codepoint i=
n the IP header, this ensures they receive the same treatment as other DCTP=
 packets when forwarded by a switching fabric under load. For DCTCP connect=
ions, the sender SHOULD set ECT for SYN, SYN-ACK and RST packets."

This is just an ACK to say that I agree.

Cheers,
Michael


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From: Praveen Balasubramanian <pravb@microsoft.com>
To: Praveen Balasubramanian <pravb@microsoft.com>, Michael Welzl <michawe@ifi.uio.no>, "<gorry@erg.abdn.ac.uk> Fairhurst" <gorry@erg.abdn.ac.uk>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/d_0ehXLvUhs24rzgfHoJCuunE_c>
Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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We are way past the deadline for WGLC which was March 3rd :). Since the com=
ments can be never ending and I have limited time, henceforth I will only f=
ix feedback that is critical in nature - i.e. something blatantly wrong or =
very unclear language. Thanks for all the feedback so far, I think the docu=
ment is in great shape now. You can expect to see the 05 revision being pos=
ted soon.=20

-----Original Message-----
From: Praveen Balasubramanian [mailto:pravb@microsoft.com]=20
Sent: Thursday, March 16, 2017 11:37 AM
To: Michael Welzl <michawe@ifi.uio.no>; <gorry@erg.abdn.ac.uk> Fairhurst <g=
orry@erg.abdn.ac.uk>
Cc: tcpm@ietf.org; draft-ietf-tcpm-dctcp@ietf.org
Subject: RE: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04

Changed to : " If SYN , SYN-ACK and RST packets for DCTCP connections have =
ECT set in the IP header, they will receive the same treatment as other DCT=
CP packets when forwarded by a switching fabric under load. Lack of ECT in =
these packets may result in a higher drop rate depending on the switching f=
abric configuration. Hence for DCTCP connections, the sender SHOULD set ECT=
 for SYN, SYN-ACK and RST packets."

-----Original Message-----
From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
Sent: Thursday, March 16, 2017 1:58 AM
To: <gorry@erg.abdn.ac.uk> Fairhurst <gorry@erg.abdn.ac.uk>
Cc: Praveen Balasubramanian <pravb@microsoft.com>; tcpm@ietf.org; draft-iet=
f-tcpm-dctcp@ietf.org
Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04


> On 16 Mar 2017, at 09:52, Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:
>=20
> On 16/03/2017, 08:26, Michael Welzl wrote:
>> Hi,
>>=20
>> I haven't read the draft since long ago (one of the first versions), but=
 I just read this dialogue and I have only one comment, so I'm removing eve=
rything else:
>>=20
>>=20
>>> ---
>>>=20
>>> 3.4.  Handling of SYN, SYN-ACK, RST Packets
>>>=20
>>>   " The switching fabric can drop TCP packets that do not have the ECT
>>>    set in the IP header.  If SYN and SYN-ACK packets for DCTCP
>>>    connections do not have ECT set, they will be dropped with high
>>>    probability.  For DCTCP connections, the sender SHOULD set ECT for
>>>    SYN, SYN-ACK and RST packets."
>>>=20
>>> - I'd take the position that the fabric can and will drop any packets u=
nder overload, as per RFC7567. I'd prefer to explicitly state that to avoid=
 a misconception that ECT eliminates all drop (rather than nearly all drops=
).
>>>=20
>>>=20
>>>>> Change this to : " If SYN and SYN-ACK packets for DCTCP connections d=
o not have ECT set in the IP header, they will likely be dropped by the swi=
tching fabric under load. For DCTCP connections, the sender SHOULD set ECT =
for SYN, SYN-ACK and RST packets."
>> To me, "will likely be dropped by the switching fabric under load" doesn=
't sound quite right - as if they would most probably be dropped whenever t=
here is any form of load (no matter how much). I would suggest to change th=
is to "...they are more likely to be dropped by the switching fabric under =
load."
>>=20
>> Cheers,
>> Michael
> I agree with Michael, but also I don't see it is correct for transport pr=
otocolsto need a very specific network behaviour. The choice of which packe=
ts are dropped is a design choice of the AQM: If we were to see FQ-Codel or=
 FQ-PIE (for exmaple) deployed, then a common behaviour would be to separat=
ely queue new flows (even if non-ECT) - Switch design can change - I theref=
ore think we should not be assuming a specific switch behaviour, but also I=
 do think the ID does not need to go there ....  because I suspect the clau=
se could perhaps be written differently...
>=20
> - I'd suggest to change to:
> "SYN and SYN-ACK packets for DCTCP connections set the ECT(0) codepoint i=
n the IP header, this ensures they receive the same treatment as other DCTP=
 packets when forwarded by a switching fabric under load. For DCTCP connect=
ions, the sender SHOULD set ECT for SYN, SYN-ACK and RST packets."

This is just an ACK to say that I agree.

Cheers,
Michael


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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tcpm] WGLC comments for draft-ietf-tcpm-dctcp-04
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Thanks,

Gorry

On 16/03/2017 18:30, Praveen Balasubramanian wrote:
> It is pretty clear that all the considerations are for ACK packet loss:
>
> " However, if one or more ACK packets are dropped, it is possible that a subsequent ACK will cumulatively acknowledge
>       a mix of CE and non-CE segments. This will, of course, result in a less accurate congestion estimate. There are some potential considerations:"
>
> The fact that CE is being set also implies this is a response packet.
>
> "If packet loss mostly occurs under heavy congestion, most drops will occur during an unbroken string of CE packets,
>           and the estimate will be unaffected."
>
> I wasn't sure this needed further qualification but I don't see the harm adding ACK to qualify the packet loss, so fixed.
>
> -----Original Message-----
> From: Gorry Fairhurst [mailto:gorry@erg.abdn.ac.uk]
> Sent: Thursday, March 16, 2017 2:01 AM
> To: Praveen Balasubramanian <pravb@microsoft.com>
> Cc: tcpm@ietf.org; draft-ietf-tcpm-dctcp@ietf.org
> Subject: Re: WGLC comments for draft-ietf-tcpm-dctcp-04
>
> I wonder if the following could be easily also resolved, see suggestion below.
>
> On 16/03/2017, 02:52, Praveen Balasubramanian wrote:
>> In section 6:
>>
>> This statement could be better written:
>>
>> "If the estimation gain is small relative to the packet loss rate, the estimate may not be too inaccurate."
>>
>> - First, I think the loss rate noted, may be the loss rate for the
>> return path (i.e., the opposite direction to the packets bing
>> CE-marked?)
>> - Second, may not be too inacurrate sounds odd - do you think the accuracy is acceptable for normal use?
>>
>> If this relates to ACK loss, the following statement isn't necessarily true. And especially wculd be far from true for an internet with asymmetric routing:
>>
>> "o  If packet loss mostly occurs under heavy congestion, most drops
>>         will occur during an unbroken string of CE packets, and the
>>         estimate will be unaffected.
>> "
>>
>>>>>>  Yes these considerations are for the case where ACKs are lost reducing the accuracy of Alpha. The further sentence says that the impact of such loss has not been measured so its hard for me to say whether it will be "acceptable for normal use". Also, none of this applies to the internet and AFAIK routing is symmetric in the datacenter. I propose that we leave this text as is.
> - Is it worth inserting "ACK" in the following, to disambiguate packet loss from ACK loss.
>
>     "o  If ACK packet loss mostly occurs under heavy congestion, most drops
>         will occur during an unbroken string of CE packets, and the
>         estimate will be unaffected.
>
> Gorry
>


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From: Jonathan Looney <jtl.ietf@gmail.com>
Date: Mon, 20 Mar 2017 14:12:30 -0400
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Subject: Re: [tcpm] 64-bit sequence numbers
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--001a113e8788799bc4054b2d78bf
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Dear Joe,

Thanks for taking the time to provide feedback!

On Wed, Mar 15, 2017 at 4:25 PM, Joe Touch <touch@isi.edu> wrote:

> Hi, Jonathan (et al.),
>
> A few issues:
>
> 1) why do you want to use all 64 bits for the TCP-AO MAC, but only the
> low 32 bits for the KDF? The latter won't help with backward
> compatibility, so why do this? FWIW, one mode of TCP-AO ignores options;
> I don't see how you can say "ignore options" (experimental - RFC6978).
> This doc probably needs to explain what happens when "ignore options" is
> set (for NAT traversal), i.e., the sequence number extensions will be
> ignored completely for TCP-AO in that case, AFAICT.
>

Because the KDFs are expected to take 32-bit sequence numbers, it seemed to
make sense to only use 32-bit sequence numbers as input to these functions.
If there is a desire to take 64-bit sequence numbers as input to the KDFs,
we could certainly specify this; however, I believe this would require
modifying the specifications for KDFs, and I wanted to avoid going farther
than necessary in this draft.

The MAC only takes 64-bit sequence numbers once we are reasonably sure that
the session supports 64-bit sequence numbers. Because the MAC already used
a 64-bit sequence number, there were not the same concerns with MACs as the
KDF.

I don't see an automatic conflict between ignoring TCP options while
constructing the MAC on the one hand, and using 64-bit sequence numbers in
the MAC on the other hand. Can you explain why you think this is a conflict?



> 2) the doc should address how to handle RSTs, esp. if the state goes
> away. It seems like a connection negotiated with 64-bit sequence numbers
> can't ever be reset if the connection state is lost (all the RSTs will
> be treated as out of window
>

I'm not sure I understand this comment. If the host sending the RST
understands 64-bit sequence numbers, it can use full 64-bit sequence
numbers in the RST. If the host doesn't understand 64-bit sequence numbers,
it will use 32-bit sequence numbers and (after the three-way handshake)
that RST will be ignored. While that *may* result in ignoring "legitimate"
RSTs from middleboxes that don't understand 64-bit sequence numbers, I'm
not sure this is a fatal flaw. After all, in today's Internet, it seems
better to err on the side of ignoring legitimate RSTs rather than accepting
illegitimate RSTs.



> 3) there are more things than just TCP-AO that use ISNs as input; it
> seems like you need a blanket statement on how they're handled. This
> also then seems to kill automatic fall-back, AFAICT.
>

I'm not sure which features you are referring to. If you can provide
specifics, I can try to provide a more specific answer.

The general principal is to use 32-bit sequence numbers until you are sure
you can successfully negotiate 64-bit sequence numbers. So, if a feature
needs to use ISNs in the three-way handshake, it will need to use 32-bit
sequence numbers. If a feature uses ISNs after that point, it can use
64-bit sequence numbers.

More details may be needed. (e.g. Does a feature switch to using 64-bit
sequence numbers as soon as it detects support?) A general rule will need
to be conservative (if you need to use ISNs in the three-way handshake, you
need to always use 32-bit sequence numbers); however, specific rules can be
more tailored to the specific feature. If there are features that would
benefit from the use of 64-bit sequence numbers, we should be sure to
specify how they can use them.

If this is not explained thoroughly enough in the document, I would be
happy to modify it to explain the general rule in more detail. (Again, more
specific examples will help, since there are limits to how specific you can
be in a "general rule".)


4) the doc needs to explain the impact of using this feature on each
> segment on the TCP option space
>

I'm not sure what you mean by this. Can you elaborate?


5) the doc should probably address the impact of odd middleboxes, e.g.,
> that split or splice segments or change sequence numbers
>

Fair enough. IMHO, such middleboxes are already broken, since they can't
possibly know how to properly modify the TCP options. However, it is a good
point that the document should consider this.


That's just scratching the surface; I fear this is a Gordian knot of
> issues for which there is no simple sword.
>

It may be that there are challenges to using 64-bit sequence numbers.
However, I think it is important to find a way forward on this. Otherwise,
I fear we will "soon" (in TCP modification timelines) run into performance
limitations.

Thanks!

Jonathan

--001a113e8788799bc4054b2d78bf
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">Dear=
 Joe,</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">=
Thanks for taking the time to provide feedback!</div><div class=3D"gmail_qu=
ote"><br></div><div class=3D"gmail_quote">On Wed, Mar 15, 2017 at 4:25 PM, =
Joe Touch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" target=3D"=
_blank">touch@isi.edu</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,2=
04);padding-left:1ex">Hi, Jonathan (et al.),<br>
<br>
A few issues:<br>
<br>
1) why do you want to use all 64 bits for the TCP-AO MAC, but only the<br>
low 32 bits for the KDF? The latter won&#39;t help with backward<br>
compatibility, so why do this? FWIW, one mode of TCP-AO ignores options;<br=
>
I don&#39;t see how you can say &quot;ignore options&quot; (experimental - =
RFC6978).<br>
This doc probably needs to explain what happens when &quot;ignore options&q=
uot; is<br>
set (for NAT traversal), i.e., the sequence number extensions will be<br>
ignored completely for TCP-AO in that case, AFAICT.<br></blockquote><div><b=
r></div><div>Because the KDFs are expected to take 32-bit sequence numbers,=
 it seemed to make sense to only use 32-bit sequence numbers as input to th=
ese functions. If there is a desire to take 64-bit sequence numbers as inpu=
t to the KDFs, we could certainly specify this; however, I believe this wou=
ld require modifying the specifications for KDFs, and I wanted to avoid goi=
ng farther than necessary in this draft.</div><div><br></div><div>The MAC o=
nly takes 64-bit sequence numbers once we are reasonably sure that the sess=
ion supports 64-bit sequence numbers. Because the MAC already used a 64-bit=
 sequence number, there were not the same concerns with MACs as the KDF.</d=
iv><div><br></div><div>I don&#39;t see an automatic conflict between ignori=
ng TCP options while constructing the MAC on the one hand, and using 64-bit=
 sequence numbers in the MAC on the other hand. Can you explain why you thi=
nk this is a conflict?</div><div><br></div><div>=C2=A0</div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex">
2) the doc should address how to handle RSTs, esp. if the state goes<br>
away. It seems like a connection negotiated with 64-bit sequence numbers<br=
>
can&#39;t ever be reset if the connection state is lost (all the RSTs will<=
br>
be treated as out of window<br></blockquote><div><br></div><div>I&#39;m not=
 sure I understand this comment. If the host sending the RST understands 64=
-bit sequence numbers, it can use full 64-bit sequence numbers in the RST. =
If the host doesn&#39;t understand 64-bit sequence numbers, it will use 32-=
bit sequence numbers and (after the three-way handshake) that RST will be i=
gnored. While that *may* result in ignoring &quot;legitimate&quot; RSTs fro=
m middleboxes that don&#39;t understand 64-bit sequence numbers, I&#39;m no=
t sure this is a fatal flaw. After all, in today&#39;s Internet, it seems b=
etter to err on the side of ignoring legitimate RSTs rather than accepting =
illegitimate RSTs.</div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex">
3) there are more things than just TCP-AO that use ISNs as input; it<br>
seems like you need a blanket statement on how they&#39;re handled. This<br=
>
also then seems to kill automatic fall-back, AFAICT.<br></blockquote><div><=
br></div><div>I&#39;m not sure which features you are referring to. If you =
can provide specifics, I can try to provide a more specific answer.</div><d=
iv>=C2=A0</div><div>The general principal is to use 32-bit sequence numbers=
 until you are sure you can successfully negotiate 64-bit sequence numbers.=
 So, if a feature needs to use ISNs in the three-way handshake, it will nee=
d to use 32-bit sequence numbers. If a feature uses ISNs after that point, =
it can use 64-bit sequence numbers.</div><div><br></div><div>More details m=
ay be needed. (e.g. Does a feature switch to using 64-bit sequence numbers =
as soon as it detects support?) A general rule will need to be conservative=
 (if you need to use ISNs in the three-way handshake, you need to always us=
e 32-bit sequence numbers); however, specific rules can be more tailored to=
 the specific feature. If there are features that would benefit from the us=
e of 64-bit sequence numbers, we should be sure to specify how they can use=
 them.</div><div><br></div><div>If this is not explained thoroughly enough =
in the document, I would be happy to modify it to explain the general rule =
in more detail. (Again, more specific examples will help, since there are l=
imits to how specific you can be in a &quot;general rule&quot;.)</div><div>=
<br></div><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">
4) the doc needs to explain the impact of using this feature on each<br>
segment on the TCP option space<br></blockquote><div><br></div><div>I&#39;m=
 not sure what you mean by this. Can you elaborate?=C2=A0</div><div><br></d=
iv><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">
5) the doc should probably address the impact of odd middleboxes, e.g.,<br>
that split or splice segments or change sequence numbers<br></blockquote><d=
iv><br></div><div>Fair enough. IMHO, such middleboxes are already broken, s=
ince they can&#39;t possibly know how to properly modify the TCP options. H=
owever, it is a good point that the document should consider this.</div><di=
v>=C2=A0</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1=
ex">
That&#39;s just scratching the surface; I fear this is a Gordian knot of<br=
>
issues for which there is no simple sword.<br></blockquote><div><br></div><=
div>It may be that there are challenges to using 64-bit sequence numbers. H=
owever, I think it is important to find a way forward on this. Otherwise, I=
 fear we will &quot;soon&quot; (in TCP modification timelines) run into per=
formance limitations.</div><div><br></div><div>Thanks!</div><div><br></div>=
<div>Jonathan=C2=A0</div></div></div></div>

--001a113e8788799bc4054b2d78bf--


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] 64-bit sequence numbers
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See below...

