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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Reminder: the WG last call for rfc1323bis ends on Monday 13th.

Below is an initial draft write-up. Please let me know if you think it =
does not present things correctly. Note that the below write-up is only =
a draft, and can change until we request publication.

- Pasi



(1) What type of RFC is being requested (BCP, Proposed Standard,
Internet Standard, Informational, Experimental, or Historic)?  Why
is this the proper type of RFC?  Is this type of RFC indicated in the
title page header?

	Proposed Standard. This draft replaces earlier RFC 1323 that
	also was a Proposed Standard. The title page header currently
	says "Standards Track".

	(TBD: title page should be updated to indicate "Proposed
	Standard" as intended status)

(2) The IESG approval announcement includes a Document Announcement
Write-Up. Please provide such a Document Announcement Write-Up. Recent
examples can be found in the "Action" announcements for approved
documents. The approval announcement contains the following sections:

Technical Summary

	This document replaces RFC 1323, making minor fixes and
	clarifications to the original document. The document
	specifies three performance enhancing extensions to TCP, using
	two TCP options. The first is window scale option, that allows
	representation of receive window sizes of up to 2^30 bytes
	using the 16-bit window field. The second option is TCP
	timestamps option that allows round-trip time measurement on
	each TCP segment, and third is an algorithm to detect and
	reject old duplicate segments that happen to match the current
	TCP window (Protect Against Wrapped Sequence numbers).

Working Group Summary

	This document has been a chartered TCPM working group item
	since year 2008. It has not been under significant
	controversy, but the progress has been slow because of earlier
	lack of WG (and authoring) cycles. During the last 12 months,
	with the help of an additional co-editor, the progress on the
	draft became faster.

Document Quality

	The predecessor of this document, RFC 1323, was published in
	1992, and is deployed in most TCP implementations. This
	document includes fixes and clarifications based on the gained
	deployment experience. The recent versions of the document
	have been reviewed and discussed by multiple working group
	participants.

Personnel

	Document Shepherd is Pasi Sarolahti.
	Reponsible Area Director TBA

(3) Briefly describe the review of this document that was performed by
the Document Shepherd.  If this version of the document is not ready
for publication, please explain why the document is being forwarded to
the IESG.

	The document shepherd has read the latest version of the
	document and the recent mailing list discussion, and thinks
	the document is ready for publication.

(4) Does the document Shepherd have any concerns about the depth or
breadth of the reviews that have been performed?

	No. =20

(5) Do portions of the document need review from a particular or from
broader perspective, e.g., security, operational complexity, AAA, DNS,
DHCP, XML, or internationalization? If so, describe the review that
took place.

	No.

(6) Describe any specific concerns or issues that the Document Shepherd
has with this document that the Responsible Area Director and/or the
IESG should be aware of? For example, perhaps he or she is uncomfortable
with certain parts of the document, or has concerns whether there really
is a need for it. In any event, if the WG has discussed those issues and
has indicated that it still wishes to advance the document, detail those
concerns here.

	Document shepherd does not have concerns about the document

(7) Has each author confirmed that any and all appropriate IPR
disclosures required for full conformance with the provisions of BCP 78
and BCP 79 have already been filed. If not, explain why.

	To be confirmed.

(8) Has an IPR disclosure been filed that references this document?
If so, summarize any WG discussion and conclusion regarding the IPR
disclosures.

	No IPR disclosures have been filed on this document

(9) How solid is the WG consensus behind this document? Does it=20
represent the strong concurrence of a few individuals, with others
being silent, or does the WG as a whole understand and agree with it?  =20=


	The recent versions of the document were actively discussed on
	the TCPM mailing list, by several members of the community,
	for example regarding whether negotiating the use of
	timestamps should be allowed later in the TCP
	connection. While the current approach may not have been
	unanimously supported in the past discussions, the WG chairs
	believe the current version represents the rough consensus of
	the WG.

	There were also some opinions suggesting that some of the
	benefits could be achieved with less TCP option overhead. The
	WG chairs concluded that considering the original scope of
	the document (fixes and clarifications to RFC 1323), and its
	large existing deployment base, it is important to publish the
	current document, but keep the discussion open for future
	enhancements.

(10) Has anyone threatened an appeal or otherwise indicated extreme=20
discontent? If so, please summarise the areas of conflict in separate
email messages to the Responsible Area Director. (It should be in a
separate email because this questionnaire is publicly available.)=20

	No.

(11) Identify any ID nits the Document Shepherd has found in this
document. (See http://www.ietf.org/tools/idnits/ and the Internet-Drafts
Checklist). Boilerplate checks are not enough; this check needs to be
thorough.

	Few nits found -- TBD: authors to fix before submitting

(12) Describe how the document meets any required formal review
criteria, such as the MIB Doctor, media type, and URI type reviews.

	No formal review required.

(13) Have all references within this document been identified as
either normative or informative?

	Yes

(14) Are there normative references to documents that are not ready for
advancement or are otherwise in an unclear state? If such normative
references exist, what is the plan for their completion?

	No

(15) Are there downward normative references references (see RFC 3967)?
If so, list these downward references to support the Area Director in =
the
Last Call procedure.

	No

(16) Will publication of this document change the status of any
existing RFCs? Are those RFCs listed on the title page header, listed
in the abstract, and discussed in the introduction? If the RFCs are not
listed in the Abstract and Introduction, explain why, and point to the
part of the document where the relationship of this document to the
other RFCs is discussed. If this information is not in the document,
explain why the WG considers it unnecessary.

	No

(17) Describe the Document Shepherd's review of the IANA considerations
section, especially with regard to its consistency with the body of the
document. Confirm that all protocol extensions that the document makes
are associated with the appropriate reservations in IANA registries.
Confirm that any referenced IANA registries have been clearly
identified. Confirm that newly created IANA registries include a
detailed specification of the initial contents for the registry, that
allocations procedures for future registrations are defined, and a
reasonable name for the new registry has been suggested (see RFC 5226).

	There are no IANA considerations, even though the document
	specifies two TCP options, because the needed numbers have
	already been allocated when publishing RFC 1323.

(18) List any new IANA registries that require Expert Review for future
allocations. Provide any public guidance that the IESG would find
useful in selecting the IANA Experts for these new registries.

	No new IANA registeries needed.

(19) Describe reviews and automated checks performed by the Document
Shepherd to validate sections of the document written in a formal
language, such as XML code, BNF rules, MIB definitions, etc.

	No automated checks needed.


From wwwrun@rfc-editor.org  Thu May  9 13:44:03 2013
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Subject: [tcpm] RFC 6937 on Proportional Rate Reduction for TCP
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6937

        Title:      Proportional Rate Reduction for TCP 
        Author:     M. Mathis,
                    N. Dukkipati,
                    Y. Cheng
        Status:     Experimental
        Stream:     IETF
        Date:       May 2013
        Mailbox:    mattmathis@google.com, 
                    nanditad@google.com, 
                    ycheng@google.com
        Pages:      16
        Characters: 39437
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-tcpm-proportional-rate-reduction-04.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6937.txt

This document describes an experimental Proportional Rate Reduction
(PRR) algorithm as an alternative to the widely deployed Fast
Recovery and Rate-Halving algorithms.  These algorithms determine the
amount of data sent by TCP during loss recovery.  PRR minimizes
excess window adjustments, and the actual window size at the end of
recovery will be as close as possible to the ssthresh, as determined
by the congestion control algorithm.

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


EXPERIMENTAL: This memo defines an Experimental Protocol for the
Internet community.  It does not specify an Internet standard of any
kind. Discussion and suggestions for improvement are requested.
Distribution of this memo is unlimited.

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From mallman@icir.org  Fri May 10 08:37:06 2013
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I adamantly do not support 1323bis being published in its current
state.

  - My overall notion is that bis documents should take into account
    what has been learned since the initial publication.  Sometimes
    we learn the documents are unclear.  Sometimes we learn about
    refinements to techniques that can make things better in some
    fashion.  Sometimes we learn that additional mechanisms would be
    handy.  Sometimes we learn that what we previously wrote is just
    wrong.  We should take all this into account when we update
    documents.

  - 1323bis does not take into account much of what has been learned
    since the publication of 1323.  Sure, it fixes a few small
    things, but it just completely ignores the big things.  There
    are forests.  And, there are trees.  And, 1323bis is concerned
    with bark.

  - 1.1, item (1): This does not say *why* the window is a
    "fundamental performance problem".  I mean, I know it is, but it
    seems the document should at least sketch and cite some notion
    that the maximum window and the RTT constrain the performance.

  - 1.1, item (2): Citing 6675 is fine.  That is how to use SACKs.
    But, seemingly if you want to cite *SACK* you should also cite
    2018 and 2883 as these define SACK.

  - 1.1, item (3): This is not a "fundamental performance problem".
    This should be removed from the list (see below).

  - 1.3 states "it is not expected that most TCP implementations
    will properly handle unknown options".  Why do we have to frame
    this as an expectation when there is empirical evidence that
    this is the case?  Again, bis documents should be open to what
    we have *learned*.

    At least these two papers speak directly to the issue:

      Alberto Medina, Mark Allman, Sally Floyd.  Measuring
      Interactions Between Transport Protocols and Middleboxes.  ACM
      SIGCOMM/USENIX Internet Measurement Conference.  October 2004.

      Alberto Medina, Mark Allman, Sally Floyd.  Measuring the
      Evolution of Transport Protocols in the Internet.  ACM
      Computer Communication Review, 35(2), April 2005.

    There are probably others.  (It is not my intention to flog my
    papers, I just readily know of them.)

  - 1.3: "We recognize there is a tradeoff between the bandwidth
    saved by reducing unnecessary retransmission timeouts, and the
    extra header bandwidth used by this option.  It is required that
    this TCP option will be sent on non-<SYN> segments only after an
    exchange of options on the <SYN> segments has indicated that
    both sides understand this extension."

    Two issues:

      - First, you **never** establish that RTTM "reduc[es]
        unnecessary retransmission timeouts".  You do not cite
        anything that says that.  You do not present data that says
        that.  You are simply waving your hands and hoping that more
        RTT samples means a better RTO and that leads to fewer
        spurious retransmissions.

      - Second, I find the above ambiguous.  When I read it I
        wondered if this was wiggle room to let TCPs not send TS on
        all segments after the 3WHS.  Later in the document you
        unequivocally say that TS must be sent on all segments after
        negotiated.  I think you want "will be sent on ALL
        non-<SYN>" in the above to leave no doubt what you mean.

  - 2.1--2.3: The whole discussion of the WS option is a bit hard to
    follow.  

    The first sentence of the section says: "The window scale
    extension expands the definition of the TCP window to 32 bits
    and then uses a scale factor to carry this 32-bit value in the
    16-bit Window field of the TCP header".

    That is not true because at most the option dictates you can
    scale the window by at most 15 bits---or to an overall window
    size of 31 bits.

    As much is admitted later.

    And, then it is further refined such that the window must
    actually be *less* than 2^31 and so the max WS factor should be
    14 to yield a max window of 2^30.

    The discussion of why this is and all is fine and I have no
    problem with it.  But, don't say the window can be 32 bits or 31
    bits when it can't be.  It just leads to confusion.

  - 2.2: It is confusing that you introduce a "scale factor" that is
    different from the encoded "window scale".  I.e., a "scale
    factor" of 1 seemingly means the window size is not scaled at
    all.  But, this requires a "window scale" of zero.  The
    discussion could just be cleaned up to get rid of the "scale
    factor" notion.  Or, **at least** you could define the scale
    factor as one plus the window scale.

  - 2.2: "A Window Scale option in a segment without a SYN bit
    SHOULD be ignored."  Why is this?  Why not "MUST be ignored"?
    Is there some case where the TCP should really pay attention to
    it?  I can't think of one as things are presently defined.

  - In 2.3 you note (in a somewhat tortured way) that if the WS
    arrives as 15 then it should be assumed to be 14.  You might
    sketch why this is safe.  I.e., even if the other side is
    offering a window of 2^31 bytes you'll only use half that and so
    all will be fine.

  - 2.3: You should cite something for congestion avoidance and slow
    start.  I usually cite both Jac88 and the RFC.  But, either is
    fine.

  - 3.1: "Many TCP implementations base their RTT measurements upon
    a sample of one segment per window or less.  While this yields
    an adequate approximation to the RTT for small windows, it
    results in an unacceptably poor RTT estimate for a LFN."

    Do you have evidence of this?  We have evidence you're wrong:

      Mark Allman, Vern Paxson.  On Estimating End-to-End Network
      Path Properties.  Proceedings of the ACM SIGCOMM Technical
      Symposium, Cambridge, MA, September 1999.

    That shows that the number of samples per RTT is pretty
    immaterial to the effectiveness of the RTO.

  - 3.1: "If we look at RTT estimation as a signal processing
    problem (which it is), a data signal at some frequency, the
    packet rate, is being sampled at a lower frequency, the window
    rate.  This lower sampling frequency violates Nyquist's criteria
    and may therefore introduce "aliasing" artifacts into the
    estimated RTT [Hamming77]."

    This is hand waving.  At best.

    I buy that the RTT is a signal.  The mistake above is trying to
    tightly couple the frequency of that signal to the "packet
    rate".  If that was true then the conclusion that *any* RTT
    measurement strategy that relied only on TCP packets themselves
    would in fact violate Nyquist.  But, why should I think its
    true?  It might be true for one flow over a dumbbell where every
    packet directly and materially influences the RTT.  But, over an
    network with even a little statmux it quickly becomes clear that
    the frequency of the RTT process is not dependent on the packet
    rate of any particular source and so the above reasoning is just
    not sound.

    Or, to put it a different way: when the math and the world
    disagree, the world is right.  And the world says something
    different from what the document says:

      Mark Allman, Vern Paxson.  On Estimating End-to-End Network
      Path Properties.  Proceedings of the ACM SIGCOMM Technical
      Symposium, Cambridge, MA, September 1999.

  - 3.1: "RTT estimator".  Note, TCP does not have an "RTT
    estimator".  Scrub this from the document.  We have an "RTO
    estimator".  These are different things.  Confusing them is a
    fundamental mistake.

  - 3.1: "it becomes effectively impossible to obtain a valid RTT
    measurement".  This is FUD.  The RTO backoff [RFC6298] stays in
    place until you can take an unambiguous RTT sample.  So, this
    statement is not true unless you get to the point when you
    cannot get a single packet through the network per maximum RTO.
    And, in that case, RTTM isn't going to save you.  Please remove
    this notion.

  - 3.1: "It is vitally important to use the RTTM mechanism with big
    windows; otherwise, the door is opened to some dangerous
    instabilities due to aliasing."

    Show me.  Where is the evidence?  Cite something.  This document
    makes a whole lot of statements that seem to come from thin air.
    In contrast to when 1323 was published, we have studied this
    stuff for quite a long time now and we have a better idea how
    these things work.  So, you should be able to readily establish
    such statements or you should not make them.  (Or, **at least**
    explain how you think this "dangerous instability" is going to
    come to pass.)

    This is a whole additional level of ridiculous.  Its one thing
    to say you get a more accurate RTO with more RTT samples.  You'd
    be wrong, but at least all we're talking about is a little more
    or a little less time waiting on retransmissions.  But, to say
    this leads to *instability* is not right.  The backoff in the
    RTO backstops busted RTO (congestion) decisions.  Even if the
    RTO is quite wrong the backoff will kick in to prevent
    "dangerous instabilities".

  - End of 3.3 on RTT sample weighting factors.

    (1) The problem with the history being truncated when using RTTM
        was independently highlighted by Ludwig and Floyd.  We
        should at least have the common courtesy to cite Sally's
        note to e2e and Reiner's paper.

    (2) Instead of some nebulous reference to RTO algorithms you
        could point to the standard one, anyway.  I fully understand
        that some implementations don't use it, but at least as a
        concrete example of an algorithm that has this problem.

    (3) ARE YOU KIDDING ME?  

        This document goes through a tortured and wrong analysis
        telling me how "vitally important" RTTM is to address a
        "fundamental performance problem" of TCP over LFNs and how
        my performance is going to be "unacceptably poor" without
        it. 

        And, then we have this pesky problem that the history in the
        RTO depends on the window size when you actually do the
        thing you suggest we do (use RTTM).

        So, the first way to address this problem with the RTO is:

          "an implementation could choose to just use one sample per
          RTT to update the RTO estimator"

        Thanks for the whiplash!  Holy shit.  Really?! 

        Which one is it?  If I take this advice then aren't I in
        massive violation of Nyquist and causing dangerous
        instability to the network?

        This negates whole tracts of the damn draft.  Be wrong if
        you want, but at least have the decency to stay wrong.

  - Again, bis documents should reflect our understanding of the
    world.  RTTM does not *hurt* anything.  It just doesn't *help*
    anything either.  We should be honest enough to take this into
    account in our documents.

    RTTM should not be deprecated.  It should be a MAY.

    RTTM should not be discussed with breathless bullshit about hand
    wavy math and un-demonstrated stability issues and whatnot.  

    We should say that RTTM is absolutely within compliance of the
    spec and that it will not hurt your RTO.

    We should also say that RTTM is unlikely to help your RTO.

    We should leave it to implementers to decide if RTTM is useful
    for their purposes.

    We should specify a way to vary the gains in the standard RTO
    algorithm based on the current cwnd.

    And, we should absolutely state that there are other uses for
    the timestamp option (like Eifel, like PAWS) and there is
    nothing wrong with the *option* for that purpose.

  - 3.4, (A): Why are we discussing this in terms of the "Kth"
    segment?  Delayed ACKs per the standard is "2nd".  Why do we
    have to make the discussion in terms of some theory rather than
    in terms of what we have specified?

  - 3.2 insinuates that you should not include a timestamp on an
    RST: "TSopt MUST be sent in every non-<RST> segment", implying
    it should not be sent on an <RST> (or you'd have just said
    "every segment").  But, then 4.2 goes on to (rightly IMO)
    develop why we should include it on <RST> segments.  This
    inconsistency needs fixed.

Sorry ... there is just no way this is close to ready, IMO.

allman




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My feeling about the new bis is it just tries to change/add too many
things, like discussing the weighting factor in RTO formula,
middle-boxes, etc.

As a developer what I am looking for in a -bis is critical changes in
the protocol or implementations for well-identified problems. Like the
sequence check Dave identified. I don't know any existing real big
problem in the RTTM area other than some minutia corner cases. It's
always a MAY to do RTTM with TS to me. It's a why-not if regular
sample is not available due to Karn's check. In the end of the day, we
implement what makes sense on REAL networks, and listen to IETF if
they make good suggestions.

A clear and to the point -bis is more appreciated. If there is any
caveat or caution that's not intuitively clear, please back up with
scientific data.

So after these complaints, maybe we can start with a draft copy of the
minimal changes that address the absolutely critical problems?










On Fri, May 10, 2013 at 8:37 AM, Mark Allman <mallman@icir.org> wrote:
>
> I adamantly do not support 1323bis being published in its current
> state.
>
>   - My overall notion is that bis documents should take into account
>     what has been learned since the initial publication.  Sometimes
>     we learn the documents are unclear.  Sometimes we learn about
>     refinements to techniques that can make things better in some
>     fashion.  Sometimes we learn that additional mechanisms would be
>     handy.  Sometimes we learn that what we previously wrote is just
>     wrong.  We should take all this into account when we update
>     documents.
>
>   - 1323bis does not take into account much of what has been learned
>     since the publication of 1323.  Sure, it fixes a few small
>     things, but it just completely ignores the big things.  There
>     are forests.  And, there are trees.  And, 1323bis is concerned
>     with bark.
>
>   - 1.1, item (1): This does not say *why* the window is a
>     "fundamental performance problem".  I mean, I know it is, but it
>     seems the document should at least sketch and cite some notion
>     that the maximum window and the RTT constrain the performance.
>
>   - 1.1, item (2): Citing 6675 is fine.  That is how to use SACKs.
>     But, seemingly if you want to cite *SACK* you should also cite
>     2018 and 2883 as these define SACK.
>
>   - 1.1, item (3): This is not a "fundamental performance problem".
>     This should be removed from the list (see below).
>
>   - 1.3 states "it is not expected that most TCP implementations
>     will properly handle unknown options".  Why do we have to frame
>     this as an expectation when there is empirical evidence that
>     this is the case?  Again, bis documents should be open to what
>     we have *learned*.
>
>     At least these two papers speak directly to the issue:
>
>       Alberto Medina, Mark Allman, Sally Floyd.  Measuring
>       Interactions Between Transport Protocols and Middleboxes.  ACM
>       SIGCOMM/USENIX Internet Measurement Conference.  October 2004.
>
>       Alberto Medina, Mark Allman, Sally Floyd.  Measuring the
>       Evolution of Transport Protocols in the Internet.  ACM
>       Computer Communication Review, 35(2), April 2005.
>
>     There are probably others.  (It is not my intention to flog my
>     papers, I just readily know of them.)
>
>   - 1.3: "We recognize there is a tradeoff between the bandwidth
>     saved by reducing unnecessary retransmission timeouts, and the
>     extra header bandwidth used by this option.  It is required that
>     this TCP option will be sent on non-<SYN> segments only after an
>     exchange of options on the <SYN> segments has indicated that
>     both sides understand this extension."
>
>     Two issues:
>
>       - First, you **never** establish that RTTM "reduc[es]
>         unnecessary retransmission timeouts".  You do not cite
>         anything that says that.  You do not present data that says
>         that.  You are simply waving your hands and hoping that more
>         RTT samples means a better RTO and that leads to fewer
>         spurious retransmissions.
>
>       - Second, I find the above ambiguous.  When I read it I
>         wondered if this was wiggle room to let TCPs not send TS on
>         all segments after the 3WHS.  Later in the document you
>         unequivocally say that TS must be sent on all segments after
>         negotiated.  I think you want "will be sent on ALL
>         non-<SYN>" in the above to leave no doubt what you mean.
>
>   - 2.1--2.3: The whole discussion of the WS option is a bit hard to
>     follow.
>
>     The first sentence of the section says: "The window scale
>     extension expands the definition of the TCP window to 32 bits
>     and then uses a scale factor to carry this 32-bit value in the
>     16-bit Window field of the TCP header".
>
>     That is not true because at most the option dictates you can
>     scale the window by at most 15 bits---or to an overall window
>     size of 31 bits.
>
>     As much is admitted later.
>
>     And, then it is further refined such that the window must
>     actually be *less* than 2^31 and so the max WS factor should be
>     14 to yield a max window of 2^30.
>
>     The discussion of why this is and all is fine and I have no
>     problem with it.  But, don't say the window can be 32 bits or 31
>     bits when it can't be.  It just leads to confusion.
>
>   - 2.2: It is confusing that you introduce a "scale factor" that is
>     different from the encoded "window scale".  I.e., a "scale
>     factor" of 1 seemingly means the window size is not scaled at
>     all.  But, this requires a "window scale" of zero.  The
>     discussion could just be cleaned up to get rid of the "scale
>     factor" notion.  Or, **at least** you could define the scale
>     factor as one plus the window scale.
>
>   - 2.2: "A Window Scale option in a segment without a SYN bit
>     SHOULD be ignored."  Why is this?  Why not "MUST be ignored"?
>     Is there some case where the TCP should really pay attention to
>     it?  I can't think of one as things are presently defined.
>
>   - In 2.3 you note (in a somewhat tortured way) that if the WS
>     arrives as 15 then it should be assumed to be 14.  You might
>     sketch why this is safe.  I.e., even if the other side is
>     offering a window of 2^31 bytes you'll only use half that and so
>     all will be fine.
>
>   - 2.3: You should cite something for congestion avoidance and slow
>     start.  I usually cite both Jac88 and the RFC.  But, either is
>     fine.
>
>   - 3.1: "Many TCP implementations base their RTT measurements upon
>     a sample of one segment per window or less.  While this yields
>     an adequate approximation to the RTT for small windows, it
>     results in an unacceptably poor RTT estimate for a LFN."
>
>     Do you have evidence of this?  We have evidence you're wrong:
>
>       Mark Allman, Vern Paxson.  On Estimating End-to-End Network
>       Path Properties.  Proceedings of the ACM SIGCOMM Technical
>       Symposium, Cambridge, MA, September 1999.
>
>     That shows that the number of samples per RTT is pretty
>     immaterial to the effectiveness of the RTO.
>
>   - 3.1: "If we look at RTT estimation as a signal processing
>     problem (which it is), a data signal at some frequency, the
>     packet rate, is being sampled at a lower frequency, the window
>     rate.  This lower sampling frequency violates Nyquist's criteria
>     and may therefore introduce "aliasing" artifacts into the
>     estimated RTT [Hamming77]."
>
>     This is hand waving.  At best.
>
>     I buy that the RTT is a signal.  The mistake above is trying to
>     tightly couple the frequency of that signal to the "packet
>     rate".  If that was true then the conclusion that *any* RTT
>     measurement strategy that relied only on TCP packets themselves
>     would in fact violate Nyquist.  But, why should I think its
>     true?  It might be true for one flow over a dumbbell where every
>     packet directly and materially influences the RTT.  But, over an
>     network with even a little statmux it quickly becomes clear that
>     the frequency of the RTT process is not dependent on the packet
>     rate of any particular source and so the above reasoning is just
>     not sound.
>
>     Or, to put it a different way: when the math and the world
>     disagree, the world is right.  And the world says something
>     different from what the document says:
>
>       Mark Allman, Vern Paxson.  On Estimating End-to-End Network
>       Path Properties.  Proceedings of the ACM SIGCOMM Technical
>       Symposium, Cambridge, MA, September 1999.
>
>   - 3.1: "RTT estimator".  Note, TCP does not have an "RTT
>     estimator".  Scrub this from the document.  We have an "RTO
>     estimator".  These are different things.  Confusing them is a
>     fundamental mistake.
>
>   - 3.1: "it becomes effectively impossible to obtain a valid RTT
>     measurement".  This is FUD.  The RTO backoff [RFC6298] stays in
>     place until you can take an unambiguous RTT sample.  So, this
>     statement is not true unless you get to the point when you
>     cannot get a single packet through the network per maximum RTO.
>     And, in that case, RTTM isn't going to save you.  Please remove
>     this notion.
>
>   - 3.1: "It is vitally important to use the RTTM mechanism with big
>     windows; otherwise, the door is opened to some dangerous
>     instabilities due to aliasing."
>
>     Show me.  Where is the evidence?  Cite something.  This document
>     makes a whole lot of statements that seem to come from thin air.
>     In contrast to when 1323 was published, we have studied this
>     stuff for quite a long time now and we have a better idea how
>     these things work.  So, you should be able to readily establish
>     such statements or you should not make them.  (Or, **at least**
>     explain how you think this "dangerous instability" is going to
>     come to pass.)
>
>     This is a whole additional level of ridiculous.  Its one thing
>     to say you get a more accurate RTO with more RTT samples.  You'd
>     be wrong, but at least all we're talking about is a little more
>     or a little less time waiting on retransmissions.  But, to say
>     this leads to *instability* is not right.  The backoff in the
>     RTO backstops busted RTO (congestion) decisions.  Even if the
>     RTO is quite wrong the backoff will kick in to prevent
>     "dangerous instabilities".
>
>   - End of 3.3 on RTT sample weighting factors.
>
>     (1) The problem with the history being truncated when using RTTM
>         was independently highlighted by Ludwig and Floyd.  We
>         should at least have the common courtesy to cite Sally's
>         note to e2e and Reiner's paper.
>
>     (2) Instead of some nebulous reference to RTO algorithms you
>         could point to the standard one, anyway.  I fully understand
>         that some implementations don't use it, but at least as a
>         concrete example of an algorithm that has this problem.
>
>     (3) ARE YOU KIDDING ME?
>
>         This document goes through a tortured and wrong analysis
>         telling me how "vitally important" RTTM is to address a
>         "fundamental performance problem" of TCP over LFNs and how
>         my performance is going to be "unacceptably poor" without
>         it.
>
>         And, then we have this pesky problem that the history in the
>         RTO depends on the window size when you actually do the
>         thing you suggest we do (use RTTM).
>
>         So, the first way to address this problem with the RTO is:
>
>           "an implementation could choose to just use one sample per
>           RTT to update the RTO estimator"
>
>         Thanks for the whiplash!  Holy shit.  Really?!
>
>         Which one is it?  If I take this advice then aren't I in
>         massive violation of Nyquist and causing dangerous
>         instability to the network?
>
>         This negates whole tracts of the damn draft.  Be wrong if
>         you want, but at least have the decency to stay wrong.
>
>   - Again, bis documents should reflect our understanding of the
>     world.  RTTM does not *hurt* anything.  It just doesn't *help*
>     anything either.  We should be honest enough to take this into
>     account in our documents.
>
>     RTTM should not be deprecated.  It should be a MAY.
>
>     RTTM should not be discussed with breathless bullshit about hand
>     wavy math and un-demonstrated stability issues and whatnot.
>
>     We should say that RTTM is absolutely within compliance of the
>     spec and that it will not hurt your RTO.
>
>     We should also say that RTTM is unlikely to help your RTO.
>
>     We should leave it to implementers to decide if RTTM is useful
>     for their purposes.
>
>     We should specify a way to vary the gains in the standard RTO
>     algorithm based on the current cwnd.
>
>     And, we should absolutely state that there are other uses for
>     the timestamp option (like Eifel, like PAWS) and there is
>     nothing wrong with the *option* for that purpose.
>
>   - 3.4, (A): Why are we discussing this in terms of the "Kth"
>     segment?  Delayed ACKs per the standard is "2nd".  Why do we
>     have to make the discussion in terms of some theory rather than
>     in terms of what we have specified?
>
>   - 3.2 insinuates that you should not include a timestamp on an
>     RST: "TSopt MUST be sent in every non-<RST> segment", implying
>     it should not be sent on an <RST> (or you'd have just said
>     "every segment").  But, then 4.2 goes on to (rightly IMO)
>     develop why we should include it on <RST> segments.  This
>     inconsistency needs fixed.
>
> Sorry ... there is just no way this is close to ready, IMO.
>
> allman
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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My pushback ...

> As a developer what I am looking for in a -bis is critical changes in
> the protocol or implementations for well-identified problems. Like the
> sequence check Dave identified. I don't know any existing real big
> problem in the RTTM area other than some minutia corner cases. It's
> always a MAY to do RTTM with TS to me.

Of course *everything* is a MAY at the end of the day.  But, when the
document says:

   It is recommended that a modern TCP stack implements and make use of
   the extensions described in this document.

I think it is saying more than "eh, if ya want".  It is saying "we
really think there is a good reason to use these things [including
RTTM]".  And, that isn't so.  Or, at least it isn't demonstrated. 

If the document is staked around a bunch of hand waves and we have
actual data that speaks to the point we should use the data.

And, we should insist that AT LEAST the document be internally
consistent and not say "we had problem P and so we advocate solution X,
but X has a little problem Y and so to get around Y just don't do X".
Just say "we don't know how to soundly solve P" and leave it at that.

allman




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From touch@isi.edu  Fri May 10 10:37:26 2013
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I'm still digging through the entire doc, but I disagree on the 
following point:

On 5/10/2013 8:37 AM, Mark Allman wrote:
>    - 3.2 insinuates that you should not include a timestamp on an
>      RST: "TSopt MUST be sent in every non-<RST> segment", implying
>      it should not be sent on an <RST> (or you'd have just said
>      "every segment").

"MUST in non-RST" != "MUST NOT" or even "SHOULD NOT" in RST

At best, it might imply "NOT 'MUST' in RST".

The idea with RST is that you can include it or not, and there's nothing 
that tips the balance (i.e., no SHOULD).

 > But, then 4.2 goes on to (rightly IMO)
>      develop why we should include it on <RST> segments.  This
>      inconsistency needs fixed.

It might be arguable that:
	- senders SHOULD include TSopt in RST

but, in order to prevent reboots from leaving stale connections around:
	- receivers MUST accept RST with or without TSopt equivalently

Joe

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> On 5/10/2013 8:37 AM, Mark Allman wrote:
> >    - 3.2 insinuates that you should not include a timestamp on an
> >      RST: "TSopt MUST be sent in every non-<RST> segment", implying
> >      it should not be sent on an <RST> (or you'd have just said
> >      "every segment").
> 
> "MUST in non-RST" != "MUST NOT" or even "SHOULD NOT" in RST
> 
> At best, it might imply "NOT 'MUST' in RST".

That is why I used the words "insinuates" and "implying".  

Irrespective of whether timestamps ought to be included on RST packets,
the document should be clear on the point and consistent throughout.

allman




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From touch@isi.edu  Fri May 10 10:46:32 2013
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On 5/10/2013 10:41 AM, Mark Allman wrote:
>
>> On 5/10/2013 8:37 AM, Mark Allman wrote:
>>>     - 3.2 insinuates that you should not include a timestamp on an
>>>       RST: "TSopt MUST be sent in every non-<RST> segment", implying
>>>       it should not be sent on an <RST> (or you'd have just said
>>>       "every segment").
>>
>> "MUST in non-RST" != "MUST NOT" or even "SHOULD NOT" in RST
>>
>> At best, it might imply "NOT 'MUST' in RST".
>
> That is why I used the words "insinuates" and "implying".
>
> Irrespective of whether timestamps ought to be included on RST packets,
> the document should be clear on the point and consistent throughout.

Agreed; IMO, they cannot be required by the receiver. Doing so prevents 
cleanup or old state, and because this isn't a security mechanism, it is 
inappropriate to prevent that cleanup.

Joe

From nishida@sfc.wide.ad.jp  Fri May 10 11:03:47 2013
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Hello,

I've been thinking that we might want to have a "RTTM considertation" draft.
I've seen lots of valuable comments during 1323bis discussions. I'll
be disappointed if they are just put in the ML archives.

For me, it's fine to say It is recommended that a modern TCP stack
implements and make use of the extensions" in 1323bis.
Because AFAIK, it is the only standardized way to solve PAWS.

So, my personal preference is something like:
"we had problem P and so we advocate solution X, but we know it's not
a perfect solution. Please check Y for more detail"
This way, we don't have to put many stuff in the bis draft.

Thanks,
--
Yoshifumi

On Fri, May 10, 2013 at 9:59 AM, Mark Allman <mallman@icir.org> wrote:
>
> My pushback ...
>
>> As a developer what I am looking for in a -bis is critical changes in
>> the protocol or implementations for well-identified problems. Like the
>> sequence check Dave identified. I don't know any existing real big
>> problem in the RTTM area other than some minutia corner cases. It's
>> always a MAY to do RTTM with TS to me.
>
> Of course *everything* is a MAY at the end of the day.  But, when the
> document says:
>
>    It is recommended that a modern TCP stack implements and make use of
>    the extensions described in this document.
>
> I think it is saying more than "eh, if ya want".  It is saying "we
> really think there is a good reason to use these things [including
> RTTM]".  And, that isn't so.  Or, at least it isn't demonstrated.
>
> If the document is staked around a bunch of hand waves and we have
> actual data that speaks to the point we should use the data.
>
> And, we should insist that AT LEAST the document be internally
> consistent and not say "we had problem P and so we advocate solution X,
> but X has a little problem Y and so to get around Y just don't do X".
> Just say "we don't know how to soundly solve P" and leave it at that.
>
> allman
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From mallman@icir.org  Fri May 10 11:27:41 2013
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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I think you're mashing things together that should not be mashed
together. 

> For me, it's fine to say It is recommended that a modern TCP stack
> implements and make use of the extensions" in 1323bis.  Because AFAIK,
> it is the only standardized way to solve PAWS.

This doesn't exactly parse for me.  But, don't confuse timestamps and
RTTM.  Small windows (without WS) are certainly a problem.  WS solves
that.  We know that we need some sort of PAWS when using larger windows
and we have a perfectly reasonable PAWS that depends on the timestamp
option.  That is all well and fine.

But, none of that says we need recommend RTTM---which is using the
timestamp option for a whole different reason.

And, it might even be OK with me to continue to recommend RTTM if that
was even remotely well argued and took into account the things we have
learned since the publication of 1323.  But, 1323bis just parrots out
what 1323 says without any indication that we as a community have worked
on these issues in any way in the intervening 20 years.  That is
ridiculous. 

allman




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From rs@netapp.com  Mon May 13 08:04:52 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>, Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Date: Mon, 13 May 2013 15:04:45 +0000
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Hi Mark,


>   - My overall notion is that bis documents should take into account
>     what has been learned since the initial publication.  Sometimes
>     we learn the documents are unclear.  Sometimes we learn about
>     refinements to techniques that can make things better in some
>     fashion.  Sometimes we learn that additional mechanisms would be
>     handy.  Sometimes we learn that what we previously wrote is just
>     wrong.  We should take all this into account when we update
>     documents.

Agreed.

=20
>   - 1323bis does not take into account much of what has been learned
>     since the publication of 1323.  Sure, it fixes a few small
>     things, but it just completely ignores the big things.=20

The bulk of the text was indeed carried over, without updating,=20
from RFC1323 (and the stalled attempts to publish updates). I've=20
started to edit these text sections to be more=20
in line with actual findings since the conception of the text.
=20

>     There are forests.  And, there are trees.  And, 1323bis is=20
>     concerned with bark.

To remain in that metaphor, a tree cann't live without bark, thus bark=20
is an important aspect to a forest as well :)

>=20
>   - 1.1, item (1): This does not say *why* the window is a
>     "fundamental performance problem".  I mean, I know it is, but it
>     seems the document should at least sketch and cite some notion
>     that the maximum window and the RTT constrain the performance.

Adressed.


>   - 1.1, item (2): Citing 6675 is fine.  That is how to use SACKs.
>     But, seemingly if you want to cite *SACK* you should also cite
>     2018 and 2883 as these define SACK.

Added.

=20
>   - 1.1, item (3): This is not a "fundamental performance problem".
>     This should be removed from the list (see below).

Done; Added a reference to the timestamp option at the mentioning of PAWS i=
n the reliability section.

>   - 1.3 states "it is not expected that most TCP implementations
>     will properly handle unknown options".  Why do we have to frame
>     this as an expectation when there is empirical evidence that
>     this is the case?  Again, bis documents should be open to what
>     we have *learned*.
>=20
>     At least these two papers speak directly to the issue:
>=20
>       Alberto Medina, Mark Allman, Sally Floyd.  Measuring
>       Interactions Between Transport Protocols and Middleboxes.  ACM
>       SIGCOMM/USENIX Internet Measurement Conference.  October 2004.
>=20
>       Alberto Medina, Mark Allman, Sally Floyd.  Measuring the
>       Evolution of Transport Protocols in the Internet.  ACM
>       Computer Communication Review, 35(2), April 2005.
>=20
>     There are probably others.  (It is not my intention to flog my
>     papers, I just readily know of them.)

Fixed the text.


>   - 1.3: "We recognize there is a tradeoff between the bandwidth
>     saved by reducing unnecessary retransmission timeouts, and the
>     extra header bandwidth used by this option.  It is required that
>     this TCP option will be sent on non-<SYN> segments only after an
>     exchange of options on the <SYN> segments has indicated that
>     both sides understand this extension."
>=20
>     Two issues:
>=20
>       - First, you **never** establish that RTTM "reduc[es]
>         unnecessary retransmission timeouts".  You do not cite
>         anything that says that.  You do not present data that says
>         that.  You are simply waving your hands and hoping that more
>         RTT samples means a better RTO and that leads to fewer
>         spurious retransmissions.

Updated the text to reflect the findings since it's first inception.


>       - Second, I find the above ambiguous.  When I read it I
>         wondered if this was wiggle room to let TCPs not send TS on
>         all segments after the 3WHS.  Later in the document you
>         unequivocally say that TS must be sent on all segments after
>         negotiated.  I think you want "will be sent on ALL
>         non-<SYN>" in the above to leave no doubt what you mean.

Fixed.

=20
>   - 2.1--2.3: The whole discussion of the WS option is a bit hard to
>     follow.
>=20
>     The first sentence of the section says: "The window scale
>     extension expands the definition of the TCP window to 32 bits
>     and then uses a scale factor to carry this 32-bit value in the
>     16-bit Window field of the TCP header".
>=20
>     That is not true because at most the option dictates you can
>     scale the window by at most 15 bits---or to an overall window
>     size of 31 bits.
>=20
>     As much is admitted later.
>=20
>     And, then it is further refined such that the window must
>     actually be *less* than 2^31 and so the max WS factor should be
>     14 to yield a max window of 2^30.
>=20
>     The discussion of why this is and all is fine and I have no
>     problem with it.  But, don't say the window can be 32 bits or 31
>     bits when it can't be.  It just leads to confusion.

Updated all these references to 32/31 bits to 30 bits... (with the exceptio=
n of the implementation hint, where a 32 bit value is still mentioned).


=20
>   - 2.2: It is confusing that you introduce a "scale factor" that is
>     different from the encoded "window scale".  I.e., a "scale
>     factor" of 1 seemingly means the window size is not scaled at
>     all.  But, this requires a "window scale" of zero.  The
>     discussion could just be cleaned up to get rid of the "scale
>     factor" notion.  Or, **at least** you could define the scale
>     factor as one plus the window scale.

Well, that apparently hasn't led to wrong implementations; the window scale=
 is the (integer) exponent, base-two, of the scale factor; and the scale fa=
ctor is limited to powers of two (1, 2, 4, 8, .. 2^14). As a non-native spe=
aker, I'm struggling re-writing this text only with referenced to "scale ex=
ponets", as they are much less natural to speak about than factors. But the=
 factor simply isn't signaled...

Unless there is additional feedback, ideally with a suggestion for an updat=
ed text segment, I'd retain this part...


=20
>   - 2.2: "A Window Scale option in a segment without a SYN bit
>     SHOULD be ignored."  Why is this?  Why not "MUST be ignored"?
>     Is there some case where the TCP should really pay attention to
>     it?  I can't think of one as things are presently defined.

Fixed.

=20
>   - In 2.3 you note (in a somewhat tortured way) that if the WS
>     arrives as 15 then it should be assumed to be 14.  You might
>     sketch why this is safe.  I.e., even if the other side is
>     offering a window of 2^31 bytes you'll only use half that and so
>     all will be fine.

I don't know how to remove the tortured way of describing things; I added "=
This extends the effective bandwidth * delay product, over which TCP can op=
erate effectively, to 1 GiB. Paths with even higher bandwidths * delay prod=
ucts will not run at maximum possible performance using this mechanism." I =
hope that clarifies what to expect when one want'S to signal a TCP receive =
window > 1GiB.


=20
>   - 2.3: You should cite something for congestion avoidance and slow
>     start.  I usually cite both Jac88 and the RFC.  But, either is
>     fine.

Done.


Not going into Section 3 for now.


> allman

Thanks for the valuable feedback!

Incorporating your valid critic of RTTM and re-wording large sections of te=
xt will take longer than sections 1+2.

Best regards,
   Richard

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Just one quick clarification ...

> >   - In 2.3 you note (in a somewhat tortured way) that if the WS
> >     arrives as 15 then it should be assumed to be 14.  You might
> >     sketch why this is safe.  I.e., even if the other side is
> >     offering a window of 2^31 bytes you'll only use half that and so
> >     all will be fine.
> 
> I don't know how to remove the tortured way of describing things; 

The text is in terms of WS values > 14.  Just say "15" because that is
the only such value.  There are just more words that lead to more
thinking than are necessary here.

> I added "This extends the effective bandwidth * delay product, over
> which TCP can operate effectively, to 1 GiB. Paths with even higher
> bandwidths * delay products will not run at maximum possible
> performance using this mechanism." I hope that clarifies what to
> expect when one want'S to signal a TCP receive window > 1GiB.

I think I was unclear or you missed the point because that does not
address my comment.

The text says, effectively, if you receive a window scale of 15 then you
should instead use a window scale of 14 (because you previously worked
out that window scales of 15 are problematic).  I am saying that you
should note that this silent downgrading from 15 to 14 is safe even if
the endpoints are a little out-of-sync.  I.e., if the receiver thinks
the window is 2^31 and the sender thinks it is 2^30 then all is well
because half the buffer will just never be used because the sender will
not exceed a window of 2^30 bytes.  It took me a minute to work this out
and figure out that this silent reduction is OK and so it just seems
like a nice place to throw a sentence to say that this is indeed fine.

allman




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From john@jlc.net  Mon May 13 17:21:41 2013
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Mark Allman <mallman@icir.org> wrote:
> 
> The text says, effectively, if you receive a window scale of 15 then you
> should instead use a window scale of 14 (because you previously worked
> out that window scales of 15 are problematic).

   I am fine with forbidding the use of scale 15.

   I am uncomfortable with silently changing it to 14.

> I am saying that you should note that this silent downgrading from
> 15 to 14 is safe even if the endpoints are a little out-of-sync.

   I don't understand why we need to allow the silent change to 14;
but if we do so, we really should give an argument about why it is
safe!

--
John Leslie <john@jlc.net>

From mallman@icir.org  Mon May 13 18:23:26 2013
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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> Mark Allman <mallman@icir.org> wrote:
> >=20
> > The text says, effectively, if you receive a window scale of 15 then you
> > should instead use a window scale of 14 (because you previously worked
> > out that window scales of 15 are problematic).
>=20
>    I am fine with forbidding the use of scale 15.
>=20
>    I am uncomfortable with silently changing it to 14.
>=20
> > I am saying that you should note that this silent downgrading from
> > 15 to 14 is safe even if the endpoints are a little out-of-sync.
>=20
>    I don't understand why we need to allow the silent change to 14;
> but if we do so, we really should give an argument about why it is
> safe!

John-

It doesn't hurt anything.  If you tell me you have 2^31 bytes of buffer
and I only ever use 2^30 bytes of buffer then all is well.  Even without
WS you could give me a window of X and I could use X/2 for whatever
reason.=20

I mean, I suppose you could make the case that if someone does something
out of spec like advertise 15 then maybe it comes from some bug and
maybe there isn't near that much buffer because the whole thing comes
From=20being screwed up.  Well, OK.  I mean it doesn't seem that this has
caused big issues to date (20 years down the line) and so I would think
with an explanation of it is fine.

allman




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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>, John Leslie <john@jlc.net>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis 
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Hi John,

The only potential bad effect would be a reduction in bandwidth, in the unl=
ikely case that the delay * bandwidth product really exceeds 1 GiB (laser l=
ink to a moon station, 100 Gbit/s transatlantic link)?

So, how about a sentence like=20

     "This is safe as a sender can always
      choose to use any signaled receive window only partially."



Richard Scheffenegger


> -----Original Message-----
> From: mallman@icir.org [mailto:mallman@icir.org]
> Sent: Dienstag, 14. Mai 2013 03:23
> To: John Leslie
> Cc: Scheffenegger, Richard; tcpm (tcpm@ietf.org)
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
>=20
>=20
> > Mark Allman <mallman@icir.org> wrote:
> > >
> > > The text says, effectively, if you receive a window scale of 15 then
> > > you should instead use a window scale of 14 (because you previously
> > > worked out that window scales of 15 are problematic).
> >
> >    I am fine with forbidding the use of scale 15.
> >
> >    I am uncomfortable with silently changing it to 14.
> >
> > > I am saying that you should note that this silent downgrading from
> > > 15 to 14 is safe even if the endpoints are a little out-of-sync.
> >
> >    I don't understand why we need to allow the silent change to 14;
> > but if we do so, we really should give an argument about why it is
> > safe!
>=20
> John-
>=20
> It doesn't hurt anything.  If you tell me you have 2^31 bytes of buffer
> and I only ever use 2^30 bytes of buffer then all is well.  Even without
> WS you could give me a window of X and I could use X/2 for whatever
> reason.
>=20
> I mean, I suppose you could make the case that if someone does something
> out of spec like advertise 15 then maybe it comes from some bug and maybe
> there isn't near that much buffer because the whole thing comes From bein=
g
> screwed up.  Well, OK.  I mean it doesn't seem that this has caused big
> issues to date (20 years down the line) and so I would think with an
> explanation of it is fine.
>=20
> allman
>=20
>=20


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>, Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Hi Mark,

>   - 3.1: "Many TCP implementations base their RTT measurements upon
>     a sample of one segment per window or less.  While this yields
>     an adequate approximation to the RTT for small windows, it
>     results in an unacceptably poor RTT estimate for a LFN."
>=20
>     Do you have evidence of this?  We have evidence you're wrong:
>=20
>       Mark Allman, Vern Paxson.  On Estimating End-to-End Network
>       Path Properties.  Proceedings of the ACM SIGCOMM Technical
>       Symposium, Cambridge, MA, September 1999.
>=20
>     That shows that the number of samples per RTT is pretty
>     immaterial to the effectiveness of the RTO.

I'd like to point out, that the whole section 3 does not make a clear disti=
nction between *timely measuring* RTT, and using that measured RTT for furt=
her purposes, such as arriving at a useful value for RTO.

We can also argue about the specific wording in the 2nd sentence; But it ap=
pears to me, that you read "poor RTT estimate =3D=3D poor RTO estimate"?

>   - 3.1: "RTT estimator".  Note, TCP does not have an "RTT
>     estimator".  Scrub this from the document.  We have an "RTO
>     estimator".  These are different things.  Confusing them is a
>     fundamental mistake.

Correct. And IMHO 1323 is looking at RTT, but, agreed, only with the intend=
ed purpose of refining RTO (where these improvements haven't been as benefi=
cial as originally envisioned).

I rewrote the section 3.1 entirely:

3.  TCP Timestamp Option

3.1.  Introduction

   TCP measures the round trip time (RTT), primarily for the purpose of
   arriving at a decent value for the Retransmission Timeout (RTO) timer
   interval.  Accurate and current RTT estimates are necessary to adapt
   to changing traffic conditions, while a conservative estimate of the
   RTO inveral is necessary to minimize spurious RTOs.

   When [RFC1323] was originally written, it was perceived that more
   timely and accurate RTT measurements would contribute to reducing
   spurious RTOs, while maintaining their timeliness.  At the time, RTO
   was also the only mechanism to make use of the measured RTT.  It has
   been shown, that taking more RTT samples has only a very limited
   effect to optimize RTOs [Allman99].

   This document makes a clear distinction between the round trip time
   measurement (RTTM) mechanism, and subsequent mechanisms using the RTT
   signal as input, such as RTO (see Section 3.4).

   It is important to use the timestamp option with big windows, to
   allow the use of the PAWS mechanism (see Section 4).  Furthermore,
   the option is useful for all TCP's, since it simplifies the sender
   and allows the use of additional optimizations such as Eifel
   ([RFC3522], [RFC4015]) and others.

3.2.  Timestamp Option

And made the last paragraph of 3.3 a dedicated section, for the separation =
between RTTM and RTO:

3.4.  Updating the RTO value

   [KL04] has highlighted the problem that an unmodified RTO
   calculation, which is updated with per-packet RTT samples, will
   truncate the path history too soon.  This can lead to an increase in
   spurious retransmissions, when the path properties vary in the order
   of a few RTTs, but a high number of RTT samples are taken on a much
   shorter timescale.

   Implementers should note that with timestamps multiple RTTMs can be
   taken per RTT.  The [RFC6298] RTO estimator has weighting factors,
   alpha and beta, based on an implicit assumption that at most one RTTM
   will be sampled per RTT.  When using multiple RTTMs per RTT to update
   the RTO estimator, the weighting factor SHOULD be decreased to take
   into account the more frequent RTTMs.

   For example, an implementation could choose to

   o  just use one sample per RTT to update the RTO estimator, or

   o  vary the gain based on the congestion window, or

   o  take an average of all the RTT measurements (and the maximum of
      the variance) received over one RTT,

   and then use that value to update the RTO estimator.  This document
   does not prescribe any particular method for modifying the RTO
   estimator.







>   - End of 3.3 on RTT sample weighting factors.
>=20
>     (1) The problem with the history being truncated when using RTTM
>         was independently highlighted by Ludwig and Floyd.  We
>         should at least have the common courtesy to cite Sally's
>         note to e2e and Reiner's paper.

I tried to find Sally's comment (also trying to find the reference given in=
 your "Using Spurious Retransmissios to Adapt the Retransmission Timeout" p=
aper), but was unsuccessful... Do you have a link?



>   - 3.4, (A): Why are we discussing this in terms of the "Kth"
>     segment?  Delayed ACKs per the standard is "2nd".  Why do we
>     have to make the discussion in terms of some theory rather than
>     in terms of what we have specified?

fixed

>   - 3.2 insinuates that you should not include a timestamp on an
>     RST: "TSopt MUST be sent in every non-<RST> segment", implying
>     it should not be sent on an <RST> (or you'd have just said
>     "every segment").  But, then 4.2 goes on to (rightly IMO)
>     develop why we should include it on <RST> segments.  This
>     inconsistency needs fixed.


fixed

>     RTTM should not be deprecated.  It should be a MAY.
>=20
>     RTTM should not be discussed with breathless bullshit about hand
>     wavy math and un-demonstrated stability issues and whatnot.
>=20
>     We should say that RTTM is absolutely within compliance of the
>     spec and that it will not hurt your RTO.
>=20
>     We should also say that RTTM is unlikely to help your RTO.
>=20
>     We should leave it to implementers to decide if RTTM is useful
>     for their purposes.
>=20
>     We should specify a way to vary the gains in the standard RTO
>     algorithm based on the current cwnd.
>=20
>     And, we should absolutely state that there are other uses for
>     the timestamp option (like Eifel, like PAWS) and there is
>     nothing wrong with the *option* for that purpose.

Best regards,
   Richard


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From: David Borman <David.Borman@quantum.com>
To: "Scheffenegger, Richard" <rs@netapp.com>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On May 13, 2013, at 10:04 AM, "Scheffenegger, Richard" <rs@netapp.com> wrot=
e:

>>  - 2.1--2.3: The whole discussion of the WS option is a bit hard to
>>    follow.
>>=20
>>    The first sentence of the section says: "The window scale
>>    extension expands the definition of the TCP window to 32 bits
>>    and then uses a scale factor to carry this 32-bit value in the
>>    16-bit Window field of the TCP header".
>>=20
>>    That is not true because at most the option dictates you can
>>    scale the window by at most 15 bits---or to an overall window
>>    size of 31 bits.
>>=20
>>    As much is admitted later.
>>=20
>>    And, then it is further refined such that the window must
>>    actually be *less* than 2^31 and so the max WS factor should be
>>    14 to yield a max window of 2^30.
>>=20
>>    The discussion of why this is and all is fine and I have no
>>    problem with it.  But, don't say the window can be 32 bits or 31
>>    bits when it can't be.  It just leads to confusion.
>=20
> Updated all these references to 32/31 bits to 30 bits... (with the except=
ion of the implementation hint, where a 32 bit value is still mentioned).

I'd like to see what you changed to 30 bits.  Section 2.1, ok.  Section 2.3,
first bullet, ok.  I didn't see anything else that should be changed to 30 =
bits,
did I miss something?

>=20
>=20
>=20
>>  - 2.2: It is confusing that you introduce a "scale factor" that is
>>    different from the encoded "window scale".  I.e., a "scale
>>    factor" of 1 seemingly means the window size is not scaled at
>>    all.  But, this requires a "window scale" of zero.  The

No, a scale factor of 1 means a shift.cnt of zero.  In general,
a scale factor of 2^N means a shift.cnt of N.  Now that I'm thinking
about this distinction between scale factor vs. shift count:

   Please change Snd.Wind.Scale and Rcv.Wind.Scale to Snd.Wind.Shift
   and Rcv.Wind.Shift throughout the document, as these variables
   contain the shift count, not the scale factor.
=09

>>    discussion could just be cleaned up to get rid of the "scale
>>    factor" notion.  Or, **at least** you could define the scale
>>    factor as one plus the window scale.

That's not right, the scale factor is 2^N, not 1+N.  We should make
this distinction at first mention in section 2.1, rewrite it as:

   The window scale extension expands the definition of the TCP window
   to 32 bits and then uses a scale factor to carry this 32-bit value in
   the 16-bit Window field of the TCP header (SEG.WND in RFC 793).  The
   scale factor, encoded as a shift count, is carried in a TCP option,
   Window Scale.  This option is sent only in a <SYN> segment (a segment
   with the SYN bit on), hence the window scale is fixed in each direction
   when a connection is opened.

The change is the insertion of ", encoded as a shift count,"


>=20
> Well, that apparently hasn't led to wrong implementations; the window sca=
le is the (integer) exponent, base-two, of the scale factor; and the scale =
factor is limited to powers of two (1, 2, 4, 8, .. 2^14). As a non-native s=
peaker, I'm struggling re-writing this text only with referenced to "scale =
exponets", as they are much less natural to speak about than factors. But t=
he factor simply isn't signaled...
>=20
> Unless there is additional feedback, ideally with a suggestion for an upd=
ated text segment, I'd retain this part=85

I've looked through the 1323bis-11 to see if there are places where it
uses just "window scale", but aside from Rcv.Wind.Scale and Snd.Wind.Scale,
it seems to be pretty consistent using the the term "scale factor".
Changing Wind.Scale to Wind.Shift should clean things up.

...
>> - In 2.3 you note (in a somewhat tortured way) that if the WS
>>    arrives as 15 then it should be assumed to be 14.  You might
>>    sketch why this is safe.  I.e., even if the other side is
>>    offering a window of 2^31 bytes you'll only use half that and so
>>    all will be fine.
>=20
> I don't know how to remove the tortured way of describing things; I added=
 "This extends the effective bandwidth * delay product, over which TCP can =
operate effectively, to 1 GiB. Paths with even higher bandwidths * delay pr=
oducts will not run at maximum possible performance using this mechanism." =
I hope that clarifies what to expect when one want'S to signal a TCP receiv=
e window > 1GiB.

I think the current text is fine.  Keep in mind that the
clamping of the WS value to 14 applies not only to 15, but also to
16, 17 ... 255.  For any value > 14, log the error but continue the
connection using WS=3D14.  What else is there to do?  Drop the connection?
Generate an ICMP Parameter Problem message?  The goal of clamping the
WS to 14 is to allow the connection to continue, even if in a crippled
state.

My other comment is that the WS is *not* the advertised window, just
the scaling of the advertised window.  If they are only advertising
a window of 32K (WS=3D15, window=3D1), the other side will calculate the
window as 16K (WS=3D14, window=3D1).  So for small window values it could
cut the throughput in half.

But the reality, as has been noted, is that WS>14 has not been a
problem for the past 20 years, so implementors have figured it out,
and I see no reason to change from the 1323 clamping of all WS>14.
The text is there to cover the bases from a standards definition
point, as opposed to leaving the behavior undefined and up to the
implementor to decide how to handle WS>14.

At most, a one-line comment after the line that clamps the WS to 14:

   This should allow TCP connections to continue, even if
   operating in a degraded state due to the incorrect WS value.

But I'm not convinced we need that, this hasn't been an area of
problems or confusion over the past 20 years, so perhaps its best
to just leave it alone without trying to explain it.

			-David Borman

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To: Richard Scheffenegger <rs@netapp.com>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Richard,

Some minor cleanup for Appendix F:

> Appendix F.  Window Retraction Example
>=20
>   Consider a established TCP connection with WSCALE=3D7 (128 byte
>   receiver window quantization), that is running with a very small

Along with my previous message of changing to Wind.Scale to Wind.Shift
throughout the document, change the previous 2 lines to:

Consider an established TCP connection using a scale factor of
128 bytes, Snd.Wind.Shift=3D7 and Rcv.Wind.Shift=3D7, that is running
with very small

>   windows because the receiver is bottlenecked and both ends are doing
>   small reads and writes.
>=20
>   Consider the ACKs coming back:
>=20
>   SEG.ACK  SEG.WIN computed SND.WIN   receiver's actual window
>   1000     2       1256               1300
>=20
>   The sender writes 40 bytes and receiver ACKs:
>=20
>   1040     2       1296               1300
>=20
>   The sender writes 5 additional bytes and the receiver has a problem.
>   Two choices:
>=20
>   1045     2       1301               1300   - BEYOND BUFFER
>=20
>   1045     1       1173               1300   - RETRACTED WINDOW
>=20
>   This problems is completely general and can in principle happen any
>   time the sender does a write which is smaller than the window scale
>   quanta.

Change the previous 3 lines to:

This is a general problem and can happen any time the sender
does a write which is smaller than the scale factor.

>=20
>   In most stacks it is at least partially obscured when the window size
>   is larger than some small number of segments because the stacks
>   prefer to announce windows that are integral numbers of segments
>   (rounded up to the next window quanta).  This plus silly window

Change the previous 2 lines to:

 prefer to announce windows that are an integral numbers of segments,
 rounded to the next scale factor.  This plus silly window

>   suppression tends to cause less frequent, larger window updates.  If
>   the window was rounded down to a segment size there is more
>   opportunity to advance it ("beyond buffer" case above) rather than

>   retracting it.

Change the previous 2 lines to:

 opportunity to advance the window, the BEYOND BUFFER case above, rather
 than retracting the window.

			-David Borman


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From mallman@icir.org  Tue May 14 07:33:19 2013
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> I'd like to point out, that the whole section 3 does not make a clear
> distinction between *timely measuring* RTT, and using that measured
> RTT for further purposes, such as arriving at a useful value for RTO. 

Correct.  1323 and 1323bis are both very sloppy in this regard.

But, standard TCP only uses RTT samples for one thing: computing an
RTO.  And, there are enough words about the RTO in 1323bis that it is
clear that the intent of RTTM is to feed a whole lot more RTT samples
into the RTO estimator.  So, trying to somehow say RTTM isn't about the
RTO is at best splitting hairs. 

> We can also argue about the specific wording in the 2nd sentence; But
> it appears to me, that you read "poor RTT estimate == poor RTO
> estimate"? 

Yes.  See above.  There is no "RTT estimate" in TCP.  I think this
equivalence is the clear intent of the document.

(Not that it shouldn't be cleaned up ... it should be ...)

Also, note, that nothing I have said precludes someone from needing many
RTT samples per RTT for something else.  E.g., it may well be needed for
delay-based congestion control.  And, if so, the timestamp option can
provide a steady stream of RTT samples.  But, it seems clear to me that
a steady stream of RTT samples is not what the aim of RTTM in 1323bis.

> I rewrote the section 3.1 entirely:

I am not going to try to piece together what you have done from email.
This is not a small change.  When you're ready, issue another draft and
we can all take a look.

> >   - End of 3.3 on RTT sample weighting factors.
> > 
> >     (1) The problem with the history being truncated when using RTTM
> >         was independently highlighted by Ludwig and Floyd.  We
> >         should at least have the common courtesy to cite Sally's
> >         note to e2e and Reiner's paper.
> 
> I tried to find Sally's comment (also trying to find the reference
> given in your "Using Spurious Retransmissios to Adapt the
> Retransmission Timeout" paper), but was unsuccessful... Do you have a
> link? 

I figured that'd be easy to track down, but I couldn't find it quickly.
I was wrong, it was not from end2end it was from the tcp-lw list.  But,
even with that bit of information I can't find it.  I guess you can just
forget that Sally worked this out since the Internet has! :)

allman




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> No, a scale factor of 1 means a shift.cnt of zero.  In general,
> a scale factor of 2^N means a shift.cnt of N.  

Yeah- OK.  I am not good at math.  But, the point stands: make the
distinction concrete in the text.  Especially for people who are bad at
math. :-)

> Keep in mind that the
> clamping of the WS value to 14 applies not only to 15, but also to
> 16, 17 ... 255.

Yeah- I am bad at math.  Somehow I was mapping the WS to 4 bits and not
to 8.

> But the reality, as has been noted, is that WS>14 has not been a
> problem for the past 20 years, so implementors have figured it out,
> and I see no reason to change from the 1323 clamping of all WS>14.

Me either.