Joe


On 3/20/2017 11:12 AM, Jonathan Looney wrote:
> Dear Joe,
>
> Thanks for taking the time to provide feedback!
>
> On Wed, Mar 15, 2017 at 4:25 PM, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
>
>     Hi, Jonathan (et al.),
>
>     A few issues:
>
>     1) why do you want to use all 64 bits for the TCP-AO MAC, but only =
the
>     low 32 bits for the KDF? The latter won't help with backward
>     compatibility, so why do this? FWIW, one mode of TCP-AO ignores
>     options;
>     I don't see how you can say "ignore options" (experimental - RFC697=
8).
>     This doc probably needs to explain what happens when "ignore
>     options" is
>     set (for NAT traversal), i.e., the sequence number extensions will =
be
>     ignored completely for TCP-AO in that case, AFAICT.
>
>
> Because the KDFs are expected to take 32-bit sequence numbers, it
> seemed to make sense to only use 32-bit sequence numbers as input to
> these functions. If there is a desire to take 64-bit sequence numbers
> as input to the KDFs, we could certainly specify this; however, I
> believe this would require modifying the specifications for KDFs, and
> I wanted to avoid going farther than necessary in this draft.
I don't see a rationale for treating the two differently... (see the
next note)

> The MAC only takes 64-bit sequence numbers once we are reasonably sure
> that the session supports 64-bit sequence numbers. Because the MAC
> already used a 64-bit sequence number, there were not the same
> concerns with MACs as the KDF.
The MAC uses an *internal* SNE; you're overriding that by using the
extension in the option. That's a change too.

>
> I don't see an automatic conflict between ignoring TCP options while
> constructing the MAC on the one hand, and using 64-bit sequence
> numbers in the MAC on the other hand. Can you explain why you think
> this is a conflict?

I don't think you should be say two conflicting things to the protocol -
"ignore options" means ignore them all, not "ignore all except this one".=


> =20
>
>     2) the doc should address how to handle RSTs, esp. if the state goe=
s
>     away. It seems like a connection negotiated with 64-bit sequence
>     numbers
>     can't ever be reset if the connection state is lost (all the RSTs w=
ill
>     be treated as out of window
>
>
> I'm not sure I understand this comment. If the host sending the RST
> understands 64-bit sequence numbers, it can use full 64-bit sequence
> numbers in the RST. If the host doesn't understand 64-bit sequence
> numbers, it will use 32-bit sequence numbers and (after the three-way
> handshake) that RST will be ignored. While that *may* result in
> ignoring "legitimate" RSTs from middleboxes that don't understand
> 64-bit sequence numbers, I'm not sure this is a fatal flaw.

It means you can't reset a connection. Remember that TCP state doesn't
go away by itself. So you'll accumulate dead connections that will
interfere with new ones - and the new one won't reset the old one away
either. This can be handled, but requires timeouts to cleanup dead
connections, which are not part of the TCP spec. This was discussed in
TCP-AO.

> After all, in today's Internet, it seems better to err on the side of
> ignoring legitimate RSTs rather than accepting illegitimate RSTs.

Agreed, but you need to discuss it and find a way to reset connections
when state is lost.

>  3) there are more things than just TCP-AO that use ISNs as input; it
>
>     seems like you need a blanket statement on how they're handled. Thi=
s
>     also then seems to kill automatic fall-back, AFAICT.
>
>
> I'm not sure which features you are referring to. If you can provide
> specifics, I can try to provide a more specific answer.
Search the RFCs for ISNs. Some crypto algs use ISNs as a source or
"randomness" (incorrectly, of course). There are also RFCs that talk
about generating ISNs. A scrub of these will give you a list of what to
address.

> =20
> The general principal is to use 32-bit sequence numbers until you are
> sure you can successfully negotiate 64-bit sequence numbers. So, if a
> feature needs to use ISNs in the three-way handshake, it will need to
> use 32-bit sequence numbers. If a feature uses ISNs after that point,
> it can use 64-bit sequence numbers.

If that's true, then TCP-AO MAC cannot use the 64-bit ISNs in MAC
calculations until after the handshake completes! Otherwise, you can't
fall-back.

>
> More details may be needed. (e.g. Does a feature switch to using
> 64-bit sequence numbers as soon as it detects support?) A general rule
> will need to be conservative (if you need to use ISNs in the three-way
> handshake, you need to always use 32-bit sequence numbers); however,
> specific rules can be more tailored to the specific feature. If there
> are features that would benefit from the use of 64-bit sequence
> numbers, we should be sure to specify how they can use them.

This is a dangerous pit. TCP should have one set of assumptions at the
start of a connection, and another after the TWHS completes. Doing
changes at any other time is perilous and ill-defined.

> If this is not explained thoroughly enough in the document, I would be
> happy to modify it to explain the general rule in more detail. (Again,
> more specific examples will help, since there are limits to how
> specific you can be in a "general rule".)
>
>
>     4) the doc needs to explain the impact of using this feature on eac=
h
>     segment on the TCP option space
>
>
> I'm not sure what you mean by this. Can you elaborate?

You're using bytes for this option, and option space is very limited.
See the discussion of option space in TCP-AO.

>
>     5) the doc should probably address the impact of odd middleboxes,
>     e.g.,
>     that split or splice segments or change sequence numbers
>
>
> Fair enough. IMHO, such middleboxes are already broken, since they
> can't possibly know how to properly modify the TCP options. However,
> it is a good point that the document should consider this.

Right - the issue is whether you give false positives and whether you
can detect these and fall back (or should).

> =20
>
>     That's just scratching the surface; I fear this is a Gordian knot o=
f
>     issues for which there is no simple sword.
>
>
> It may be that there are challenges to using 64-bit sequence numbers.
> However, I think it is important to find a way forward on this.
> Otherwise, I fear we will "soon" (in TCP modification timelines) run
> into performance limitations.

"Important" doesn't mean "possible". Don't get ahead of that issue, pleas=
e.

Joe

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<html>
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    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <p>See below...</p>
    <p>Joe<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 3/20/2017 11:12 AM, Jonathan Looney
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">Dear Joe,</div>
          <div class="gmail_quote"><br>
          </div>
          <div class="gmail_quote">Thanks for taking the time to provide
            feedback!</div>
          <div class="gmail_quote"><br>
          </div>
          <div class="gmail_quote">On Wed, Mar 15, 2017 at 4:25 PM, Joe
            Touch <span dir="ltr">&lt;<a moz-do-not-send="true"
                href="mailto:touch@isi.edu" target="_blank">touch@isi.edu</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">Hi, Jonathan (et al.),<br>
              <br>
              A few issues:<br>
              <br>
              1) why do you want to use all 64 bits for the TCP-AO MAC,
              but only the<br>
              low 32 bits for the KDF? The latter won't help with
              backward<br>
              compatibility, so why do this? FWIW, one mode of TCP-AO
              ignores options;<br>
              I don't see how you can say "ignore options" (experimental
              - RFC6978).<br>
              This doc probably needs to explain what happens when
              "ignore options" is<br>
              set (for NAT traversal), i.e., the sequence number
              extensions will be<br>
              ignored completely for TCP-AO in that case, AFAICT.<br>
            </blockquote>
            <div><br>
            </div>
            <div>Because the KDFs are expected to take 32-bit sequence
              numbers, it seemed to make sense to only use 32-bit
              sequence numbers as input to these functions. If there is
              a desire to take 64-bit sequence numbers as input to the
              KDFs, we could certainly specify this; however, I believe
              this would require modifying the specifications for KDFs,
              and I wanted to avoid going farther than necessary in this
              draft.</div>
          </div>
        </div>
      </div>
    </blockquote>
    I don't see a rationale for treating the two differently... (see the
    next note)<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>The MAC only takes 64-bit sequence numbers once we are
              reasonably sure that the session supports 64-bit sequence
              numbers. Because the MAC already used a 64-bit sequence
              number, there were not the same concerns with MACs as the
              KDF.</div>
          </div>
        </div>
      </div>
    </blockquote>
    The MAC uses an *internal* SNE; you're overriding that by using the
    extension in the option. That's a change too.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div><br>
            </div>
            <div>I don't see an automatic conflict between ignoring TCP
              options while constructing the MAC on the one hand, and
              using 64-bit sequence numbers in the MAC on the other
              hand. Can you explain why you think this is a conflict?</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I don't think you should be say two conflicting things to the
    protocol - "ignore options" means ignore them all, not "ignore all
    except this one". <br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>Â </div>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
              0.8ex;border-left:1px solid
              rgb(204,204,204);padding-left:1ex">
              2) the doc should address how to handle RSTs, esp. if the
              state goes<br>
              away. It seems like a connection negotiated with 64-bit
              sequence numbers<br>
              can't ever be reset if the connection state is lost (all
              the RSTs will<br>
              be treated as out of window<br>
            </blockquote>
            <div><br>
            </div>
            <div>I'm not sure I understand this comment. If the host
              sending the RST understands 64-bit sequence numbers, it
              can use full 64-bit sequence numbers in the RST. If the
              host doesn't understand 64-bit sequence numbers, it will
              use 32-bit sequence numbers and (after the three-way
              handshake) that RST will be ignored. While that *may*
              result in ignoring "legitimate" RSTs from middleboxes that
              don't understand 64-bit sequence numbers, I'm not sure
              this is a fatal flaw. </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    It means you can't reset a connection. Remember that TCP state
    doesn't go away by itself. So you'll accumulate dead connections
    that will interfere with new ones - and the new one won't reset the
    old one away either. This can be handled, but requires timeouts to
    cleanup dead connections, which are not part of the TCP spec. This
    was discussed in TCP-AO.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>After all, in today's Internet, it seems better to err
              on the side of ignoring legitimate RSTs rather than
              accepting illegitimate RSTs.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Agreed, but you need to discuss it and find a way to reset
    connections when state is lost.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>Â 3) there are more things than just TCP-AO that use
              ISNs as input; it<br>
            </div>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
              0.8ex;border-left:1px solid
              rgb(204,204,204);padding-left:1ex">
              seems like you need a blanket statement on how they're
              handled. This<br>
              also then seems to kill automatic fall-back, AFAICT.<br>
            </blockquote>
            <div><br>
            </div>
            <div>I'm not sure which features you are referring to. If
              you can provide specifics, I can try to provide a more
              specific answer.</div>
          </div>
        </div>
      </div>
    </blockquote>
    Search the RFCs for ISNs. Some crypto algs use ISNs as a source or
    "randomness" (incorrectly, of course). There are also RFCs that talk
    about generating ISNs. A scrub of these will give you a list of what
    to address.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>Â </div>
            <div>The general principal is to use 32-bit sequence numbers
              until you are sure you can successfully negotiate 64-bit
              sequence numbers. So, if a feature needs to use ISNs in
              the three-way handshake, it will need to use 32-bit
              sequence numbers. If a feature uses ISNs after that point,
              it can use 64-bit sequence numbers.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    If that's true, then TCP-AO MAC cannot use the 64-bit ISNs in MAC
    calculations until after the handshake completes! Otherwise, you
    can't fall-back.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div><br>
            </div>
            <div>More details may be needed. (e.g. Does a feature switch
              to using 64-bit sequence numbers as soon as it detects
              support?) A general rule will need to be conservative (if
              you need to use ISNs in the three-way handshake, you need
              to always use 32-bit sequence numbers); however, specific
              rules can be more tailored to the specific feature. If
              there are features that would benefit from the use of
              64-bit sequence numbers, we should be sure to specify how
              they can use them.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    This is a dangerous pit. TCP should have one set of assumptions at
    the start of a connection, and another after the TWHS completes.
    Doing changes at any other time is perilous and ill-defined.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>If this is not explained thoroughly enough in the
              document, I would be happy to modify it to explain the
              general rule in more detail. (Again, more specific
              examples will help, since there are limits to how specific
              you can be in a "general rule".)</div>
            <div><br>
            </div>
            <div><br>
            </div>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
              0.8ex;border-left:1px solid
              rgb(204,204,204);padding-left:1ex">
              4) the doc needs to explain the impact of using this
              feature on each<br>
              segment on the TCP option space<br>
            </blockquote>
            <div><br>
            </div>
            <div>I'm not sure what you mean by this. Can you elaborate?
              <br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    You're using bytes for this option, and option space is very
    limited. See the discussion of option space in TCP-AO.<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div><br>
            </div>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
              0.8ex;border-left:1px solid
              rgb(204,204,204);padding-left:1ex">
              5) the doc should probably address the impact of odd
              middleboxes, e.g.,<br>
              that split or splice segments or change sequence numbers<br>
            </blockquote>
            <div><br>
            </div>
            <div>Fair enough. IMHO, such middleboxes are already broken,
              since they can't possibly know how to properly modify the
              TCP options. However, it is a good point that the document
              should consider this.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Right - the issue is whether you give false positives and whether
    you can detect these and fall back (or should).<br>
    <br>
    <blockquote
cite="mid:CAJrujPm1WQpqU0ALTqCm9iYgkER_KeFS6=dH0E7M0KU8jV2A2w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>Â </div>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
              0.8ex;border-left:1px solid
              rgb(204,204,204);padding-left:1ex">
              That's just scratching the surface; I fear this is a
              Gordian knot of<br>
              issues for which there is no simple sword.<br>
            </blockquote>
            <div><br>
            </div>
            <div>It may be that there are challenges to using 64-bit
              sequence numbers. However, I think it is important to find
              a way forward on this. Otherwise, I fear we will "soon"
              (in TCP modification timelines) run into performance
              limitations.<br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    "Important" doesn't mean "possible". Don't get ahead of that issue,
    please.<br>
    <br>
    Joe<br>
  </body>
</html>

--------------72171B8D55E90B5DA483F794--


From nobody Tue Mar 21 05:16:21 2017
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Dear all,

I just read the document and have two comments:

1. In section 2 the usage of ">>" is specified but ">>" is never used. I =
guess
   the paragraph can be removed.

2. It is stated two times:
   Additionally, TCB interdependence can be applied to any protocol
   with congestion state, including SCTP [RFC4960] and DCCP [RFC434],
   as well as for individual subflows in Multipath TCP [RFC6824].
   I think it should be made clear that information can also be
   shared between protocols. I also don't see why you can only do
   this between protocols having a congestion control. You might
   even want to share PMTU, MMS_S, MMS_R, RTT, RTTvar with protocols
   not providing a congestion control.

Best regards
Michael, as an individual=


From nobody Tue Mar 21 10:30:47 2017
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From: Joe Touch <touch@isi.edu>
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Hi,  Michael,


On 3/21/2017 5:16 AM, Michael Tuexen wrote:
> Dear all,
>
> I just read the document and have two comments:
>
> 1. In section 2 the usage of ">>" is specified but ">>" is never used. I guess
>    the paragraph can be removed.
We'll wait on that for now. If we go BCP, then this might be useful.

> 2. It is stated two times:
>    Additionally, TCB interdependence can be applied to any protocol
>    with congestion state, including SCTP [RFC4960] and DCCP [RFC434],
>    as well as for individual subflows in Multipath TCP [RFC6824].
>    I think it should be made clear that information can also be
>    shared between protocols. I also don't see why you can only do
>    this between protocols having a congestion control. You might
>    even want to share PMTU, MMS_S, MMS_R, RTT, RTTvar with protocols
>    not providing a congestion control.
We'll check. This is complicated by policy routing, of course. Also, we
need to check which of these values are truly transport, or merely
derived from network values already shared.

Thanks for the suggestion!

Joe



From ietf@weinrank.net  Tue Mar 21 15:53:04 2017
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To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
From: Felix Weinrank <ietf@weinrank.net>
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Subject: [tcpm] draft-ietf-tcpm-rack-02 - TLP algorithm question
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This is a multi-part message in MIME format.
--------------B89A9BC1984E11EC964DBEDD
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Hey all,

I have a question regarding the TLP algorithm in draft-ietf-tcpm-rack-02
In 6.2.1.  Phase 1: Scheduling a loss probe:

>         If an SRTT estimate is available:
>             PTO = 2 * SRTT
>         Else:
>             PTO = initial RTO of 1 sec
>         If FlightSize = 1:
>             PTO = max(PTO, 1.5 * SRTT + WCDelAckT)
>         PTO = max(10ms, PTO)
>         PTO = max(RTO, PTO)
The last line "PTO = max(RTO, PTO)" makes no sense to me.
It has been changed from "min" to "max" in the 02 version.

The explaining paragraph below also hints for "min"
> If RTO is smaller than the computed value for PTO, then a probe is scheduled to be sent
> at the RTO time.

Greetings
Felix Weinrank

--------------B89A9BC1984E11EC964DBEDD
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=utf-8">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <p>Hey all,<br>
      <br>
      I have a question regarding the TLP algorithm in
      draft-ietf-tcpm-rack-02<br>
      In 6.2.1.Â  Phase 1: Scheduling a loss probe:<br>
      <br>
    </p>
    <blockquote type="cite">
      <pre class="newpage" style="font-size: 13.3333px; margin-top: 0px; margin-bottom: 0px; break-before: page; color: rgb(0, 0, 0); font-style: normal; font-variant-ligatures: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration-style: initial; text-decoration-color: initial;">       If an SRTT estimate is available:
           PTO = 2 * SRTT
       Else:
           PTO = initial RTO of 1 sec
       If FlightSize = 1:
           PTO = max(PTO, 1.5 * SRTT + WCDelAckT)
       PTO = max(10ms, PTO)
       PTO = max(RTO, PTO)</pre>
    </blockquote>
    The last line "PTO = max(RTO, PTO)" makes no sense to me.<br>
    It has been changed from "min" to "max" in the 02 version.<br>
    <br>
    The explaining paragraph below also hints for "min"<br>
    <blockquote type="cite">
      <pre class="newpage" style="font-size: 13.3333px; margin-top: 0px; margin-bottom: 0px; break-before: page; color: rgb(0, 0, 0); font-style: normal; font-variant-ligatures: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration-style: initial; text-decoration-color: initial;">If RTO is smaller than the computed value for PTO, then a probe is scheduled to be sent
at the RTO time.</pre>
    </blockquote>
    <br>
    Greetings<br>
    Felix Weinrank<br>
  </body>
</html>

--------------B89A9BC1984E11EC964DBEDD--


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From: Neal Cardwell <ncardwell@google.com>
Date: Tue, 21 Mar 2017 19:51:07 -0400
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To: Felix Weinrank <ietf@weinrank.net>
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Subject: Re: [tcpm] draft-ietf-tcpm-rack-02 - TLP algorithm question
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On Tue, Mar 21, 2017 at 6:53 PM, Felix Weinrank <ietf@weinrank.net> wrote:
>
> Hey all,
>
> I have a question regarding the TLP algorithm in draft-ietf-tcpm-rack-02
> In 6.2.1.  Phase 1: Scheduling a loss probe:
...
>        PTO = max(10ms, PTO)
>        PTO = max(RTO, PTO)
>
> The last line "PTO = max(RTO, PTO)" makes no sense to me.
> It has been changed from "min" to "max" in the 02 version.

Yes, I believe you are right that a bug crept in between
draft-ietf-tcpm-rack-01 and draft-ietf-tcpm-rack-02. I have reverted
to:

   PTO = min(RTO, PTO)

 in our internal copy, so this fix should be reflected in the next
version of the RACK Internet Draft.

Thanks for catching that!

neal


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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-cubic-04
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Hello,
The WGLC for this draft has already finished, but we haven't seen many
comments on this draft.
We are trying to interpret this is an indication that people are satisfied
with the current draft, but we just would like to be sure.
Please let us know in case you have very last minute comments.
We'll do similar final check at the venue. If nothing happens, we will try
to proceed the doc after reviewing the updated version.

Thanks,
--
tcpm co-chairs

On Wed, Feb 15, 2017 at 1:13 AM, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
wrote:

> Hello,
>
> In addition to the dctcp draft, TCP chairs conclude that
> draft-ietf-tcpm-cubic-04 is ready for WGLC as well and decided to initiate
> the call.
>
> The WGLC for this draft runs until * Friday March 3 *.
> Please send your feedback to the ML.
>
> BTW, the intended status of the draft is Informational.
> The draft is available from the following URL.
> https://tools.ietf.org/html/draft-ietf-tcpm-cubic-04
>
> Thanks,
> --
> Yoshi
>

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<div dir=3D"ltr">Hello,<div><div class=3D"gmail_extra">The WGLC for this dr=
aft has already finished, but we haven&#39;t seen many comments on this dra=
ft.</div><div class=3D"gmail_extra">We are trying to interpret this is an i=
ndication that people are satisfied with the current draft, but we just wou=
ld like to be sure.=C2=A0</div><div class=3D"gmail_extra">Please let us kno=
w in case you have very last minute comments.</div><div class=3D"gmail_extr=
a">We&#39;ll do similar final check at the venue. If nothing happens, we wi=
ll try to proceed the doc after reviewing the updated version.<br></div><di=
v class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Thanks,</div><=
div class=3D"gmail_extra">--</div><div class=3D"gmail_extra">tcpm co-chairs=
</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Feb=
 15, 2017 at 1:13 AM, Yoshifumi Nishida <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:nishida@sfc.wide.ad.jp" target=3D"_blank">nishida@sfc.wide.ad.jp</a>&g=
t;</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"ltr">Hello,<=
div><br></div><div>In addition to the dctcp draft, TCP chairs conclude that=
 draft-ietf-tcpm-cubic-04 is ready for WGLC as well and decided to initiate=
 the call.</div><div><br></div><div>The WGLC for this draft runs until * Fr=
iday March 3 *.=C2=A0<br></div><div>Please send your feedback to the ML.</d=
iv><div><br></div><div><div>BTW, the intended status of the draft is Inform=
ational.=C2=A0</div><div>The draft is available from the following URL.</di=
v></div><div><div><a href=3D"https://tools.ietf.org/html/draft-ietf-tcpm-cu=
bic-04" target=3D"_blank">https://tools.ietf.org/html/dr<wbr>aft-ietf-tcpm-=
cubic-04</a></div></div><div><br></div><div>Thanks,</div><div>--</div><div>=
Yoshi</div></div>
</blockquote></div><br></div></div></div>

--001a113d4a58f0647d054b6c7c8a--


From nobody Fri Mar 24 14:59:45 2017
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Subject: [tcpm] Comments for draft-ietf-tcpm-rto-consider-05.txt
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Hi Mark,

I read the ID with a special focus on SCTP and have three comments:

1. S.3 states:
   Protocols such as SCTP [RFC4960] and MP-TCP [RFC6182] that
   communicate in a unicast fashion with multiple specific
   endpoints can leverage the requirements in this document
   provided they track state and follow the requirements for each
   endpoint independently.

   SCTP requires this and I think MP-TCP also for its subflows.
   So it seems strange that you restrict the statement to
   SCTP implementations fulfilling a requirement which has to be true. 

2. S.3 states:
   I.e., if host A communicates with hosts B and C, A must use
   independent RTOs for traffic sent to B and C.
   
   SCTP association are between two hosts. Each host can have
   multiple IP-addresses. But you can't have an SCTP association
   between host A on one side and multiple hosts (B and C) on the
   other side. I think the same applies to MP-TCP. SO you might want
   to use a term different from host.

3. S.5 states:
   E.g., a protocol like SCTP [RFC4960] could leverage the
   requirements for data that is sent only "partially reliably".

   It might be useful to add a reference to RFC 3758.

Best regards
Michael (as an individual)


   

   



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> I read the ID with a special focus on SCTP and have three comments:

Many thanks!

> 1. S.3 states:
>    Protocols such as SCTP [RFC4960] and MP-TCP [RFC6182] that
>    communicate in a unicast fashion with multiple specific
>    endpoints can leverage the requirements in this document
>    provided they track state and follow the requirements for each
>    endpoint independently.
>=20
>    SCTP requires this and I think MP-TCP also for its subflows.
>    So it seems strange that you restrict the statement to
>    SCTP implementations fulfilling a requirement which has to be true.=20

The statement is meant to be general not really specific to SCTP or
MP-TCP.  That is, this is not normatively saying something that SCTP
already says, somehow.  The "such as" was meant to indicate these
are just two examples.  I can reword this to more explicitly
indicate these are exemplars that follow the guidance, but the
guidance is general.  Make sense?

> 2. S.3 states:
>    I.e., if host A communicates with hosts B and C, A must use
>    independent RTOs for traffic sent to B and C.
>=20=20=20=20
>    SCTP association are between two hosts. Each host can have
>    multiple IP-addresses. But you can't have an SCTP association
>    between host A on one side and multiple hosts (B and C) on the
>    other side. I think the same applies to MP-TCP. SO you might want
>    to use a term different from host.

Do you have a favorite word?  "Endpoint"?

> 3. S.5 states:
>    E.g., a protocol like SCTP [RFC4960] could leverage the
>    requirements for data that is sent only "partially reliably".
>=20
>    It might be useful to add a reference to RFC 3758.

Will do.

Many thanks!

allman




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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/91UceoliFoOn3tBS4tkqfG7fz1U>
Subject: [tcpm] On ACK rate issues and remedies in transport protocols
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Hi
I have asked for a slot at TSVWG for a presentation on issued with transpor=
t protocol ACKs. What I am in interested is to know what earlier work such =
as RFC5690 exist and also if there are possibilities to reduce the ACK rate=
 in light of recent additions such as RACK and packet pacing.

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

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

A mistake is to commit a misunderstanding
                     Bob Dylan
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D


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<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal">I have asked for a slot at TSVWG for a presentation =
on issued with transport protocol ACKs. What I am in interested is to know =
what earlier work such as RFC5690 exist and also if there are possibilities=
 to reduce the ACK rate in light of
 recent additions such as RACK and packet pacing.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">/Ingemar<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Ingemar Johansson&nbsp;=
 M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Master Researcher<o:p><=
/o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif"><o:p>&nbsp;</o:p></span=
></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Ericsson AB<o:p></o:p><=
/span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Wireless Access Network=
s<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Labratoriegr=E4nd 11<o:=
p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">971 28, Lule=E5, Sweden=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Phone &#43;46-1071 4304=
2<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">SMS/MMS &#43;46-73 078 =
3289<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><a href=3D"mailto:inge=
mar.s.johansson@ericsson.com"><span style=3D"font-size:10.0pt;font-family:&=
quot;Arial&quot;,sans-serif;color:blue">ingemar.s.johansson@ericsson.com</s=
pan></a><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,sans-=
serif"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><a href=3D"www.ericsso=
n.com"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,sans-s=
erif;color:blue">www.ericsson.com</span></a><span style=3D"font-size:10.0pt=
;font-family:&quot;Arial&quot;,sans-serif"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif;background:#CCCCDD"><o:p=
>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif;color:#333333;background=
:white">A mistake is to commit a misunderstanding</span><span style=3D"font=
-size:10.0pt;font-family:&quot;Arial&quot;,sans-serif;color:#333333"><br>
<span style=3D"background:white">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; Bob Dylan<br>
</span>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<span style=3D"background:white"><o:p><=
/o:p></span></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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From nobody Sat Mar 25 13:25:40 2017
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
In-Reply-To: <74855.1490410093@lawyers.icir.org>
Date: Sat, 25 Mar 2017 21:25:31 +0100
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/sb4SUBa1TcHSY3gYmoPzn9qriE8>
Subject: Re: [tcpm] Comments for draft-ietf-tcpm-rto-consider-05.txt
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> On 25 Mar 2017, at 03:48, Mark Allman <mallman@icir.org> wrote:
>=20
>=20
>> I read the ID with a special focus on SCTP and have three comments:
>=20
> Many thanks!
>=20
>> 1. S.3 states:
>>   Protocols such as SCTP [RFC4960] and MP-TCP [RFC6182] that
>>   communicate in a unicast fashion with multiple specific
>>   endpoints can leverage the requirements in this document
>>   provided they track state and follow the requirements for each
>>   endpoint independently.
>>=20
>>   SCTP requires this and I think MP-TCP also for its subflows.
>>   So it seems strange that you restrict the statement to
>>   SCTP implementations fulfilling a requirement which has to be true.=20=

>=20
> The statement is meant to be general not really specific to SCTP or
> MP-TCP.  That is, this is not normatively saying something that SCTP
> already says, somehow.  The "such as" was meant to indicate these
> are just two examples.  I can reword this to more explicitly
> indicate these are exemplars that follow the guidance, but the
> guidance is general.  Make sense?
Yepp.

Something like

  Protocols that communicate in a unicast fashion with multiple
  specific transport addresses (pairs of an IP address and a port =
number)
  can leverage the requirements in this document provided they track =
state
  and follow the requirements for each transport address independently.
  Therefore, SCTP [RFC4960] and MP-TCP [RFC6182] can leverage the =
requirements
  in this document.

>=20
>> 2. S.3 states:
>>   I.e., if host A communicates with hosts B and C, A must use
>>   independent RTOs for traffic sent to B and C.
>>=20
>>   SCTP association are between two hosts. Each host can have
>>   multiple IP-addresses. But you can't have an SCTP association
>>   between host A on one side and multiple hosts (B and C) on the
>>   other side. I think the same applies to MP-TCP. SO you might want
>>   to use a term different from host.
>=20
> Do you have a favorite word?  "Endpoint"?
Transport address? Something like

  I.e., if an endpoint with transport address A communicates with
  an endpoint having transport addresses B and C, it must use
  independent RTOs for traffic sent to B and C.

>=20
>> 3. S.5 states:
>>   E.g., a protocol like SCTP [RFC4960] could leverage the
>>   requirements for data that is sent only "partially reliably".
>>=20
>>   It might be useful to add a reference to RFC 3758.
>=20
> Will do.
Great.
>=20
> Many thanks!
You are welcome!

Best regards
Michael
>=20
> allman
>=20
>=20
>=20


From nobody Mon Mar 27 08:58:12 2017
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-dctcp-05.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           : Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters
        Authors         : Stephen Bensley
                          Dave Thaler
                          Praveen Balasubramanian
                          Lars Eggert
                          Glenn Judd
	Filename        : draft-ietf-tcpm-dctcp-05.txt
	Pages           : 15
	Date            : 2017-03-27

Abstract:
   This informational memo describes Datacenter TCP (DCTCP), an
   improvement to TCP congestion control for datacenter traffic.  DCTCP
   uses improved Explicit Congestion Notification (ECN) processing to
   estimate the fraction of bytes that encounter congestion, rather than
   simply detecting that some congestion has occurred.  DCTCP then
   scales the TCP congestion window based on this estimate.  This method
   achieves high burst tolerance, low latency, and high throughput with
   shallow-buffered switches.  This memo also discusses deployment
   issues related to the coexistence of DCTCP and conventional TCP, the
   lack of a negotiating mechanism between sender and receiver, and
   presents some possible mitigations.  DCTCP as described in this draft
   is applicable to deployments in controlled environments like
   datacenters but it MUST NOT be deployed over the public Internet
   without additional measures, as detailed in Section 5.