I just wanted a better explanation that it was indeed safe
(conservative) to clamp.

allman




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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On Tue, May 14, 2013 at 6:58 AM, Scheffenegger, Richard <rs@netapp.com> wrote:
> Hi Mark,
>
>>   - 3.1: "Many TCP implementations base their RTT measurements upon
>>     a sample of one segment per window or less.  While this yields
>>     an adequate approximation to the RTT for small windows, it
>>     results in an unacceptably poor RTT estimate for a LFN."
>>
>>     Do you have evidence of this?  We have evidence you're wrong:
>>
>>       Mark Allman, Vern Paxson.  On Estimating End-to-End Network
>>       Path Properties.  Proceedings of the ACM SIGCOMM Technical
>>       Symposium, Cambridge, MA, September 1999.
>>
>>     That shows that the number of samples per RTT is pretty
>>     immaterial to the effectiveness of the RTO.
>
> I'd like to point out, that the whole section 3 does not make a clear distinction between *timely measuring* RTT, and using that measured RTT for further purposes, such as arriving at a useful value for RTO.
>
> We can also argue about the specific wording in the 2nd sentence; But it appears to me, that you read "poor RTT estimate == poor RTO estimate"?
>
>>   - 3.1: "RTT estimator".  Note, TCP does not have an "RTT
>>     estimator".  Scrub this from the document.  We have an "RTO
>>     estimator".  These are different things.  Confusing them is a
>>     fundamental mistake.
>
> Correct. And IMHO 1323 is looking at RTT, but, agreed, only with the intended purpose of refining RTO (where these improvements haven't been as beneficial as originally envisioned).
>
> I rewrote the section 3.1 entirely:
>
> 3.  TCP Timestamp Option
>
> 3.1.  Introduction
>
>    TCP measures the round trip time (RTT), primarily for the purpose of
>    arriving at a decent value for the Retransmission Timeout (RTO) timer
>    interval.  Accurate and current RTT estimates are necessary to adapt
>    to changing traffic conditions, while a conservative estimate of the
>    RTO inveral is necessary to minimize spurious RTOs.
>
>    When [RFC1323] was originally written, it was perceived that more
>    timely and accurate RTT measurements would contribute to reducing
>    spurious RTOs, while maintaining their timeliness.  At the time, RTO
>    was also the only mechanism to make use of the measured RTT.  It has
>    been shown, that taking more RTT samples has only a very limited
>    effect to optimize RTOs [Allman99].
>
>    This document makes a clear distinction between the round trip time
>    measurement (RTTM) mechanism, and subsequent mechanisms using the RTT
>    signal as input, such as RTO (see Section 3.4).
>
>    It is important to use the timestamp option with big windows, to
>    allow the use of the PAWS mechanism (see Section 4).  Furthermore,
>    the option is useful for all TCP's, since it simplifies the sender
>    and allows the use of additional optimizations such as Eifel
>    ([RFC3522], [RFC4015]) and others.
>
> 3.2.  Timestamp Option
>
> And made the last paragraph of 3.3 a dedicated section, for the separation between RTTM and RTO:
>
> 3.4.  Updating the RTO value
>
>    [KL04] has highlighted the problem that an unmodified RTO
>    calculation, which is updated with per-packet RTT samples, will
>    truncate the path history too soon.  This can lead to an increase in
>    spurious retransmissions, when the path properties vary in the order
>    of a few RTTs, but a high number of RTT samples are taken on a much
>    shorter timescale.
>
>    Implementers should note that with timestamps multiple RTTMs can be
>    taken per RTT.  The [RFC6298] RTO estimator has weighting factors,
So is RTTM without timestamps. It occurs to me this issue belongs to
the RTO RFC 6298.

>    alpha and beta, based on an implicit assumption that at most one RTTM
>    will be sampled per RTT.  When using multiple RTTMs per RTT to update
>    the RTO estimator, the weighting factor SHOULD be decreased to take
>    into account the more frequent RTTMs.
>
>    For example, an implementation could choose to
>
>    o  just use one sample per RTT to update the RTO estimator, or
>
>    o  vary the gain based on the congestion window, or
>
>    o  take an average of all the RTT measurements (and the maximum of
>       the variance) received over one RTT,
have the latter two designs or examples been tested in the minimal
scale. can we stop proposing designs that don't really belong to this
RFC?

IMO the editor has tendency to keep adding new things or logic that
are either untested (e.g., sack idea) or not strongly relevant.

>
>    and then use that value to update the RTO estimator.  This document
>    does not prescribe any particular method for modifying the RTO
>    estimator.
it's suffice to me to kill the text of three examples above and have
the last sentence.

>
>
>
>
>
>
>
>>   - End of 3.3 on RTT sample weighting factors.
>>
>>     (1) The problem with the history being truncated when using RTTM
>>         was independently highlighted by Ludwig and Floyd.  We
>>         should at least have the common courtesy to cite Sally's
>>         note to e2e and Reiner's paper.
>
> I tried to find Sally's comment (also trying to find the reference given in your "Using Spurious Retransmissios to Adapt the Retransmission Timeout" paper), but was unsuccessful... Do you have a link?
>
>
>
>>   - 3.4, (A): Why are we discussing this in terms of the "Kth"
>>     segment?  Delayed ACKs per the standard is "2nd".  Why do we
>>     have to make the discussion in terms of some theory rather than
>>     in terms of what we have specified?
>
> fixed
>
>>   - 3.2 insinuates that you should not include a timestamp on an
>>     RST: "TSopt MUST be sent in every non-<RST> segment", implying
>>     it should not be sent on an <RST> (or you'd have just said
>>     "every segment").  But, then 4.2 goes on to (rightly IMO)
>>     develop why we should include it on <RST> segments.  This
>>     inconsistency needs fixed.
>
>
> fixed
>
>>     RTTM should not be deprecated.  It should be a MAY.
>>
>>     RTTM should not be discussed with breathless bullshit about hand
>>     wavy math and un-demonstrated stability issues and whatnot.
>>
>>     We should say that RTTM is absolutely within compliance of the
>>     spec and that it will not hurt your RTO.
>>
>>     We should also say that RTTM is unlikely to help your RTO.
>>
>>     We should leave it to implementers to decide if RTTM is useful
>>     for their purposes.
>>
>>     We should specify a way to vary the gains in the standard RTO
>>     algorithm based on the current cwnd.
>>
>>     And, we should absolutely state that there are other uses for
>>     the timestamp option (like Eifel, like PAWS) and there is
>>     nothing wrong with the *option* for that purpose.
>
> Best regards,
>    Richard
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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> >    When using multiple RTTMs per RTT to update
> >    the RTO estimator, the weighting factor SHOULD be decreased to take
> >    into account the more frequent RTTMs.
> >
> >    For example, an implementation could choose to
> >
> >    o  just use one sample per RTT to update the RTO estimator, or
> >
> >    o  vary the gain based on the congestion window, or
> >
> >    o  take an average of all the RTT measurements (and the maximum of
> >       the variance) received over one RTT,
>
> have the latter two designs or examples been tested in the minimal
> scale.

I think the second is fine.  The idea here is that if you take N samples
per RTT then you weight each sample such that the overall gain adds up
to the gain you would have used if you only took one sample (e.g., use
gain / N).  So, overall the EWMA history moves about like it does for
one sample / RTT.  Not exactly, but close.  So, this is just a little
math and I don't think it is a big departure from what is specified now.

I completely agree with you about the third option.  That seems pulled
out of thin air and without any sort of analysis I don't know why we'd
suggest it.

(I will note there is a contradiction in the above text ... "When using
multiple RTTMs per RTT to update the RTO estimator" [...] "just use one
sample per RTT".  I.e., the first bullet doesn't really address the
situation that was teed up.)

allman




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From prvs=38466dd104=david.borman@quantum.com  Tue May 14 12:51:11 2013
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From: David Borman <David.Borman@quantum.com>
To: "mallman@icir.org" <mallman@icir.org>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On May 14, 2013, at 1:52 PM, Mark Allman <mallman@icir.org> wrote:

>=20
>>>   When using multiple RTTMs per RTT to update
>>>   the RTO estimator, the weighting factor SHOULD be decreased to take
>>>   into account the more frequent RTTMs.
>>>=20
>>>   For example, an implementation could choose to
>>>=20
>>>   o  just use one sample per RTT to update the RTO estimator, or
>>>=20
>>>   o  vary the gain based on the congestion window, or
>>>=20
>>>   o  take an average of all the RTT measurements (and the maximum of
>>>      the variance) received over one RTT,
>>=20
>> have the latter two designs or examples been tested in the minimal
>> scale.
>=20
> I think the second is fine.  The idea here is that if you take N samples
> per RTT then you weight each sample such that the overall gain adds up
> to the gain you would have used if you only took one sample (e.g., use
> gain / N).  So, overall the EWMA history moves about like it does for
> one sample / RTT.  Not exactly, but close.  So, this is just a little
> math and I don't think it is a big departure from what is specified now.
>=20
> I completely agree with you about the third option.  That seems pulled
> out of thin air and without any sort of analysis I don't know why we'd
> suggest it.
>=20
> (I will note there is a contradiction in the above text ... "When using
> multiple RTTMs per RTT to update the RTO estimator" [...] "just use one
> sample per RTT".  I.e., the first bullet doesn't really address the
> situation that was teed up.)

Ok, I wrote the original text of this section, along with the examples.
The primary issue is addressed by the text just before the above snippit,
which now reads:

  Implementers should note that with timestamps multiple RTTMs can be
  taken per RTT.  The [RFC6298] RTO estimator has weighting factors,
  alpha and beta, based on an implicit assumption that at most one RTTM
  will be sampled per RTT.

Rather than leaving it with just that statement, it seemed prudent to
have at least one example to give implementors some guidance.  But just
one example didn't seem wise, as that would then become the default, and
we just wanted these to be examples, subject to change over time based
on experience.  The primary goal is to make sure that implementors don't
just blindly start injecting multiple RTTM per RTT into an RTO estimator
that has an implicit assumption of at most one RTTM per RTT.

The third example was to get a single RTTM value based on all the
received RTTM within one RTT, as opposed to just picking one and
ignoring the rest, and inject that averaged information into the
RTO estimator, which assumed only one RTTM per RTT.  It was also
the thought that it would help with getting a different number of
RTTM samples in each RTT.  You don't know a priori how many RTTM
samples there will be in any given RTT, hence you don't really know
how to properly weight them as they arrive, but it's easy to keep
a running average as they arrive.

So, for this section, the primary goal is to make sure that implementors
don't just blindly start injecting multiple RTTM samples per RTT into
the RTO estimator without accounting for the increased sample rate.
The examples are secondary to this.

If there are examples of how to address this issue, there should be
more than one example, unless there now exists one preferred method
for dealing with this issue.  Otherwise, eliminate the examples
altogether, and the implementor will have to get guidance for this
issue from other sources.

			-David Borman


>=20
> allman
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-12.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : TCP Extensions for High Performance
	Author(s)       : David Borman
                          Bob Braden
                          Van Jacobson
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-1323bis-12.txt
	Pages           : 45
	Date            : 2013-05-14

Abstract:
   This document specifies a set of TCP extensions to improve
   performance over paths with a large bandwidth * delay product and to
   provide reliable operation over very high-speed paths.  It defines
   TCP options for scaled windows and timestamps.  The timestamps are
   used for two distinct mechanisms, RTTM (Round Trip Time Measurement)
   and PAWS (Protection Against Wrapped Sequences).

   This document updates and obsoletes RFC 1323.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-12

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-1323bis-12


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


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis 
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Hi Mark,

>> I tried to find Sally's comment (also trying to find the reference
>> given in your "Using Spurious Retransmissios to Adapt the
>> Retransmission Timeout" paper), but was unsuccessful... Do you have a
>> link?
>=20
> I figured that'd be easy to track down, but I couldn't find it quickly.
> I was wrong, it was not from end2end it was from the tcp-lw list.  But,
> even with that bit of information I can't find it.  I guess you can just
> forget that Sally worked this out since the Internet has! :)


I think I found it: :)

http://www.ietf.org/mail-archive/web/tcpm/current/msg02508.html



Here is the link to the diff from revision 11:

http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-ietf-t=
cpm-1323bis-11.txt&url2=3Dhttp://tools.ietf.org/id/draft-ietf-tcpm-1323bis-=
12.txt



Richard Scheffenegger



> -----Original Message-----
> From: mallman@icir.org [mailto:mallman@icir.org]
> Sent: Dienstag, 14. Mai 2013 16:33
> To: Scheffenegger, Richard
> Cc: Pasi Sarolahti; tcpm (tcpm@ietf.org)
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
>=20
>=20
> > I'd like to point out, that the whole section 3 does not make a clear
> > distinction between *timely measuring* RTT, and using that measured
> > RTT for further purposes, such as arriving at a useful value for RTO.
>=20
> Correct.  1323 and 1323bis are both very sloppy in this regard.
>=20
> But, standard TCP only uses RTT samples for one thing: computing an RTO.
> And, there are enough words about the RTO in 1323bis that it is clear tha=
t
> the intent of RTTM is to feed a whole lot more RTT samples into the RTO
> estimator.  So, trying to somehow say RTTM isn't about the RTO is at best
> splitting hairs.
>=20
> > We can also argue about the specific wording in the 2nd sentence; But
> > it appears to me, that you read "poor RTT estimate =3D=3D poor RTO
> > estimate"?
>=20
> Yes.  See above.  There is no "RTT estimate" in TCP.  I think this
> equivalence is the clear intent of the document.
>=20
> (Not that it shouldn't be cleaned up ... it should be ...)
>=20
> Also, note, that nothing I have said precludes someone from needing many
> RTT samples per RTT for something else.  E.g., it may well be needed for
> delay-based congestion control.  And, if so, the timestamp option can
> provide a steady stream of RTT samples.  But, it seems clear to me that a
> steady stream of RTT samples is not what the aim of RTTM in 1323bis.
>=20
> > I rewrote the section 3.1 entirely:
>=20
> I am not going to try to piece together what you have done from email.
> This is not a small change.  When you're ready, issue another draft and w=
e
> can all take a look.
>=20
> > >   - End of 3.3 on RTT sample weighting factors.
> > >
> > >     (1) The problem with the history being truncated when using RTTM
> > >         was independently highlighted by Ludwig and Floyd.  We
> > >         should at least have the common courtesy to cite Sally's
> > >         note to e2e and Reiner's paper.
> >
>=20
> allman
>=20
>=20


From mallman@icir.org  Tue May 14 18:42:58 2013
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> If there are examples of how to address this issue, there should be
> more than one example, unless there now exists one preferred method
> for dealing with this issue.

I think there is a preferred method: the method we have been using for
25 years.  I think you are making this too difficult.  I don't think
this is somehow a deeply mysterious problem.  Let me pop up and walk
through the situation.

(0) We have a standard RTO for TCP.  It is Van's RTO algorithm.  It was
    crudely specified in RFC1122.  We wrote a better version of it in
    RFC2988.  It was updated in RFC6298.

(1) TCP's RTO spec uses EWMAs for both SRTT and RTTVAR.  These then make
    up the RTO.  The goal of the EWMA is to lend some weight to the most
    recent RTT sample while also keeping some amount of history of
    previous samples.  The EWMAs use gains of alpha=1/8 for SRTT and
    beta=1/4 for RTTVAR.

(2) Finally, RFC6298 is for a TCP that takes one RTT sample per RTT.  If
    the gains are held constant and more samples are taken per RTT then
    the recent samples force out the history more rapidly.

(3) The RTTM algorithm suggests taking more than one RTT sample per
    RTT.  So, we end up with an RTO estimator that remembers relatively
    less and less history as the congestion window grows.

(4) If we want to keep roughly the same amount of history (in terms of
    a time span) regardless of the number of samples taken per RTT we
    need to adjust the gains.  E.g., if we were to take two samples per
    RTT then we'd use an alpha of 1/8*1/2 for each sample and that would
    approximate the case of one sample with a gain of 1/8.

(5) We know roughly how many RTT samples a congestion window worth of
    data will yield.  So, something like this will approximate RFC6298
    regardless of how many samples are pushed through the EWMAs per RTT:

    ExpectedSamples = ceiling (FlightSize / SMSS / 2)
    alpha' = alpha / ExpectedSamples
    beta' = beta / ExpectedSamples

    Now just run the algorithm in RFC6298 with alpha' and beta' instead
    of alpha and beta.  I.e.,

        RTTVAR <- (1 - beta') * RTTVAR + beta' * |SRTT - R'|
        SRTT <- (1 - alpha') * SRTT + alpha' * R'

        (for each RTT sample R')

(6) That isn't **exactly** the same as RFC6298, but it is absolutely
    **almost** the same.  And, it is absolutely within the spirit of
    TCP's standardized RTO.  I.e., we keep about the same amount of
    history and we weight the recent RTT samples with about the same
    weight.

    So, I don't see why 1323bis shouldn't say "if you use this RTTM we
    codify then you make the following adjustments to TCP's standard RTO
    procedure [...insert step (5) here...].  To me that is well
    understood and straightforward.  This notion of taking the average
    and using the max variance or something is not as well understood.
    Nor would anything else you cook up.  E.g., why the mean?  Why not
    the median?  Why not the mode?  I don't understand why we would not
    simply take a very small step to say "continue to use TCP's standard
    RTO" in favor of something unknown.  And, I also don't understand
    the argument that we need to give implementers a bunch of choices
    here.  I mean, I am fine with folks going off and playing with
    things in their research.  But, what is the argument that says we
    for sure need to deviate from the standard here?

allman




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From pasi.sarolahti@iki.fi  Wed May 15 03:29:51 2013
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On May 15, 2013, at 4:42 AM, Mark Allman <mallman@icir.org> wrote:
>=20
>    So, I don't see why 1323bis shouldn't say "if you use this RTTM we
>    codify then you make the following adjustments to TCP's standard =
RTO
>    procedure [...insert step (5) here...].  To me that is well
>    understood and straightforward.

When you update the RTO RFC the next time, you should remember to =
include your step (5) :-) RTO calculation is orthogonal to the method of =
collecting RTT samples, so in my mind it would be better to keep the =
algorithm in one place. Multiple RTT samples per window can also be =
taken without timestamps.

The modification seems straightforward enough, but some practical =
experience might be useful before writing it down in a standards track =
document. Timestamps have been around for quite a while, so what do =
implementations currently do about the RTO? Linux has its own tweaks to =
address this, but how about other implementations? Are they just fine =
with the current algorithm (maybe because RTO is often clamped to 1 =
sec)?

- Pasi


From mallman@icir.org  Wed May 15 04:50:00 2013
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Pasi-

> When you update the RTO RFC the next time, you should remember to
> include your step (5) :-) RTO calculation is orthogonal to the method
> of collecting RTT samples, so in my mind it would be better to keep
> the algorithm in one place. Multiple RTT samples per window can also
> be taken without timestamps. 

Well, maybe ... 

  - The above is certainly fair pushback.

  - Given that taking multiple RTT samples per RTT doesn't buy anything
    for the RTO estimator codified in RFC6298 I am not sure why we'd
    specify what to do in that case (beyond some "address it" statement,
    which is already there) in a bis of that document.

    But, OK, OK.... we could argue about that for a long time over
    beers.

  - It seems to me that if you are saying take multiple RTT samples per
    RTT then it should be incumbent upon you to address this point and
    not to simply kick it off to somewhere else.  And, the RTTM
    algorithm is saying exactly this: take more RTT samples per RTT.

> The modification seems straightforward enough, but some practical
> experience might be useful before writing it down in a standards track
> document.

That sort of generic pushback doesn't hold a lot of water with me.

  - What "practical experience" do you want?  We have a very long
    history with the alpha and beta gains specified in RFC6298 (and
    previous versions and implementations before that).  All I did was
    to apply those same gains to get the same history window for the
    case when multiple RTT samples are collected per RTT.  Its just
    straightforward math to get to roughly the standardized behavior
    from the estimator.  And, we have experience with that behavior.
    So, I am not sure what you are looking for here.

  - What do you suggest we do?  You say that because we haven't used
    this gain adjustment we can't write it down in a standards track
    document.  But, then what can we write down?  Any scheme we write
    down will have the same issue (e.g., the average the RTTs and feed
    the average to the estimator notion David sketched).  And, writing
    down nothing seems at least dubious when it is so straightforward to
    adjust the gains to make the estimator's behavior consistent with
    long-standing practice and the standard.

We have a standard and long-standing TCP RTO estimator.  All I am
suggesting is that we apply it here.  I have a hard time understanding
why that seems like such a big deal.

allman




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From michawe@ifi.uio.no  Wed May 15 06:27:01 2013
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, David Borman <David.Borman@quantum.com>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On 15. mai 2013, at 13:49, Mark Allman wrote:

>=20
> Pasi-
>=20
>> When you update the RTO RFC the next time, you should remember to
>> include your step (5) :-) RTO calculation is orthogonal to the method
>> of collecting RTT samples, so in my mind it would be better to keep
>> the algorithm in one place. Multiple RTT samples per window can also
>> be taken without timestamps.=20
>=20
> Well, maybe ...=20
>=20
>  - The above is certainly fair pushback.
>=20
>  - Given that taking multiple RTT samples per RTT doesn't buy anything
>    for the RTO estimator codified in RFC6298 I am not sure why we'd
>    specify what to do in that case (beyond some "address it" =
statement,
>    which is already there) in a bis of that document.
>=20
>    But, OK, OK.... we could argue about that for a long time over
>    beers.
>=20
>  - It seems to me that if you are saying take multiple RTT samples per
>    RTT then it should be incumbent upon you to address this point and
>    not to simply kick it off to somewhere else.  And, the RTTM
>    algorithm is saying exactly this: take more RTT samples per RTT.
>=20
>> The modification seems straightforward enough, but some practical
>> experience might be useful before writing it down in a standards =
track
>> document.
>=20
> That sort of generic pushback doesn't hold a lot of water with me.
>=20
>  - What "practical experience" do you want?  We have a very long
>    history with the alpha and beta gains specified in RFC6298 (and
>    previous versions and implementations before that).  All I did was
>    to apply those same gains to get the same history window for the
>    case when multiple RTT samples are collected per RTT.  Its just
>    straightforward math to get to roughly the standardized behavior
>    from the estimator.  And, we have experience with that behavior.
>    So, I am not sure what you are looking for here.

I may be wrong, but I think that the behavior with more samples + =
changed alpha and beta as you suggested can be significantly different.

By averaging over a larger amount of samples, the significance of =
outliers becomes diminished, even if we average over an accordingly =
longer time period. I played with Excel a bit to confirm this, and have =
calculated the following example: R=3D200ms, and alpha, beta as you =
suggest / from RFC6298 (1/8 / number_of_samples, 1/4 / =
number_of_samples). Let's assume that the connection has just begun, =
i.e. RTTVAR is 100ms. Say that, after 10 RTTs, a delay spike happens, =
seen by only 1 packet, giving it an RTT of 300ms.

Case 1, 1 sample per RTT: in the next RTT (RTT#11), the RTO is at 329ms =
(followed by 311 and 296 for the next RTTs).

Case 2, 2 samples per RTT: to have the same situation (but still =
assuming only one packet is affected), 300ms sample has to be packet =
#20. Then, the samples of the next packets (one taken every half RTT) =
are 280, 274, 268, 263, 258, 253. That is, the RTOs in RTT #11, are 280 =
/ 274, in RTT #12 they are 268/263, in RTT #13 they are 258/253.

Using the average of the two samples in case 2, this means that you have =
52ms (about 1/4 base RTT) difference in the 1st RTT after the delay =
spike, followed by 46ms in the 2nd and 41 in the 3rd. Using these =
specific values, this difference may or may not seem truly significant, =
but the effect does get amplified when using a larger number of =
samples...

I have a tendency to always make mistakes in such emails, so chances =
are, I'm wrong for some reason... in this case, very sorry for the =
noise.

Cheers,
Michael


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On May 15, 2013, at 2:49 PM, Mark Allman <mallman@icir.org> wrote:

>> The modification seems straightforward enough, but some practical
>> experience might be useful before writing it down in a standards =
track
>> document.
>=20
> That sort of generic pushback doesn't hold a lot of water with me.
>=20
>  - What "practical experience" do you want?  We have a very long
>    history with the alpha and beta gains specified in RFC6298 (and
>    previous versions and implementations before that).  All I did was
>    to apply those same gains to get the same history window for the
>    case when multiple RTT samples are collected per RTT.  Its just
>    straightforward math to get to roughly the standardized behavior
>    from the estimator.  And, we have experience with that behavior.
>    So, I am not sure what you are looking for here.

Well, if I'm not mistaken, we have long experience using RFC1323 =
together with RFC6298 as it is currently specified (the default FreeBSD =
behavior, I suppose, but I haven't followed BSD closely) and nothing has =
collapsed. But I fully agree with your earlier mail that rfc1323bis =
should refer to Reiner's paper and Sally's mail about the problems of =
RTTM and RTO calculation. On the other hand, I think Reiner's paper also =
proposed a slightly different fix.

I don't think the adjusted RTO algorithm would be "such a big deal", but =
I'm a bit hesitant adding new RTO algorithm specifications in the =
context of rfc1323bis, even if it was a minor adjustment to follow the =
original intent of RFC6298.

(Just a voice of one individual. If people support including the fixed =
RTO calculation, so be it.)

- Pasi


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From: David Borman <David.Borman@quantum.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On May 15, 2013, at 8:58 AM, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote:

> On May 15, 2013, at 2:49 PM, Mark Allman <mallman@icir.org> wrote:
>=20
>>> The modification seems straightforward enough, but some practical
>>> experience might be useful before writing it down in a standards track
>>> document.
>>=20
>> That sort of generic pushback doesn't hold a lot of water with me.
>>=20
>> - What "practical experience" do you want?  We have a very long
>>   history with the alpha and beta gains specified in RFC6298 (and
>>   previous versions and implementations before that).  All I did was
>>   to apply those same gains to get the same history window for the
>>   case when multiple RTT samples are collected per RTT.  Its just
>>   straightforward math to get to roughly the standardized behavior
>>   from the estimator.  And, we have experience with that behavior.
>>   So, I am not sure what you are looking for here.
>=20
> Well, if I'm not mistaken, we have long experience using RFC1323 together=
 with RFC6298 as it is currently specified (the default FreeBSD behavior, I=
 suppose, but I haven't followed BSD closely) and nothing has collapsed. Bu=
t I fully agree with your earlier mail that rfc1323bis should refer to Rein=
er's paper and Sally's mail about the problems of RTTM and RTO calculation.=
 On the other hand, I think Reiner's paper also proposed a slightly differe=
nt fix.
>=20
> I don't think the adjusted RTO algorithm would be "such a big deal", but =
I'm a bit hesitant adding new RTO algorithm specifications in the context o=
f rfc1323bis, even if it was a minor adjustment to follow the original inte=
nt of RFC6298.
>=20
> (Just a voice of one individual. If people support including the fixed RT=
O calculation, so be it.)

Ok, given the recent discussion, I propose the following:

1) Add an appendix based on Mark's message about modifying the RFC6298 algo=
rithm.
2) Replace the examples in section 3.4 with a reference to the new appendix.

As long as 3.4 points out there is a problem of feeding more RTTM samples p=
er RTT into an unmodified RTO estimator, we've achieved the primary objecti=
ve.  Examples of how to address that has always been secondary, and moving =
it to an appendix explicitly makes it not part of the spec.

			-David
=20
>=20
> - Pasi
>=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-1323bis
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On 5/15/2013 7:16 AM, David Borman wrote:
> Ok, given the recent discussion, I propose the following:
>
> 1) Add an appendix based on Mark's message about modifying the RFC6298 algorithm.

On the issue, maybe, but on the new code Mark provides, definitely no.

As a current standards-track doc, changes should reflect current 
experience and deployment experience, not intended future practice.

> 2) Replace the examples in section 3.4 with a reference to the new appendix.
>
> As long as 3.4 points out there is a problem of feeding more RTTM
> samples per RTT into an unmodified RTO estimator, we've achieved the
> primary objective.

That's useful to point out because it's current experience.

> Examples of how to address that has always been
> secondary, and moving it to an appendix explicitly makes it not part of
> the spec.

Those should be part of either another document or a later update, once 
there is WG consensus on a specific solution.

Joe

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From: David Borman <David.Borman@quantum.com>
To: "mallman@icir.org" <mallman@icir.org>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On May 14, 2013, at 8:42 PM, Mark Allman <mallman@icir.org> wrote:
...
>    ... I don't understand why we would not
>    simply take a very small step to say "continue to use TCP's standard
>    RTO" in favor of something unknown.  And, I also don't understand
>    the argument that we need to give implementers a bunch of choices
>    here.  I mean, I am fine with folks going off and playing with
>    things in their research.  But, what is the argument that says we
>    for sure need to deviate from the standard here?

If we have one preferred recommendation, then giving that as the only
example is just fine.  At the time the examples went into the earlier
versions of 1323bis, we wanted to avoid codifying only one way to
address the issue. It appears things have changed.

Here's an excerpt from the 2008 Philly IETF TCPM meeting minutes,
back when Mark was one of the TCPM co-chairs. :-)

>   Joe Touch: If you supply one example in section 3.3 on RTTM, this
>    will become the default. David, we could choose an alternative
>    RTTM implementation style.=20
>   Mark: A few people have solved this by changing the weights on the
>    sample.=20
>   David said that he expected that this should then illustrate that
>    there were multiple approaches. ... We should be careful that we
>    don't suggest something that is too extreme, we need to support
>    this with research.=20
>   Mark said there were published papers that talk about weight
>    changes.=20
>   Joe said we should recommend alternatives that we know are safe -
>    but not too much importance to one unless we know how to
>    choose).=20
>   David said that he should tell people what to do, and not
>    prescribe an algorithm.=20
>   Mark said we needed to keep the variance and adjust the gain based
>    on the cwnd.

So, I'll continue to argue that the primary goal is to tell people they
need to adapt their RTO estimator if they start feeding in multiple RTTM
samples per RTT.  We shouldn't get hung up on the examples, and should
just remove them or move them to an appendix if they are a stumbling block.

Ironically, the next sentence in the 2008 minutes is:
>   Hope to make the next version ready for a WGLC. Lars said we
>   should add milestones for the two documents (Ted to do).

			-David

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From rs@netapp.com  Wed May 15 08:23:35 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Joe Touch <touch@isi.edu>, David Borman <David.Borman@quantum.com>, "<mallman@icir.org>" <mallman@icir.org>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Hi,

I'd like to point to RFC6298, section 3:

   Traditionally, TCP implementations have taken one RTT measurement at
   a time (typically, once per RTT).  However, when using the timestamp
   option, each ACK can be used as an RTT sample.  RFC 1323 [JBB92]
   suggests that TCP connections utilizing large congestion windows
   should take many RTT samples per window of data to avoid aliasing
   effects in the estimated RTT.  A TCP implementation MUST take at
   least one RTT measurement per RTT (unless that is not possible per
   Karn's algorithm).

   For fairly modest congestion window sizes, research suggests that
   timing each segment does not lead to a better RTT estimator [AP99].
   Additionally, when multiple samples are taken per RTT, the alpha and
   beta defined in Section 2 may keep an inadequate RTT history.  A
   method for changing these constants is currently an open research
   question.

So, it is in fact pointed out in the proper document (that is concerned wit=
h calculating the RTO).

However, I looked up recent BSD and Linux:

FreeBSD 9.1 does *NOT* modify alpha/beta when RTT samples are performed on =
every ACK.

Linux 3.2.4 has apparently removed the earlier variable rttvar update stuff=
, and is updating sRTT once per window, and RTTvar with the maximum varianc=
e observed in the last window (instead of an average/mean/last variance).

Perhaps Kevin or others can comment on other widely used stacks...


To get back to 1323bis: As I have stated, that document concerns itself wit=
h obtaining more frequent RTT samples; there used to be an implicit assumpt=
ion, that the only consumer of that signal will be the RTO calculation (tha=
t much has been true when 1323 was published), but nowadays there are many =
more (potential) uses for timely RTT measurements (cubic - the lack of a se=
curity mechanism in 1323TS turned out to be a big problem; chirp, ledbat). =
Also, there are other uses for timestamps themselves (eifel, passive measur=
ements).

As there is no clear guidance in RFC6298 as to how to deal with more sample=
s per window, it might be prudent to include some text in 1323bis now. My p=
ersonal preference would be to put that into an appendix though...


One last observation:=20

RTO calculations are often performed using fixed-comma (integer) arithmetic=
s.

High timestamp clock granularities and simple adjustments of the alpha/beta=
 factors as suggested might not work properly, when the window sizes grow t=
oo large (ie. >16 segments), and sRTT would no longer be updated at all... =
Based on the papers Mark referred to, that might not be too much of a probl=
em, further masking this issue, but nevertheless a simple "fix" might prove=
 to be worse than the suggestion by David early on, to use only a single RT=
T sample per window (even though that was commented very strongly by Mark).



Regards,

Richard Scheffenegger


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Joe Touch
> Sent: Mittwoch, 15. Mai 2013 16:29
> To: David Borman
> Cc: tcpm (tcpm@ietf.org); <mallman@icir.org>
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
>=20
>=20
>=20
> On 5/15/2013 7:16 AM, David Borman wrote:
> > Ok, given the recent discussion, I propose the following:
> >
> > 1) Add an appendix based on Mark's message about modifying the RFC6298
> algorithm.
>=20
> On the issue, maybe, but on the new code Mark provides, definitely no.
>=20
> As a current standards-track doc, changes should reflect current
> experience and deployment experience, not intended future practice.
>=20
> > 2) Replace the examples in section 3.4 with a reference to the new
> appendix.
> >
> > As long as 3.4 points out there is a problem of feeding more RTTM
> > samples per RTT into an unmodified RTO estimator, we've achieved the
> > primary objective.
>=20
> That's useful to point out because it's current experience.
>=20
> > Examples of how to address that has always been secondary, and moving
> > it to an appendix explicitly makes it not part of the spec.
>=20
> Those should be part of either another document or a later update, once
> there is WG consensus on a specific solution.
>=20
> Joe
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From mallman@icir.org  Wed May 15 09:32:34 2013
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Michael-

> I may be wrong, but I think that the behavior with more samples +
> changed alpha and beta as you suggested can be significantly
> different. 

Your comparison is not sound because you are conflating two things: (1)
adjusted gains and (2) multiple samples.

So, yes, you find the RTOs that pop out are different.  E.g., in the
case of one sample per RTT does that get the max sample or the min
sample during the given RTT?  It matters to the result.  Clearly taking
1 and taking N are going to be different.  The argument behind RTTM is
that this difference is important, i.e., more samples gives you a more
accurate assessment of the RTT process by taking into account the
fluctuations that can be observed within an RTT.  Our paper / data
suggests this is wrong.

All I am saying is that the length of the history kept in the EWMA
should be roughly the same regardless of the number of samples taken.

If you do the same test and keep a constant RTT for each RTT you'll see
that the RTOs that pop out are roughly the same.

I.e., pick a series of RTT measurements for each round of the transfer.
I dunno, say 0.1, 0.11, 0.09, 0.85, 0.13 for the first five RTTs.  Now,
use the RFC6298 gains and run each of those samples through the RTO
calculation once.  Call that X.  Now, pick some number of samples N per
RTT and run each of the above numbers through the RTO calculation N
times, but with alpha/N and beta/N as the gains (i.e., you are
simulating multiple samples that are all the same; that isn't realistic,
but shows the math).  Call the last RTO Y.  What you'll find is that the
X and Y are pretty similar.  I.e., we have adjusted the gains to keep
roughly the same amount of history in the overall RTO.  And, that is the
point.

allman




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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On 5/15/2013 8:23 AM, Scheffenegger, Richard wrote:
> As there is no clear guidance in RFC6298 as to how to deal with more
> samples per window, it might be prudent to include some text in 1323bis
> now. My personal preference would be to put that into an appendix though...

I agree that it's important to point out that this is an important known 
issue, but I don't think we should include suggested solutions that we 
come up with on-the-fly right now.

Joe

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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On Wed, May 15, 2013 at 8:23 AM, Scheffenegger, Richard <rs@netapp.com>wrote:

>
> Hi,
>
> I'd like to point to RFC6298, section 3:
>
>    Traditionally, TCP implementations have taken one RTT measurement at
>    a time (typically, once per RTT).  However, when using the timestamp
>    option, each ACK can be used as an RTT sample.  RFC 1323 [JBB92]
>    suggests that TCP connections utilizing large congestion windows
>    should take many RTT samples per window of data to avoid aliasing
>    effects in the estimated RTT.  A TCP implementation MUST take at
>    least one RTT measurement per RTT (unless that is not possible per
>    Karn's algorithm).
>
>    For fairly modest congestion window sizes, research suggests that
>    timing each segment does not lead to a better RTT estimator [AP99].
>    Additionally, when multiple samples are taken per RTT, the alpha and
>    beta defined in Section 2 may keep an inadequate RTT history.  A
>    method for changing these constants is currently an open research
>    question.
>
> So, it is in fact pointed out in the proper document (that is concerned
> with calculating the RTO).
>
> However, I looked up recent BSD and Linux:
>
> FreeBSD 9.1 does *NOT* modify alpha/beta when RTT samples are performed on
> every ACK.
>
> Linux 3.2.4 has apparently removed the earlier variable rttvar update
> stuff, and is updating sRTT once per window, and RTTvar with the maximum
> variance observed in the last window (instead of an average/mean/last
> variance).
>
That's simply wrong. I have been working on that piece of code in Linux
everyday since this year. Linux RTO formula is different than the RFC but
it's not what you described. I am happy to point out the details but don't
want to derail the thread.


>
> Perhaps Kevin or others can comment on other widely used stacks...
>
>
> To get back to 1323bis: As I have stated, that document concerns itself
> with obtaining more frequent RTT samples; there used to be an implicit
> assumption, that the only consumer of that signal will be the RTO
> calculation (that much has been true when 1323 was published), but nowadays
> there are many more (potential) uses for timely RTT measurements (cubic -
> the lack of a security mechanism in 1323TS turned out to be a big problem;
> chirp, ledbat). Also, there are other uses for timestamps themselves
> (eifel, passive measurements).
>
> As there is no clear guidance in RFC6298 as to how to deal with more
> samples per window, it might be prudent to include some text in 1323bis
> now. My personal preference would be to put that into an appendix though...
>
>
> One last observation:
>
> RTO calculations are often performed using fixed-comma (integer)
> arithmetics.
>
> High timestamp clock granularities and simple adjustments of the
> alpha/beta factors as suggested might not work properly, when the window
> sizes grow too large (ie. >16 segments), and sRTT would no longer be
> updated at all... Based on the papers Mark referred to, that might not be
> too much of a problem, further masking this issue, but nevertheless a
> simple "fix" might prove to be worse than the suggestion by David early on,
> to use only a single RTT sample per window (even though that was commented
> very strongly by Mark).
>
>
>
> Regards,
>
> Richard Scheffenegger
>
>
> > -----Original Message-----
> > From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> > Joe Touch
> > Sent: Mittwoch, 15. Mai 2013 16:29
> > To: David Borman
> > Cc: tcpm (tcpm@ietf.org); <mallman@icir.org>
> > Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
> >
> >
> >
> > On 5/15/2013 7:16 AM, David Borman wrote:
> > > Ok, given the recent discussion, I propose the following:
> > >
> > > 1) Add an appendix based on Mark's message about modifying the RFC6298
> > algorithm.
> >
> > On the issue, maybe, but on the new code Mark provides, definitely no.
> >
> > As a current standards-track doc, changes should reflect current
> > experience and deployment experience, not intended future practice.
> >
> > > 2) Replace the examples in section 3.4 with a reference to the new
> > appendix.
> > >
> > > As long as 3.4 points out there is a problem of feeding more RTTM
> > > samples per RTT into an unmodified RTO estimator, we've achieved the
> > > primary objective.
> >
> > That's useful to point out because it's current experience.
> >
> > > Examples of how to address that has always been secondary, and moving
> > > it to an appendix explicitly makes it not part of the spec.
> >
> > Those should be part of either another document or a later update, once
> > there is WG consensus on a specific solution.
> >
> > Joe
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--001a11c33428f4f11904dcc4613e
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, May 15, 2013 at 8:23 AM, Scheffenegger, Richard <span dir=
=3D"ltr">&lt;<a href=3D"mailto:rs@netapp.com" target=3D"_blank">rs@netapp.c=
om</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
Hi,<br>
<br>
I&#39;d like to point to RFC6298, section 3:<br>
<br>
=A0 =A0Traditionally, TCP implementations have taken one RTT measurement at=
<br>
=A0 =A0a time (typically, once per RTT). =A0However, when using the timesta=
mp<br>
=A0 =A0option, each ACK can be used as an RTT sample. =A0RFC 1323 [JBB92]<b=
r>
=A0 =A0suggests that TCP connections utilizing large congestion windows<br>
=A0 =A0should take many RTT samples per window of data to avoid aliasing<br=
>
=A0 =A0effects in the estimated RTT. =A0A TCP implementation MUST take at<b=
r>
=A0 =A0least one RTT measurement per RTT (unless that is not possible per<b=
r>
=A0 =A0Karn&#39;s algorithm).<br>
<br>
=A0 =A0For fairly modest congestion window sizes, research suggests that<br=
>
=A0 =A0timing each segment does not lead to a better RTT estimator [AP99].<=
br>
=A0 =A0Additionally, when multiple samples are taken per RTT, the alpha and=
<br>
=A0 =A0beta defined in Section 2 may keep an inadequate RTT history. =A0A<b=
r>
=A0 =A0method for changing these constants is currently an open research<br=
>
=A0 =A0question.<br>
<br>
So, it is in fact pointed out in the proper document (that is concerned wit=
h calculating the RTO).<br>
<br>
However, I looked up recent BSD and Linux:<br>
<br>
FreeBSD 9.1 does *NOT* modify alpha/beta when RTT samples are performed on =
every ACK.<br>
<br>
Linux 3.2.4 has apparently removed the earlier variable rttvar update stuff=
, and is updating sRTT once per window, and RTTvar with the maximum varianc=
e observed in the last window (instead of an average/mean/last variance).<b=
r>

</blockquote><div style>That&#39;s simply wrong. I have been working on tha=
t piece of code in Linux everyday since this year. Linux RTO formula is dif=
ferent than the RFC but it&#39;s not what you described. I am happy to poin=
t out the details but don&#39;t want to derail the thread.</div>

<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
<br>
Perhaps Kevin or others can comment on other widely used stacks...<br>
<br>
<br>
To get back to 1323bis: As I have stated, that document concerns itself wit=
h obtaining more frequent RTT samples; there used to be an implicit assumpt=
ion, that the only consumer of that signal will be the RTO calculation (tha=
t much has been true when 1323 was published), but nowadays there are many =
more (potential) uses for timely RTT measurements (cubic - the lack of a se=
curity mechanism in 1323TS turned out to be a big problem; chirp, ledbat). =
Also, there are other uses for timestamps themselves (eifel, passive measur=
ements).<br>


<br>
As there is no clear guidance in RFC6298 as to how to deal with more sample=
s per window, it might be prudent to include some text in 1323bis now. My p=
ersonal preference would be to put that into an appendix though...<br>


<br>
<br>
One last observation:<br>
<br>
RTO calculations are often performed using fixed-comma (integer) arithmetic=
s.<br>
<br>
High timestamp clock granularities and simple adjustments of the alpha/beta=
 factors as suggested might not work properly, when the window sizes grow t=
oo large (ie. &gt;16 segments), and sRTT would no longer be updated at all.=
.. Based on the papers Mark referred to, that might not be too much of a pr=
oblem, further masking this issue, but nevertheless a simple &quot;fix&quot=
; might prove to be worse than the suggestion by David early on, to use onl=
y a single RTT sample per window (even though that was commented very stron=
gly by Mark).<br>


<br>
<br>
<br>
Regards,<br>
<br>
Richard Scheffenegger<br>
<div class=3D"im HOEnZb"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:tcpm-bounces@ietf.org">tcpm-bounces@ietf.org</=
a> [mailto:<a href=3D"mailto:tcpm-bounces@ietf.org">tcpm-bounces@ietf.org</=
a>] On Behalf Of<br>
&gt; Joe Touch<br>
&gt; Sent: Mittwoch, 15. Mai 2013 16:29<br>
&gt; To: David Borman<br>
&gt; Cc: tcpm (<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a>); &lt;<a =
href=3D"mailto:mallman@icir.org">mallman@icir.org</a>&gt;<br>
&gt; Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">&gt; On 5/15/2013 7:16 AM, Da=
vid Borman wrote:<br>
&gt; &gt; Ok, given the recent discussion, I propose the following:<br>
&gt; &gt;<br>
&gt; &gt; 1) Add an appendix based on Mark&#39;s message about modifying th=
e RFC6298<br>
&gt; algorithm.<br>
&gt;<br>
&gt; On the issue, maybe, but on the new code Mark provides, definitely no.=
<br>
&gt;<br>
&gt; As a current standards-track doc, changes should reflect current<br>
&gt; experience and deployment experience, not intended future practice.<br=
>
&gt;<br>
&gt; &gt; 2) Replace the examples in section 3.4 with a reference to the ne=
w<br>
&gt; appendix.<br>
&gt; &gt;<br>
&gt; &gt; As long as 3.4 points out there is a problem of feeding more RTTM=
<br>
&gt; &gt; samples per RTT into an unmodified RTO estimator, we&#39;ve achie=
ved the<br>
&gt; &gt; primary objective.<br>
&gt;<br>
&gt; That&#39;s useful to point out because it&#39;s current experience.<br=
>
&gt;<br>
&gt; &gt; Examples of how to address that has always been secondary, and mo=
ving<br>
&gt; &gt; it to an appendix explicitly makes it not part of the spec.<br>
&gt;<br>
&gt; Those should be part of either another document or a later update, onc=
e<br>
&gt; there is WG consensus on a specific solution.<br>
&gt;<br>
&gt; Joe<br>
&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div>

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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On Wed, May 15, 2013 at 6:58 AM, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote:
> I don't think the adjusted RTO algorithm would be "such a big deal", but I'm a bit hesitant adding new RTO algorithm specifications in the context of rfc1323bis, even if it was a minor adjustment to follow the original intent of RFC6298.

+1

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Mark,=20

do you know of any stack that is deployed and uses the suggested adjustable=
 gain? You mentioned that there are stacks doing this...


I'm not questioning the math (and that the numbers aren't exactly the same =
is fine), but as Joe mentioned, the bar for a standards track document need=
s to be demonstrated deployment...


If there isn't a widely deployed stack using the adjusted gains, we are pro=
bably left with "use only one sample per RTT to update RTO" as that is wide=
ly deployed in the linux stack...

Also, that would not change the calculation of 6298 a bit, which seems to b=
e the other requirement...



Richard Scheffenegger


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[answering sporadically between meetings ... I see a large number of
 messages ... ugh]

> do you know of any stack that is deployed and uses the suggested
> adjustable gain? You mentioned that there are stacks doing this... 

I do not.  But, I have lost the bubble on this.

> I'm not questioning the math (and that the numbers aren't exactly the
> same is fine), but as Joe mentioned, the bar for a standards track
> document needs to be demonstrated deployment... 

Well, that is clearly not correct or we'd never publish anything new on
the standard track.

> If there isn't a widely deployed stack using the adjusted gains, we
> are probably left with "use only one sample per RTT to update RTO" as
> that is widely deployed in the linux stack... 

Well, the problem is "use only one RTT to update the RTO" is isomorphic
with "SHOULD NOT use RTTM".  I am personally fine with deprecating RTTM
completely (note: RTTM is not the same as the TS option and it is not
the same as PAWS).  In my view RTTM does not add anything and so we can
just say that and move on.  But, my sense of the WG is that making it a
MAY is probably more likely to obtain consensus.  So, if you MAY use
RTTM then we're left with this question of how to actually update the
RTO with the multiple samples.  There seem to be two viable choices:

(1) Make no comment and let all samples flow through the RFC6298
    estimator.  We know this produces an aggressive RTO that keeps less
    history and ends up waiting a bit less time for needed RTOs and also
    tripped a whole lot more spurious RTOs.

(2) We recommend adjusting the gains to adapt the history to the spirit
    of TCP's standard RTO algorithm.

allman




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On 5/15/2013 10:29 AM, Mark Allman wrote:
>> I'm not questioning the math (and that the numbers aren't exactly the
>> >same is fine), but as Joe mentioned, the bar for a standards track
>> >document needs to be demonstrated deployment...

FWIW, I actually said "As a current standards-track doc, changes should 
reflect current experience and deployment experience, not intended 
future practice."

> Well, that is clearly not correct or we'd never publish anything new on
> the standard track.

I agree, but we shouldn't be introducing new mechanism during WGLC either.

Further, this isn't a new standards-track doc; it's an update to an 
existing one. The bar for that ought to be a little higher, and ought to 
include some experience.

Joe

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> I agree, but we shouldn't be introducing new mechanism during WGLC
> either.

We shouldn't be havin' an effin' WGLC.

allman




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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On Wed, May 15, 2013 at 10:29 AM, Mark Allman <mallman@icir.org> wrote:
[snip]
> the same as PAWS).  In my view RTTM does not add anything and so we can
> just say that and move on.  But, my sense of the WG is that making it a
> MAY is probably more likely to obtain consensus.  So, if you MAY use
+1

> RTTM then we're left with this question of how to actually update the
> RTO with the multiple samples.  There seem to be two viable choices:
>
> (1) Make no comment and let all samples flow through the RFC6298
>     estimator.  We know this produces an aggressive RTO that keeps less
>     history and ends up waiting a bit less time for needed RTOs and also
>     tripped a whole lot more spurious RTOs.
+1

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> Ok, given the recent discussion, I propose the following:
> 
> 1) Add an appendix based on Mark's message about modifying the RFC6298
>    algorithm. 
> 2) Replace the examples in section 3.4 with a reference to the new
>    appendix. 
> 
> As long as 3.4 points out there is a problem of feeding more RTTM
> samples per RTT into an unmodified RTO estimator, we've achieved the
> primary objective.  Examples of how to address that has always been
> secondary, and moving it to an appendix explicitly makes it not part
> of the spec. 

I can abide.

And, shut up. :-)

allman




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From pasi.sarolahti@iki.fi  Wed May 15 11:56:53 2013
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On May 15, 2013, at 9:05 PM, Mark Allman <mallman@icir.org> wrote:

>> I agree, but we shouldn't be introducing new mechanism during WGLC
>> either.
>=20
> We shouldn't be havin' an effin' WGLC.

It seems sometimes you need a WGLC to get people really read a document. =
This discussion thread started near the end of the WGLC. While there =
have been good discussions on individual issues earlier, I don't recall =
many as thorough reviews as yours, at least so that they had reached the =
list. So many thanks for your comments so far! (hopefully you still have =
energy to follow up that they are addressed well)

In the end the main thing is that we are making progress, with or =
without WGLC. Hopefully in 2018 we don't need to review the old =
discussions on this draft anymore :-)

- Pasi


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>, "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Hi,

Before I again post an update with disputed sections of text, here is my cu=
rrent version of section 3. Note that the title was also changed to put the=
 emphasis away from the RTTM/RTO update part.

I've tried to keep all the comments reflected in this updated text, but thi=
s might not yet reflect the concensus!

Best regards,
  Richard



3.  TCP Timestamp Option

3.1.  Introduction

   TCP measures the round trip time (RTT), primarily for the purpose of
   arriving at a reasonable value for the Retransmission Timeout (RTO)
   timer interval.  Accurate and current RTT estimates are necessary to
   adapt to changing traffic conditions, while a conservative estimate
   of the RTO inveral is necessary to minimize spurious RTOs.

   When [RFC1323] was originally written, it was perceived that taking
   RTT measurements for each segment, and also during retransmissions,
   would contribute to reduce spurious RTOs, while maintaining the
   timeliness of necessary RTOs.  At the time, RTO was also the only
   mechanism to make use of the measured RTT.  It has been shown, that
   taking more RTT samples has only a very limited effect to optimize
   RTOs [Allman99].

   This document makes a clear distinction between the round trip time
   measurement (RTTM) mechanism, and subsequent mechanisms using the RTT
   signal as input, such as RTO (see Section 3.4).

   The timestamp option is important when large receive windows are
   used, to allow the use of the PAWS mechanism (see Section 4).
   Furthermore, the option is useful for all TCP's, since it simplifies
   the sender and allows the use of additional optimizations such as
   Eifel ([RFC3522], [RFC4015]) and others.

3.2.  Timestamp Option

   TCP is a symmetric protocol, allowing data to be sent at any time in
   either direction, and therefore timestamp echoing may occur in either
   direction.  For simplicity and symmetry, we specify that timestamps
   always be sent and echoed in both directions.  For efficiency, we
   combine the timestamp and timestamp reply fields into a single TCP
   Timestamp Option.

   TCP Timestamp Option (TSopt):

   Kind: 8

   Length: 10 bytes

          +-------+-------+---------------------+---------------------+
          |Kind=3D8 |  10   |   TS Value (TSval)  |TS Echo Reply (TSecr)|
          +-------+-------+---------------------+---------------------+
              1       1              4                     4



   The Timestamp Option carries two four-byte timestamp fields.  The
   Timestamp Value field (TSval) contains the current value of the
   timestamp clock of the TCP sending the option.

   The Timestamp Echo Reply field (TSecr) is valid if the ACK bit is set
   in the TCP header; if it is valid, it echoes a timestamp value that
   was sent by the remote TCP in the TSval field of a Timestamp option.
   When TSecr is not valid, its value MUST be zero.  However, a value of
   zero does not imply TSecr being invalid.  The TSecr value will
   generally be from the most recent Timestamp Option that was received;
   however, there are exceptions that are explained below.

   A TCP MAY send the Timestamp option (TSopt) in an initial <SYN>
   segment (i.e., segment containing a SYN bit and no ACK bit), and MAY
   send a TSopt in other segments only if it received a TSopt in the
   initial <SYN> or <SYN,ACK> segment for the connection.

   Once TSopt has been successfully negotiated (sent and received)
   during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
   non-<RST> segment for the duration of the connection, and SHOULD be
   sent in a <RST> segment (see Section 4.2 for details).  If a non-
   <RST> segment is received without a TSopt, a TCP MAY drop the segment
   and send an <ACK> for the last in-sequence segment.  A TCP MUST NOT
   abort a TCP connection if a non-<RST> segment is received without a
   TSopt.

   If a TSopt is received on a connection where TSopt was not negotiated
   in the initial three-way handshake, the TSopt MUST be ignored and the
   packet processed normally.

   In the case of crossing <SYN> segments where one <SYN> contains a
   TSopt and the other doesn't, both sides MAY send a TSopt in the
   <SYN,ACK> segment.

   TSopt is required for the two mechanisms described in sections 3.3
   and 4.2.  There are also other mechanisms that rely on the presence
   of the TSopt, e.g.  [RFC3522].  If a TCP stopped sending TSopt at any
   time during an established session, it interferes with these
   mechanisms.  This update to [RFC1323] describes explicitly the
   previous assumption (see Section 4.2), that each TCP segment must
   have TSopt, once negotiated.

3.3.  The RTTM Mechanism

   RTTM places a Timestamp Option in every segment, with a TSval that is
   obtained from a (virtual) "timestamp clock".  Values of this clock
   MUST be at least approximately proportional to real time, in order to
   measure actual RTT.

   These TSval values are echoed in TSecr values in the reverse
   direction.  The difference between a received TSecr value and the
   current timestamp clock value provides a RTT measurement.

   When timestamps are used, every segment that is received will contain
   a TSecr value.  However, these values cannot all be used to update
   the measured RTT.  The following example illustrates why.  It shows a
   one-way data flow with segments arriving in sequence without loss.
   Here A, B, C... represent data blocks occupying successive blocks of
   sequence numbers, and ACK(A),... represent the corresponding
   cumulative acknowledgments.  The two timestamp fields of the
   Timestamp Option are shown symbolically as <TSval=3Dx,TSecr=3Dy>.  Each
   TSecr field contains the value most recently received in a TSval
   field.

              TCP  A                                     TCP B

                              <A,TSval=3D1,TSecr=3D120> ----->

                   <---- <ACK(A),TSval=3D127,TSecr=3D1>

                              <B,TSval=3D5,TSecr=3D127> ----->

                   <---- <ACK(B),TSval=3D131,TSecr=3D5>

                . . . . . . . . . . . . . . . . . . . . . .

                              <C,TSval=3D65,TSecr=3D131> ---->

                   <---- <ACK(C),TSval=3D191,TSecr=3D65>

                                  (etc.)

   The dotted line marks a pause (60 time units long) in which A had
   nothing to send.  Note that this pause inflates the RTT which B could
   infer from receiving TSecr=3D131 in data segment C. Thus, in one-way
   data flows, RTTM in the reverse direction measures a value that is
   inflated by gaps in sending data.  However, the following rule
   prevents a resulting inflation of the measured RTT:

   RTTM Rule: A TSecr value received in a segment MAY be used to update
              the averaged RTT measurement only if the segment advances
              the left edge of the send window, i.e.  SND.UNA is
              increased.

   Since TCP B is not sending data, the data segment C does not
   acknowledge any new data when it arrives at B. Thus, the inflated
   RTTM measurement is not used to update B's RTTM measurement.


3.4.  Updating the RTO value

   [Ludwig00] and [Floyd05] have highlighted the problem that an
   unmodified RTO calculation, which is updated with per-packet RTT
   samples, will truncate the path history too soon.  This can lead to
   an increase in spurious retransmissions, when the path properties
   vary in the order of a few RTTs, but a high number of RTT samples are
   taken on a much shorter timescale.

   Implementers should note that with timestamps multiple RTTMs can be
   taken per RTT.  The [RFC6298] RTO estimator has weighting factors,
   alpha and beta, based on an implicit assumption that at most one RTTM
   will be sampled per RTT.  When multiple RTTMs per RTT are available
   to update the RTO estimator, this implicit assumption must be
   considered.  An implementation suggestion is detailed in Appendix G.


{ 3.5. - former 3.4. - not changed }


Appendix G.  RTO calculation modification

   This document RECOMMENDS that the standard RTO calculation
   ([RFC6298]) is modified in the following way.  We roughly know how
   many samples a congestion window worth of data will yield, not
   accounting for ACK compression, and ACK losses.  Such events will
   result in more history of the path being reflected in the final value
   for RTO, and are uncritical.  This modification will approximate the
   RTO estimator described in [RFC6298], regardless how many samples are
   taken per window:

      ExpectedSamples =3D ceiling(FlightSize / (SMSS * 2))

      alpha' =3D alpha / ExpectedSamples

      beta' =3D beta / ExpectedSamples

   Note that the factor 2 in ExpectedSamples is due to "Delayed ACKs".

   Instead of using alpha and beta in the algorithm of [RFC6298], use
   alpha' and beta' instead:

      RTTVAR <- (1 - beta') * RTTVAR + beta' * |SRTT - R'|

      SRTT <- (1 - alpha') * SRTT + alpha' * R'

      (for each sample R')=20


From prvs=38475306d6=david.borman@quantum.com  Wed May 15 13:34:42 2013
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From: David Borman <David.Borman@quantum.com>
To: Richard Scheffenegger <rs@netapp.com>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
Thread-Index: AQHOUauYPnzg5qLOKEuqOpXGnU/i6Q==
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On Apr 29, 2013, at 4:05 AM, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote:

> Hi,
>=20
> WG chairs believe that the earlier outstanding issues have been addressed=
 in the latest version of 1323bis, and the document starts to be ready to b=
e submitted for publication. Therefore, this mail starts a working group la=
st call for draft-ietf-tcpm-1323bis-11 ("TCP Extensions for High Performanc=
e").
>=20
> The WG last call runs for two weeks, until Monday, May 13th. Please send =
any comments on the mailing list by then. Also notifications of approving t=
he current version (even without other comments) are helpful.
>=20
> The draft is available at http://tools.ietf.org/html/draft-ietf-tcpm-1323=
bis-11

I've been reviewing the latest version, -12.

First, I missed this earlier, but between -07 and -08, the "Timestamps" opt=
ion was renamed to the "Timestamp" option,  did I miss the discussion on ma=
king that change?  I'd like to see that put back to the original 1323 name.=
  It's plural because the option holds two timestamps.  Additionally, there=
 are many instances of "timestamp option" that should all be capitalized as=
 "Timestamps option."

Here are some other nits that I found:

Page 10:

  o  The window field (SEG.WND) of every outgoing segment, with the
     exception of <SYN> segments, is right-shifted by Rcv.Wind.Shift
     bits:

                   SND.WND =3D RCV.WND >> Rcv.Wind.Shift

That should be "SEG.WND =3D ..."


Page 14:
	... clock value provides a RTT measurement.
should be:
	... clock value provides an RTT measurement.


Page 13, 19:
Multiple instances of "a <RST>" should be "an <RST>"


Page 27:
The second bullet, the last 2 sentences:
        These middle boxes must also support the Window Scale option
        and apply the scaling when processing segments.  If the window
        scale cannot be determined, it must not do window based
        processing.
should be changed to to use "scale factor":
        These middle boxes must also support the Window Scale option
        and apply the scaling factor when processing segments.  If the scale
        factor cannot be determined, it must not do window based
        processing.


Page 31:
	"The following layouts are recommended for..."
should be:
	"The following layout is recommended for..."
(we only have one layout)

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From: "Scheffenegger, Richard" <rs@netapp.com>
To: David Borman <David.Borman@quantum.com>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Date: Wed, 15 May 2013 21:00:37 +0000
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Hi David,

> First, I missed this earlier, but between -07 and -08, the "Timestamps"
> option was renamed to the "Timestamp" option,  did I miss the discussion
> on making that change?  I'd like to see that put back to the original 132=
3
> name.  It's plural because the option holds two timestamps.  Additionally=
,
> there are many instances of "timestamp option" that should all be
> capitalized as "Timestamps option."

It wasn't consistent (but the majority was Timestamps);=20

I have reverted that change, but the text surrounding every instance should=
 be checked by a native speaker if the plual/singular form (will they apply=
 in plural to "timestamps" or singular to the "option"?) are consistent...


> Page 14:
> 	... clock value provides a RTT measurement.
> should be:
> 	... clock value provides an RTT measurement.
>=20
>=20
> Page 13, 19:
> Multiple instances of "a <RST>" should be "an <RST>"

Ok, after my simplistic approach of vowel vs. consonant doesn't work (appar=
ently, in American English, it's the initial sound rather than letter of a =
proper word, that counts), and the use of a vs an for acronyms seems to be =
a discussed topic (some count acronyms as proper words, others use the init=
ial word) - is there a guideline from the WG/IETF/RFC-Editor here?




Fixed all the other nits!

Thanks a lot!