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

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-05
https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-dctcp-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-dctcp-05


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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From: "Scharf, Michael (Nokia - DE/Stuttgart)" <michael.scharf@nokia.com>
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Subject: [tcpm] Notetakers?
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Hi all,

Would somebody volunteer to take notes in TCPM tomorrow?

It seems that more and more groups rely collaborative note-taking on Etherp=
ad - but we only have seldomly used it in TCPM. Are thoughts on moving to E=
therpad?

Help would be greatly appreciated!

Thanks

Michael

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<p class=3D"MsoNormal"><span lang=3D"DE">Hi all,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"DE"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">Would somebody volunteer to take notes in TCPM tomor=
row?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It seems that more and more groups rely collaborativ=
e note-taking on Etherpad
<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8211;</span=
> but we only have seldomly used it in TCPM. Are thoughts on moving to Ethe=
rpad?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Help would be greatly appreciated!<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Michael<o:p></o:p></p>
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From: "Scharf, Michael (Nokia - DE/Stuttgart)" <michael.scharf@nokia.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: WGLC completion of DCTCP
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Date: Tue, 28 Mar 2017 20:22:44 +0000
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Subject: [tcpm] WGLC completion of DCTCP
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Folks,

Could the contributors who have commented during the WGLC please have a loo=
k at -05 and let us know if the comments are sufficiently addressed?

Thanks

Michael


-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of internet-drafts@ietf=
.org
Sent: Montag, 27. M=E4rz 2017 17:58
To: i-d-announce@ietf.org
Cc: tcpm@ietf.org
Subject: [tcpm] I-D Action: draft-ietf-tcpm-dctcp-05.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
This draft is a work item of the TCP Maintenance and Minor Extensions of th=
e IETF.

        Title           : Datacenter TCP (DCTCP): TCP Congestion Control fo=
r Datacenters
        Authors         : Stephen Bensley
                          Dave Thaler
                          Praveen Balasubramanian
                          Lars Eggert
                          Glenn Judd
	Filename        : draft-ietf-tcpm-dctcp-05.txt
	Pages           : 15
	Date            : 2017-03-27

Abstract:
   This informational memo describes Datacenter TCP (DCTCP), an
   improvement to TCP congestion control for datacenter traffic.  DCTCP
   uses improved Explicit Congestion Notification (ECN) processing to
   estimate the fraction of bytes that encounter congestion, rather than
   simply detecting that some congestion has occurred.  DCTCP then
   scales the TCP congestion window based on this estimate.  This method
   achieves high burst tolerance, low latency, and high throughput with
   shallow-buffered switches.  This memo also discusses deployment
   issues related to the coexistence of DCTCP and conventional TCP, the
   lack of a negotiating mechanism between sender and receiver, and
   presents some possible mitigations.  DCTCP as described in this draft
   is applicable to deployments in controlled environments like
   datacenters but it MUST NOT be deployed over the public Internet
   without additional measures, as detailed in Section 5.


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

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-05
https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-dctcp-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-dctcp-05


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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-05.txt
	Pages           : 95
	Date            : 2017-03-28

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 are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-05
https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rfc793bis-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-05


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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From: Wesley Eddy <wes@mti-systems.com>
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Date: Tue, 28 Mar 2017 18:01:53 -0400
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-05.txt
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There are some changes in this update that the WG should review.

A summary of what to look for is below, but I also recommend scanning 
the diff:

https://tools.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-05.txt


1. added CWR and ECE bits from ECN to the header format diagram, and 
reference to 3168 for definitions

2. instead of RFC1122 requirement MUST implement slowstart + congestion 
avoidance, added "MUST implement RFC 5681" ---- please review this!

3. added "SHOULD implement ECN" --- please review this!  This is outside 
what 1122 or other documents say, but I think it is the intent of 
publishing 3168 on standards track and having it update 793.

4. updated 1122 text about source quench to reflect that 6633 deprecated it

5. modified text on ICMP messages to also mention ICMPv6 (not covered by 
1122) --  please check this!

6. split 1122 discussion of ICMP handling into soft and hard errors per 
5461, and added notes to reference other discussion in 5461 about 
widespread implementation behavior





On 3/28/2017 5:50 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-05.txt
> 	Pages           : 95
> 	Date            : 2017-03-28
>
> 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 are also htmlized versions available at:
> https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-05
> https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rfc793bis-05
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rfc793bis-05
>
>
> 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 Tue Mar 28 15:06:48 2017
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-05.txt
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Some observations:

3.1 currently defined options
    this either needs to be the complete current list or indicate that
the list is elsewhere (the latter seems better to me)
    if list elsewhere, the discussion should be clear that this doc
defines only the following options; others are defined elsewhere
    is there any notion of the minimum required options? if so, is this
that set? (it might be useful to say so)

throughout
    the discussion of PMTU and MSS might benefit from an update as per
draft-iett-intarea-tunnels
    key points:
        PMTU discovery assumes TCP set IP to prevent both on-path (for
IPv4) and source fragmentation (IPv4 and IPv6), otherwise it optimizes
source fragment size rather than avoiding source fragmentation
        MSS is defined in terms of EMTU_R, which is independent of PMTU
        there are two effective MTUs - EMTU_S and EMTU_R. The former is
set and may or may not be limited by PMTU.

I can coordinate off-list on both if useful.

Joe


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On 3/28/2017 6:06 PM, Joe Touch wrote:
> Some observations:
>
> 3.1 currently defined options
>      this either needs to be the complete current list or indicate that
> the list is elsewhere (the latter seems better to me)
>      if list elsewhere, the discussion should be clear that this doc
> defines only the following options; others are defined elsewhere
>      is there any notion of the minimum required options? if so, is this
> that set? (it might be useful to say so)


I propose that this should be described as the minimum required set, and 
that we add pointer to the IANA registry of TCP Option Kind Numbers for 
the complete set:
https://www.iana.org/assignments/tcp-parameters/tcp-parameters.xhtml


> throughout
>      the discussion of PMTU and MSS might benefit from an update as per
> draft-iett-intarea-tunnels
>      key points:
>          PMTU discovery assumes TCP set IP to prevent both on-path (for
> IPv4) and source fragmentation (IPv4 and IPv6), otherwise it optimizes
> source fragment size rather than avoiding source fragmentation
>          MSS is defined in terms of EMTU_R, which is independent of PMTU
>          there are two effective MTUs - EMTU_S and EMTU_R. The former is
> set and may or may not be limited by PMTU.


If clarity can be improved in 793bis, that sounds good to me.  The text 
currently is patched together from bits of many other RFCs.


> I can coordinate off-list on both if useful.

I think on-list is preferable to make sure everyone else agrees with us :)


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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rfc793bis-05.txt
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Hi,

I have a question / concern that=E2=80=99s different from the things Wes =
mentioned about this diff:

for both the old and new RFC, if I want to implement a TCP that is able =
to send data on the first SYN (not TFO-style, but as allowed by this =
spec), how do I make this possible using the API in section 3.9?

This needs application data to be handed over before the handshake is =
completed. I think this could only be done with =E2=80=9CSEND=E2=80=9D =
triggering =E2=80=9COPEN=E2=80=9D, as described in this text:=20

***
Some implementations may allow users to SEND first; in
         which case, an automatic OPEN would be done.  If the calling
         process is not authorized to use this connection, an error is
         returned.
***

As it stands, when I read this, it makes me think =E2=80=9Cwhy would =
anyone do this?=E2=80=9D. I believe there should be some clarification =
there regarding the point of this - to explain that this can make it =
possible to send data together with the SYN.

Cheers,
Michael



> On Mar 28, 2017, at 4:50 PM, Internet-Drafts@ietf.org wrote:
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions =
of the IETF.
>=20
>        Title           : Transmission Control Protocol Specification
>        Author          : Wesley M. Eddy
> 	Filename        : draft-ietf-tcpm-rfc793bis-05.txt
> 	Pages           : 95
> 	Date            : 2017-03-28
>=20
> 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).
>=20
>   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.
>=20
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rfc793bis/
>=20
> There are also htmlized versions available at:
> https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-05
> https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rfc793bis-05
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rfc793bis-05
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-cubic-04
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Hi all,

I have a comment:

When we did tests with Cubic in a local testbed, it exhibited nonlinear =
behavior as we varied the BDP (e.g. changing capacity or RTT).

We found the =E2=80=9CFast convergence=E2=80=9D mechanism in section 4.6 =
to be the culprit: it assumes that, when W_max < W_last_max, new flows =
have joined and Cubic should make room for them.
However, when doing tests of only a single flow in a controlled =
environment, and the BDP is e.g. around 10.5 packets, in reality, a flow =
may get W_max of 10, 11, 10, 11, =E2=80=A6 etc. This means that the =
mechanism keeps kicking in. Next, you do a test with a BDP of 0.5 =
packets more, and then W_max is statically 11 =E2=80=A6 so the behavior =
fluctuates quite a bit as a result of small changes to the BDP.

When testing flows in controlled condition and in isolation, I think =
this mechanism doesn=E2=80=99t make sense to use anyway. So I think it =
would be useful to insert a comment about this in section 4.6 for people =
doing such experiments.

Besides, I think the last paragraph of section 4.6 is hard to =
understand. It just doesn=E2=80=99t seem clear to say that flows with =
smaller shares can better increase their windows because flows with =
larger shares =E2=80=9Cspend more time around the plateau=E2=80=9D.

The comments in the code snippet are also confusing: =E2=80=9Ccheck =
downward trend=E2=80=9D and =E2=80=9Ccheck upward trend=E2=80=9D, what =
is that supposed to mean? I think =E2=80=9Ccheck downward trend=E2=80=9D =
could be =E2=80=9Cshould we make room for other flows?=E2=80=9D or =
something like that.

Cheers,
Michael


> On Mar 23, 2017, at 4:23 PM, Yoshifumi Nishida =
<nishida@sfc.wide.ad.jp> wrote:
>=20
> Hello,
> The WGLC for this draft has already finished, but we haven't seen many =
comments on this draft.
> We are trying to interpret this is an indication that people are =
satisfied with the current draft, but we just would like to be sure.=20
> Please let us know in case you have very last minute comments.
> We'll do similar final check at the venue. If nothing happens, we will =
try to proceed the doc after reviewing the updated version.
>=20
> Thanks,
> --
> tcpm co-chairs
>=20
> On Wed, Feb 15, 2017 at 1:13 AM, Yoshifumi Nishida =
<nishida@sfc.wide.ad.jp <mailto:nishida@sfc.wide.ad.jp>> wrote:
> Hello,
>=20
> In addition to the dctcp draft, TCP chairs conclude that =
draft-ietf-tcpm-cubic-04 is ready for WGLC as well and decided to =
initiate the call.
>=20
> The WGLC for this draft runs until * Friday March 3 *.=20
> Please send your feedback to the ML.
>=20
> BTW, the intended status of the draft is Informational.=20
> The draft is available from the following URL.
> https://tools.ietf.org/html/draft-ietf-tcpm-cubic-04 =
<https://tools.ietf.org/html/draft-ietf-tcpm-cubic-04>
>=20
> Thanks,
> --
> Yoshi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


--Apple-Mail=_BE676EDC-CC91-4D07-980D-237B9495DBC1
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	charset=utf-8

<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"">Hi all,<div class=3D""><br class=3D""></div><div class=3D"">I =
have a comment:</div><div class=3D""><br class=3D""></div><div =
class=3D"">When we did tests with Cubic in a local testbed, it exhibited =
nonlinear behavior as we varied the BDP (e.g. changing capacity or =
RTT).</div><div class=3D""><br class=3D""></div><div class=3D"">We found =
the =E2=80=9CFast convergence=E2=80=9D mechanism in section 4.6 to be =
the culprit: it assumes that, when W_max &lt; W_last_max, new flows have =
joined and Cubic should make room for them.</div><div class=3D"">However, =
when doing tests of only a single flow in a controlled environment, and =
the BDP is e.g. around 10.5 packets, in reality, a flow may get W_max of =
10, 11, 10, 11, =E2=80=A6 etc. This means that the mechanism keeps =
kicking in. Next, you do a test with a BDP of 0.5 packets more, and then =
W_max is statically 11 =E2=80=A6 so the behavior fluctuates quite a bit =
as a result of small changes to the BDP.</div><div class=3D""><br =
class=3D""></div><div class=3D"">When testing flows in controlled =
condition and in isolation, I think this mechanism doesn=E2=80=99t make =
sense to use anyway. So I think it would be useful to insert a comment =
about this in section 4.6 for people doing such experiments.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Besides, I think the =
last paragraph of section 4.6 is hard to understand. It just doesn=E2=80=99=
t seem clear to say that flows with smaller shares can better increase =
their windows because flows with larger shares =E2=80=9Cspend more time =
around the plateau=E2=80=9D.</div><div class=3D""><br =
class=3D""></div><div class=3D"">The comments in the code snippet are =
also confusing: =E2=80=9Ccheck downward trend=E2=80=9D and =E2=80=9Ccheck =
upward trend=E2=80=9D, what is that supposed to mean? I think =E2=80=9Cche=
ck downward trend=E2=80=9D could be =E2=80=9Cshould we make room for =
other flows?=E2=80=9D or something like that.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Cheers,</div><div =
class=3D"">Michael</div><div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Mar 23, 2017, at 4:23 PM, Yoshifumi Nishida &lt;<a =
href=3D"mailto:nishida@sfc.wide.ad.jp" =
class=3D"">nishida@sfc.wide.ad.jp</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D"">Hello,<div class=3D""><div class=3D"gmail_extra">The WGLC for =
this draft has already finished, but we haven't seen many comments on =
this draft.</div><div class=3D"gmail_extra">We are trying to interpret =
this is an indication that people are satisfied with the current draft, =
but we just would like to be sure.&nbsp;</div><div =
class=3D"gmail_extra">Please let us know in case you have very last =
minute comments.</div><div class=3D"gmail_extra">We'll do similar final =
check at the venue. If nothing happens, we will try to proceed the doc =
after reviewing the updated version.<br class=3D""></div><div =
class=3D"gmail_extra"><br class=3D""></div><div =
class=3D"gmail_extra">Thanks,</div><div class=3D"gmail_extra">--</div><div=
 class=3D"gmail_extra">tcpm co-chairs</div><div class=3D"gmail_extra"><br =
class=3D""><div class=3D"gmail_quote">On Wed, Feb 15, 2017 at 1:13 AM, =
Yoshifumi Nishida <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank" =
class=3D"">nishida@sfc.wide.ad.jp</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"><div dir=3D"ltr" =
class=3D"">Hello,<div class=3D""><br class=3D""></div><div class=3D"">In =
addition to the dctcp draft, TCP chairs conclude that =
draft-ietf-tcpm-cubic-04 is ready for WGLC as well and decided to =
initiate the call.</div><div class=3D""><br class=3D""></div><div =
class=3D"">The WGLC for this draft runs until * Friday March 3 =
*.&nbsp;<br class=3D""></div><div class=3D"">Please send your feedback =
to the ML.</div><div class=3D""><br class=3D""></div><div class=3D""><div =
class=3D"">BTW, the intended status of the draft is =
Informational.&nbsp;</div><div class=3D"">The draft is available from =
the following URL.</div></div><div class=3D""><div class=3D""><a =
href=3D"https://tools.ietf.org/html/draft-ietf-tcpm-cubic-04" =
target=3D"_blank" class=3D"">https://tools.ietf.org/html/dr<wbr =
class=3D"">aft-ietf-tcpm-cubic-04</a></div></div><div class=3D""><br =
class=3D""></div><div class=3D"">Thanks,</div><div class=3D"">--</div><div=
 class=3D"">Yoshi</div></div>
</blockquote></div><br class=3D""></div></div></div>
_______________________________________________<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"">https://www.ietf.org/mailman/listinfo/tcpm<br =
class=3D""></div></blockquote></div><br class=3D""></div></body></html>=

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From: "Scharf, Michael (Nokia - DE/Stuttgart)" <michael.scharf@nokia.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: RFC 793 (bis) vs. rto-consider
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Subject: [tcpm] RFC 793 (bis) vs. rto-consider
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<chair hat off>

Maybe I should expand on what I have orally mentioned in the meeting:

draft-ietf-tcpm-rfc793bis-05 states:


   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 [8], including
   Karn's algorithm for taking RTT samples.