Richard Scheffenegger


> -----Original Message-----
> From: David Borman [mailto:David.Borman@quantum.com]
> Sent: Mittwoch, 15. Mai 2013 22:34
> To: Scheffenegger, Richard
> Cc: tcpm (tcpm@ietf.org)
> Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
>=20
>=20
> On Apr 29, 2013, at 4:05 AM, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote=
:
>=20
> > Hi,
> >
> > WG chairs believe that the earlier outstanding issues have been
> addressed in the latest version of 1323bis, and the document starts to be
> ready to be submitted for publication. Therefore, this mail starts a
> working group last call for draft-ietf-tcpm-1323bis-11 ("TCP Extensions
> for High Performance").
> >
> > The WG last call runs for two weeks, until Monday, May 13th. Please sen=
d
> any comments on the mailing list by then. Also notifications of approving
> the current version (even without other comments) are helpful.
> >
> > The draft is available at http://tools.ietf.org/html/draft-ietf-tcpm-
> 1323bis-11
>=20
> I've been reviewing the latest version, -12.
>=20
> Here are some other nits that I found:
>=20
> Page 10:
>=20
>   o  The window field (SEG.WND) of every outgoing segment, with the
>      exception of <SYN> segments, is right-shifted by Rcv.Wind.Shift
>      bits:
>=20
>                    SND.WND =3D RCV.WND >> Rcv.Wind.Shift
>=20
> That should be "SEG.WND =3D ..."
>=20
>=20
> Page 14:
> 	... clock value provides a RTT measurement.
> should be:
> 	... clock value provides an RTT measurement.
>=20
>=20
> Page 13, 19:
> Multiple instances of "a <RST>" should be "an <RST>"
>=20
>=20
> Page 27:
> The second bullet, the last 2 sentences:
>         These middle boxes must also support the Window Scale option
>         and apply the scaling when processing segments.  If the window
>         scale cannot be determined, it must not do window based
>         processing.
> should be changed to to use "scale factor":
>         These middle boxes must also support the Window Scale option
>         and apply the scaling factor when processing segments.  If the
> scale
>         factor cannot be determined, it must not do window based
>         processing.
>=20
>=20
> Page 31:
> 	"The following layouts are recommended for..."
> should be:
> 	"The following layout is recommended for..."
> (we only have one layout)
>=20
> ----------------------------------------------------------------------
> The information contained in this transmission may be confidential. Any
> disclosure, copying, or further distribution of confidential information
> is not permitted unless such privilege is explicitly granted in writing b=
y
> Quantum. Quantum reserves the right to have electronic communications,
> including email and attachments, sent across its networks filtered throug=
h
> anti virus and spam software programs and retain such messages in order t=
o
> comply with applicable data security and retention requirements. Quantum
> is not responsible for the proper and complete transmission of the
> substance of this communication or for any delay in its receipt.

From michawe@ifi.uio.no  Thu May 16 04:32:37 2013
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Hi,

Sorry everyone for reacting to this one so late, I think it's a bit off =
track of the main discussion now - I'm answering anyway:


On 15. mai 2013, at 18:32, Mark Allman wrote:

>=20
> Michael-
>=20
>> I may be wrong, but I think that the behavior with more samples +
>> changed alpha and beta as you suggested can be significantly
>> different.=20
>=20
> Your comparison is not sound because you are conflating two things: =
(1)
> adjusted gains and (2) multiple samples.

No, that was in fact the point I was trying to make...


> So, yes, you find the RTOs that pop out are different.  E.g., in the
> case of one sample per RTT does that get the max sample or the min
> sample during the given RTT?  It matters to the result.  Clearly =
taking
> 1 and taking N are going to be different.  The argument behind RTTM is
> that this difference is important, i.e., more samples gives you a more
> accurate assessment of the RTT process by taking into account the
> fluctuations that can be observed within an RTT.  Our paper / data
> suggests this is wrong.

I should look at this paper again, I'm sure I read it long ago. Anyway, =
it seems logical that using more data cannot be a bad thing - I do agree =
with the hand-wavery in the original RFC1323 in that respect. If that =
turns out to lead to poor behavior, it tells us that the process behind =
it (using EWMA, and EWMA+4*EWMA(Var)) is probably not so great.

But that's a different story altogether, I don't want to suggest =
updating the RTO calculation here... just saying that this notion of =
"using less information is better" is kinda odd.


> All I am saying is that the length of the history kept in the EWMA
> should be roughly the same regardless of the number of samples taken.

Yeah, I got the idea.


> If you do the same test and keep a constant RTT for each RTT you'll =
see
> that the RTOs that pop out are roughly the same.
>=20
> I.e., pick a series of RTT measurements for each round of the =
transfer.
> I dunno, say 0.1, 0.11, 0.09, 0.85, 0.13 for the first five RTTs.  =
Now,
> use the RFC6298 gains and run each of those samples through the RTO
> calculation once.  Call that X.  Now, pick some number of samples N =
per
> RTT and run each of the above numbers through the RTO calculation N
> times, but with alpha/N and beta/N as the gains (i.e., you are
> simulating multiple samples that are all the same; that isn't =
realistic,
> but shows the math).  Call the last RTO Y.  What you'll find is that =
the
> X and Y are pretty similar.  I.e., we have adjusted the gains to keep
> roughly the same amount of history in the overall RTO.  And, that is =
the
> point.

I see what you're trying to achieve, but I don't think the method works =
out that well. With init. R=3D0.1 and assuming that the first =
measurement was already taken, i.e. SRTT=3D0.1, RTTVAR=3D0.05, RTO=3D0.3 =
N =3D 1, after 5 RTTs using the RTT values you give above, with N=3D1 =
the RTO is 0,87, and with N=3D3 it is 0,81. I did another calculation =
where I assumed that these values appear after the RTO has stabilized, =
i.e. initial RTTVAR=3D0. Then, with N=3D1, you get 0,822438965 and with =
N=3D3 you get 0,752924008. I don't know if that would qualify to be =
"roughly the same". I think the problem is that a EWMA with halved alpha =
isn't quite the same as doubling an SMA, and the error is probably =
magnified by the 4*RTTVAR.

So this was only the favorable case of equal values in every RTT and =
just some random numbers you provided for RTT samples... who knows how =
far off things could get in reality. Essentially, the one-sample-per-RTT =
estimation assumes that it doesn't matter whatever sample you pick =
within an RTT. Due to the abs() function in RTO calculation, any =
deviation from SRTT will make the RTO grow, and if you pick one sample =
in an RTT, chances are that this gives you more variation than the =
average of all the measurements you could take in an RTT, and hence a =
more conservative -- "better working" -- RTO. Duh! :-)  Averaging over =
that is going to give you something else.

So I don't have the impression that it's valid to say that this is =
roughly the same, but by all means, let's do it! It's going to flatten =
out some outliers, and hence make the RTO more aggressive, great! Maybe =
then it's even going to fire spuriously more often - all the better, =
then we have a reason to fix it   :-)))

Cheers,
Michael

PS: seriously, I do believe that the problem with the RTO is in the =
trade-off that it entails: too short =3D horrible things happen. too =
long =3D inefficient. What we should have is a more aggressive timer for =
retransmitting with less serious consequences, and then we can make the =
RTO as conservative as we wish. TLP has this timer, I guess I'm arguing =
for applying it in general, not just to the last 2 packets of a window =
or so.

PPS: just out of historical curiosity, whatever was the rationale for =
the number 4 in "+4*RTTAVR"? Was it the number the dice showed?


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> > Your comparison is not sound because you are conflating two things:
> > (1) adjusted gains and (2) multiple samples.
> 
> No, that was in fact the point I was trying to make...

What I am saying is that there are two questions---one of whether
running multiple samples per RTT helps and one of how long the history
should be---here.  You are trying to answer both of them at the same
time and I am focused in this discussion only on one of the
questions---keeping the history similar irrespective of the number of
RTTs.  So, we're talking a bit of apples and oranges here.

> just saying that this notion of "using less information is better" is
> kinda odd. 

Re: multiple samples per RTT: it isn't that you come up with a better
estimator with one sample, it is that you do not come up with a better
estimator with multiple samples.  They are similar estimators.  (And,
one takes more work, has more on-the-wire overhead, etc.  So, the
"better" is not in terms of the result, but result-for-the-effort.)

> > I.e., pick a series of RTT measurements for each round of the
> > transfer.  I dunno, say 0.1, 0.11, 0.09, 0.85, 0.13 for the first
> > five RTTs.  Now, use the RFC6298 gains and run each of those samples
> > through the RTO calculation once.  Call that X.  Now, pick some
> > number of samples N per RTT and run each of the above numbers
> > through the RTO calculation N times, but with alpha/N and beta/N as
> > the gains (i.e., you are simulating multiple samples that are all
> > the same; that isn't realistic, but shows the math).  Call the last
> > RTO Y.  What you'll find is that the X and Y are pretty similar.
> > I.e., we have adjusted the gains to keep roughly the same amount of
> > history in the overall RTO.  And, that is the point.
> 
> I see what you're trying to achieve, but I don't think the method
> works out that well. 

It does.  I have done the calculations.  I didn't check your work, but
two things ...

  - The history applies to the EWMAs.  So, doing the entire RTO
    calculation makes things look a little worse because the discrepancy
    in the RTTVAR term is multiplied (as you noted).

  - I think that 0.87 & 0.81 (your numbers) are in fact pretty close.
    That is 60msec difference on a number that exceeds 800msec.  I never
    claimed these things were *exactly* equivalent.  But, just chopping
    the gain by N is *approximately* right.  I am sure if you wanted to
    have the samples from the present RTT count the *exact* same in
    terms of history irrespective of the number of samples taken then
    you could sit down with a pencil and figure out how to adjust the
    gain to make that happen.  And, hey, maybe that is a good exercise.
    But, to me the simple division of the gain by N looks pretty close
    and so it doesn't seem terribly important to make things more
    difficult by making it more complex.  But, eh, if you want to do it,
    we can have the discussion.

> So this was only the favorable case of equal values in every RTT and
> just some random numbers you provided for RTT samples... who knows how
> far off things could get in reality.

Again, the "favorable case" is meant to illustrate that the history is
roughly the same the regardless of the number of RTT samples if you make
a straightforward tweak to the gains.  The illustration was meant to
show you that the math works as I have characterized it.

The illustration is absolutely far too simplistic to comment on the
efficacy of the resulting RTOs.  We have published an empirical study to
deal with that question.

> PS: seriously, I do believe that the problem with the RTO is in the
> trade-off that it entails: too short = horrible things happen. too
> long = inefficient.

Right.  And, the tradeoff is fundamental.  We could not find a sweet
spot that optimized for both less waiting time and fewer spurious
retransmits.

> PPS: just out of historical curiosity, whatever was the rationale for
> the number 4 in "+4*RTTAVR"? Was it the number the dice showed? 

I think this is discussed in Jacobson's 1988 paper.  But, I cannot at
the moment conjure the rationale.

allman




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From michawe@ifi.uio.no  Thu May 16 05:51:23 2013
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Hi,


On 16. mai 2013, at 14:12, Mark Allman wrote:

>=20
>>> Your comparison is not sound because you are conflating two things:
>>> (1) adjusted gains and (2) multiple samples.
>>=20
>> No, that was in fact the point I was trying to make...
>=20
> What I am saying is that there are two questions---one of whether
> running multiple samples per RTT helps and one of how long the history
> should be---here.  You are trying to answer both of them at the same
> time and I am focused in this discussion only on one of the
> questions---keeping the history similar irrespective of the number of
> RTTs.  So, we're talking a bit of apples and oranges here.

well, my argument is that your focus leaves out an important aspect of =
reality - the possibility of e.g. two packets in one RTT experiencing a =
different RTT.
It's like saying "if we assume that this doesn't happen, the values are =
roughly the same". Fine, as a model, where we can make all kinds of =
assumptions all the time, but you started out saying that this gives you =
roughly the same numbers, and I'm saying that only in your perfect model =
world, it would (plus there's my less relevant (because less =
significant) criticism of the actual process below).


>> just saying that this notion of "using less information is better" is
>> kinda odd.=20
>=20
> Re: multiple samples per RTT: it isn't that you come up with a better
> estimator with one sample, it is that you do not come up with a better
> estimator with multiple samples.  They are similar estimators.  (And,
> one takes more work, has more on-the-wire overhead, etc.  So, the
> "better" is not in terms of the result, but result-for-the-effort.)

ok, I can see that


>>> I.e., pick a series of RTT measurements for each round of the
>>> transfer.  I dunno, say 0.1, 0.11, 0.09, 0.85, 0.13 for the first
>>> five RTTs.  Now, use the RFC6298 gains and run each of those samples
>>> through the RTO calculation once.  Call that X.  Now, pick some
>>> number of samples N per RTT and run each of the above numbers
>>> through the RTO calculation N times, but with alpha/N and beta/N as
>>> the gains (i.e., you are simulating multiple samples that are all
>>> the same; that isn't realistic, but shows the math).  Call the last
>>> RTO Y.  What you'll find is that the X and Y are pretty similar.
>>> I.e., we have adjusted the gains to keep roughly the same amount of
>>> history in the overall RTO.  And, that is the point.
>>=20
>> I see what you're trying to achieve, but I don't think the method
>> works out that well.=20
>=20
> It does.  I have done the calculations.  I didn't check your work, but
> two things ...
>=20
>  - The history applies to the EWMAs.  So, doing the entire RTO
>    calculation makes things look a little worse because the =
discrepancy
>    in the RTTVAR term is multiplied (as you noted).

Indeed! I also looked at the pure EWMA values, which indeed (as =
suspected) look better, but that's just not what it is.


>  - I think that 0.87 & 0.81 (your numbers) are in fact pretty close.
>    That is 60msec difference on a number that exceeds 800msec.  I =
never

Pretty close for N=3D3, which was a random number I picked, and for the =
RTT values you've given.


>    claimed these things were *exactly* equivalent.  But, just chopping
>    the gain by N is *approximately* right.  I am sure if you wanted to
>    have the samples from the present RTT count the *exact* same in
>    terms of history irrespective of the number of samples taken then
>    you could sit down with a pencil and figure out how to adjust the
>    gain to make that happen.  And, hey, maybe that is a good exercise.
>    But, to me the simple division of the gain by N looks pretty close
>    and so it doesn't seem terribly important to make things more
>    difficult by making it more complex.  But, eh, if you want to do =
it,
>    we can have the discussion.

Nah, let's go with what you suggest and see what happens. I don't buy =
the "roughly similar" bit mainly because of the sampling issue above, =
but who cares. Small is beautiful.


>> So this was only the favorable case of equal values in every RTT and
>> just some random numbers you provided for RTT samples... who knows =
how
>> far off things could get in reality.
>=20
> Again, the "favorable case" is meant to illustrate that the history is
> roughly the same the regardless of the number of RTT samples if you =
make
> a straightforward tweak to the gains.  The illustration was meant to
> show you that the math works as I have characterized it.
>=20
> The illustration is absolutely far too simplistic to comment on the
> efficacy of the resulting RTOs.  We have published an empirical study =
to
> deal with that question.

>=20
>> PS: seriously, I do believe that the problem with the RTO is in the
>> trade-off that it entails: too short =3D horrible things happen. too
>> long =3D inefficient.
>=20
> Right.  And, the tradeoff is fundamental.  We could not find a sweet
> spot that optimized for both less waiting time and fewer spurious
> retransmits.

... and you left out my suggestion to incorporate a second, more =
aggressive timer, in TLP style. What do you think, wouldn't that be =
good?
(yes I know - it's research... do the work first... but well, the Google =
folks have started, with TLP)

Cheers,
Michael


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> well, my argument is that your focus leaves out an important aspect of
> reality - the possibility of e.g. two packets in one RTT experiencing
> a different RTT. 

No.

Not at all.

You are completely missing the point.  

Or, trying to cloud it.

We use EWMAs to incorporate both the new sample at some weight and also
some sense of history of what the samples have been in the past.  

Consider three cases ... 

(1) In RTT#N using one sample per RTT the RTT sample will contribute 1/8
    to the SRTT EWMA (alpha in RFC6298).  And, the SRTT history from
    previous rounds will contribute 7/8.

(2) Now consider taking two samples in RTT#N.  Each sample will
    contribute 1/8 to the SRTT EWMA.  So, the total contribution from
    RTT#N is roughly 1/4.  And, the historical contribution is 3/4.

(3) Now consider taking two samples in RTT#N and also adjusting the
    gains to take this into account.  So, we use alpha'=(1/8)/2=1/16.
    So, each of the samples contributes 1/16 to the SRTT.  Together the
    two samples contribute about 1/8 to the SRTT.  And, the historical
    SRTT contributes about 7/8.

All I am saying is that case (2) ages the history out more rapidly and
that (1) and (3) provide roughly the same history in the EWMA.

I am not claiming that the math dictates the RTO (or the SRTT or the
RTTVAR) is the same if you take multiple real-world RTT samples per
RTT.  What I am claiming is that with the gain adjustment we retain
roughly the same amount of history (in terms of time span) in the
estimator regardless of the number of samples we take per RTT because
each round is weighted roughly the same.

None of that says the RTO will be the same with multiple samples.  None
of that means I am living in some fantasy world.  None of that means I
don't appreciate that multiple samples might change the RTO.  All it
says is that a simple adjustment to the gains means that the
contribution to the EWMA is roughly the same per RTT irrespective of the
number of samples.

allman




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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, David Borman <David.Borman@quantum.com>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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On 16. mai 2013, at 15:12, Mark Allman wrote:

>=20
>> well, my argument is that your focus leaves out an important aspect =
of
>> reality - the possibility of e.g. two packets in one RTT experiencing
>> a different RTT.=20
>=20
> No.
>=20
> Not at all.
>=20
> You are completely missing the point. =20
>=20
> Or, trying to cloud it.

Missing, perhaps, trying to cloud it, definitely not. Why in the world =
would I?


> We use EWMAs to incorporate both the new sample at some weight and =
also
> some sense of history of what the samples have been in the past. =20
>=20
> Consider three cases ...=20
>=20
> (1) In RTT#N using one sample per RTT the RTT sample will contribute =
1/8
>    to the SRTT EWMA (alpha in RFC6298).  And, the SRTT history from
>    previous rounds will contribute 7/8.
>=20
> (2) Now consider taking two samples in RTT#N.  Each sample will
>    contribute 1/8 to the SRTT EWMA.  So, the total contribution from
>    RTT#N is roughly 1/4.  And, the historical contribution is 3/4.
>=20
> (3) Now consider taking two samples in RTT#N and also adjusting the
>    gains to take this into account.  So, we use alpha'=3D(1/8)/2=3D1/16.=

>    So, each of the samples contributes 1/16 to the SRTT.  Together the
>    two samples contribute about 1/8 to the SRTT.  And, the historical
>    SRTT contributes about 7/8.

All of that is correct, and I didn't question any of that;


> All I am saying is that case (2) ages the history out more rapidly and
> that (1) and (3) provide roughly the same history in the EWMA.

but my interpretation of what you said (see below) is that the result =
(which is the RTO) is roughly the same. What you say here is more =
limiting. Sorry, I didn't want to put words in your mouth. Anyway, I'm =
concerned about the case where the two samples in (3) are different, =
which does not at all seem unlikely to me. Then an RTO derived from (1) =
and (3) can become quite different too.


> I am not claiming that the math dictates the RTO (or the SRTT or the
> RTTVAR) is the same if you take multiple real-world RTT samples per
> RTT.  What I am claiming is that with the gain adjustment we retain
> roughly the same amount of history (in terms of time span) in the
> estimator regardless of the number of samples we take per RTT because
> each round is weighted roughly the same.
>=20
> None of that says the RTO will be the same with multiple samples.  =
None
> of that means I am living in some fantasy world.  None of that means I
> don't appreciate that multiple samples might change the RTO.  All it
> says is that a simple adjustment to the gains means that the
> contribution to the EWMA is roughly the same per RTT irrespective of =
the
> number of samples.

This is what you now say  :-)   in your original email, you said:

  That isn't **exactly** the same as RFC6298, but it is absolutely
   **almost** the same.  And, it is absolutely within the spirit of
   TCP's standardized RTO.  I.e., we keep about the same amount of
   history and we weight the recent RTT samples with about the same
   weight.

   So, I don't see why 1323bis shouldn't say "if you use this RTTM we
   codify then you make the following adjustments to TCP's standard RTO
   procedure [...insert step (5) here...].  To me that is well


Granted, "That" may be various things, but I thought it refers to the =
overall calculation and, with it, the result you get from it, which is =
the RTO. =46rom this text and your recommendation for 1323bis, I took it =
as "the resulting RTO is roughly the same". Now you limit that =
statement, but that also weakens the argument for changing 1323bis in =
this way, I'd say.

Anyway, as I said, you have my vote for changing it, because it makes =
the RTO less aggressive by smoothing out stuff. Within the limits of =
avoiding to change the actual calculation method, I do see it as a =
reasonable approach.

Cheers,
Michael


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> > We use EWMAs to incorporate both the new sample at some weight and also
> > some sense of history of what the samples have been in the past.  
> > 
> > Consider three cases ... 
> > 
> > (1) In RTT#N using one sample per RTT the RTT sample will contribute 1/8
> >    to the SRTT EWMA (alpha in RFC6298).  And, the SRTT history from
> >    previous rounds will contribute 7/8.
> > 
> > (2) Now consider taking two samples in RTT#N.  Each sample will
> >    contribute 1/8 to the SRTT EWMA.  So, the total contribution from
> >    RTT#N is roughly 1/4.  And, the historical contribution is 3/4.
> > 
> > (3) Now consider taking two samples in RTT#N and also adjusting the
> >    gains to take this into account.  So, we use alpha'=(1/8)/2=1/16.
> >    So, each of the samples contributes 1/16 to the SRTT.  Together the
> >    two samples contribute about 1/8 to the SRTT.  And, the historical
> >    SRTT contributes about 7/8.
> 
> All of that is correct, and I didn't question any of that;
> 
> 
> > All I am saying is that case (2) ages the history out more rapidly and
> > that (1) and (3) provide roughly the same history in the EWMA.
> 
> but my interpretation of what you said (see below) is that the result
> (which is the RTO) is roughly the same. 

No- I did not mean to say the RTO would be the same.  I don't think I
said that---even in re-reading what you pasted in.  If what I wrote was
unclear to you, sorry.  All I am getting at is we should include roughly
the same amount of history in the SRTT and RTTVAR (and hence the RTO).

> Then an RTO derived from
> (1) and (3) can become quite different too. 

Sure.  And, that is the original point of RTTM.  I.e., the claim was
that by taking multiple samples the resulting RTO would be better.  Our
data suggests this hypothesis is not true.

(And, note, that even two RTOs that are "different" might work
similarly.  E.g., say one is X and one is 2X.  If neither ever triggers
a retransmission then we don't really see a difference in the
performance even if the numbers are in fact different.  But, when we do
see retransmissions---needed or not---then we don't see much difference
in the estimators.)

allman




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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>, Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-1323bis
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Hi,


>=20
> Sure.  And, that is the original point of RTTM.  I.e., the claim was that
> by taking multiple samples the resulting RTO would be better.  Our data
> suggests this hypothesis is not true.

Within the confines of the specified linear EWMA calculation given in RFC62=
98. (Which is also the implicit assumption in this discussion, correct?).



>> just saying that this notion of "using less information is better" is=20
>> kinda odd.
>=20
> Re: multiple samples per RTT: it isn't that you come up with a better=20
> estimator with one sample, it is that you do not come up with a better=20
> estimator with multiple samples.  They are similar estimators.  (And,=20
> one takes more work, has more on-the-wire overhead, etc.  So, the=20
> "better" is not in terms of the result, but result-for-the-effort.)

I looked at a couple of papers about TCP/RTO in the past days, to refresh m=
y memory.=20

I think the relevance for 1323bis is limited to stating the fact, that RFC6=
298 RTO constants work for 1 sample per RTT, and when taking more samples p=
er RTT, either the RTO calculation method needs to be adjusted (huge volume=
 of papers on that, none at the IETF), or the RFC6298 gains need to fixed t=
o keep approximately similar history (current text in the Appendix of 1323b=
is), or throw out most of the additional information gained by the RTT samp=
les, and run the legacy RTO calculation (e.g. what Linux does).


Taking a step back, I think since the turn of the century, additional mecha=
nisms that unwind the "catastrophic" effects of a spurious RTO (F-RTO, Eife=
l Response, DSACK) are widely deployed, thus a RTO on the more aggressive s=
ide (such as not adjusting the gains and sampling nearly every segment) no =
longer have as dire consequences as they used to have.

This is also, why I would suggest to disentangle in 1323bis the RTT samplin=
g, from the specific RTO calculation method, to allow a higher frequency RT=
TM signal become available for other mechanisms as well.


Best regards,
  Richard




From pasi.sarolahti@iki.fi  Fri May 17 02:48:08 2013
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Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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On May 15, 2013, at 10:06 PM, "Scheffenegger, Richard" <rs@netapp.com> =
wrote:

> Appendix G.  RTO calculation modification
>=20
>   This document RECOMMENDS that the standard RTO calculation

RECOMMENDS is not a RFC 2119 keyword. "RECOMMENDED" would be, but my =
preference would rather be just not to recommend anything, nor have RFC =
2119 keywords in Appendix. Just state the fact that taking multiple =
samples in RTT would shorten the window of history calculated in RFC =
6298, and the below algorithm aims to maintain similar window than =
originally intended by RFC 6298.

- Pasi


>   ([RFC6298]) is modified in the following way.  We roughly know how
>   many samples a congestion window worth of data will yield, not
>   accounting for ACK compression, and ACK losses.  Such events will
>   result in more history of the path being reflected in the final =
value
>   for RTO, and are uncritical.  This modification will approximate the
>   RTO estimator described in [RFC6298], regardless how many samples =
are
>   taken per window:
>=20
>      ExpectedSamples =3D ceiling(FlightSize / (SMSS * 2))
>=20
>      alpha' =3D alpha / ExpectedSamples
>=20
>      beta' =3D beta / ExpectedSamples
>=20
>   Note that the factor 2 in ExpectedSamples is due to "Delayed ACKs".
>=20
>   Instead of using alpha and beta in the algorithm of [RFC6298], use
>   alpha' and beta' instead:
>=20
>      RTTVAR <- (1 - beta') * RTTVAR + beta' * |SRTT - R'|
>=20
>      SRTT <- (1 - alpha') * SRTT + alpha' * R'
>=20
>      (for each sample R')=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Date: Fri, 17 May 2013 18:54:22 +0000
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Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Hi Pasi,

I've taken your text donation and updated the appendix as follows:


Appendix G.  RTO calculation modification

   Taking multiple RTT samples per window would shorten the history
   calculated by the RTO mechanism in [RFC6298], and the below algorithm
   aims to maintain a similar history as originally intended by
   [RFC6298].

   We roughly know how many samples a congestion window worth of data
   will yield, not accounting for ACK compression, and ACK losses.  Such
   events will result in more history of the path being reflected in the
   final value for RTO, and are uncritical.  This modification will
   approximate the RTO estimator described in [RFC6298], regardless how
   many samples are taken per window:

      ExpectedSamples =3D ceiling(FlightSize / (SMSS * 2))

      alpha' =3D alpha / ExpectedSamples

      beta' =3D beta / ExpectedSamples

   Note that the factor 2 in ExpectedSamples is due to "Delayed ACKs".

   Instead of using alpha and beta in the algorithm of [RFC6298], use
   alpha' and beta' instead:

      RTTVAR <- (1 - beta') * RTTVAR + beta' * |SRTT - R'|

      SRTT <- (1 - alpha') * SRTT + alpha' * R'

      (for each sample R')



Richard Scheffenegger


> -----Original Message-----
> From: Pasi Sarolahti [mailto:pasi.sarolahti@iki.fi]
> Sent: Freitag, 17. Mai 2013 11:48
> To: Scheffenegger, Richard
> Cc: mallman@icir.org; tcpm (tcpm@ietf.org); David Borman
> (David.Borman@quantum.com)
> Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
>=20
> On May 15, 2013, at 10:06 PM, "Scheffenegger, Richard" <rs@netapp.com>
> wrote:
>=20
> > Appendix G.  RTO calculation modification
> >
> >   This document RECOMMENDS that the standard RTO calculation
>=20
> RECOMMENDS is not a RFC 2119 keyword. "RECOMMENDED" would be, but my
> preference would rather be just not to recommend anything, nor have RFC
> 2119 keywords in Appendix. Just state the fact that taking multiple
> samples in RTT would shorten the window of history calculated in RFC 6298=
,
> and the below algorithm aims to maintain similar window than originally
> intended by RFC 6298.
>=20
> - Pasi
>=20
>=20
> >   ([RFC6298]) is modified in the following way.  We roughly know how
> >   many samples a congestion window worth of data will yield, not
> >   accounting for ACK compression, and ACK losses.  Such events will
> >   result in more history of the path being reflected in the final value
> >   for RTO, and are uncritical.  This modification will approximate the
> >   RTO estimator described in [RFC6298], regardless how many samples are
> >   taken per window:
> >
> >      ExpectedSamples =3D ceiling(FlightSize / (SMSS * 2))
> >
> >      alpha' =3D alpha / ExpectedSamples
> >
> >      beta' =3D beta / ExpectedSamples
> >
> >   Note that the factor 2 in ExpectedSamples is due to "Delayed ACKs".
> >
> >   Instead of using alpha and beta in the algorithm of [RFC6298], use
> >   alpha' and beta' instead:
> >
> >      RTTVAR <- (1 - beta') * RTTVAR + beta' * |SRTT - R'|
> >
> >      SRTT <- (1 - alpha') * SRTT + alpha' * R'
> >
> >      (for each sample R')
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm


From michawe@ifi.uio.no  Fri May 17 13:28:37 2013
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Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Hi,


On May 17, 2013, at 8:54 PM, "Scheffenegger, Richard" <rs@netapp.com> =
wrote:

> Hi Pasi,
>=20
> I've taken your text donation and updated the appendix as follows:
>=20
>=20
> Appendix G.  RTO calculation modification
>=20
>   Taking multiple RTT samples per window would shorten the history
>   calculated by the RTO mechanism in [RFC6298], and the below =
algorithm
>   aims to maintain a similar history as originally intended by
>   [RFC6298].
>=20
>   We roughly know how many samples a congestion window worth of data
>   will yield, not accounting for ACK compression, and ACK losses.  =
Such
>   events will result in more history of the path being reflected in =
the
>   final value for RTO, and are uncritical.  This modification will
>   approximate the RTO estimator described in [RFC6298], regardless how
>   many samples are taken per window:

Sorry, but I disagree with "will approximate the RTO estimator described =
in=85"
=3D> I haven't seen any strong indication that this is really valid to =
say, and Mark has made it clear (in my understanding) that he never said =
this.

I suggest: "This modification will ensure that a similar amount of time =
is taken into account for the RTO estimation, regardless of how many =
samples are taken per window:"

Cheers,
Michael


From internet-drafts@ietf.org  Sat May 18 08:58:16 2013
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : TCP Extensions for High Performance
	Author(s)       : David Borman
                          Bob Braden
                          Van Jacobson
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-1323bis-13.txt
	Pages           : 46
	Date            : 2013-05-18

Abstract:
   This document specifies a set of TCP extensions to improve
   performance over paths with a large bandwidth * delay product and to
   provide reliable operation over very high-speed paths.  It defines
   TCP options for scaled windows and timestamps.  The timestamps are
   used for two distinct mechanisms, RTTM (Round Trip Time Measurement)
   and PAWS (Protection Against Wrapped Sequences).

   This document updates and obsoletes RFC 1323.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-13

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-1323bis-13


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


From rs@netapp.com  Sat May 18 09:45:04 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, "David Borman \(David.Borman@quantum.com\)" <David.Borman@quantum.com>, "mallman@icir.org" <mallman@icir.org>
Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
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Thanks Michael,

I've also updated the auxiliary stuff (changes section), and submitted a ne=
w version.


http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-ietf-t=
cpm-1323bis-11.txt&url2=3Dhttp://tools.ietf.org/id/draft-ietf-tcpm-1323bis-=
13.txt

Richard Scheffenegger


> -----Original Message-----
> From: Michael Welzl [mailto:michawe@ifi.uio.no]
> Sent: Freitag, 17. Mai 2013 22:28
> To: Scheffenegger, Richard
> Cc: Pasi Sarolahti; tcpm (tcpm@ietf.org); David Borman
> (David.Borman@quantum.com); mallman@icir.org
> Subject: Re: [tcpm] Updated Section 3 of draft-ietf-tcpm-1323bis
>=20
> Hi,
>=20
>=20
> On May 17, 2013, at 8:54 PM, "Scheffenegger, Richard" <rs@netapp.com>
> wrote:
>=20
> > Hi Pasi,
> >
> > I've taken your text donation and updated the appendix as follows:
> >
> >
> > Appendix G.  RTO calculation modification
> >
> >   Taking multiple RTT samples per window would shorten the history
> >   calculated by the RTO mechanism in [RFC6298], and the below algorithm
> >   aims to maintain a similar history as originally intended by
> >   [RFC6298].
> >
> >   We roughly know how many samples a congestion window worth of data
> >   will yield, not accounting for ACK compression, and ACK losses.  Such
> >   events will result in more history of the path being reflected in the
> >   final value for RTO, and are uncritical.  This modification will
> >   approximate the RTO estimator described in [RFC6298], regardless how
> >   many samples are taken per window:
>=20
> Sorry, but I disagree with "will approximate the RTO estimator described
> in..."
> =3D> I haven't seen any strong indication that this is really valid to sa=
y,
> and Mark has made it clear (in my understanding) that he never said this.
>=20
> I suggest: "This modification will ensure that a similar amount of time i=
s
> taken into account for the RTO estimation, regardless of how many samples
> are taken per window:"
>=20
> Cheers,
> Michael


From ycheng@google.com  Mon May 20 12:13:57 2013
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 20 May 2013 12:13:31 -0700
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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I suggest removing all these texts;

"""If a non-<RST> segment is received without a TSopt, a TCP MAY drop
the segment and send an <ACK> for the last in-sequence segment.  ""
What's the point to ignore a perfectly fine data packet w/o TS-opt and
respond a (DUP)ACK? the sender is probably may trigger fast recovery
falsely. Responding ACKs is not going to make any forward progress  on
buggy implementations or middle-boxes.

""" A TCP MUST NOT abort a TCP connection if a non-<RST> segment is
received without a TSopt. """
TCP should abort only if it's truly in trouble. so let's not
re-emphasize the general rule of trying to keep things going until the
last minute.

"""If a TSopt is received on a connection where TSopt was not
negotiated in the initial three-way handshake, the TSopt MUST be
ignored and the packet processed normally.""" there was an RFC that
states this general rule already?

"""   In the case of crossing <SYN> segments where one <SYN> contains
a TSopt and the other doesn't, both sides MAY send a TSopt in the
<SYN,ACK> segment. """ The prior text on SYN already covers this.

In general I found section 3 has redundant MUST/SHOULD/MAY terms. An
RFC should be parsimonious on these terms so implementer can focus on
key rules.



On Sat, May 18, 2013 at 8:57 AM,  <internet-drafts@ietf.org> wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.
>
>         Title           : TCP Extensions for High Performance
>         Author(s)       : David Borman
>                           Bob Braden
>                           Van Jacobson
>                           Richard Scheffenegger
>         Filename        : draft-ietf-tcpm-1323bis-13.txt
>         Pages           : 46
>         Date            : 2013-05-18
>
> Abstract:
>    This document specifies a set of TCP extensions to improve
>    performance over paths with a large bandwidth * delay product and to
>    provide reliable operation over very high-speed paths.  It defines
>    TCP options for scaled windows and timestamps.  The timestamps are
>    used for two distinct mechanisms, RTTM (Round Trip Time Measurement)
>    and PAWS (Protection Against Wrapped Sequences).
>
>    This document updates and obsoletes RFC 1323.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-1323bis
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-13
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-1323bis-13
>
>
> 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 touch@isi.edu  Mon May 20 13:11:40 2013
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On 5/20/2013 12:13 PM, Yuchung Cheng wrote:
> I suggest removing all these texts;
>
> """If a non-<RST> segment is received without a TSopt, a TCP MAY drop
> the segment and send an <ACK> for the last in-sequence segment.  ""
> What's the point to ignore a perfectly fine data packet w/o TS-opt and
> respond a (DUP)ACK? the sender is probably may trigger fast recovery
> falsely. Responding ACKs is not going to make any forward progress  on
> buggy implementations or middle-boxes.
>
> """ A TCP MUST NOT abort a TCP connection if a non-<RST> segment is
> received without a TSopt. """
> TCP should abort only if it's truly in trouble. so let's not
> re-emphasize the general rule of trying to keep things going until the
> last minute.

The text currently says:

    Once TSopt has been successfully negotiated (sent and received)
    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
    non-<RST> segment for the duration of the connection, and SHOULD be
    sent in a <RST> segment (see Section 4.2 for details).  If a non-
    <RST> segment is received without a TSopt, a TCP MAY drop the segment
    and send an <ACK> for the last in-sequence segment.  A TCP MUST NOT
    abort a TCP connection if a non-<RST> segment is received without a
    TSopt.

This wording leaves a few ambiguities:

- The first MUST implies some sort of uniform non-RST non-TSopt 
processing, but the rest of the paragraph doesn't provide it.

- The last sentence implies that receipt of a RST non-TSopt could cause 
an abort because of the lack of TSopt; it is because 'abort' is the 
consequence of processed RST.

I suggest clarifying this as:

    Once TSopt has been successfully negotiated (sent and received)
    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
    non-<RST> segment for the duration of the connection, and SHOULD be
    sent in a <RST> segment (see Section 4.2 for details).  If a non-
    <RST> segment is received without a TSopt, a TCP MUST drop the
    segment
    and MAY also send an <ACK> for the last in-sequence segment.
    A TCP MUST NOT abort a TCP connection because any segment lacks
    an expected TSopt.

Joe

From Alexander.Zimmermann@netapp.com  Tue May 21 03:03:25 2013
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: New Version Notification for draft-zimmermann-tcpm-tcp-rfc4614bis-02.txt
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Hi,

a new version of the TCP roadmap is available:

* Incorporate feedback from Wes, Richard and Oliver (thanks to all)
* Introduction for several sections (thanks Wes!)
* Including all new RFCs

Alex

--

A new version of I-D, draft-zimmermann-tcpm-tcp-rfc4614bis-02.txt
has been successfully submitted by Martin Duke and posted to the
IETF repository.

Filename:	 draft-zimmermann-tcpm-tcp-rfc4614bis
Revision:	 02
Title:		 A Roadmap for Transmission Control Protocol (TCP) Specification Do=
cuments
Creation date:	 2013-05-21
Group:		 Individual Submission
Number of pages: 48
URL:             http://www.ietf.org/internet-drafts/draft-zimmermann-tcpm-=
tcp-rfc4614bis-02.txt
Status:          http://datatracker.ietf.org/doc/draft-zimmermann-tcpm-tcp-=
rfc4614bis
Htmlized:        http://tools.ietf.org/html/draft-zimmermann-tcpm-tcp-rfc46=
14bis-02
Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-zimmermann-tcpm-t=
cp-rfc4614bis-02

Abstract:
 This document contains a "roadmap" to the Requests for Comments (RFC)
 documents relating to the Internet's Transmission Control Protocol
 (TCP).  This roadmap provides a brief summary of the documents
 defining TCP and various TCP extensions that have accumulated in the
 RFC series.  This serves as a guide and quick reference for both TCP
 implementers and other parties who desire information contained in
 the TCP-related RFCs.




The IETF Secretariat

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On Mon, May 20, 2013 at 1:10 PM, Joe Touch <touch@isi.edu> wrote:
>
>
>
> On 5/20/2013 12:13 PM, Yuchung Cheng wrote:
>>
>> I suggest removing all these texts;
>>
>> """If a non-<RST> segment is received without a TSopt, a TCP MAY drop
>> the segment and send an <ACK> for the last in-sequence segment.  ""
>> What's the point to ignore a perfectly fine data packet w/o TS-opt and
>> respond a (DUP)ACK? the sender is probably may trigger fast recovery
>> falsely. Responding ACKs is not going to make any forward progress  on
>> buggy implementations or middle-boxes.
>>
>> """ A TCP MUST NOT abort a TCP connection if a non-<RST> segment is
>> received without a TSopt. """
>> TCP should abort only if it's truly in trouble. so let's not
>> re-emphasize the general rule of trying to keep things going until the
>> last minute.
>
>
> The text currently says:
>
>    Once TSopt has been successfully negotiated (sent and received)
>    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>    non-<RST> segment for the duration of the connection, and SHOULD be
>    sent in a <RST> segment (see Section 4.2 for details).  If a non-
>
>    <RST> segment is received without a TSopt, a TCP MAY drop the segment
>    and send an <ACK> for the last in-sequence segment.  A TCP MUST NOT
>
>    abort a TCP connection if a non-<RST> segment is received without a
>    TSopt.
>
> This wording leaves a few ambiguities:
>
> - The first MUST implies some sort of uniform non-RST non-TSopt processing, but the rest of the paragraph doesn't provide it.
>
> - The last sentence implies that receipt of a RST non-TSopt could cause an abort because of the lack of TSopt; it is because 'abort' is the consequence of processed RST.
>
> I suggest clarifying this as:
>
>    Once TSopt has been successfully negotiated (sent and received)
>    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>    non-<RST> segment for the duration of the connection, and SHOULD be
>    sent in a <RST> segment (see Section 4.2 for details).  If a non-
>    <RST> segment is received without a TSopt, a TCP MUST drop the
>    segment
>    and MAY also send an <ACK> for the last in-sequence segment.
>    A TCP MUST NOT abort a TCP connection because any segment lacks
>    an expected TSopt.
I like Joe's text better

>
> Joe

From internet-drafts@ietf.org  Thu May 23 07:09:36 2013
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the TCP Maintenance and Minor Extensions Work=
ing Group of the IETF.

	Title           : TCP Extensions for High Performance
	Author(s)       : David Borman
                          Bob Braden
                          Van Jacobson
                          Richard Scheffenegger
	Filename        : draft-ietf-tcpm-1323bis-14.txt
	Pages           : 46
	Date            : 2013-05-23

Abstract:
   This document specifies a set of TCP extensions to improve
   performance over paths with a large bandwidth * delay product and to
   provide reliable operation over very high-speed paths.  It defines
   TCP options for scaled windows and timestamps.  The timestamps are
   used for two distinct mechanisms, RTTM (Round Trip Time Measurement)
   and PAWS (Protection Against Wrapped Sequences).

   This document updates and obsoletes RFC 1323.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-1323bis-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-1323bis-14


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


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Yuchung Cheng <ycheng@google.com>, Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Thank you Yuchung!

I have submitted an update with the text clarification from Joe!



Richard Scheffenegger

> On 5/20/2013 01:35 PM, Yuchung Cheng wrote:
> >
> > I suggest clarifying this as:
> >
> >    Once TSopt has been successfully negotiated (sent and received)
> >    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
> >    non-<RST> segment for the duration of the connection, and SHOULD be
> >    sent in a <RST> segment (see Section 4.2 for details).  If a non-
> >    <RST> segment is received without a TSopt, a TCP MUST drop the
> >    segment
> >    and MAY also send an <ACK> for the last in-sequence segment.
> >    A TCP MUST NOT abort a TCP connection because any segment lacks
> >    an expected TSopt.
>
> I like Joe's text better
>=20


From olivier.bonaventure@uclouvain.be  Fri May 24 00:57:41 2013
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Joe, Yuchung,

> I suggest clarifying this as:
>
>     Once TSopt has been successfully negotiated (sent and received)
>     during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>     non-<RST> segment for the duration of the connection, and SHOULD be
>     sent in a <RST> segment (see Section 4.2 for details).

I fail to see the rationale for forcing a TCP connection to always send 
the TSopt in every segment. The timestamp aids to estimate the rtt and 
provides PAWS for long connections. Given the size of the TCP option 
space, we should not force the utilisation of the TSopt in all segments. 
Consider a TCP implementation that supports SACK, RFC1323 and TCP-AO or 
Multipath TCP. When this implementation sends an ACK segment that 
contains a SACK, is it better to encode a long SACK block or to use 
option space to encode a timestamp ? I'd vote for providing a timestamp 
from time to time and reporting accurate SACK blocks.


 > If a non-
 >     <RST> segment is received without a TSopt, a TCP MUST drop the
 >     segment
 >     and MAY also send an <ACK> for the last in-sequence segment.
 >     A TCP MUST NOT abort a TCP connection because any segment lacks
 >     an expected TSopt.

Dropping segments that do not contain the TSopt is excessive. There are 
on the Internet middleboxes that coalesce or split segments. While doing 
that, they may remove options. Dropping a segment because it does not 
contain the TSopt which is only informative appears overkill to me. 
Dropping a segment that does not contain the negotiated TCP-AO option 
makes sens, but not for the TSopt.


Olivier


-- 
INL, ICTEAM, UCLouvain, Belgium, http://inl.info.ucl.ac.be

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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Olivier.Bonaventure@uclouvain.be" <Olivier.Bonaventure@uclouvain.be>, "Joe Touch" <touch@isi.edu>
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Hi Olivier,

>=20
> I fail to see the rationale for forcing a TCP connection to always send
> the TSopt in every segment. The timestamp aids to estimate the rtt and
> provides PAWS for long connections. Given the size of the TCP option
> space, we should not force the utilisation of the TSopt in all segments.
> Consider a TCP implementation that supports SACK, RFC1323 and TCP-AO or
> Multipath TCP. When this implementation sends an ACK segment that contain=
s
> a SACK, is it better to encode a long SACK block or to use option space t=
o
> encode a timestamp ? I'd vote for providing a timestamp from time to time
> and reporting accurate SACK blocks.


We have been there; the interaction of multiple options used together was s=
pecifically stripped from this update. And randomly sending TSopt or not wo=
uld break mechanisms like PAWS (and there might be others reliant on TSopt)=
 - see below.


In that particular example you give, that implementation should choose to e=
ither limit TCP-AO to consume at most 20 bytes (to allow for 1 SACK block; =
the semantics of SACK will provide the most "important" blocks repeatedly),=
 or to not use TSopt at all - still limiting TCP-AO to use no more than 32 =
bytes to allow for a single SACK block...



>=20
>  > If a non-
>  >     <RST> segment is received without a TSopt, a TCP MUST drop the
>  >     segment
>  >     and MAY also send an <ACK> for the last in-sequence segment.
>  >     A TCP MUST NOT abort a TCP connection because any segment lacks
>  >     an expected TSopt.
>=20
> Dropping segments that do not contain the TSopt is excessive. There are o=
n
> the Internet middleboxes that coalesce or split segments. While doing
> that, they may remove options. Dropping a segment because it does not
> contain the TSopt which is only informative appears overkill to me.
> Dropping a segment that does not contain the negotiated TCP-AO option
> makes sens, but not for the TSopt.

When TS is in use, the receiver will use the PAWS mechanism. Effectively, t=
he TS becomes an extension of the sequence number. Accepting segments witho=
ut TS could be seen like accepting a segment, where some (arbitrary) number=
 of MSB bits in the sequence number are masked, and the remaining bits foun=
d to "probably" fall within the current window...


A middlebox should not strip TSopt if it ever let TSopt through. That is me=
ntioned in section 6, but perhaps your view is that this is not emphasized =
enough? If a middlebox does remove TSopt any any non-<SYN>-only segment, it=
 is broken and should be removed from the network.

If a middlebox chooses to split TCP segments (rather than IP packets), it n=
eeds to properly deal with TSopt too - putting that option, unaltered, in a=
ll the fragments created. If it doesn't, it is broken and should be removed=
.

One way of "motivating" the retirement of broken stuff is to point them out=
.=20

Do you have any recent example of a broken middlebox that does not deal pro=
perly with TSopt?

Richard

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From: Yuchung Cheng <ycheng@google.com>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On Fri, May 24, 2013 at 12:57 AM, Olivier Bonaventure
<Olivier.Bonaventure@uclouvain.be> wrote:
> Joe, Yuchung,
>
>
>> I suggest clarifying this as:
>>
>>     Once TSopt has been successfully negotiated (sent and received)
>>     during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>>     non-<RST> segment for the duration of the connection, and SHOULD be
>>     sent in a <RST> segment (see Section 4.2 for details).
>
>
> I fail to see the rationale for forcing a TCP connection to always send the
> TSopt in every segment. The timestamp aids to estimate the rtt and provides
> PAWS for long connections. Given the size of the TCP option space, we should
> not force the utilisation of the TSopt in all segments. Consider a TCP
> implementation that supports SACK, RFC1323 and TCP-AO or Multipath TCP. When
> this implementation sends an ACK segment that contains a SACK, is it better
> to encode a long SACK block or to use option space to encode a timestamp ?
> I'd vote for providing a timestamp from time to time and reporting accurate
> SACK blocks.
>
>
>
>> If a non-
>>     <RST> segment is received without a TSopt, a TCP MUST drop the
>>     segment
>>     and MAY also send an <ACK> for the last in-sequence segment.
>>     A TCP MUST NOT abort a TCP connection because any segment lacks
>>     an expected TSopt.
>
> Dropping segments that do not contain the TSopt is excessive. There are on
> the Internet middleboxes that coalesce or split segments. While doing that,
> they may remove options. Dropping a segment because it does not contain the
> TSopt which is only informative appears overkill to me. Dropping a segment
> that does not contain the negotiated TCP-AO option makes sens, but not for
> the TSopt.

These are good points. Olivier are you suggesting to change MUST to SHOULD of
these two statements?
1.  TSopt MUST be sent in every non-<RST> segment
2. ... received without a TSopt, TCP MUST drop the segment

Richard: I do see middle-boxes (or receiver!) corrupt TS options, unfortunately.

>
>
> Olivier
>
>
> --
> INL, ICTEAM, UCLouvain, Belgium, http://inl.info.ucl.ac.be

From touch@isi.edu  Fri May 24 11:39:44 2013
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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FWIW, I was not opening any new issues; I was clarifying the text only. 
I don't believe my new text changes any of the meaning from the previous 
intended version; IMO it only better handles ambiguities.

We discussed the reason for these requirements before.

Joe

On 5/24/2013 12:57 AM, Olivier Bonaventure wrote:
> Joe, Yuchung,
>
>> I suggest clarifying this as:
>>
>>     Once TSopt has been successfully negotiated (sent and received)
>>     during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>>     non-<RST> segment for the duration of the connection, and SHOULD be
>>     sent in a <RST> segment (see Section 4.2 for details).
>
> I fail to see the rationale for forcing a TCP connection to always send
> the TSopt in every segment. The timestamp aids to estimate the rtt and
> provides PAWS for long connections. Given the size of the TCP option
> space, we should not force the utilisation of the TSopt in all segments.
> Consider a TCP implementation that supports SACK, RFC1323 and TCP-AO or
> Multipath TCP. When this implementation sends an ACK segment that
> contains a SACK, is it better to encode a long SACK block or to use
> option space to encode a timestamp ? I'd vote for providing a timestamp
> from time to time and reporting accurate SACK blocks.
>
>
>  > If a non-
>  >     <RST> segment is received without a TSopt, a TCP MUST drop the
>  >     segment
>  >     and MAY also send an <ACK> for the last in-sequence segment.
>  >     A TCP MUST NOT abort a TCP connection because any segment lacks
>  >     an expected TSopt.
>
> Dropping segments that do not contain the TSopt is excessive. There are
> on the Internet middleboxes that coalesce or split segments. While doing
> that, they may remove options. Dropping a segment because it does not
> contain the TSopt which is only informative appears overkill to me.
> Dropping a segment that does not contain the negotiated TCP-AO option
> makes sens, but not for the TSopt.
>
>
> Olivier
>
>

From pasi.sarolahti@iki.fi  Mon May 27 03:19:45 2013
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On May 24, 2013, at 10:57 AM, Olivier Bonaventure =
<Olivier.Bonaventure@uclouvain.be> wrote:

> Joe, Yuchung,
>=20
>> I suggest clarifying this as:
>>=20
>>    Once TSopt has been successfully negotiated (sent and received)
>>    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
>>    non-<RST> segment for the duration of the connection, and SHOULD =
be
>>    sent in a <RST> segment (see Section 4.2 for details).
>=20
> I fail to see the rationale for forcing a TCP connection to always =
send the TSopt in every segment. The timestamp aids to estimate the rtt =
and provides PAWS for long connections. Given the size of the TCP option =
space, we should not force the utilisation of the TSopt in all segments. =
Consider a TCP implementation that supports SACK, RFC1323 and TCP-AO or =
Multipath TCP. When this implementation sends an ACK segment that =
contains a SACK, is it better to encode a long SACK block or to use =
option space to encode a timestamp ? I'd vote for providing a timestamp =
from time to time and reporting accurate SACK blocks.

Always including TSopt seems to represent the rough consensus of the WG, =
even if it might not be a uniform agreement. This was discussed quite =
extensively some time ago. (Though judging rough consensus from Emails =
is difficult... we'd need some sort of electronic hum tool :-)

About your question: if you believe Multipath TCP is going to be useful =
for you, then it might be a reasonable choice to just disable timestamps =
entirely to make room for MPTCP and SACK, assuming the current =
work-in-progress spec. What do current MPTCP implementations do about =
this, by the way?

However...

> > If a non-
> >     <RST> segment is received without a TSopt, a TCP MUST drop the
> >     segment
> >     and MAY also send an <ACK> for the last in-sequence segment.
> >     A TCP MUST NOT abort a TCP connection because any segment lacks
> >     an expected TSopt.
>=20
> Dropping segments that do not contain the TSopt is excessive. There =
are on the Internet middleboxes that coalesce or split segments. While =
doing that, they may remove options. Dropping a segment because it does =
not contain the TSopt which is only informative appears overkill to me. =
Dropping a segment that does not contain the negotiated TCP-AO option =
makes sens, but not for the TSopt.

MUST drop all (non-RST) segments without timestamp seems indeed =
excessive, and not very liberal in what you are accepting [RFC793]. Why =
is this needed? I don't see why "MUST be sent" on the sending side =
should imply "MUST drop" on the receiving side.

In the worst case this might discourage enabling timestamps at all, if =
an implementation wants to be safe against middleboxes as described.

- Pasi


From christoph.paasch@gmail.com  Mon May 27 07:53:43 2013
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Hello,

On Mon, May 27, 2013 at 01:19:25PM +0300, Pasi Sarolahti wrote:
> On May 24, 2013, at 10:57 AM, Olivier Bonaventure
> <Olivier.Bonaventure@uclouvain.be> wrote:
> >> I suggest clarifying this as:
> >> 
> >>    Once TSopt has been successfully negotiated (sent and received)
> >>    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
> >>    non-<RST> segment for the duration of the connection, and SHOULD be
> >>    sent in a <RST> segment (see Section 4.2 for details).
> > 
> > I fail to see the rationale for forcing a TCP connection to always send
> > the TSopt in every segment. The timestamp aids to estimate the rtt and
> > provides PAWS for long connections. Given the size of the TCP option
> > space, we should not force the utilisation of the TSopt in all segments.
> > Consider a TCP implementation that supports SACK, RFC1323 and TCP-AO or
> > Multipath TCP. When this implementation sends an ACK segment that
> > contains a SACK, is it better to encode a long SACK block or to use
> > option space to encode a timestamp ? I'd vote for providing a timestamp
> > from time to time and reporting accurate SACK blocks.
> 
> Always including TSopt seems to represent the rough consensus of the WG,
> even if it might not be a uniform agreement. This was discussed quite
> extensively some time ago. (Though judging rough consensus from Emails is
> difficult... we'd need some sort of electronic hum tool :-)
> 
> About your question: if you believe Multipath TCP is going to be useful
> for you, then it might be a reasonable choice to just disable timestamps
> entirely to make room for MPTCP and SACK, assuming the current
> work-in-progress spec. What do current MPTCP implementations do about
> this, by the way?

the Linux Kernel implementation of MPTCP reduces the number of SACK-blocks
per packet, if the MPTCP-options need more space.

As we add an MPTCP DATA_ACK in each packet, if timestamps are enabled we
will set a maximum of 2 SACK-blocks per packet.

But this can probably be changed as described in Appendix A of RFC 6824.


Cheers,
Christoph

From rs@netapp.com  Mon May 27 08:24:01 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Christoph Paasch <christoph.paasch@uclouvain.be>, Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Date: Mon, 27 May 2013 15:23:53 +0000
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Thanks,

I think Sally Floyd showed more than a decade ago, that the importance of s=
ubsequent SACK blocks (after the first) diminishes exponentially...

As long as there is room for one SACK block, SACK will work. The 2nd, 3rd..=
. block will only add redundant information in case of ACK loss, but that i=
nformation itself will be repeated eventually in the 1st SACK block (once t=
he hole is filled).

Again, I am not supportive of using this as a reason to overhaul TSopt sema=
ntics in 1323bis as a whole... (probabilistic TSopt? Eventual PAWS?)


Richard Scheffenegger


> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Christoph Paasch
> Sent: Montag, 27. Mai 2013 16:54
> To: Pasi Sarolahti
> Cc: tcpm@ietf.org Extensions; Joe Touch
> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
>=20
> Hello,
>=20
> On Mon, May 27, 2013 at 01:19:25PM +0300, Pasi Sarolahti wrote:
> > On May 24, 2013, at 10:57 AM, Olivier Bonaventure
> > <Olivier.Bonaventure@uclouvain.be> wrote:
> > >> I suggest clarifying this as:
> > >>
> > >>    Once TSopt has been successfully negotiated (sent and received)
> > >>    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
> > >>    non-<RST> segment for the duration of the connection, and SHOULD
> be
> > >>    sent in a <RST> segment (see Section 4.2 for details).
> > >
> > > I fail to see the rationale for forcing a TCP connection to always
> > > send the TSopt in every segment. The timestamp aids to estimate the
> > > rtt and provides PAWS for long connections. Given the size of the
> > > TCP option space, we should not force the utilisation of the TSopt in
> all segments.
> > > Consider a TCP implementation that supports SACK, RFC1323 and TCP-AO
> > > or Multipath TCP. When this implementation sends an ACK segment that
> > > contains a SACK, is it better to encode a long SACK block or to use
> > > option space to encode a timestamp ? I'd vote for providing a
> > > timestamp from time to time and reporting accurate SACK blocks.
> >
> > Always including TSopt seems to represent the rough consensus of the
> > WG, even if it might not be a uniform agreement. This was discussed
> > quite extensively some time ago. (Though judging rough consensus from
> > Emails is difficult... we'd need some sort of electronic hum tool :-)
> >
> > About your question: if you believe Multipath TCP is going to be
> > useful for you, then it might be a reasonable choice to just disable
> > timestamps entirely to make room for MPTCP and SACK, assuming the
> > current work-in-progress spec. What do current MPTCP implementations
> > do about this, by the way?
>=20
> the Linux Kernel implementation of MPTCP reduces the number of SACK-block=
s
> per packet, if the MPTCP-options need more space.
>=20
> As we add an MPTCP DATA_ACK in each packet, if timestamps are enabled we
> will set a maximum of 2 SACK-blocks per packet.
>=20
> But this can probably be changed as described in Appendix A of RFC 6824.
>=20
>=20
> Cheers,
> Christoph
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From touch@isi.edu  Mon May 27 09:54:59 2013
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On 5/27/2013 3:19 AM, Pasi Sarolahti wrote:
...
> MUST drop all (non-RST) segments without timestamp seems indeed
> In the worst case this might discourage enabling timestamps at all,
> if an implementation wants to be safe against middleboxes as
> described.

We've gone through this multiple times for different issues on this 
mailing list, and have addressed it in the context of this ID in detail.

RST is the *only* way TCP cleans up old connection state. TCP doesn't 
keep things tidy; it cleans up state only when it gets in the way of a 
new connection.

RSTs that happen after a reboot won't necessarily have any state from 
the previous connection. ANY requirement about those RSTs *will* 
interfere with TCP state cleanup, and hosts could end up with the net 
effect of a "memory leak" for all connections that are interrupted by a 
reboot.

That's why RSTs MUST NEVER require anything - even, IMO, a handshake (as 
per RFC 5691).

Joe

From pasi.sarolahti@iki.fi  Mon May 27 11:10:50 2013
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On May 27, 2013, at 7:53 PM, Joe Touch <touch@isi.edu> wrote:

> On 5/27/2013 3:19 AM, Pasi Sarolahti wrote:
> ...
>> MUST drop all (non-RST) segments without timestamp seems indeed
>> In the worst case this might discourage enabling timestamps at all,
>> if an implementation wants to be safe against middleboxes as
>> described.
>=20
[...]
> That's why RSTs MUST NEVER require anything - even, IMO, a handshake =
(as per RFC 5691).

Fine, but my (and Olivier's) comment was about normal segments that do =
not carry RST flag. Why MUST those be dropped at the receiver if they =
are missing TSopt for some reason? This is a new requirement compared to =
earlier versions of the draft and RFC1323, and I don't think current =
implementations drop such segments either.

- Pasi


From touch@isi.edu  Mon May 27 11:32:06 2013
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On 5/27/2013 11:10 AM, Pasi Sarolahti wrote:
> On May 27, 2013, at 7:53 PM, Joe Touch <touch@isi.edu> wrote:
>
>> On 5/27/2013 3:19 AM, Pasi Sarolahti wrote:
>> ...
>>> MUST drop all (non-RST) segments without timestamp seems indeed
>>> In the worst case this might discourage enabling timestamps at all,
>>> if an implementation wants to be safe against middleboxes as
>>> described.
>>
> [...]
>> That's why RSTs MUST NEVER require anything - even, IMO, a handshake (as per RFC 5691).
>
> Fine, but my (and Olivier's) comment was about normal segments that
> do not carry RST flag.

Sorry - I misread your post. However:

> Why MUST those be dropped at the receiver if
> they are missing TSopt for some reason? This is a new requirement
> compared to earlier versions of the draft and RFC1323, and I don't
> think current implementations drop such segments either.

The original requirement was here in -13:

    Once TSopt has been successfully negotiated (sent and received)
    during the <SYN>, <SYN,ACK> exchange, TSopt MUST be sent in every
    non-<RST> segment for the duration of the connection, and SHOULD be
    sent in a <RST> segment (see Section 4.2 for details).  If a non-
    <RST> segment is received without a TSopt, a TCP MAY drop the segment
    and send an <ACK> for the last in-sequence segment.  A TCP MUST NOT
    abort a TCP connection if a non-<RST> segment is received without a
    TSopt.

Regarding handling of non-RSTs lacking TSopt, we did discuss this on the 
list; you even cited this within the past day or so:

---
> Always including TSopt seems to represent the rough consensus of the
> WG, even if it might not be a uniform agreement. This was discussed
> quite extensively some time ago. (Though judging rough consensus from
> Emails is difficult... we'd need some sort of electronic hum tool
---

Once you indicate a sender-side MUST, you need to describe how the 
receiver handles exceptions. IMO, there's little point to claiming PAWS 
protection if you're going to react *in any way* to a non-RST segment 
lacking TSopt.

I.e., a sender MUST include implies - IMO - a receiver MUST silently 
discard when the 'must' isn't there.

Joe



From pasi.sarolahti@iki.fi  Mon May 27 13:41:12 2013
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On May 27, 2013, at 9:31 PM, Joe Touch <touch@ISI.EDU> wrote:

> Regarding handling of non-RSTs lacking TSopt, we did discuss this on =
the list; you even cited this within the past day or so:
>=20
> ---
>> Always including TSopt seems to represent the rough consensus of the
>> WG, even if it might not be a uniform agreement. This was discussed
>> quite extensively some time ago. (Though judging rough consensus from
>> Emails is difficult... we'd need some sort of electronic hum tool
> ---

My recollection is that this discussion concerned sender-side behavior, =
but there have been tens of Emails during the past weeks, so I might =
have missed something.

> Once you indicate a sender-side MUST, you need to describe how the =
receiver handles exceptions. IMO, there's little point to claiming PAWS =
protection if you're going to react *in any way* to a non-RST segment =
lacking TSopt.
>=20
> I.e., a sender MUST include implies - IMO - a receiver MUST silently =
discard when the 'must' isn't there.

My reading of the original robustness principle is that TCP receiver =
should have minimal required validation checks for segments, so I don't =
think sender-side "MUST include" automatically implies receiver-side =
"MUST ignore if missing", unless we can point out a clear problem in =
accepting the segment. Perhaps the possible unreliability of PAWS is =
such, then.