The first sentence with "must" is taken from RFC 793. The second sentence w=
ith "MUST" seems to follow from the MUST in RFC 6298. In my reading of the =
standards, the current wording in 793bis correctly reflects the wording of =
TCP PS standards.

draft-ietf-tcpm-rto-consider-05 states:


        (a) In steady state the RTO SHOULD be set based on recent

            observations of both the FT and the variance of the FT.



            In other words, the RTO should be based on a reasonable

            amount of time that the sender should wait for delivery

            confirmation before retransmitting the given data.



        (b) FT observations SHOULD be taken regularly.

To me, this wording seems compatible with RFC 8085, i.e., the BCP for UDP.

When rto-consider was approved as TCPM WG item, both (a) and (b) was a MUST=
.

As individual contributor, I struggle with accepting a SHOULD in (a) and (b=
) as best current practice requirement for TCP. In TCP, the well-defined pr=
otocol mechanisms IMHO allow a MUST for both cases, and this is how I read =
the "must" in RFC 793.

Also, I believe that in this case the requirements for TCP can (well, must)=
 be tighter than for UDP protocols. The difference is that for UDP protocol=
s it cannot be guaranteed that the protocol allows regular FT observations.=
 In TCP we know that this is possible. So why allowing a SHOULD for TCP?

So, I wonder if there could be a reasoning for applying these two SHOULD to=
 TCP? Most notably, could there be any reason why a TCP implementation woul=
d not measure RTT "regularly"?

Any thoughts?

Michael

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<p class=3D"MsoNormal">&lt;chair hat off&gt;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Maybe I should expand on what I have orally mentione=
d in the meeting:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">draft-ietf-tcpm-rfc793bis-05 states:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<pre> &nbsp;&nbsp;Because of the variability of the networks that compose a=
n<o:p></o:p></pre>
<pre>&nbsp;&nbsp; internetwork system and the wide range of uses of TCP con=
nections the<o:p></o:p></pre>
<pre>&nbsp;&nbsp; retransmission timeout (RTO) must be dynamically determin=
ed.<o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; The RTO MUST be computed according to the alg=
orithm in [8], including<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; Karn's algorithm for taking RTT samples.<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The first sentence with <span style=3D"font-family:&=
quot;Times New Roman&quot;,serif">
&#8220;</span>must<span style=3D"font-family:&quot;Times New Roman&quot;,se=
rif">&#8221;</span> is taken from RFC 793. The second sentence with
<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8220;</span=
>MUST<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8221;<=
/span> seems to follow from the MUST in RFC 6298. In my reading of the stan=
dards, the current wording in 793bis correctly reflects the wording
 of TCP PS standards.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">draft-ietf-tcpm-rto-consider-05 states:<o:p></o:p></=
p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<pre> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(a) In steady state the RTO=
 SHOULD be set based on recent<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; obs=
ervations of both the FT and the variance of the FT.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; In =
other words, the RTO should be based on a reasonable<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; amo=
unt of time that the sender should wait for delivery<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; con=
firmation before retransmitting the given data.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (b) FT observations SHOULD =
be taken regularly.<o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">To me, this wording seems compatible with RFC 8085, =
i.e., the BCP for UDP.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">When rto-consider was approved as TCPM WG item, both=
 (a) and (b) was a MUST.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As individual contributor, I struggle with accepting=
 a SHOULD in (a) and (b) as best current practice requirement for TCP. In T=
CP, the well-defined protocol mechanisms IMHO allow a MUST for both cases, =
and this is how I read the
<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8220;</span=
>must<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8221;<=
/span> in RFC 793.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Also, I believe that in this case the requirements f=
or TCP can (well, must) be tighter than for UDP protocols. The difference i=
s that for UDP protocols it cannot be guaranteed that the protocol allows r=
egular FT observations. In TCP we
 know that this is possible. So why allowing a SHOULD for TCP?<o:p></o:p></=
p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So, I wonder if there could be a reasoning for apply=
ing these two SHOULD to TCP? Most notably, could there be any reason why a =
TCP implementation would not measure RTT
<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8220;</span=
>regularly<span style=3D"font-family:&quot;Times New Roman&quot;,serif">&#8=
221;</span>?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Any thoughts?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Michael<o:p></o:p></p>
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To: "Scharf\, Michael \(Nokia - DE\/Stuttgart\)" <michael.scharf@nokia.com>
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I think this is all a little too pedantic for me.

To my eye the difference between a "MUST" and a "SHOULD" is a good
reason.  That is, a "SHOULD" is in fact a "MUST" except when there
is a good reason to do something else.

So, do I have a handy (good) reason why TCP would not take an RTT
sample once per RTT?  No, not really.  So, I think it basically must
do so.

That said, we have done enough work tweaking the TCP RTO to know
that the world would not end if a sample was taken every other RTT
(in fact, this often happens naturally during periods of loss
because of Karn's algorithm).  And, in fact, the RTO performance
would be pretty much the same.  Is that my druthers?  No, not
really.  I'd like to see the once per RTT sampling in TCP.  But, if
someone came to me and said "for efficiency reasons in my lightbulb
TCP implementation I only take an RTT every other RTT" I would
probably shrug and go "seems fine".

Document wording aside: Do you think the once-per-RTT sampling in
TCP is so crucial that if an implementation did not adhere to it
that it would cause big problems (for the network or that
implementation)?

allman


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Thread-Topic: [tcpm] RFC 793 (bis) vs. rto-consider 
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/vb8pqdDMTe903GQUOu6GI5cAoZM>
Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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The RFC 2119 wording is

SHOULD   This word, or the adjective "RECOMMENDED", mean that there
   may exist valid reasons in particular circumstances to ignore a
   particular item, but the full implications must be understood and
   carefully weighed before choosing a different course.

I really wonder if there would exist *any* valid reason for a TCP implement=
ation not to measure the FT "regularly".=20

The opposite to "regularly" is "not regularly". "not regularly" would e.g. =
to only measure the FT once during the lifetime of a connection. I cannot r=
eally think of particular circumstances where this would make sense in TCP.

For TCP, I think the SHOULD in rto-consider is not the right guidance (let =
alone that it clearly contradicts RFC 6298). But for UDP the SHOULD is prob=
ably appropriate as per RFC 8085. So it seems that I run into an inconsiste=
ncy between my opinion for TCP and UDP, which in turn causes more formal fo=
llow-up issues.

Note that my comment is not about the measurement frequency and the MUST fo=
r once-per-RTT in RFC 6298. I would review this wording in a RFC6298bis.

Michael
=20

> -----Original Message-----
> From: mallman@icir.org [mailto:mallman@icir.org]
> Sent: Mittwoch, 29. M=E4rz 2017 22:33
> To: Scharf, Michael (Nokia - DE/Stuttgart) <michael.scharf@nokia.com>
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
>=20
>=20
> I think this is all a little too pedantic for me.
>=20
> To my eye the difference between a "MUST" and a "SHOULD" is a good
> reason.  That is, a "SHOULD" is in fact a "MUST" except when there is a g=
ood
> reason to do something else.
>=20
> So, do I have a handy (good) reason why TCP would not take an RTT sample
> once per RTT?  No, not really.  So, I think it basically must do so.
>=20
> That said, we have done enough work tweaking the TCP RTO to know that
> the world would not end if a sample was taken every other RTT (in fact, t=
his
> often happens naturally during periods of loss because of Karn's algorith=
m).
> And, in fact, the RTO performance would be pretty much the same.  Is that
> my druthers?  No, not really.  I'd like to see the once per RTT sampling =
in TCP.
> But, if someone came to me and said "for efficiency reasons in my lightbu=
lb
> TCP implementation I only take an RTT every other RTT" I would probably
> shrug and go "seems fine".
>=20
> Document wording aside: Do you think the once-per-RTT sampling in TCP is
> so crucial that if an implementation did not adhere to it that it would c=
ause
> big problems (for the network or that implementation)?
>=20
> allman
>=20
>=20
> --
> https://www.icir.org/mallman/
> @mallman_icsi
>=20
>=20


From nobody Wed Mar 29 14:33:36 2017
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Subject: [tcpm] About CUBIC and buffer draining
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Hi,

In Praveen=E2=80=99s very interesting presentation on TCP in Windows, =
today:
=
https://www.ietf.org/proceedings/98/slides/slides-98-tcpm-tcp-improvements=
-in-windows-01.pdf
he stated that =E2=80=9CCUBIC builds up large buffer in absence of =
AQM=E2=80=9D  (slide 4).

In the absence of an AQM algorithm, even a single Reno flow alone will =
not let a queue drain when it=E2=80=99s (for several practical reasons, =
slightly more than) a BDP in length. To allow a BDP worth of queuing to =
drain, the sender needs a backoff factor of 0.5 or less. CUBIC=E2=80=99s =
backoff (multiplication) factor is 0.7, so it can even produce a =
standing queue when the queue is smaller than a BDP.

For reasonably configured networks, and with background load =
fluctuations, that=E2=80=99s maybe not such a huge problem (after all, =
the Internet still works  ;-)  ).
However, CUBIC=E2=80=99s backoff factor choice is =E2=80=9Cblind=E2=80=9D =
- CUBIC has no idea of the length of the queue.

That=E2=80=99s what we=E2=80=99re trying to fix with ABE =
(draft-khademi-tcpm-alternativebackoff-ecn): because ECN is a hint that =
the queue was probably small, we can use a larger backoff factor (i.e. =
reduce by less).  =3D> if the whole Internet would support ECN, and ABE =
was implemented as a way to back off, I would wonder: should CUBIC use =
0.5 when it reacts to loss?

These are researchy futuristic musings. Getting back down to earth, I =
think it may be good to add a sentence to the CUBIC draft, where the =
backoff factor of 0.7 is introduced (last paragraph of section 3). =
There, the text now only says:

***
   CUBIC sets the multiplicative window decrease factor to 0.7 while
   Standard TCP uses 0.5.  While this improves the scalability of the
   protocol, a side effect of this decision is slower convergence
   especially under low statistical multiplexing environments.
***

=E2=80=A6 but I think the fact that it is more likely to produce a =
standing queue is a bigger issue, and worth mentioning here.

Cheers,
Michael


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Subject: [tcpm] on resetting backoff in rto-consider
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--94eb2c05d0d4d962a0054be95e74
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    (3) Each time the RTO is used to detect a loss and a retransmission
        is scheduled, the value of the RTO MUST be exponentially backed
        off such that the next firing requires a longer interval.  The
        backoff SHOULD be removed after the successful repair of the
        lost data and subsequent transmission of non-retransmitted data.



I wanted to ask in mic but we ran out of time: AFAIK, the backoff is reset
whenever a new RTT sample is obtained, not necessarily the segment being
repaired, either via new data being acked or TS-opt approach. The rationale
that is exp-backoff is not necessary if ack clocking has been restored.

Since we are talking about wiggle rooms: I would propose we just say
backoff SHOULD be removed once a new FT is obtained.

Is this critical? probably not. But I just want to finish my questions.

--94eb2c05d0d4d962a0054be95e74
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<div dir=3D"ltr"><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;=
margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x;color:rgb(0,0,0)"><br></pre><pre class=3D"gmail-newpage" style=3D"font-si=
ze:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">    (3) Eac=
h time the RTO is used to detect a loss and a retransmission
        is scheduled, the value of the RTO MUST be exponentially backed
        off such that the next firing requires a longer interval.  The
        backoff SHOULD be removed after the successful repair of the
        lost data and subsequent transmission of non-retransmitted data.
</pre><div><br></div><div><br></div><div>I wanted to ask in mic but we ran =
out of time: AFAIK, the backoff is reset whenever a new RTT sample is obtai=
ned, not necessarily the segment being repaired, either via new data being =
acked or TS-opt approach. The rationale that is exp-backoff is not necessar=
y if ack clocking has been restored.</div><div><br></div><div>Since we are =
talking about wiggle rooms: I would propose we just say backoff SHOULD be r=
emoved once a new FT is obtained.</div><div><br></div><div>Is this critical=
? probably not. But I just want to finish my questions.</div></div>

--94eb2c05d0d4d962a0054be95e74--


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From nobody Wed Mar 29 19:48:24 2017
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> The RFC 2119 wording is
>=20
> SHOULD   This word, or the adjective "RECOMMENDED", mean that there
>    may exist valid reasons in particular circumstances to ignore a
>    particular item, but the full implications must be understood and
>    carefully weighed before choosing a different course.

Right... like I said: the difference between a MUST and a SHOULD is
a good reason.  (At least when we're talking about things that don't
involve interoperability.)

> I really wonder if there would exist *any* valid reason for a TCP
> implementation not to measure the FT "regularly".=20=20
>=20
> The opposite to "regularly" is "not regularly". "not regularly"
> would e.g. to only measure the FT once during the lifetime of a
> connection. I cannot really think of particular circumstances
> where this would make sense in TCP.=20

Look, that is just another point in the tradeoff space.  We looked
at it in [AP99].  It turns out this sort of "take first" approach
reduces the waiting time for needed retransmissions compared to
6298.  It also follows the fundamental tradeoff and does this at the
expense of increasing spurious retransmissions.  But, that is no
different than many changes one can make---and have been made in
running code (e.g., hacking at the EWMA gains or the variance
multiplier or whatever).  And, the take-first approach is sort of
middling.  It isn't particularly good or bad in either direction.
It certainly isn't ludicrous by any stretch.

Do I think this is a great spot in the tradeoff space?  Not really.
Would I want to say that implementations "MAY take a single RTT
measurement per connection"?  No, not really.  I think RTOs benefit
From=20taking regular RTT measurements and feeding those into an RTO
estimator.  But, will the world end if they don't?  Nope.  I would
certainly not make some categorical statement like there could never
be any valid reason to do this.  Certainly it'd be a simpler
implementation.  And, maybe for some things that is an over-riding
concern.

But, whether Michael or I or anyone would do this is absolutely
beside the point.  To me the question is whether we think the impact
of doing this is so egregiously bad that we are compelled to say
"never do it" it the strongest possible way.  Maybe Michael has
evidence I haven't seen, but I don't think we're anywhere even close
to that point.

> (let alone that it clearly contradicts RFC 6298)

I think "contradict" is the wrong word here.  It is different to be
sure.  But, the **POINT** of rto-consider is to be different and to
loosen things up.  If it wasn't different than 6298 then there
wouldn't be any point.

If you want things nice and tight then rto-consider probably runs
counter to your thinking for the entire nine pages.

And, if consensus is that we want precise and tight algorithms that
are well specified in formal language but divorced from reality, OK
(which is where we're at now, BTW).  I'll accept that.  I don't
agree with that notion.  But, I've been on the wrong side of
consensus before.  However, that is not at all my read of consensus.

allman




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> On Mar 29, 2017, at 12:44 PM, Scharf, Michael (Nokia - DE/Stuttgart) <mich=
ael.scharf@nokia.com> wrote:
>=20
> So, I wonder if there could be a reasoning for applying these two SHOULD t=
o TCP? Most notably, could there be any reason why a TCP implementation woul=
d not measure RTT =E2=80=9Cregularly=E2=80=9D?

Overhead and complexity for small footprint devices. IMO SHOULD is sufficien=
t. MUST needs to be reserved for protocol correctness or behavior truly haza=
rdous to others.