But there are possible deployment implications in additional receiver =
check, too. In a perfect world this might work, but RFC 1323 did not =
have as strict MUST requirement about including TSopt in all segments =
after handshake, and there are possible middlebox issues as Olivier =
mentioned. So I think it would be worthwhile addition to the middlebox =
section, then, to say if a middlebox removes TSopt option, or a =
resegmenting middlebox does not include it in all segments, that breaks =
the TCP connection entirely, if this version of spec is followed.

As a mere data point, I did a quick check on Linux that currently seems =
to first check if timestamp option was in incoming segment, and only =
then do PAWS check. It does not reject segments because of lack of =
TSopt, at least based on my quick reading.

- Pasi


From touch@isi.edu  Mon May 27 14:39:18 2013
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On 5/27/2013 1:40 PM, Pasi Sarolahti wrote:
> On May 27, 2013, at 9:31 PM, Joe Touch <touch@ISI.EDU> wrote:
>
>> Regarding handling of non-RSTs lacking TSopt, we did discuss this
>> onthe list; you even cited this within the past day or so:
>>
>> ---
>>> Always including TSopt seems to represent the rough consensus of the
>>> WG, even if it might not be a uniform agreement. This was discussed
>>> quite extensively some time ago. (Though judging rough consensus from
>>> Emails is difficult... we'd need some sort of electronic hum tool
>> ---
>
> My recollection is that this discussion concerned sender-side
> behavior, but there have been tens of Emails during the past weeks, so I
> might have missed something.

It was the part about PAWS; if you accept segments omitting TSopt, those
segments aren't PAWS-protected, and so the connection isn't protectd.

>> Once you indicate a sender-side MUST, you need to describe how the
>> receiver handles exceptions. IMO, there's little point to claiming PAWS
>> protection if you're going to react *in any way* to a non-RST segment
>> lacking TSopt.
>>
>> I.e., a sender MUST include implies - IMO - a receiver MUST
>> silentlydiscard when the 'must' isn't there.
>
> My reading of the original robustness principle is that TCP receiver
> should have minimal required validation checks for segments,

There are four alternatives here regarding robustness:

	1. should the segment be accepted fully?

	2 should the segment generate a response (resend last ACK)?

	3. should the segment be silently ignored?

	4. should the segment generate an error?

Robustness suggests preferring these in order, where appropriate. 
However, if the point of TSopt is PAWS protection, then the best you can 
do is nothing (#3).

 > so I don't
> think sender-side "MUST include" automatically implies receiver-side
> "MUST ignore if missing", unless we can point out a clear problem in
> accepting the segment.

If you cannot point out such a problem, the MUST was clearly 
inappropriate. If there is *any* situation in which a receiver should do 
more than ignore a segment or report an error, then the condition is, at 
best, a SHOULD.

 > Perhaps the possible unreliability of PAWS is such, then.

That was my impression of the list discussion on this issue.

> But there are possible deployment implications in additional
> receiver check, too. In a perfect world this might work, but RFC 1323
> did not have as strict MUST requirement about including TSopt in all
> segments after handshake, and there are possible middlebox issues as
> Olivier mentioned. So I think it would be worthwhile addition to the
> middlebox section, then, to say if a middlebox removes TSopt option,
> or a resegmenting middlebox does not include it in all segments, that
> breaks the TCP connection entirely, if this version of spec is
> followed.

Sure. It's defeating the value of the TSopt. It's inappropriate for a 
connection to claim TSopt protection and not have it.

> As a mere data point, I did a quick check on Linux that currently
> seems to first check if timestamp option was in incoming segment, and
> only then do PAWS check. It does not reject segments because of lack of
> TSopt, at least based on my quick reading.

If there's a safe way to interpret segments without TSopt in a 
TSopt-protected connection, then sure, do it. I don't recall any being 
raised, which is why I suggested the way forward I did.

Joe

From rs@netapp.com  Mon May 27 15:05:58 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>, Joe Touch <touch@ISI.EDU>
Thread-Topic: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Hi Pasi,


Yes, there was no formal requirement in 1323 (predating 2119), but its in 1=
323's PAWS section:

  4.2  The PAWS Mechanism
[...]
      The basic idea
      is that a segment can be discarded as an old duplicate if it is
      received with a timestamp SEG.TSval less than some timestamp
      recently received on this connection.

(an undefined TSval might be numeric zero, or TSrecent... 1323 doesn't say.=
)

We also had discussed the precedence of processing (1323 PAWS vs. 793 in-wi=
ndow acceptance testing), and the 1323 had some ambiguity how to go about t=
his (the 1323 text would actually allow the omission of TSopt sometime alon=
g a session, voiding PAWS protection at that time, and potentially undefine=
d (receiver) behavior then...

Regards,

Richard Scheffenegger



> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Pasi Sarolahti
> Sent: Montag, 27. Mai 2013 22:41
> To: Joe Touch
> Cc: tcpm@ietf.org Extensions
> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
>=20
> On May 27, 2013, at 9:31 PM, Joe Touch <touch@ISI.EDU> wrote:
>=20
> > Regarding handling of non-RSTs lacking TSopt, we did discuss this on th=
e
> list; you even cited this within the past day or so:
> >
> > ---
> >> Always including TSopt seems to represent the rough consensus of the
> >> WG, even if it might not be a uniform agreement. This was discussed
> >> quite extensively some time ago. (Though judging rough consensus from
> >> Emails is difficult... we'd need some sort of electronic hum tool
> > ---
>=20
> My recollection is that this discussion concerned sender-side behavior,
> but there have been tens of Emails during the past weeks, so I might have
> missed something.
>=20
> > Once you indicate a sender-side MUST, you need to describe how the
> receiver handles exceptions. IMO, there's little point to claiming PAWS
> protection if you're going to react *in any way* to a non-RST segment
> lacking TSopt.
> >
> > I.e., a sender MUST include implies - IMO - a receiver MUST silently
> discard when the 'must' isn't there.
>=20
> My reading of the original robustness principle is that TCP receiver
> should have minimal required validation checks for segments, so I don't
> think sender-side "MUST include" automatically implies receiver-side "MUS=
T
> ignore if missing", unless we can point out a clear problem in accepting
> the segment. Perhaps the possible unreliability of PAWS is such, then.
>=20
> But there are possible deployment implications in additional receiver
> check, too. In a perfect world this might work, but RFC 1323 did not have
> as strict MUST requirement about including TSopt in all segments after
> handshake, and there are possible middlebox issues as Olivier mentioned.
> So I think it would be worthwhile addition to the middlebox section, then=
,
> to say if a middlebox removes TSopt option, or a resegmenting middlebox
> does not include it in all segments, that breaks the TCP connection
> entirely, if this version of spec is followed.
>=20
> As a mere data point, I did a quick check on Linux that currently seems t=
o
> first check if timestamp option was in incoming segment, and only then do
> PAWS check. It does not reject segments because of lack of TSopt, at leas=
t
> based on my quick reading.
>=20
> - Pasi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From michawe@ifi.uio.no  Tue May 28 01:03:31 2013
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Subject: [tcpm] Can't we use newcwv for IW too?
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Hi,

Recently, the fact that the most recent HTTP2.0 draft lets a sender =
transfer TCP's cwnd to the receiver caused an interesting discussion in =
the HTTPBIS group. To see the thread, look for "cwnd" at:
http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/thread.html

I agree with folks saying that it's high time for transport people to =
get involved. I just subscribed to the list. Anyway, as Lars Eggert =
mentioned in this thread, this group (TCPM) is a better home for such =
discussions; this problem really is a TCP problem, not an app layer =
problem. This message that he wrote:
http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
made me think: irrespective of the size of your "normal" IW, why can't =
we construct a rule like IW =3D max(IW, cwnd derived from new-cwv from a =
previous connection) for a connection to the same host as the previous =
one?

As Lars has pointed out in his message, this idea isn't new. However, it =
seems to me that the key problem of previous approaches (let alone the =
current HTTP2.0 one!) is related to cwnd dynamics - we will have to =
collapse or in some way decay it after a while of course - how exactly =
do we do that? What's new is that, with newcwv ( =
draft-fairhurst-tcpm-newcwv-05 ), we now have an approach that reuses =
cwnd after a rate-limited *or idle* time, and that approach has some =
support behind it. It's a TCPM WG item, it has gone through some =
evaluation-update rounds (cf. =
http://trac.tools.ietf.org/group/irtf/trac/wiki/ICCRG_newcwv ), and is =
backed by research (cf. http://dl.acm.org/citation.cfm?id=3D2413230 ).

There really isn't much difference between newcwv restarting after an =
idle period and a new connection using an IW based on a previous =
connection's newcwv.
We may have a new port number (although we could also be extra =
restrictive here, if we really must), and we had the FIN + SYN =
handshakes. These things may lead to different treatment in middle-boxes =
and they may give us a new path, using ECMP, so yes, it's not exactly =
the same as restarting from idle, but then again, consider:
- it takes only 1 RTT to notice the mistake and react to it; this is =
quite close to the Jump Start idea ( =
http://www.icir.org/mallman/papers/jumpstart-pfldnet07.pdf ), but backed =
by prior knowledge about the path whenever stuff like ECMP didn't kick =
in
- we do this only for one connection following a previously terminated =
one  (sure, if N just terminated, we could do it N times, within a =
certain time frame...)

Thoughts?

Michael


--Apple-Mail=_C6111B64-5255-4C4D-96EB-AB2166896C96
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Hi,<div><br></div><div>Recently, the fact that the most recent HTTP2.0 =
draft lets a sender transfer TCP's cwnd to the receiver caused an =
interesting discussion in the HTTPBIS group. To see the thread, look for =
"cwnd" at:</div><div><a =
href=3D"http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/thread=
.html">http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/thread.=
html</a></div><div><br></div><div>I agree with folks saying that it's =
high time for transport people to get involved. I just subscribed to the =
list. Anyway, as Lars Eggert mentioned in this thread, this group (TCPM) =
is a better home for such discussions; this problem really is a TCP =
problem, not an app layer problem. This message that he =
wrote:</div><div><a =
href=3D"http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.h=
tml">http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html=
</a></div><div>made me think: irrespective of the size of your "normal" =
IW, why can't we construct a rule like&nbsp;IW =3D max(IW, cwnd derived =
from&nbsp;new-cwv from a previous connection) for a connection to the =
same host as the previous one?</div><div><br></div><div>As Lars has =
pointed out in his message, this idea isn't new. However, it seems to me =
that the key problem of previous approaches (let alone the current =
HTTP2.0 one!) is related to cwnd dynamics - we will have to collapse or =
in some way decay it after a while of course - how exactly do we do =
that? What's new is that, with newcwv ( =
draft-fairhurst-tcpm-newcwv-05&nbsp;), we now have an approach that =
reuses cwnd after a rate-limited *or idle* time, and that approach has =
some support behind it. It's a TCPM WG item, it has gone through some =
evaluation-update rounds (cf.&nbsp;<a =
href=3D"http://trac.tools.ietf.org/group/irtf/trac/wiki/ICCRG_newcwv">http=
://trac.tools.ietf.org/group/irtf/trac/wiki/ICCRG_newcwv</a>&nbsp;), and =
is backed by research (cf.&nbsp;<a =
href=3D"http://dl.acm.org/citation.cfm?id=3D2413230">http://dl.acm.org/cit=
ation.cfm?id=3D2413230</a>&nbsp;).</div><div><br></div><div>There really =
isn't much difference between newcwv restarting after an idle period and =
a new connection using an IW based on a previous connection's =
newcwv.</div><div>We may have a new port number (although we could also =
be extra restrictive here, if we really must), and we had the&nbsp;FIN + =
SYN handshakes. These things may lead to different treatment in =
middle-boxes and they may give us a new path, using ECMP, so yes, it's =
not exactly the same as restarting from idle, but then again, =
consider:</div><div>- it takes only 1 RTT to notice the mistake and =
react to it; this is quite close to the Jump Start idea (&nbsp;<a =
href=3D"http://www.icir.org/mallman/papers/jumpstart-pfldnet07.pdf">http:/=
/www.icir.org/mallman/papers/jumpstart-pfldnet07.pdf</a> ), but backed =
by prior knowledge about the path whenever stuff like ECMP didn't kick =
in</div><div>- we do this only for one connection following a previously =
terminated one &nbsp;(sure, if N just terminated, we could do it N =
times, within a certain time =
frame...)</div><div><br></div><div>Thoughts?</div><div><br></div><div>Mich=
ael</div><div><br></div></body></html>=

--Apple-Mail=_C6111B64-5255-4C4D-96EB-AB2166896C96--

From pasi.sarolahti@iki.fi  Tue May 28 01:45:37 2013
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To: Joe Touch <touch@ISI.EDU>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On May 28, 2013, at 12:38 AM, Joe Touch <touch@ISI.EDU> wrote:

> There are four alternatives here regarding robustness:
>=20
> 	1. should the segment be accepted fully?
>=20
> 	2 should the segment generate a response (resend last ACK)?
>=20
> 	3. should the segment be silently ignored?
>=20
> 	4. should the segment generate an error?
>=20
> Robustness suggests preferring these in order, where appropriate. =
However, if the point of TSopt is PAWS protection, then the best you can =
do is nothing (#3).


After the latest addition, there is some inconsistency between Sec. 3.2 =
and 4.3.

* In the PAWS algorithm in sec. 4.3., there should be an additional step =
before R1 for checking if Timestamps option is present, and if not, tell =
to discard the segment (if negotiation had been successful).

* PAWS algorithm tells to send acknowledgment if the check fails. =
Shouldn't we then, for consistency, require sending acknowledgment also =
if TSopt is missing? Current MAY in Sec. 3.2. is not very informative in =
my mind.

But based on my quick Linux check, I believe that currently =
implementations follow what sec. 4.3. currently says, and what it said =
in RFC 1323: "if there is Timestamps option in the segment, then [do =
PAWS check]....". So this would be relevant change compared to RFC 1323 =
that should also be highlighted in Appendix H, because it may require =
action from implementors.

>> But there are possible deployment implications in additional
>> receiver check, too. In a perfect world this might work, but RFC 1323
>> did not have as strict MUST requirement about including TSopt in all
>> segments after handshake, and there are possible middlebox issues as
>> Olivier mentioned. So I think it would be worthwhile addition to the
>> middlebox section, then, to say if a middlebox removes TSopt option,
>> or a resegmenting middlebox does not include it in all segments, that
>> breaks the TCP connection entirely, if this version of spec is
>> followed.
>=20
> Sure. It's defeating the value of the TSopt. It's inappropriate for a =
connection to claim TSopt protection and not have it.

So, to reflect the comment Olivier made earlier, I'd suggest expanding =
the third bullet in the Security Considerations section with something =
like following:

OLD:
" *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
         segment, high speed connections that need PAWS would not have
         that protection.  Middleboxes should not remove the Timestamps
         option."

NEW:
" *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
         segment, high speed connections that need PAWS would not have
         that protection. Successful negotiation of Timestamps option =
enforces a stricter verification of incoming segments at receiver. If =
the Timestamps option was removed from a subsequent data segment after a =
successful negotiation (for example, as part of re-segmentation), the =
segment is discarded by the receiver without further processing. =
Middleboxes should not remove the Timestamps option. One study found =
that 6% of measured HTTP connections had the Timestamps option removed =
from data segments [Honda11]."

[Honda11]  M. Honda, Y. Nishida, C. Raiciu, A. Greenhalgh, M. Handley, =
and H. Tokuda. Is it still possible to extend TCP?. In Proc. of ACM =
Internet Measurement Conference (IMC) '11, Berlin, Germany, November =
2011.=20

- Pasi


From touch@isi.edu  Wed May 29 11:39:59 2013
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Subject: Re: [tcpm] Can't we use newcwv for IW too?
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On 5/28/2013 1:03 AM, Michael Welzl wrote:
> I agree with folks saying that it's high time for transport people to
> get involved. I just subscribed to the list. Anyway, as Lars Eggert
> mentioned in this thread, this group (TCPM) is a better home for such
> discussions; this problem really is a TCP problem, not an app layer
> problem. This message that he wrote:
> http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
> made me think: irrespective of the size of your "normal" IW, why can't
> we construct a rule like IW = max(IW, cwnd derived from new-cwv from a
> previous connection) for a connection to the same host as the previous one?

There are reasons this might work (it's a bit like RFC2140), and reasons 
it might not (IP address != host these days, esp. for web server farms).

> There really isn't much difference between newcwv restarting after an
> idle period and a new connection using an IW based on a previous
> connection's newcwv.

That depends on how many new connections you open using this 
information, and whether the previous connection is closed. If it's not 
1:1, then yes, it's different. A more complete consideration of the 
issues of 2140 would handle that, but just importing the info from a 
previous connection isn't sufficient by itself.

You already pointed out some potential for alternate paths due to port 
number changes, but that's less of a concern AFAICT.

Joe

From michawe@ifi.uio.no  Wed May 29 12:55:44 2013
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Subject: Re: [tcpm] Can't we use newcwv for IW too?
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Hi,

Just to clarify:

On May 29, 2013, at 8:38 PM, Joe Touch <touch@isi.edu> wrote:

>=20
>=20
> On 5/28/2013 1:03 AM, Michael Welzl wrote:
>> I agree with folks saying that it's high time for transport people to
>> get involved. I just subscribed to the list. Anyway, as Lars Eggert
>> mentioned in this thread, this group (TCPM) is a better home for such
>> discussions; this problem really is a TCP problem, not an app layer
>> problem. This message that he wrote:
>> http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
>> made me think: irrespective of the size of your "normal" IW, why =
can't
>> we construct a rule like IW =3D max(IW, cwnd derived from new-cwv =
from a
>> previous connection) for a connection to the same host as the =
previous one?
>=20
> There are reasons this might work (it's a bit like RFC2140), and =
reasons it might not (IP address !=3D host these days, esp. for web =
server farms).
>=20
>> There really isn't much difference between newcwv restarting after an
>> idle period and a new connection using an IW based on a previous
>> connection's newcwv.
>=20
> That depends on how many new connections you open using this =
information, and whether the previous connection is closed. If it's not =
1:1, then yes, it's different. A more complete consideration of the =
issues of 2140 would handle that, but just importing the info from a =
previous connection isn't sufficient by itself.

I did mean 1:1.


> You already pointed out some potential for alternate paths due to port =
number changes, but that's less of a concern AFAICT.
>=20
> Joe

Cheers,
Michael


From touch@isi.edu  Wed May 29 12:58:59 2013
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Subject: Re: [tcpm] Can't we use newcwv for IW too?
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On 5/29/2013 12:55 PM, Michael Welzl wrote:
> Hi,
>
> Just to clarify:
>
> On May 29, 2013, at 8:38 PM, Joe Touch <touch@isi.edu> wrote:
>
>>
>>
>> On 5/28/2013 1:03 AM, Michael Welzl wrote:
>>> I agree with folks saying that it's high time for transport people to
>>> get involved. I just subscribed to the list. Anyway, as Lars Eggert
>>> mentioned in this thread, this group (TCPM) is a better home for such
>>> discussions; this problem really is a TCP problem, not an app layer
>>> problem. This message that he wrote:
>>> http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
>>> made me think: irrespective of the size of your "normal" IW, why can't
>>> we construct a rule like IW = max(IW, cwnd derived from new-cwv from a
>>> previous connection) for a connection to the same host as the previous one?
>>
>> There are reasons this might work (it's a bit like RFC2140), and reasons it might not (IP address != host these days, esp. for web server farms).
>>
>>> There really isn't much difference between newcwv restarting after an
>>> idle period and a new connection using an IW based on a previous
>>> connection's newcwv.
>>
>> That depends on how many new connections you open using this information, and whether the previous connection is closed. If it's not 1:1, then yes, it's different. A more complete consideration of the issues of 2140 would handle that, but just importing the info from a previous connection isn't sufficient by itself.
>
> I did mean 1:1.

Ahh; then the algorithm would need to be more specifically described as 
"only co-opting the window of a recently closed TCP connection", and 
explain how long "recently" would imply ;-)

In that case, though, IMO it's just a subset of 2140, some versions of 
which are already in widespread use.

Joe

>> You already pointed out some potential for alternate paths due to port number changes, but that's less of a concern AFAICT.
>>
>> Joe
>
> Cheers,
> Michael
>

From michawe@ifi.uio.no  Wed May 29 13:02:56 2013
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Subject: Re: [tcpm] Can't we use newcwv for IW too?
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On May 29, 2013, at 9:58 PM, Joe Touch <touch@isi.edu> wrote:

>=20
>=20
> On 5/29/2013 12:55 PM, Michael Welzl wrote:
>> Hi,
>>=20
>> Just to clarify:
>>=20
>> On May 29, 2013, at 8:38 PM, Joe Touch <touch@isi.edu> wrote:
>>=20
>>>=20
>>>=20
>>> On 5/28/2013 1:03 AM, Michael Welzl wrote:
>>>> I agree with folks saying that it's high time for transport people =
to
>>>> get involved. I just subscribed to the list. Anyway, as Lars Eggert
>>>> mentioned in this thread, this group (TCPM) is a better home for =
such
>>>> discussions; this problem really is a TCP problem, not an app layer
>>>> problem. This message that he wrote:
>>>> =
http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
>>>> made me think: irrespective of the size of your "normal" IW, why =
can't
>>>> we construct a rule like IW =3D max(IW, cwnd derived from new-cwv =
from a
>>>> previous connection) for a connection to the same host as the =
previous one?
>>>=20
>>> There are reasons this might work (it's a bit like RFC2140), and =
reasons it might not (IP address !=3D host these days, esp. for web =
server farms).
>>>=20
>>>> There really isn't much difference between newcwv restarting after =
an
>>>> idle period and a new connection using an IW based on a previous
>>>> connection's newcwv.
>>>=20
>>> That depends on how many new connections you open using this =
information, and whether the previous connection is closed. If it's not =
1:1, then yes, it's different. A more complete consideration of the =
issues of 2140 would handle that, but just importing the info from a =
previous connection isn't sufficient by itself.
>>=20
>> I did mean 1:1.
>=20
> Ahh; then the algorithm would need to be more specifically described =
as "only co-opting the window of a recently closed TCP connection", and =
explain how long "recently" would imply ;-)
>=20
> In that case, though, IMO it's just a subset of 2140, some versions of =
which are already in widespread use.

Indeed! But using rules from new-cwv for "recently". But yes, it's not =
much more than that  (this being said, I don't think 2140 did =
specifically *recommend* re-using cwnd for later connections, it just =
hinted at the possibility, I thought=85 but I'd have to read it again).

Cheers,
Michael


From touch@isi.edu  Wed May 29 13:09:00 2013
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Hi, Pasi,

On 5/28/2013 1:45 AM, Pasi Sarolahti wrote:
> On May 28, 2013, at 12:38 AM, Joe Touch <touch@ISI.EDU> wrote:
>
>> There are four alternatives here regarding robustness:
>>
>> 	1. should the segment be accepted fully?
>>
>> 	2 should the segment generate a response (resend last ACK)?
>>
>> 	3. should the segment be silently ignored?
>>
>> 	4. should the segment generate an error?
>>
> > Robustness suggests preferring these in order, where appropriate.
>> However, if the point of TSopt is PAWS protection, then the best you can
>> do is nothing (#3).
>
>
> After the latest addition, there is some inconsistency between Sec.
> 3.2 and 4.3.
>
> * In the PAWS algorithm in sec. 4.3., there should be an additional
> step before R1 for checking if Timestamps option is present, and if not,
> tell to discard the segment (if negotiation had been successful).

IMO, yes.

> * PAWS algorithm tells to send acknowledgment if the check fails.
> Shouldn't we then, for consistency, require sending acknowledgment also
> if TSopt is missing? Current MAY in Sec. 3.2. is not very informative in
> my mind.

I don't think it's useful to send the ACK in that case. The point of 
PAWS is to ignore segments that are not protected. Even sending an ACK 
defeats that. At best, send a dup ack for last segment received 
correctly, but I still think that doing *anything* other than 
silent-drop isn't what PAWS protection intends.

> But based on my quick Linux check, I believe that currently
> implementations follow what sec. 4.3. currently says, and what it said
> in RFC 1323: "if there is Timestamps option in the segment, then [do
> PAWS check]....". So this would be relevant change compared to RFC 1323
> that should also be highlighted in Appendix H, because it may require
> action from implementors.

Yes, if so.

>>> But there are possible deployment implications in additional
>>> receiver check, too. In a perfect world this might work, but RFC 1323
>>> did not have as strict MUST requirement about including TSopt in all
>>> segments after handshake, and there are possible middlebox issues as
>>> Olivier mentioned. So I think it would be worthwhile addition to the
>>> middlebox section, then, to say if a middlebox removes TSopt option,
>>> or a resegmenting middlebox does not include it in all segments, that
>>> breaks the TCP connection entirely, if this version of spec is
>>> followed.
>>
>> Sure. It's defeating the value of the TSopt. It's inappropriate for
>> a  connection to claim TSopt protection and not have it.
>
> So, to reflect the comment Olivier made earlier, I'd suggest
> expanding  the third bullet in the Security Considerations section with something
> like following:
> OLD:
> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>           segment, high speed connections that need PAWS would not have
>           that protection.  Middleboxes should not remove the Timestamps
>           option."
>
> NEW:
> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>           segment, high speed connections that need PAWS would not have
> that protection. Successful negotiation of Timestamps option
> enforces a stricter verification of incoming segments at receiver. If
> the Timestamps option was removed from a subsequent data segment
> after a successful negotiation (for example, as part of
> re-segmentation), the segment is discarded by the receiver without
> further processing. Middleboxes should not remove the Timestamps
> option. One study found that 6% of measured HTTP connections had the
> Timestamps option removed from data segments [Honda11]."

The last sentence seems out of place; why are you including that? If 
there are implications to the 6% drop then you should mention that, but 
I would expect that elsewhere than the security considerations section.

Joe

From touch@isi.edu  Wed May 29 13:12:10 2013
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Subject: Re: [tcpm] Can't we use newcwv for IW too?
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On 5/29/2013 1:02 PM, Michael Welzl wrote:
>
> On May 29, 2013, at 9:58 PM, Joe Touch <touch@isi.edu> wrote:
>
>>
>>
>> On 5/29/2013 12:55 PM, Michael Welzl wrote:
>>> Hi,
>>>
>>> Just to clarify:
>>>
>>> On May 29, 2013, at 8:38 PM, Joe Touch <touch@isi.edu> wrote:
>>>
>>>>
>>>>
>>>> On 5/28/2013 1:03 AM, Michael Welzl wrote:
>>>>> I agree with folks saying that it's high time for transport people to
>>>>> get involved. I just subscribed to the list. Anyway, as Lars Eggert
>>>>> mentioned in this thread, this group (TCPM) is a better home for such
>>>>> discussions; this problem really is a TCP problem, not an app layer
>>>>> problem. This message that he wrote:
>>>>> http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
>>>>> made me think: irrespective of the size of your "normal" IW, why can't
>>>>> we construct a rule like IW = max(IW, cwnd derived from new-cwv from a
>>>>> previous connection) for a connection to the same host as the previous one?
>>>>
>>>> There are reasons this might work (it's a bit like RFC2140), and reasons it might not (IP address != host these days, esp. for web server farms).
>>>>
>>>>> There really isn't much difference between newcwv restarting after an
>>>>> idle period and a new connection using an IW based on a previous
>>>>> connection's newcwv.
>>>>
>>>> That depends on how many new connections you open using this information, and whether the previous connection is closed. If it's not 1:1, then yes, it's different. A more complete consideration of the issues of 2140 would handle that, but just importing the info from a previous connection isn't sufficient by itself.
>>>
>>> I did mean 1:1.
>>
>> Ahh; then the algorithm would need to be more specifically described as "only co-opting the window of a recently closed TCP connection", and explain how long "recently" would imply ;-)
>>
>> In that case, though, IMO it's just a subset of 2140, some versions of which are already in widespread use.
>
> Indeed! But using rules from new-cwv for "recently".

Sure.

> But yes, it's not much more than that  (this being said, I don't
> think 2140 did specifically *recommend* re-using cwnd for later
> connections, it just hinted at the possibility, I thought… but I'd
> have to read it again).

It was informational, but it did talk about reusing cwnd both across 
concurrent connections (spatial sharing) and across sequences of 
connections (temporal sharing).

Joe

>
> Cheers,
> Michael
>

From gorry@erg.abdn.ac.uk  Wed May 29 23:30:55 2013
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Date: Thu, 30 May 2013 07:30:48 +0100
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To: "Joe Touch" <touch@isi.edu>
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [tcpm] Can't we use newcwv for IW too?
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So here are some thoughts - if we were talking about "bulk" flows where
the cwnd reflects the network capacity to a specific destination, then
sharing of the cwnd is something that is being done by many stacks. In
reality the cwnd that is "shared" is often higher than that sustained on
the path, since this is inflated each RTT (the extra between 1 MSS and
cwnd).

Now, if we have a flow that does not fully utilise cwnd (variously called
thin,variable rate, bursty, intermittent, ...) then the cwnd can be much
higher than the that sustained on the path. RFC 2861, attempted to keep
the cwnd close to the actual used capacity, but this restricts the
dynamics o an application and adds latency to applications that are
bursty. Hence if we desire to enable these applications we should not
aggressively trim the cwnd to what was used.

New-cwv tries to avoid unecessary cwnd growth, but proposes to avoid
forcing a low cwnd on a bursty application. Instead it creates a new phase
(the non-validated phase) where cwnd exceeds the actual path capacity
used, and where the cwnd growth is capped, but importantly the congestion
response is changed to reduce more I think this helps motivate persistent
use of TCP.

Now, my question is what cwnd value should be "shared" when the flow is in
the  non-validated phase? - If a flow was to share non-valiudated cwnd,
and all flows implemented the new-cwv algorithm, then the flow that
received the "extra" cwnd would be non-validated, and that would be OK. If
it were a normal TCP flow, then I think this inflates cwnd in an
unreasonable way.

Gorry

>
> On 5/29/2013 1:02 PM, Michael Welzl wrote:
>>
>> On May 29, 2013, at 9:58 PM, Joe Touch <touch@isi.edu> wrote:
>>
>>>
>>>
>>> On 5/29/2013 12:55 PM, Michael Welzl wrote:
>>>> Hi,
>>>>
>>>> Just to clarify:
>>>>
>>>> On May 29, 2013, at 8:38 PM, Joe Touch <touch@isi.edu> wrote:
>>>>
>>>>>
>>>>>
>>>>> On 5/28/2013 1:03 AM, Michael Welzl wrote:
>>>>>> I agree with folks saying that it's high time for transport people
>>>>>> to
>>>>>> get involved. I just subscribed to the list. Anyway, as Lars Eggert
>>>>>> mentioned in this thread, this group (TCPM) is a better home for
>>>>>> such
>>>>>> discussions; this problem really is a TCP problem, not an app layer
>>>>>> problem. This message that he wrote:
>>>>>> http://lists.w3.org/Archives/Public/ietf-http-wg/2013AprJun/0277.html
>>>>>> made me think: irrespective of the size of your "normal" IW, why
>>>>>> can't
>>>>>> we construct a rule like IW = max(IW, cwnd derived from new-cwv from
>>>>>> a
>>>>>> previous connection) for a connection to the same host as the
>>>>>> previous one?
>>>>>
>>>>> There are reasons this might work (it's a bit like RFC2140), and
>>>>> reasons it might not (IP address != host these days, esp. for web
>>>>> server farms).
>>>>>
>>>>>> There really isn't much difference between newcwv restarting after
>>>>>> an
>>>>>> idle period and a new connection using an IW based on a previous
>>>>>> connection's newcwv.
>>>>>
>>>>> That depends on how many new connections you open using this
>>>>> information, and whether the previous connection is closed. If it's
>>>>> not 1:1, then yes, it's different. A more complete consideration of
>>>>> the issues of 2140 would handle that, but just importing the info
>>>>> from a previous connection isn't sufficient by itself.
>>>>
>>>> I did mean 1:1.
>>>
>>> Ahh; then the algorithm would need to be more specifically described as
>>> "only co-opting the window of a recently closed TCP connection", and
>>> explain how long "recently" would imply ;-)
>>>
>>> In that case, though, IMO it's just a subset of 2140, some versions of
>>> which are already in widespread use.
>>
>> Indeed! But using rules from new-cwv for "recently".
>
> Sure.
>
>> But yes, it's not much more than that  (this being said, I don't
>> think 2140 did specifically *recommend* re-using cwnd for later
>> connections, it just hinted at the possibility, I thought… but I'd
>> have to read it again).
>
> It was informational, but it did talk about reusing cwnd both across
> concurrent connections (spatial sharing) and across sequences of
> connections (temporal sharing).
>
> Joe
>
>>
>> Cheers,
>> Michael
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



From pasi.sarolahti@iki.fi  Thu May 30 01:36:38 2013
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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To: Joe Touch <touch@ISI.EDU>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On May 29, 2013, at 11:07 PM, Joe Touch <touch@ISI.EDU> wrote:

>> * PAWS algorithm tells to send acknowledgment if the check fails.
>> Shouldn't we then, for consistency, require sending acknowledgment =
also
>> if TSopt is missing? Current MAY in Sec. 3.2. is not very informative =
in
>> my mind.
>=20
> I don't think it's useful to send the ACK in that case. The point of =
PAWS is to ignore segments that are not protected. Even sending an ACK =
defeats that. At best, send a dup ack for last segment received =
correctly, but I still think that doing *anything* other than =
silent-drop isn't what PAWS protection intends.