Joe=

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div></div><div><br></div><div><br>On Mar 2=
9, 2017, at 12:44 PM, Scharf, Michael (Nokia - DE/Stuttgart) &lt;<a href=3D"=
mailto:michael.scharf@nokia.com">michael.scharf@nokia.com</a>&gt; wrote:<br>=
<br></div><blockquote type=3D"cite"><div>So, I wonder if there could be a re=
asoning for applying these two SHOULD to TCP? Most notably, could there be a=
ny reason why a TCP implementation would not measure RTT
<span style=3D"font-family:&quot;Times New Roman&quot;,serif">=E2=80=9C</spa=
n>regularly<span style=3D"font-family:&quot;Times New Roman&quot;,serif">=E2=
=80=9D</span>?</div></blockquote><br><div>Overhead and complexity for small f=
ootprint devices. IMO SHOULD is sufficient. MUST needs to be reserved for pr=
otocol correctness or behavior truly hazardous to others.</div><div><br></di=
v><div>Joe</div></body></html>=

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Sigh, copy + paste error fix, sorry - in line:

> On Mar 29, 2017, at 4:33 PM, Michael Welzl <michawe@ifi.uio.no> wrote:
>=20
> Hi,
>=20
> In Praveen=E2=80=99s very interesting presentation on TCP in Windows, =
today:
> =
https://www.ietf.org/proceedings/98/slides/slides-98-tcpm-tcp-improvements=
-in-windows-01.pdf
> he stated that =E2=80=9CCUBIC builds up large buffer in absence of =
AQM=E2=80=9D  (slide 4).
>=20
> In the absence of an AQM algorithm, even a single Reno flow alone will =
not let a queue drain when it=E2=80=99s (for several practical reasons, =
slightly more than) a BDP in length. To allow a BDP worth of queuing to =
drain, the sender needs a backoff factor of 0.5 or less. CUBIC=E2=80=99s =
backoff (multiplication) factor is 0.7, so it can even produce a =
standing queue when the queue is smaller than a BDP.
>=20
> For reasonably configured networks, and with background load =
fluctuations, that=E2=80=99s maybe not such a huge problem (after all, =
the Internet still works  ;-)  ).
> However, CUBIC=E2=80=99s backoff factor choice is =E2=80=9Cblind=E2=80=9D=
 - CUBIC has no idea of the length of the queue.
>=20
> That=E2=80=99s what we=E2=80=99re trying to fix with ABE =
(draft-khademi-tcpm-alternativebackoff-ecn):

That should be: draft-ietf-tcpm-alternativebackoff-ecn-00

Cheers,
Michael

> because ECN is a hint that the queue was probably small, we can use a =
larger backoff factor (i.e. reduce by less).  =3D> if the whole Internet =
would support ECN, and ABE was implemented as a way to back off, I would =
wonder: should CUBIC use 0.5 when it reacts to loss?
>=20
> These are researchy futuristic musings. Getting back down to earth, I =
think it may be good to add a sentence to the CUBIC draft, where the =
backoff factor of 0.7 is introduced (last paragraph of section 3). =
There, the text now only says:
>=20
> ***
>   CUBIC sets the multiplicative window decrease factor to 0.7 while
>   Standard TCP uses 0.5.  While this improves the scalability of the
>   protocol, a side effect of this decision is slower convergence
>   especially under low statistical multiplexing environments.
> ***
>=20
> =E2=80=A6 but I think the fact that it is more likely to produce a =
standing queue is a bigger issue, and worth mentioning here.
>=20
> Cheers,
> Michael
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Nokia - DE/Stuttgart)" <michael.scharf@nokia.com>
To: Joe Touch <touch@isi.edu>
CC: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] RFC 793 (bis) vs. rto-consider
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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It comes down to what "regularly" means, e.g. vs "occasionally" or "more tha=
n just once". There is no measurement method that ensures avoiding false rep=
orting except "continuous", could be excessive.

If you have a lot of iot devices and they all measure once, they would only c=
ause a storm on spurious timeouts if they all got it wrong. I think the law o=
f large numbers ought to work in your favor as long as you assume that the r=
tts measurements don't coincide - I.e., it's useful to decorrelate to avoid s=
ynchronization after a power recovery.

The issue of downgrading may drive this, but we should be clear that we don'=
t know that MUST is really required.

Joe

> On Mar 30, 2017, at 4:41 AM, Scharf, Michael (Nokia - DE/Stuttgart) <micha=
el.scharf@nokia.com> wrote:
>=20
> Since the context is a small footprint device (IoT and the like=E2=80=A6):=
 What happens if the device has a relatively simply TCP stack (no SACK=E2=80=
=A6) and runs on a low-bandwidth shared medium channel (wireless) that has h=
ighly variable RTT?
> =20
> If it measures the RTT once (to reduce complexity) and if that sample happ=
ens to be way below the average RTT, it can run repeatedly into spurious tim=
eouts, no? Then the exponential backoff and congestion control kicks in, but=
 if there are many of these devices (sensors=E2=80=A6) on the medium, transm=
itting every message multiple times could be hazardous, no?
> =20
> I think it comes down to the question if measuring the RTT more than once d=
uring a connection is needed for congestion control purposes, or not.
> =20
> Since there has always been the =E2=80=9Cmust=E2=80=9D wording in RFC 793 (=
and it has been a MUST in rto-consider in many revisions as well), I am not s=
ure if we can downgrade this now to SHOULD, e.g., without experimental data.=

> =20
> I think it is somehow inherently non-trivial to formally apply 2119 langua=
ge when the statement is not really about protocol correctness and interoper=
ability.
> =20
> Michael
> =20
> From: Joe Touch [mailto:touch@isi.edu]=20
> Sent: Donnerstag, 30. M=C3=A4rz 2017 08:33
> To: Scharf, Michael (Nokia - DE/Stuttgart) <michael.scharf@nokia.com>
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
> =20
> =20
>=20
> On Mar 29, 2017, at 12:44 PM, Scharf, Michael (Nokia - DE/Stuttgart) <mich=
ael.scharf@nokia.com> wrote:
>=20
> So, I wonder if there could be a reasoning for applying these two SHOULD t=
o TCP? Most notably, could there be any reason why a TCP implementation woul=
d not measure RTT =E2=80=9Cregularly=E2=80=9D?
> =20
> Overhead and complexity for small footprint devices. IMO SHOULD is suffici=
ent. MUST needs to be reserved for protocol correctness or behavior truly ha=
zardous to others.
> =20
> Joe

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div></div><div>It comes down to what "regu=
larly" means, e.g. vs "occasionally" or "more than just once". There is no m=
easurement method that ensures avoiding false reporting except "continuous",=
 could be excessive.</div><div><br></div><div>If you have a lot of iot devic=
es and they all measure once, they would only cause a storm on spurious time=
outs if they all got it wrong. I think the law of large numbers ought to wor=
k in your favor as long as you assume that the rtts measurements don't coinc=
ide - I.e., it's useful to decorrelate to avoid synchronization after a powe=
r recovery.</div><div><br></div><div>The issue of downgrading may drive this=
, but we should be clear that we don't know that MUST is really required.</d=
iv><div><br></div><div>Joe</div><div><br>On Mar 30, 2017, at 4:41 AM, Scharf=
, Michael (Nokia - DE/Stuttgart) &lt;<a href=3D"mailto:michael.scharf@nokia.=
com">michael.scharf@nokia.com</a>&gt; wrote:<br><br></div><blockquote type=3D=
"cite"><div>

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif">Since the context is a small footprint device (IoT an=
d the like</span><span style=3D"font-size:11.0pt">=E2=80=A6</span><span styl=
e=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif">):
 What happens if the device has a relatively simply TCP stack (no SACK</span=
><span style=3D"font-size:11.0pt">=E2=80=A6</span><span style=3D"font-size:1=
1.0pt;font-family:&quot;Calibri&quot;,sans-serif">) and runs on a low-bandwi=
dth shared medium channel (wireless) that has highly
 variable RTT?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif">If it measures the RTT once (to reduce complexity) an=
d if that sample happens to be way below the average RTT, it can run repeate=
dly into spurious timeouts, no? Then the exponential
 backoff and congestion control kicks in, but if there are many of these dev=
ices (sensors</span><span style=3D"font-size:11.0pt">=E2=80=A6</span><span s=
tyle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif">) on th=
e medium, transmitting every message multiple times
 could be hazardous, no?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif">I think it comes down to the question if measuring th=
e RTT more than once during a connection is needed for congestion control pu=
rposes, or not.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif">Since there has always been the
</span><span style=3D"font-size:11.0pt">=E2=80=9C</span><span style=3D"font-=
size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif">must</span><span sty=
le=3D"font-size:11.0pt">=E2=80=9D</span><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,sans-serif"> wording in RFC 793 (and it has been=

 a MUST in rto-consider in many revisions as well), I am not sure if we can d=
owngrade this now to SHOULD, e.g., without experimental data.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif">I think it is somehow inherently non-trivial to forma=
lly apply 2119 language when the statement is not really about protocol corr=
ectness and interoperability.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif">Michael<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,sans-serif"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4=
.0pt">
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0=
cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot;=
Calibri&quot;,sans-serif">From:</span></b><span style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,sans-serif"> Joe Touch [<a href=3D"mailto:touc=
h@isi.edu">mailto:touch@isi.edu</a>]
<br>
<b>Sent:</b> Donnerstag, 30. M=C3=A4rz 2017 08:33<br>
<b>To:</b> Scharf, Michael (Nokia - DE/Stuttgart) &lt;<a href=3D"mailto:mich=
ael.scharf@nokia.com">michael.scharf@nokia.com</a>&gt;<br>
<b>Cc:</b> <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<b>Subject:</b> Re: [tcpm] RFC 793 (bis) vs. rto-consider<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
On Mar 29, 2017, at 12:44 PM, Scharf, Michael (Nokia - DE/Stuttgart) &lt;<a h=
ref=3D"mailto:michael.scharf@nokia.com">michael.scharf@nokia.com</a>&gt; wro=
te:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">So, I wonder if there could be a reasoning for applyi=
ng these two SHOULD to TCP? Most notably, could there be any reason why a TC=
P implementation would not measure RTT =E2=80=9Cregularly=E2=80=9D?<o:p></o:=
p></p>
</div>
</blockquote>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Overhead and complexity for small footprint devices. I=
MO SHOULD is sufficient. MUST needs to be reserved for protocol correctness o=
r behavior truly hazardous to others.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Joe<o:p></o:p></p>
</div>
</div>
</div>


</div></blockquote></body></html>=

--Apple-Mail-CA17C088-92D9-4784-99D2-7CE9362B3F21--


From nobody Thu Mar 30 13:22:30 2017
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Date: Thu, 30 Mar 2017 23:22:15 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Scharf, Michael (Nokia - DE/Stuttgart)" <michael.scharf@nokia.com>
Cc: "Joe Touch" <touch@isi.edu>, "tcpm@ietf.org" <tcpm@ietf.org>, draft-ietf-core-cocoa@ietf.org
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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Hi,

(CC'ing the authors of CoCoA [1]...)

Hoping not to deviate too much from the main topic, I just wanted to
explain that in our work with CoCoA (which is used for CoAP over UDP), we
encountered issues in IoT experimental testbeds such as those described by
Michael.

RTT may experience high variance in different types of "typical" IoT
scenarios. Wireless links used in IoT are typically error-prone (due to
radio impairments such as multipath fading, interference, etc.) which may
lead to link layer retries that increase the RTT. Even wired links in IoT,
such as Power Line Communication ones, offer similar performance (for
different reasons).

In mesh networks, which are common in IoT, RTT variability increases.

Furthermore, some traffic patterns in IoT may complicate the scenario. 
Quite often, packet transmission is relatively infrequent. However,
several sensors may simultaneously detect some global event which needs to
be communicated, and a burst of messages will take place as a result
(increasing network load during that interval).

For these reasons, in CoCoA we have an "aging" mechanism for RTO estimates
that have not been updated for a while.

(We even have a 'weak' estimator in addition to a 'strong' estimator,
which may actually help improving the sometimes limited knowledge that a
'strong' estimator may obtain.)

Thanks,

Carles

[1] https://tools.ietf.org/html/draft-ietf-core-cocoa-01


> Since the context is a small footprint device (IoT and the likeâĤ): What
> happens if the device has a relatively simply TCP stack (no SACKâĤ) and
> runs on a low-bandwidth shared medium channel (wireless) that has highly
> variable RTT?
>
> If it measures the RTT once (to reduce complexity) and if that sample
> happens to be way below the average RTT, it can run repeatedly into
> spurious timeouts, no? Then the exponential backoff and congestion control
> kicks in, but if there are many of these devices (sensorsâĤ) on the
> medium, transmitting every message multiple times could be hazardous, no?
>
> I think it comes down to the question if measuring the RTT more than once
> during a connection is needed for congestion control purposes, or not.
>
> Since there has always been the âmustâ wording in RFC 793 (and it has
> been a MUST in rto-consider in many revisions as well), I am not sure if
> we can downgrade this now to SHOULD, e.g., without experimental data.
>
> I think it is somehow inherently non-trivial to formally apply 2119
> language when the statement is not really about protocol correctness and
> interoperability.
>
> Michael
>
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Donnerstag, 30. M¤rz 2017 08:33
> To: Scharf, Michael (Nokia - DE/Stuttgart) <michael.scharf@nokia.com>
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
>
>
>
> On Mar 29, 2017, at 12:44 PM, Scharf, Michael (Nokia - DE/Stuttgart)
> <michael.scharf@nokia.com<mailto:michael.scharf@nokia.com>> wrote:
> So, I wonder if there could be a reasoning for applying these two SHOULD
> to TCP? Most notably, could there be any reason why a TCP implementation
> would not measure RTT âregularlyâ?
>
> Overhead and complexity for small footprint devices. IMO SHOULD is
> sufficient. MUST needs to be reserved for protocol correctness or behavior
> truly hazardous to others.
>
> Joe
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



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From: William Herrin <bill@herrin.us>
Date: Thu, 30 Mar 2017 17:03:28 -0400
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Subject: Re: [tcpm] Anycast TCP
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--94eb2c0533a482c25c054bf90746
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On Wed, Mar 29, 2017 at 10:46 PM, William Herrin <bill@herrin.us> wrote:

>
> http://bill.herrin.us/network/anycasttcp.html
>
> Enjoy!
>
> -Bill Herrin
>

Greetings,

If there's a more appropriate WG to introduce this to, I sure would
appreciate a pointer.

Regards,
Bill Herrin

-- 
William Herrin ................ herrin@dirtside.com  bill@herrin.us
Dirtside Systems ......... Web: <http://www.dirtside.com/>

--94eb2c0533a482c25c054bf90746
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 29, 2017 at 10:46 PM, William Herrin <span dir=3D"ltr">&lt;<a href=
=3D"mailto:bill@herrin.us" target=3D"_blank">bill@herrin.us</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><br><div dir=3D"ltr"><div><div><a =
href=3D"http://bill.herrin.us/network/anycasttcp.html" target=3D"_blank">ht=
tp://bill.herrin.us/network/<wbr>anycasttcp.html</a><br><br></div>Enjoy!<br=
><br></div>-Bill Herrin<span class=3D"HOEnZb"><font color=3D"#888888"><br c=
lear=3D"all"></font></span></div>
</blockquote></div><br></div><div class=3D"gmail_extra">Greetings,<br><br><=
/div><div class=3D"gmail_extra">If there&#39;s a more appropriate WG to int=
roduce this to, I sure would appreciate a pointer.<br><br></div><div class=
=3D"gmail_extra">Regards,<br></div><div class=3D"gmail_extra">Bill Herrin<b=
r clear=3D"all"></div><div class=3D"gmail_extra"><br>-- <br><div class=3D"g=
mail_signature" data-smartmail=3D"gmail_signature">William Herrin .........=
....... <a href=3D"mailto:herrin@dirtside.com" target=3D"_blank">herrin@dir=
tside.com</a> =C2=A0<a href=3D"mailto:bill@herrin.us" target=3D"_blank">bil=
l@herrin.us</a><br>Dirtside Systems ......... Web: &lt;<a href=3D"http://ww=
w.dirtside.com/" target=3D"_blank">http://www.dirtside.com/</a>&gt;</div>
</div></div>

--94eb2c0533a482c25c054bf90746--


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you should list an optimization flood D, which is has a relationship to C like B 
does to A

no manipulation of sequence numbers, when you get something unknown, send it to 
all peers, but a peer can respond and say "I own that connection", eliminating 
future floods for that connection.

As a further optimization (of B and D), let the peers that you flood to send a 
nack, not just an ack, if you get a nack from all peers, you reset the 
connection.

It may be worth the slowdown to have the "do you own this connection" query be 
explicit, not implicit by forwarding the packets. This increases delays when 
something needs to be forwarded (at least one more round trip between the 
peers), but should greatly simplify the logic needed to communicate the status.

One question, if you use the "flood when not a known connection" procedure, do 
you still need all the kernel hooks? It seems to me that most of them could go 
away if you accepted the potential for a collision if multiple machines pick the 
same starting sequence number.

On Linux, IPTables already has a hook that can send packets to userspace. If you 
have a rule that accepts packets with known connections, and then follow that 
with a rule that forwards non-syn packets to the userspace program, do you still 
need a hook? The conntrack hooks allow you to manipulate the connection tables 
(so that you can take over a TCP connection that is on a cluster peer, very 
similar to this use case). So it seems to me that you may be able to implement a 
PoC of this on existing systems. Then measure where the bottlenecks are and what 
the 'miss rate' really is in the real world

David Lang

On Wed, 29 Mar 2017, William Herrin wrote:

> Date: Wed, 29 Mar 2017 22:46:49 -0400
> From: William Herrin <bill@herrin.us>
> To: tcpm@ietf.org
> Subject: [tcpm] Anycast TCP
> 
> http://bill.herrin.us/network/anycasttcp.html
>
> Enjoy!
>
> -Bill Herrin
>
>


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From: "Eggert, Lars" <lars@netapp.com>
To: William Herrin <bill@herrin.us>
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On 2017-3-30, at 16:03, William Herrin <bill@herrin.us> wrote:
> If there's a more appropriate WG to introduce this to, I sure would =
appreciate a pointer.

Until you submit an ID, which with it carries an obligation to disclose =
IPR, I expect you will not be able to engage in a discussion with many.

Lars

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From: Yuchung Cheng <ycheng@google.com>
Date: Thu, 30 Mar 2017 15:16:37 -0700
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To: Michael Welzl <michawe@ifi.uio.no>
Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, tsvwg IETF list <tsvwg@ietf.org>,  "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Subject: Re: [tcpm] [tsvwg] TCP ACK-CC (re: Transport protocol feedback overhead )
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regarding the question in the slide 5.
"Is immediate ACK needed for out of order segments if RACK is used ?"

The answer is both yes and no: a timely feedback is appreciated (yes). But
RACK does not need 100 DUPACKs each telling one more packet is delivered
out of order (no). In other words if the ACK is timely and precise about
the delivery process, RACK would work well (so does BBR). However this can
go terribly wrong if the middle-box sends only the last ACK/SACK. It has
happened in one major cloud platform.

On Thu, Mar 30, 2017 at 2:14 PM, Michael Welzl <michawe@ifi.uio.no> wrote:

> Hi,
>
> Thanks Ingemar for your interesting presentation today:
> https://www.ietf.org/proceedings/98/slides/slides-98-tsvwg-s
> essb-7-transport-protocol-feedback-overhead-issues-and-solutions-00.pdf
>
> I wanted to share with you, and the list, that David Ros  (one of the
> original authors of ACK-CC, RFC 5690) and I once advised a master student
> to:
> - implement ACK-CC
> - test it
> - analyze all RFC 2119 - style keywords in RFC 5690, to see how they
> should be changed (if at all) in a possible update
>
> Back then, our thinking was that we could write a paper about it, and try
> to get this deployed as a way of doing the Experiment and moving this to =
PS
> status.
> However, we were less and less sure about the real value of this - whethe=
r
> actual networks need this? Then, we had less and less time available and =
at
> some point dropped the ball.
> I of course found it interesting to hear that it *is* a real problem, eve=
n
> today.
>
> Anyway: this master thesis is available here:
> http://heim.ifi.uio.no/michawe/teaching/dipls/marius-olsen/
> mastersthesis-mariusno.pdf
> and I think I can safely say on behalf of David, Marius (the master
> student who did this) and myself that we=E2=80=99re happy if someone pick=
s this up
> and continues the work.
> It would require some digging, but I might also still have the code lying
> around somewhere - for some outdated version of Linux=E2=80=A6
>
> Cheers,
> Michael
>
>

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<div dir=3D"ltr">regarding the question in the slide 5.<div>&quot;Is immedi=
ate ACK needed for out of order segments if RACK is used ?&quot;</div><div>=
<br></div><div>The answer is both yes and no: a timely feedback is apprecia=
ted (yes). But RACK does not need 100 DUPACKs each telling one more packet =
is delivered out of order (no). In other words if the ACK is timely and pre=
cise about the delivery process, RACK would work well (so does BBR). Howeve=
r this can go terribly wrong if the middle-box sends only the last ACK/SACK=
. It has happened in one major cloud platform.</div><div><br></div><div>On =
Thu, Mar 30, 2017 at 2:14 PM, Michael Welzl <span dir=3D"ltr">&lt;<a href=
=3D"mailto:michawe@ifi.uio.no" target=3D"_blank">michawe@ifi.uio.no</a>&gt;=
</span> wrote:<br></div><div class=3D"gmail_extra"><div class=3D"gmail_quot=
e"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word">Hi,<d=
iv><br></div><div>Thanks Ingemar for your interesting presentation today:</=
div><div><a href=3D"https://www.ietf.org/proceedings/98/slides/slides-98-ts=
vwg-sessb-7-transport-protocol-feedback-overhead-issues-and-solutions-00.pd=
f" target=3D"_blank">https://www.ietf.org/proceedin<wbr>gs/98/slides/slides=
-98-tsvwg-s<wbr>essb-7-transport-protocol-feed<wbr>back-overhead-issues-and=
-solut<wbr>ions-00.pdf</a></div><div><br></div><div>I wanted to share with =
you, and the list, that David Ros =C2=A0(one of the original authors of ACK=
-CC, RFC 5690) and I once advised a master student to:</div><div>- implemen=
t ACK-CC</div><div>- test it</div><div>- analyze all RFC 2119 - style keywo=
rds in RFC 5690, to see how they should be changed (if at all) in a possibl=
e update</div><div><br></div><div>Back then, our thinking was that we could=
 write a paper about it, and try to get this deployed as a way of doing the=
 Experiment and moving this to PS status.</div><div>However, we were less a=
nd less sure about the real value of this - whether actual networks need th=
is? Then, we had less and less time available and at some point dropped the=
 ball.</div><div>I of course found it interesting to hear that it *is* a re=
al problem, even today.</div><div><br></div><div>Anyway: this master thesis=
 is available here: =C2=A0<a href=3D"http://heim.ifi.uio.no/michawe/teachin=
g/dipls/marius-olsen/mastersthesis-mariusno.pdf" target=3D"_blank">http://h=
eim.ifi.uio.no/michaw<wbr>e/teaching/dipls/marius-olsen/<wbr>mastersthesis-=
mariusno.pdf</a></div><div>and I think I can safely say on behalf of David,=
 Marius (the master student who did this) and myself that we=E2=80=99re hap=
py if someone picks this up and continues the work.</div><div>It would requ=
ire some digging, but I might also still have the code lying around somewhe=
re - for some outdated version of Linux=E2=80=A6</div><div><br></div><div>C=
heers,</div><div>Michael</div><div><br></div></div></blockquote></div><br><=
/div></div>

--001a114220ac567be4054bfa0e23--


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From: William Herrin <bill@herrin.us>
Date: Thu, 30 Mar 2017 23:00:30 -0400
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On Thu, Mar 30, 2017 at 5:28 PM, David Lang <david@lang.hm> wrote:

> you should list an optimization flood D, which is has a relationship to C
> like B does to A
>
> no manipulation of sequence numbers, when you get something unknown, send
> it to all peers, but a peer can respond and say "I own that connection",
> eliminating future floods for that connection.
>

Hi David,

Agreed. I didn't flesh out the flood versions because they all suffer from
a trivial attack vector: forged payload packets with randomized source
addresses and ports get flooded everywhere. That escalates a DOS against
one node into a DOS against all of them. If you see a way to mitigate the
problem then I think it'll be worth diving deeper in to the flood approach.



> As a further optimization (of B and D), let the peers that you flood to
> send a nack, not just an ack, if you get a nack from all peers, you reset
> the connection.
>

For completeness. I agree that would restore an ability to send a rst in
the flood case at the cost of briefly holding some discardable state.


It may be worth the slowdown to have the "do you own this connection" query
> be explicit, not implicit by forwarding the packets. This increases delays
> when something needs to be forwarded (at least one more round trip between
> the peers), but should greatly simplify the logic needed to communicate the
> status.
>

I don't follow how that would simplify the logic. Would you clarify?


One question, if you use the "flood when not a known connection" procedure,
> do you still need all the kernel hooks? It seems to me that most of them
> could go away if you accepted the potential for a collision if multiple
> machines pick the same starting sequence number.
>

Some of the flood versions need only the RST hook, as noted in Flood C.



> On Linux, IPTables already has a hook that can send packets to userspace.
> If you have a rule that accepts packets with known connections, and then
> follow that with a rule that forwards non-syn packets to the userspace
> program, do you still need a hook?


Outside a test harness, yes. Conntrack doubles the state information kept
for the TCP connections. However, that could be a reasonable shortcut for
prototyping and proving out the variants that don't adjust the sequence
number.


> The conntrack hooks allow you to manipulate the connection tables (so that
> you can take over a TCP connection that is on a cluster peer, very similar
> to this use case). So it seems to me that you may be able to implement a
> PoC of this on existing systems.


Good idea. Something like this:

sysctl net.netfilter.nf_conntrack_tcp_loose=0
sysctl net.netfilter.nf_conntrack_helper=0
iptables -A INPUT -p tcp --dport 80 -m conntrack \
  --ctstate NEW,ESTABLISHED -j ACCEPT
iptables -A INPUT -p tcp --dport 80 -j NFQUEUE --queue-num 0 --queue-bypass

And then use libnetfilter_queue to catch the packets in userspace.

However, that only simulates the RST hook so could only be used to evaluate
the flood versions.


Then measure where the bottlenecks are and what the 'miss rate' really is
> in the real world
>

We already have pretty good miss-rate numbers from research that's been
done for anycast DNS. I don't have a link handy but the short version is
that better than 99% of clients consistently hit the same anycast node.

Regards,
Bill Herrin



-- 
William Herrin ................ herrin@dirtside.com  bill@herrin.us
Dirtside Systems ......... Web: <http://www.dirtside.com/>

--94eb2c11ac32672d7a054bfe04ef
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 T=
hu, Mar 30, 2017 at 5:28 PM, David Lang <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:david@lang.hm" target=3D"_blank">david@lang.hm</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left:1px solid rgb(204,204,204);padding-left:1ex">you should list an opti=
mization flood D, which is has a relationship to C like B does to A<br>
<br>
no manipulation of sequence numbers, when you get something unknown, send i=
t to all peers, but a peer can respond and say &quot;I own that connection&=
quot;, eliminating future floods for that connection.<br></blockquote><div>=
<br></div><div>Hi David,<br><br></div><div>Agreed. I didn&#39;t flesh out t=
he flood versions because they all suffer from a trivial attack vector: for=
ged payload packets with randomized source addresses and ports get flooded =
everywhere. That escalates a DOS against one node into a DOS against all of=
 them. If you see a way to mitigate the problem then I think it&#39;ll be w=
orth diving deeper in to the flood approach.<br><br></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">

As a further optimization (of B and D), let the peers that you flood to sen=
d a nack, not just an ack, if you get a nack from all peers, you reset the =
connection.<br></blockquote><div><br></div><div>For completeness. I agree t=
hat would restore an ability to send a rst in the flood case at the cost of=
 briefly holding some discardable state.<br><br><br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex">

It may be worth the slowdown to have the &quot;do you own this connection&q=
uot; query be explicit, not implicit by forwarding the packets. This increa=
ses delays when something needs to be forwarded (at least one more round tr=
ip between the peers), but should greatly simplify the logic needed to comm=
unicate the status.<br></blockquote><div><br></div><div>I don&#39;t follow =
how that would simplify the logic. Would you clarify?<br></div><div>=C2=A0<=
br><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">

One question, if you use the &quot;flood when not a known connection&quot; =
procedure, do you still need all the kernel hooks? It seems to me that most=
 of them could go away if you accepted the potential for a collision if mul=
tiple machines pick the same starting sequence number.<br></blockquote><div=
><br></div><div>Some of the flood versions need only the RST hook, as noted=
 in Flood C.<br><br></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">

On Linux, IPTables already has a hook that can send packets to userspace. I=
f you have a rule that accepts packets with known connections, and then fol=
low that with a rule that forwards non-syn packets to the userspace program=
, do you still need a hook? </blockquote><div><br></div><div>Outside a test=
 harness, yes. Conntrack doubles the state information kept for the TCP con=
nections. However, that could be a reasonable shortcut for prototyping and =
proving out the variants that don&#39;t adjust the sequence number. <br></d=
iv><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">The c=
onntrack hooks allow you to manipulate the connection tables (so that you c=
an take over a TCP connection that is on a cluster peer, very similar to th=
is use case). So it seems to me that you may be able to implement a PoC of =
this on existing systems.</blockquote><div><br>Good idea. Something like th=
is:<br><br>sysctl net.netfilter.nf_conntrack_tcp_loose=3D0<br>sysctl net.ne=
tfilter.nf_conntrack_helper=3D0<br>iptables -A INPUT -p tcp --dport 80 -m c=
onntrack \<br>=C2=A0 --ctstate NEW,ESTABLISHED -j ACCEPT<br>iptables -A INP=
UT -p tcp --dport 80 -j NFQUEUE --queue-num 0 --queue-bypass<br><br><div>An=
d then use libnetfilter_queue to catch the packets in userspace.<br><br></d=
iv>However, that only simulates the RST hook so could only be used to evalu=
ate the flood versions.<br><br><br></div><blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);pa=
dding-left:1ex"> Then measure where the bottlenecks are and what the &#39;m=
iss rate&#39; really is in the real world<br></blockquote><div></div><div><=
br></div><div>We already have pretty good miss-rate numbers from research t=
hat&#39;s been done for anycast DNS. I don&#39;t have a link handy but the =
short version is that better than 99% of clients consistently hit the same =
anycast node.<br><br></div><div>Regards,<br></div><div>Bill Herrin<span></s=
pan><span></span></div></div><br><br clear=3D"all"><br>-- <br><div class=3D=
"gmail-m_1649416524807236267m_5888267306342902624m_-358028531585609481gmail=
_signature">William Herrin ................ <a href=3D"mailto:herrin@dirtsi=
de.com" target=3D"_blank">herrin@dirtside.com</a> =C2=A0<a href=3D"mailto:b=
ill@herrin.us" target=3D"_blank">bill@herrin.us</a><br>Dirtside Systems ...=
...... Web: &lt;<a href=3D"http://www.dirtside.com/" target=3D"_blank">http=
://www.dirtside.com/</a>&gt;</div>
</div></div>

--94eb2c11ac32672d7a054bfe04ef--


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From: William Herrin <bill@herrin.us>
Date: Thu, 30 Mar 2017 23:01:40 -0400
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To: "Eggert, Lars" <lars@netapp.com>
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Subject: Re: [tcpm] Anycast TCP
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On Thu, Mar 30, 2017 at 5:30 PM, Eggert, Lars <lars@netapp.com> wrote:

> On 2017-3-30, at 16:03, William Herrin <bill@herrin.us> wrote:
> > If there's a more appropriate WG to introduce this to, I sure would
> appreciate a pointer.
>
> Until you submit an ID, which with it carries an obligation to disclose
> IPR, I expect you will not be able to engage in a discussion with many.
>

Hi Lars,

That would be a discouraging shift from the open flow of ideas the last
time I participated in a WG.

For the record, while I've integrated constructive criticism from a couple
very generous individuals, the design is mine alone. I assert no IP rights
though I'd certainly appreciate being invited to join as an author of any
IDs that eventually result. To the best of my knowledge, there is no
comparable work which would allow another to assert IP rights. Certainly I
did not find any when I searched.

Regards,
Bill Herrin


-- 
William Herrin ................ herrin@dirtside.com  bill@herrin.us
Dirtside Systems ......... Web: <http://www.dirtside.com/>

--94eb2c0533a4899714054bfe082d
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 30, 2017 at 5:30 PM, Eggert, Lars <span dir=3D"ltr">&lt;<a href=3D"=
mailto:lars@netapp.com" target=3D"_blank">lars@netapp.com</a>&gt;</span> wr=
ote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex"><span>On 2017-3-30, at 16:03, Willia=
m Herrin &lt;<a href=3D"mailto:bill@herrin.us" target=3D"_blank">bill@herri=
n.us</a>&gt; wrote:<br>
&gt; If there&#39;s a more appropriate WG to introduce this to, I sure woul=
d appreciate a pointer.<br>
<br>
</span>Until you submit an ID, which with it carries an obligation to discl=
ose IPR, I expect you will not be able to engage in a discussion with many.=
<br></blockquote><div><br></div><div>Hi Lars,<br><br></div><div>That would =
be a discouraging shift from the open flow of ideas the last time I partici=
pated in a WG.<br><br></div><div>For the record, while I&#39;ve integrated =
constructive criticism from a couple very generous individuals, the design =
is mine alone. I assert no IP rights though I&#39;d certainly appreciate be=
ing invited to join as an author of any IDs that eventually result. To the =
best of my knowledge, there is no comparable work which would allow another=
 to assert IP rights. Certainly I did not find any when I searched.<br><br>=
</div><div>Regards,<br></div><div>Bill Herrin <br></div></div><br clear=3D"=
all"><br>-- <br><div class=3D"m_-1608810195695388504gmail_signature" data-s=
martmail=3D"gmail_signature">William Herrin ................ <a href=3D"mai=
lto:herrin@dirtside.com" target=3D"_blank">herrin@dirtside.com</a> =C2=A0<a=
 href=3D"mailto:bill@herrin.us" target=3D"_blank">bill@herrin.us</a><br>Dir=
tside Systems ......... Web: &lt;<a href=3D"http://www.dirtside.com/" targe=
t=3D"_blank">http://www.dirtside.com/</a>&gt;</div>
</div></div>

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CC: "tcpm@ietf.org" <tcpm@ietf.org>, "draft-ietf-core-cocoa@ietf.org" <draft-ietf-core-cocoa@ietf.org>
Thread-Topic: [tcpm] RFC 793 (bis) vs. rto-consider
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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Subject: Re: [tcpm] Anycast TCP
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On Thu, 30 Mar 2017, William Herrin wrote:

> On Thu, Mar 30, 2017 at 5:28 PM, David Lang <david@lang.hm> wrote:
>
>> you should list an optimization flood D, which is has a relationship to C
>> like B does to A
>>
>> no manipulation of sequence numbers, when you get something unknown, send
>> it to all peers, but a peer can respond and say "I own that connection",
>> eliminating future floods for that connection.
>>
>
> Hi David,
>
> Agreed. I didn't flesh out the flood versions because they all suffer from
> a trivial attack vector: forged payload packets with randomized source
> addresses and ports get flooded everywhere. That escalates a DOS against
> one node into a DOS against all of them. If you see a way to mitigate the
> problem then I think it'll be worth diving deeper in to the flood approach.

I think that the nack capability is a significant limit in the damage a flood 
can do (since the nacks will quickly shutdown the multiplication)

>
>
>> As a further optimization (of B and D), let the peers that you flood to
>> send a nack, not just an ack, if you get a nack from all peers, you reset
>> the connection.
>>
>
> For completeness. I agree that would restore an ability to send a rst in
> the flood case at the cost of briefly holding some discardable state.
>
>
> It may be worth the slowdown to have the "do you own this connection" query
>> be explicit, not implicit by forwarding the packets. This increases delays
>> when something needs to be forwarded (at least one more round trip between
>> the peers), but should greatly simplify the logic needed to communicate the
>> status.
>>
>
> I don't follow how that would simplify the logic. Would you clarify?

In your example, you were talking about sending the traffic out through tunnels 
to the nodes, and they would have to watch the traffic in the tunnel to detect 
that something new was showing up and send an ack.

If instead the node receiving the unknown traffic has an 'out of band' request 
to the other nodes "do you have connection X" with an explicit ack/nack, the 
other nodes don't have to watch the gre tunnel and decide if they need to send 
an ack based on that.

this involves a round-trip to the other nodes, but you are only sending the 
connection info (which can be packed many to a packet) and you don't send the 
actual payload packets out unless you get an ack back from someone.

This would also cut down on the dos multiplication factor as each malicious 
datastream isn't replicated in full, only it only generates a small chunk of 
traffic to the other nodes.

It wouldn't completely eliminate the vulnerability, but it would greatly reduce 
it.

>
> One question, if you use the "flood when not a known connection" procedure,
>> do you still need all the kernel hooks? It seems to me that most of them
>> could go away if you accepted the potential for a collision if multiple
>> machines pick the same starting sequence number.
>>
>
> Some of the flood versions need only the RST hook, as noted in Flood C.

if you can create a raw IP packet, can't you create one that is a rst for a 
particular connection?

>> On Linux, IPTables already has a hook that can send packets to userspace.
>> If you have a rule that accepts packets with known connections, and then
>> follow that with a rule that forwards non-syn packets to the userspace
>> program, do you still need a hook?
>
>
> Outside a test harness, yes. Conntrack doubles the state information kept
> for the TCP connections. However, that could be a reasonable shortcut for
> prototyping and proving out the variants that don't adjust the sequence
> number.
>
>
>> The conntrack hooks allow you to manipulate the connection tables (so that
>> you can take over a TCP connection that is on a cluster peer, very similar
>> to this use case). So it seems to me that you may be able to implement a
>> PoC of this on existing systems.
>
>
> Good idea. Something like this:
>
> sysctl net.netfilter.nf_conntrack_tcp_loose=0
> sysctl net.netfilter.nf_conntrack_helper=0
> iptables -A INPUT -p tcp --dport 80 -m conntrack \
>  --ctstate NEW,ESTABLISHED -j ACCEPT
> iptables -A INPUT -p tcp --dport 80 -j NFQUEUE --queue-num 0 --queue-bypass
>
> And then use libnetfilter_queue to catch the packets in userspace.
>
> However, that only simulates the RST hook so could only be used to evaluate
> the flood versions.

true, but that may be enough to show the value of the concept and evaluate if 
the flood versions can be tamed enough to be useful.

along similar lines, I'll point out multicast-mac (in linux, it's known as 
CLUSTERIP) that lets you put a number of boxes in one place with the same IP 
address and have each of them handle a subset of the traffic (the nic still sees 
the full load on each machine, but userspace only sees a subset.

so you have

big_pipe ->
   switch (multiple ->)
     server nic ->
       clusterip (traffic reduced to 1/servers at location) ->
         conntrack ->
           new/established (normal userspace handling)
           netfilter ->
             userspace, check nak list ->
               if on nak list, reset
               else send query to peers ->
                 if ack, send data to peer through gre
                 else add to nak list reset connection

It may be possible to do some trickery with conntrack to have the kernel send 
the naks (thinking out loud, if you set the conntrack state for the connection 
to fin, will it send the nak for you before the iptables new/established check?)

I don't like ideas that require any flags get set on packets that end up going 
to clients. Good firewalls block unknown flags (as we saw when ecn was 
introduced), so keeping all the tweaks local to the networks where the anycast 
servers are is a far better option.

David Lang


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From: "Scharf, Michael (Nokia - DE/Stuttgart)" <michael.scharf@nokia.com>
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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On 3/31/2017 8:02 AM, Scharf, Michael (Nokia - DE/Stuttgart) wrote:
> For what it is worth, I have doubts that in IoT an RTT measurement has to be performed *more* frequently than mandated by the TCP standards, which require a measurement roughly once per RTT. But I have no data for this.
>
> However, the discussion is whether there could be a reasonable use case for measuring the RTT *less* frequently. Examples for *less* would be to sample once-per-connection, once per minute, once per hour, etc. 

And once per N current round-trips, or even 1/N of a round trip (for
some N, e.g., 2 or 3).

> The issue is that the current version of the I-D rto-consider allows a lower frequency due to the use of the SHOULD. And this is different to my reading of RFC 793 and the standards-track specification of TCP, i.e., it is not consistent with the TCP standards. I admit that this is sort of the bureaucratic discussion but it matters specifically since we have wording on this in 793bis as well.
793 didn't have 2119 language at all.

It's up to us to interpret what are MAY/MUST/SHOULD from the language
there, even when the language has may/must/should.

I hear the argument that "793 says must, so it needs to be MUST", but
that seems to be a blind decision.

We ought to be making thoughtful ones, including MUSTs only where
necessary. It's clear that RTTs need to be measured "frequently enough",
but it also seems like requiring that to be done for EVERY packet
exchanged is both very high and rarely what a minimal TCP actually does
(esp. given TS support is a MAY anyway).

Joe


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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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Hi all,

just to clarify this point:

> Am 30.03.2017 um 06:41 schrieb Scharf, Michael (Nokia - DE/Stuttgart) =
<michael.scharf@nokia.com>:
>=20
> Since there has always been the =E2=80=9Cmust=E2=80=9D wording in RFC =
793 (and it has been a MUST in rto-consider in many revisions as well), =
I am not sure if we can downgrade this now to SHOULD, e.g., without =
experimental data.

rto-cosnider does not change/update rfc793. rfc793 is still valid and is =
simply giving a more constraint specification for one specific protocol =
which does not contradict each other. (The other way around would but =
that=E2=80=99s not the case here).

However, maybe it would be good to refer to rfc793 and rfc6298 in =
rto-consider and make the reader award that these docs make an even =
strong recommendation for TCP.

Also, given there was a discussion now if the MUST, or actually must, =
for rfc793 is still appropriate, this is a question for 793bis (and not =
the rto-cosnider draft).

Mirja



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From nobody Fri Mar 31 09:38:43 2017
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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> If it measures the RTT once (to reduce complexity) and if that
> sample happens to be way below the average RTT, it can run
> repeatedly into spurious timeouts, no? Then the exponential
> backoff and congestion control kicks in, but if there are many of
> these devices (sensors=E2=80=A6) on the medium, transmitting every message
> multiple times could be hazardous, no?

Harzardous?  No.  IMHO.

First, we have congestion control.  Nobody is suggesting we remove
that.  So, we will not collapse.

Second, if in fact the SHOULD is followed then I doubt the course
you suggest is possible because 2119 says that "the full
implications must be understood and carefully weighed before
choosing a different course".  I doubt there is an incentive for
implementers to create the situation you describe.

Third, let's be clear here ... we're saying an implementation SHOULD
take regular samples.  We are not saying "eh, if ya want" or "it is
not important to take regular samples" or any such thing.  In fact,
the draft explicitly calls out the case you are talking about and
notes that taking a single sample per connection "results in a
relatively poorly performing RTO mechanism".  So, to my mind we are
in fact warning people that they should be pretty careful if they
take this sort of approach.

Fourth, our old work on take-first doesn't suggest it is somehow
completely outrageous.

Fifth, the words in rto-consider say that one SHOULD not only
measure the RTT, but also the variance.  The variance cannot be
measured when we only have one sample, but if one initializes it
reasonably that helps the situation you describe.

There are many places in the tradeoff space that are suboptimal.
There are many ways to view "optimality".  And, there are many
disparate situations.  We know there is no one-size-fits-all.  So,
it seems the question boils down to whether this would in fact cause
a danger to the Internet.  We are a very long ways away from that
point here.=20=20

The way I look at this is that there are two MUSTs [*] in
rto-consider that I would argue are critical and have to be MUSTs.
Those are (a) exponential backoff and (b) taking a timeout as an
indication of congestion and invoking some standard CC (unless we
figure out the loss is not congestion using some standard mechanism,
e.g., F-RTO).  Nothing else rises anywhere close to the level of
importance of these two requirements, IMO.  These are the only two
things that if violated would make me wince and worry a bit.  This
is the general vibe I got from Wednesday's meeting, as well.  That
doesn't mean everything else is unimportant or has no effect or we
shouldn't care about it.  But, it means that if we give some wiggle
room and someone gets it wrong it will not be dangerous.

allman


[*] The other MUST in the document is that the initial RTO MUST be
    at least 1sec.  I think after (a) and (b) above this is probably
    the most important thing so that flows do not start off too
    aggressively and also so they at least get one good RTT sample.
    But, this also isn't all that close to (a) and (b) in my mind
    and I could see making this a SHOULD, as well.  This is all sort
    of shades of grey.  And, I go back and forth on what my druthers
    are for the requirement level of the initial RTO, but what I
    have no doubt about is that it is not nearly as important as the
    other two bits.




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From nobody Fri Mar 31 09:41:24 2017
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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Michael-

> The issue is that the current version of the I-D rto-consider
> allows a lower frequency due to the use of the SHOULD. And this is
> different to my reading of RFC 793 and the standards-track
> specification of TCP, i.e., it is not consistent with the TCP
> standards.

If it wasn't different we'd not need to write a new draft...

But, I completely disagree with your characterization of 793.  You
are torturing the words.  RFC 793 says:

  Because of the variability of the networks that compose an
  internetwork system and the wide range of uses of TCP connections
  the retransmission timeout must be dynamically determined.  One
  procedure for determining a retransmission time out is given here
  as an illustration.

First, the algorithm given is not **the** algorithm, but **an
example**.  It is called out as such.  (And, don't go and quote
793bis chapter and verse on me ... that is an effin' draft and I've
been around the block a few too many times to fall for that sort of
crapola.  We can make that say whatever is appropriate for it to
say.  And, that might be different than what it says at this exact
moment.)

Second, "must be dynamically determined" doesn't say what you claim
it says.  In particular it gives no time frame at all.  Is that
dynamic over the course of a single connection?  Over a day?  An
hour?  An RTT?  Once per connection could well fit in that we are
then not relying on a static timeout, but rather basing the timeout
on the conditions of the network path.  This is in fact dynamic.
So, you're just wrong in saying that 793 somehow says we "must"
sample at any particularly frequency.  Even the **example** given
doesn't say how often samples are taken.

Third, I agree that the wording in 6298 says TCP MUST use that
algorithm.  We can change that sentence.  I.e., to say the algorithm
is one possible algorithm that follows [rto-consider].  Again, the
goal is to **change** things, so this doesn't faze me one little
bit.  It's a sentence.  In a file.  It was not handed down as
etchings on a stone tablet.

> I admit that this is sort of the bureaucratic discussion but it
> matters specifically since we have wording on this in 793bis as
> well.

Yes- it is pedantic and bureaucratic.

I fully understand that documents like rto-consider are quite
unusual for the IETF and they run counter to our collective approach
and sensibilities, which generally favor crisp, formal, tight
specifications.  I have exactly those sensibilities, too.  And, in
fact, can remember writing RFC 2988 and applying some of those
sensibilities at the time when, e.g., Sally Floyd was pushing for a
little looser spec (although not nearly what rto-consider is).

However, I think a sign of a mature organization is the ability to
sometimes look around and decide that we have learned some things
From=20reality and can in fact let go of tight specs and crisp
formality where it is not required.  I think this in fact gives us
more credibility and makes, e.g., our MUSTs stronger because we are
not slapping them on everything in the name of conservativeness or
just-in-case.  I think a bunch of us think we're at that point with
the RTO.  I completely understand that you are not and completely
sympathize with your instincts and sensibilities.  But, I don't
think consensus is with you here.

allman




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From nobody Fri Mar 31 09:46:38 2017
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Subject: Re: [tcpm] RFC 793 (bis) vs. rto-consider
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On 3/31/2017 9:38 AM, Mark Allman wrote:
> The way I look at this is that there are two MUSTs [*] in
> rto-consider that I would argue are critical and have to be MUSTs.
> Those are (a) exponential backoff and (b) taking a timeout as an
> indication of congestion and invoking some standard CC (unless we
> figure out the loss is not congestion using some standard mechanism,
> e.g., F-RTO).  Nothing else rises anywhere close to the level of
> importance of these two requirements, IMO. 
+1


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Yuchung Cheng <ycheng@google.com>, Michael Welzl <michawe@ifi.uio.no>
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Thread-Topic: [tsvwg] TCP ACK-CC (re: Transport protocol feedback overhead )
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Subject: Re: [tcpm] [tsvwg] TCP ACK-CC (re: Transport protocol feedback overhead )
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