The PAWS algorithm motivates sending ACK with recovering from half-open =
connections (referring to fig. 10 in RFC 793), when one side crashes and =
re-opens the connection. Wouldn't this same reasoning apply also for =
missing timestamps case? It is possible that after a reboot the system =
timestamps setting is reset from enabled to disabled, if that happens to =
be the boot-time default. In such case the newly opening connection =
would have its segments discarded by the other side that still remains =
in established state, because of lack of timestamps, isn't that right?

Admittedly this is not the most likely scenario, but just trying to dig =
up the reasoning for choosing one way or the other.

Since we are on the path of clarifying the text with normative language, =
why not do the same also for Section 4.3? We should decide

1) MAY/SHOULD/MUST send response for missing timestamp

2) MAY/SHOULD/MUST send response when timestamp comparison fails =
(current text on R1 seems to want that this is MUST)

It would be good to provide these with some reasoning, too. For (2) this =
reasoning already seems to be in place.

Nit: start of section 4.3 uses "REQUIRES" that should not be capitalized =
because it is not a RFC 2119 keyword. How about just saying something =
like "if PAWS algorithm is enabled, the following steps MUST be =
followed", or something. Sorry about splitting hairs, but I think we =
should be pedantic about the keywords.

>> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>>          segment, high speed connections that need PAWS would not =
have
>> that protection. Successful negotiation of Timestamps option
>> enforces a stricter verification of incoming segments at receiver. If
>> the Timestamps option was removed from a subsequent data segment
>> after a successful negotiation (for example, as part of
>> re-segmentation), the segment is discarded by the receiver without
>> further processing. Middleboxes should not remove the Timestamps
>> option. One study found that 6% of measured HTTP connections had the
>> Timestamps option removed from data segments [Honda11]."
>=20
> The last sentence seems out of place; why are you including that? If =
there are implications to the 6% drop then you should mention that, but =
I would expect that elsewhere than the security considerations section.

Because that paragraph talks about removal of timestamps option. But I =
agree that the proposed sentence doesn't sit there very well. How about =
including the reference in the beginning of the Middleboxes section of =
the text:

"Some middleboxes have been known to remove the TCP options
      described in this document from the TCP segments [Honda11]." =
(changed <SYN> segment to TCP segments)

The main point is that the current text only talks about dropping =
options from SYN segments, but this happens also for data segments, as =
the paper shows. Since we have fairly recent quantified data about this, =
why not include the reference?

- Pasi


From michawe@ifi.uio.no  Thu May 30 06:30:18 2013
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] Can't we use newcwv for IW too?
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Hi,

Even reading this 3 times doesn't make me understand it. Could it be =
that we have a misunderstanding and you're thinking of what RFC2140 =
calls "ensemble sharing", when what I mean is what RFC2140 calls =
"temporal sharing" ? Or are you just primarily concerned with a =
situation more complicated than I envisioned?

Let's see. I was only thinking of greedy applications for now. My idea =
was just that, if you have an application that e.g. stops sending and =
starts again, greedily, after 10 seconds, new-cwv has a clear rule on =
what cwnd should be.
And, if you replace this with an application that, as it stops, also =
closes the connection and after 10 seconds starts a new connection, and =
then immediately uses it to send, greedily, then there isn't much =
difference between this case and the case above, and IW could probably =
be set using a similar rule from new-cwv.

More below:

On 30. mai 2013, at 08:30, gorry@erg.abdn.ac.uk wrote:

>=20
> So here are some thoughts - if we were talking about "bulk" flows =
where
> the cwnd reflects the network capacity to a specific destination, then
> sharing of the cwnd is something that is being done by many stacks. In

Until I sent my first email about this, I wasn't aware of that (but I =
got a private message telling me, straight away). I knew that some parts =
of RFC2140 are implemented, but I didn't expect cwnd to be reused beyond =
the end of a connection to set the next IW. Anyway, as Joe said, RFC2140 =
is informational, and I don't think it has clear rules about how long =
cwnd could be kept etc., so what I'm suggesting is to make a =
recommendation to let new-cwv consider a connection that terminates the =
same way as it considers a connection that enters the non-validated =
phase, and then allows re-using the cwnd in a following connection by =
noticing that the non-validated phase ends. If a very similar reasonable =
behavior is already out there, then I guess all I'm saying is "legalise =
it!"  :-)


> reality the cwnd that is "shared" is often higher than that sustained =
on
> the path, since this is inflated each RTT (the extra between 1 MSS and
> cwnd).

I don't get this; sorry if it should be obvious. We're talking about the =
time between the end of connection 1 and the beginning of connection 2, =
surely this cached value isn't then updated every RTT when there's =
nothing going on? And what do you mean with "the extra between 1 MSS and =
cwnd" ?


> Now, if we have a flow that does not fully utilise cwnd (variously =
called
> thin,variable rate, bursty, intermittent, ...) then the cwnd can be =
much
> higher than the that sustained on the path. RFC 2861, attempted to =
keep
> the cwnd close to the actual used capacity, but this restricts the
> dynamics o an application and adds latency to applications that are
> bursty. Hence if we desire to enable these applications we should not
> aggressively trim the cwnd to what was used.
>=20
> New-cwv tries to avoid unecessary cwnd growth, but proposes to avoid
> forcing a low cwnd on a bursty application. Instead it creates a new =
phase
> (the non-validated phase) where cwnd exceeds the actual path capacity
> used, and where the cwnd growth is capped, but importantly the =
congestion
> response is changed to reduce more I think this helps motivate =
persistent
> use of TCP.
>=20
> Now, my question is what cwnd value should be "shared" when the flow =
is in
> the  non-validated phase? - If a flow was to share non-valiudated =
cwnd,
> and all flows implemented the new-cwv algorithm, then the flow that
> received the "extra" cwnd would be non-validated, and that would be =
OK. If
> it were a normal TCP flow, then I think this inflates cwnd in an
> unreasonable way.

This last paragraph gives me the impression that you're thinking of =
ensemble sharing, not temporal sharing. I'm not suggesting giving away =
the cwnd of a flow while it's running (even if it's taking a break). Yes =
that could be reasonable as you sketch it (only for new-cwv-supporting =
flows, to make flows go from non-validated to validated), but it's just =
a more complicated beast than what I was thinking about.

Cheers,
Michael


From gorry@erg.abdn.ac.uk  Thu May 30 07:05:56 2013
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] Can't we use newcwv for IW too?
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See in-lime...

> Hi,
>
> Even reading this 3 times doesn't make me understand it. Could it be that
> we have a misunderstanding and you're thinking of what RFC2140 calls
> "ensemble sharing", when what I mean is what RFC2140 calls "temporal
> sharing" ? Or are you just primarily concerned with a situation more
> complicated than I envisioned?
>
I was initially thinking of temporal sharing when the final TCB of an
earlier session is used to initialize parameters of a new TCP connection
to the same host.

> Let's see. I was only thinking of greedy applications for now. My idea was
> just that, if you have an application that e.g. stops sending and starts
> again, greedily, after 10 seconds, new-cwv has a clear rule on what cwnd
> should be.
>
All true for a bulk flow: new-cwv would leave the flow with the same cwnd
(likely) as TCP stack without new-cwv. AND in this case cwnd would reflect
the capacity of the network path that was used - validated in the language
of CWV (within a factor of 2 or less).

> And, if you replace this with an application that, as it stops, also
> closes the connection and after 10 seconds starts a new connection, and
> then immediately uses it to send, greedily, then there isn't much
> difference between this case and the case above, and IW could probably be
> set using a similar rule from new-cwv.
>
We may agree for bulk flows.

> More below:
>
> On 30. mai 2013, at 08:30, gorry@erg.abdn.ac.uk wrote:
>
>>
>> So here are some thoughts - if we were talking about "bulk" flows where
>> the cwnd reflects the network capacity to a specific destination, then
>> sharing of the cwnd is something that is being done by many stacks. In
>
> Until I sent my first email about this, I wasn't aware of that (but I got
> a private message telling me, straight away). I knew that some parts of
> RFC2140 are implemented, but I didn't expect cwnd to be reused beyond the
> end of a connection to set the next IW. Anyway, as Joe said, RFC2140 is
> informational, and I don't think it has clear rules about how long cwnd
> could be kept etc., so what I'm suggesting is to make a recommendation to
> let new-cwv consider a connection that terminates the same way as it
> considers a connection that enters the non-validated phase, and then
> allows re-using the cwnd in a following connection by noticing that the
> non-validated phase ends. If a very similar reasonable behavior is already
> out there, then I guess all I'm saying is "legalise it!"  :-)
>
*IF* the flow was validated when it closed, then using new-cwv rules for
decay of the value with time may be useful - this could control the size
of the contribution to the temporal pool for new flows, decaying with
time. i.e. the older the close was, the fewer the bytes you allow to be
reused in an initial window for a new flow.

If the values are not right in new-cwv, then I'd welcome that we can
discuss that...

>
>> reality the cwnd that is "shared" is often higher than that sustained on
>> the path, since this is inflated each RTT (the extra between 1 MSS and
>> cwnd).
>
> I don't get this; sorry if it should be obvious. We're talking about the
> time between the end of connection 1 and the beginning of connection 2,
> surely this cached value isn't then updated every RTT when there's nothing
> going on? And what do you mean with "the extra between 1 MSS and cwnd" ?
>
This is a small point that we have not discussed with respect to the final
value of cwnd in a closed flow.

cwnd reflects the estimate of what we think is safe for the next RTT,
rather than what was actually found to be safe from ACKs returned in the
last RTT (a fundamental difference between pipeACK and cwnd). I was simply
suggesting that we shouldn't restart at cwnd but 1/2 cwnd?

>
>> Now, if we have a flow that does not fully utilise cwnd (variously
>> called
>> thin,variable rate, bursty, intermittent, ...) then the cwnd can be much
>> higher than the that sustained on the path. RFC 2861, attempted to keep
>> the cwnd close to the actual used capacity, but this restricts the
>> dynamics o an application and adds latency to applications that are
>> bursty. Hence if we desire to enable these applications we should not
>> aggressively trim the cwnd to what was used.
>>
>> New-cwv tries to avoid unecessary cwnd growth, but proposes to avoid
>> forcing a low cwnd on a bursty application. Instead it creates a new
>> phase
>> (the non-validated phase) where cwnd exceeds the actual path capacity
>> used, and where the cwnd growth is capped, but importantly the
>> congestion
>> response is changed to reduce more I think this helps motivate
>> persistent
>> use of TCP.
>>
>> Now, my question is what cwnd value should be "shared" when the flow is
>> in
>> the  non-validated phase? - If a flow was to share non-valiudated cwnd,
>> and all flows implemented the new-cwv algorithm, then the flow that
>> received the "extra" cwnd would be non-validated, and that would be OK.
>> If
>> it were a normal TCP flow, then I think this inflates cwnd in an
>> unreasonable way.
>
> This last paragraph gives me the impression that you're thinking of
> ensemble sharing, not temporal sharing. I'm not suggesting giving away the
> cwnd of a flow while it's running (even if it's taking a break). Yes that
> could be reasonable as you sketch it (only for new-cwv-supporting flows,
> to make flows go from non-validated to validated), but it's just a more
> complicated beast than what I was thinking about.
>
It applies also to temporal sharing.

Here is the issue... when an app uses TCP as currently defined, it can
send as much or as little as it chooses. The cwnd reflects an upper limit
to the maximum the app can send on one RTT. For bulk flows the effect of
sharing is clear (above).

For a flow that has been sending less than cwnd, it would not be
reasonable to say that the non-validated cwnd value of a closed flow could
be used by a new flow that did not start in a non-validated phase. If this
happens, the flow could have "far too much credit".

Of course if the new flow also used new-cwv and restored other TCB state
it would be the same as if the same TCP flow had been used persistently.
That would be nice.

Gorry


> Cheers,
> Michael
>



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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>, Joe Touch <touch@ISI.EDU>
Thread-Topic: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Date: Thu, 30 May 2013 14:28:01 +0000
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Thanks Pasi, Joe.

> Even sending an ACK
> defeats that. At best, send a dup ack for last segment received correctly=
,
> but I still think that doing *anything* other than silent-drop isn't what
> PAWS protection intends.

My understanding is, that failing the PAWS test should be treated like rece=
iving=20
a segment that is out-of-window. The reason for failing the PAWS test (miss=
ing TSopt,=20
or invalid TSval) is secondary, not?

Thus an ACK for the last in-sequence segment should be sent.

> 1) MAY/SHOULD/MUST send response for missing timestamp

This case is not currently covered. As consensus was reached that TSopt sho=
uld not be allowed to be sent arbitratily for some segments and not for oth=
ers, I would think a missing timestamp in a segment means, that segment nee=
ds to be treated like any other out-of-window segment.


> 2) MAY/SHOULD/MUST send response when timestamp comparison fails (current
> text on R1 seems to want that this is MUST)

For both, I think a MUST would be a appropriate, a SHOULD may suffice. (as =
an individual contributor).


Fixed the Nit, and updated section 6 as indicated.

Best regards,

Richard Scheffenegger



> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
> Pasi Sarolahti
> Sent: Donnerstag, 30. Mai 2013 10:36
> To: Joe Touch
> Cc: tcpm@ietf.org Extensions
> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
>=20
> On May 29, 2013, at 11:07 PM, Joe Touch <touch@ISI.EDU> wrote:
>=20
> >> * PAWS algorithm tells to send acknowledgment if the check fails.
> >> Shouldn't we then, for consistency, require sending acknowledgment
> >> also if TSopt is missing? Current MAY in Sec. 3.2. is not very
> >> informative in my mind.
> >
> > I don't think it's useful to send the ACK in that case. The point of
> PAWS is to ignore segments that are not protected. Even sending an ACK
> defeats that. At best, send a dup ack for last segment received correctly=
,
> but I still think that doing *anything* other than silent-drop isn't what
> PAWS protection intends.
>=20
> The PAWS algorithm motivates sending ACK with recovering from half-open
> connections (referring to fig. 10 in RFC 793), when one side crashes and
> re-opens the connection. Wouldn't this same reasoning apply also for
> missing timestamps case? It is possible that after a reboot the system
> timestamps setting is reset from enabled to disabled, if that happens to
> be the boot-time default. In such case the newly opening connection would
> have its segments discarded by the other side that still remains in
> established state, because of lack of timestamps, isn't that right?
>=20
> Admittedly this is not the most likely scenario, but just trying to dig u=
p
> the reasoning for choosing one way or the other.
>=20
> Since we are on the path of clarifying the text with normative language,
> why not do the same also for Section 4.3? We should decide
>=20
> 1) MAY/SHOULD/MUST send response for missing timestamp
>=20
> 2) MAY/SHOULD/MUST send response when timestamp comparison fails (current
> text on R1 seems to want that this is MUST)
>=20
> It would be good to provide these with some reasoning, too. For (2) this
> reasoning already seems to be in place.
>=20
> Nit: start of section 4.3 uses "REQUIRES" that should not be capitalized
> because it is not a RFC 2119 keyword. How about just saying something lik=
e
> "if PAWS algorithm is enabled, the following steps MUST be followed", or
> something. Sorry about splitting hairs, but I think we should be pedantic
> about the keywords.
>=20
> >> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
> >>          segment, high speed connections that need PAWS would not
> >> have that protection. Successful negotiation of Timestamps option
> >> enforces a stricter verification of incoming segments at receiver. If
> >> the Timestamps option was removed from a subsequent data segment
> >> after a successful negotiation (for example, as part of
> >> re-segmentation), the segment is discarded by the receiver without
> >> further processing. Middleboxes should not remove the Timestamps
> >> option. One study found that 6% of measured HTTP connections had the
> >> Timestamps option removed from data segments [Honda11]."
> >
> > The last sentence seems out of place; why are you including that? If
> there are implications to the 6% drop then you should mention that, but I
> would expect that elsewhere than the security considerations section.
>=20
> Because that paragraph talks about removal of timestamps option. But I
> agree that the proposed sentence doesn't sit there very well. How about
> including the reference in the beginning of the Middleboxes section of th=
e
> text:
>=20
> "Some middleboxes have been known to remove the TCP options
>       described in this document from the TCP segments [Honda11]."
> (changed <SYN> segment to TCP segments)
>=20
> The main point is that the current text only talks about dropping options
> from SYN segments, but this happens also for data segments, as the paper
> shows. Since we have fairly recent quantified data about this, why not
> include the reference?
>=20
> - Pasi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

From touch@isi.edu  Thu May 30 09:53:42 2013
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Hi, all,

On 5/30/2013 7:28 AM, Scheffenegger, Richard wrote:
> Thanks Pasi, Joe.
>
>> Even sending an ACK
>> defeats that. At best, send a dup ack for last segment received correctly,
>> but I still think that doing *anything* other than silent-drop isn't what
>> PAWS protection intends.
>
> My understanding is, that failing the PAWS test should be treated like receiving
> a segment that is out-of-window. The reason for failing the PAWS test (missing TSopt,
> or invalid TSval) is secondary, not?
>
> Thus an ACK for the last in-sequence segment should be sent.
>
>> 1) MAY/SHOULD/MUST send response for missing timestamp
>
> This case is not currently covered. As consensus was reached that
> TSopt should not be allowed to be sent arbitratily for some segments and
> not for others, I would think a missing timestamp in a segment means,
> that segment needs to be treated like any other out-of-window segment.

Missing a timestamp is a bit different; it not only means that the 
segment should be treated as out-of-window, but there's something else 
amiss with the connection.

I.e., out-of-window might be fixed at some point, but missing TSopt will 
never be fixed. So I don't see what the point of sending a response here 
would be - it's like sending the last-valid ACK when you get any other 
malformed packet (because malformed != expired timestamp) -- which we 
don't do.

>> 2) MAY/SHOULD/MUST send response when timestamp comparison fails (current
>> text on R1 seems to want that this is MUST)
>
> For both, I think a MUST would be a appropriate, a SHOULD may
> suffice.(as an individual contributor).

FWIW, I never like "SHOULD" unless we have an explicit reason for not 
selecting MUST - either a known exception (which we should mention), or 
a reason for wanting to keep the door open.

In this case, I'm not sure what the appropriate behavior would be - 
what's the point of sending a response (e.g., an ACK of the last 
correctly received data) when the timestamp fails?

I.e., I wouldn't send a message unless it provides a purpose. Does it here?

Joe

>
>
> Fixed the Nit, and updated section 6 as indicated.
>
> Best regards,
>
> Richard Scheffenegger
>
>
>
>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
>> Pasi Sarolahti
>> Sent: Donnerstag, 30. Mai 2013 10:36
>> To: Joe Touch
>> Cc: tcpm@ietf.org Extensions
>> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
>>
>> On May 29, 2013, at 11:07 PM, Joe Touch <touch@ISI.EDU> wrote:
>>
>>>> * PAWS algorithm tells to send acknowledgment if the check fails.
>>>> Shouldn't we then, for consistency, require sending acknowledgment
>>>> also if TSopt is missing? Current MAY in Sec. 3.2. is not very
>>>> informative in my mind.
>>>
>>> I don't think it's useful to send the ACK in that case. The point of
>> PAWS is to ignore segments that are not protected. Even sending an ACK
>> defeats that. At best, send a dup ack for last segment received correctly,
>> but I still think that doing *anything* other than silent-drop isn't what
>> PAWS protection intends.
>>
>> The PAWS algorithm motivates sending ACK with recovering from half-open
>> connections (referring to fig. 10 in RFC 793), when one side crashes and
>> re-opens the connection. Wouldn't this same reasoning apply also for
>> missing timestamps case? It is possible that after a reboot the system
>> timestamps setting is reset from enabled to disabled, if that happens to
>> be the boot-time default. In such case the newly opening connection would
>> have its segments discarded by the other side that still remains in
>> established state, because of lack of timestamps, isn't that right?
>>
>> Admittedly this is not the most likely scenario, but just trying to dig up
>> the reasoning for choosing one way or the other.
>>
>> Since we are on the path of clarifying the text with normative language,
>> why not do the same also for Section 4.3? We should decide
>>
>> 1) MAY/SHOULD/MUST send response for missing timestamp
>>
>> 2) MAY/SHOULD/MUST send response when timestamp comparison fails (current
>> text on R1 seems to want that this is MUST)
>>
>> It would be good to provide these with some reasoning, too. For (2) this
>> reasoning already seems to be in place.
>>
>> Nit: start of section 4.3 uses "REQUIRES" that should not be capitalized
>> because it is not a RFC 2119 keyword. How about just saying something like
>> "if PAWS algorithm is enabled, the following steps MUST be followed", or
>> something. Sorry about splitting hairs, but I think we should be pedantic
>> about the keywords.
>>
>>>> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>>>>           segment, high speed connections that need PAWS would not
>>>> have that protection. Successful negotiation of Timestamps option
>>>> enforces a stricter verification of incoming segments at receiver. If
>>>> the Timestamps option was removed from a subsequent data segment
>>>> after a successful negotiation (for example, as part of
>>>> re-segmentation), the segment is discarded by the receiver without
>>>> further processing. Middleboxes should not remove the Timestamps
>>>> option. One study found that 6% of measured HTTP connections had the
>>>> Timestamps option removed from data segments [Honda11]."
>>>
>>> The last sentence seems out of place; why are you including that? If
>> there are implications to the 6% drop then you should mention that, but I
>> would expect that elsewhere than the security considerations section.
>>
>> Because that paragraph talks about removal of timestamps option. But I
>> agree that the proposed sentence doesn't sit there very well. How about
>> including the reference in the beginning of the Middleboxes section of the
>> text:
>>
>> "Some middleboxes have been known to remove the TCP options
>>        described in this document from the TCP segments [Honda11]."
>> (changed <SYN> segment to TCP segments)
>>
>> The main point is that the current text only talks about dropping options
>> from SYN segments, but this happens also for data segments, as the paper
>> shows. Since we have fairly recent quantified data about this, why not
>> include the reference?
>>
>> - Pasi
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm

From touch@isi.edu  Thu May 30 09:59:34 2013
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Hi, all,

On 5/30/2013 1:36 AM, Pasi Sarolahti wrote:
> On May 29, 2013, at 11:07 PM, Joe Touch <touch@ISI.EDU> wrote:
...
> Nit: start of section 4.3 uses "REQUIRES" that should not be
> capitalized because it is not a RFC 2119 keyword. How about just saying
> something like "if PAWS algorithm is enabled, the following steps MUST
> be followed", or something. Sorry about splitting hairs, but I think we
> should be pedantic about the keywords.

REQUIRED is a 2119 keyword; I can't see why the present tense shouldn't 
be treated the same way, but MUST is fine too.

>>> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>>>           segment, high speed connections that need PAWS would not have
>>> that protection. Successful negotiation of Timestamps option
>>> enforces a stricter verification of incoming segments at receiver. If
>>> the Timestamps option was removed from a subsequent data segment
>>> after a successful negotiation (for example, as part of
>>> re-segmentation), the segment is discarded by the receiver without
>>> further processing. Middleboxes should not remove the Timestamps
>>> option. One study found that 6% of measured HTTP connections had the
>>> Timestamps option removed from data segments [Honda11]."
>>
>> The last sentence seems out of place; why are you including that?
>> Ifthere are implications to the 6% drop then you should mention that, but
>> I would expect that elsewhere than the security considerations section.
>
> Because that paragraph talks about removal of timestamps option. But
> I agree that the proposed sentence doesn't sit there very well. How
> about including the reference in the beginning of the Middleboxes
> section of the text:
>
> "Some middleboxes have been known to remove the TCP options
>        described in this document from the TCP segments [Honda11]." (changed <SYN> segment to TCP segments)

Yup - that's better, IMO.

> The main point is that the current text only talks about dropping
> options from SYN segments, but this happens also for data segments, as
> the paper shows. Since we have fairly recent quantified data about this,
> why not include the reference?

That's fine, it was the placement issue.

However, if your point is that they remove options from data segments 
then you might point that out explicitly in the revised text.

Question - is there a reason a midbox would let the option through on a 
SYN and block it on the data segments? Or are you worried about path 
changes whose data segments go through midboxes that the SYN didn't? 
Again, if that's the case, then you might explain that in the text above 
in the midbox section (1-2 sentences).

Joe

From l.wood@surrey.ac.uk  Thu May 30 17:37:55 2013
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From: <l.wood@surrey.ac.uk>
To: <touch@isi.edu>, <pasi.sarolahti@iki.fi>
Date: Fri, 31 May 2013 01:36:53 +0100
Thread-Topic: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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> REQUIRED is a 2119 keyword; I can't see why the present tense shouldn't
> be treated the same way, but MUST is fine too.

If the present tense is permitted, then the past tense (WAS REQUIRED) is al=
so allowed.

Ambiguous, that.

Lloyd Wood
http://sat-net.com/L.Wood/


________________________________________
From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] On Behalf Of Joe Touch =
[touch@isi.edu]
Sent: 30 May 2013 17:59
To: Pasi Sarolahti
Cc: tcpm@ietf.org Extensions
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt

Hi, all,

On 5/30/2013 1:36 AM, Pasi Sarolahti wrote:
> On May 29, 2013, at 11:07 PM, Joe Touch <touch@ISI.EDU> wrote:
...
> Nit: start of section 4.3 uses "REQUIRES" that should not be
> capitalized because it is not a RFC 2119 keyword. How about just saying
> something like "if PAWS algorithm is enabled, the following steps MUST
> be followed", or something. Sorry about splitting hairs, but I think we
> should be pedantic about the keywords.

REQUIRED is a 2119 keyword; I can't see why the present tense shouldn't
be treated the same way, but MUST is fine too.

>>> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>>>           segment, high speed connections that need PAWS would not have
>>> that protection. Successful negotiation of Timestamps option
>>> enforces a stricter verification of incoming segments at receiver. If
>>> the Timestamps option was removed from a subsequent data segment
>>> after a successful negotiation (for example, as part of
>>> re-segmentation), the segment is discarded by the receiver without
>>> further processing. Middleboxes should not remove the Timestamps
>>> option. One study found that 6% of measured HTTP connections had the
>>> Timestamps option removed from data segments [Honda11]."
>>
>> The last sentence seems out of place; why are you including that?
>> Ifthere are implications to the 6% drop then you should mention that, bu=
t
>> I would expect that elsewhere than the security considerations section.
>
> Because that paragraph talks about removal of timestamps option. But
> I agree that the proposed sentence doesn't sit there very well. How
> about including the reference in the beginning of the Middleboxes
> section of the text:
>
> "Some middleboxes have been known to remove the TCP options
>        described in this document from the TCP segments [Honda11]." (chan=
ged <SYN> segment to TCP segments)

Yup - that's better, IMO.

> The main point is that the current text only talks about dropping
> options from SYN segments, but this happens also for data segments, as
> the paper shows. Since we have fairly recent quantified data about this,
> why not include the reference?

That's fine, it was the placement issue.

However, if your point is that they remove options from data segments
then you might point that out explicitly in the revised text.

Question - is there a reason a midbox would let the option through on a
SYN and block it on the data segments? Or are you worried about path
changes whose data segments go through midboxes that the SYN didn't?
Again, if that's the case, then you might explain that in the text above
in the midbox section (1-2 sentences).

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

From touch@isi.edu  Thu May 30 17:49:37 2013
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On 5/30/2013 5:36 PM, l.wood@surrey.ac.uk wrote:
>> REQUIRED is a 2119 keyword; I can't see why the present tense shouldn't
>> be treated the same way, but MUST is fine too.
>
> If the present tense is permitted, then the past tense (WAS REQUIRED) is also allowed.

REQUIRED is an adjective, describing a state:

	Y is REQUIRED

X REQUIRES Y is the present tense verb phrase equivalent of:

	Y is REQUIRED by X

If you prefer to use only 2119 language, use the "Y is REQUIRED by X" 
variant.

"was REQUIRED" means past tense, and doesn't thus describe a current 
constraint. I don't think that makes much 2119 sense.

Joe

> ________________________________________
> From: tcpm-bounces@ietf.org [tcpm-bounces@ietf.org] On Behalf Of Joe Touch [touch@isi.edu]
> Sent: 30 May 2013 17:59
> To: Pasi Sarolahti
> Cc: tcpm@ietf.org Extensions
> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
>
> Hi, all,
>
> On 5/30/2013 1:36 AM, Pasi Sarolahti wrote:
>> On May 29, 2013, at 11:07 PM, Joe Touch <touch@ISI.EDU> wrote:
> ...
>> Nit: start of section 4.3 uses "REQUIRES" that should not be
>> capitalized because it is not a RFC 2119 keyword. How about just saying
>> something like "if PAWS algorithm is enabled, the following steps MUST
>> be followed", or something. Sorry about splitting hairs, but I think we
>> should be pedantic about the keywords.
>
> REQUIRED is a 2119 keyword; I can't see why the present tense shouldn't
> be treated the same way, but MUST is fine too.
>
>>>> " *  If the Timestamps option is removed from the <SYN> or <SYN,ACK>
>>>>            segment, high speed connections that need PAWS would not have
>>>> that protection. Successful negotiation of Timestamps option
>>>> enforces a stricter verification of incoming segments at receiver. If
>>>> the Timestamps option was removed from a subsequent data segment
>>>> after a successful negotiation (for example, as part of
>>>> re-segmentation), the segment is discarded by the receiver without
>>>> further processing. Middleboxes should not remove the Timestamps
>>>> option. One study found that 6% of measured HTTP connections had the
>>>> Timestamps option removed from data segments [Honda11]."
>>>
>>> The last sentence seems out of place; why are you including that?
>>> Ifthere are implications to the 6% drop then you should mention that, but
>>> I would expect that elsewhere than the security considerations section.
>>
>> Because that paragraph talks about removal of timestamps option. But
>> I agree that the proposed sentence doesn't sit there very well. How
>> about including the reference in the beginning of the Middleboxes
>> section of the text:
>>
>> "Some middleboxes have been known to remove the TCP options
>>         described in this document from the TCP segments [Honda11]." (changed <SYN> segment to TCP segments)
>
> Yup - that's better, IMO.
>
>> The main point is that the current text only talks about dropping
>> options from SYN segments, but this happens also for data segments, as
>> the paper shows. Since we have fairly recent quantified data about this,
>> why not include the reference?
>
> That's fine, it was the placement issue.
>
> However, if your point is that they remove options from data segments
> then you might point that out explicitly in the revised text.
>
> Question - is there a reason a midbox would let the option through on a
> SYN and block it on the data segments? Or are you worried about path
> changes whose data segments go through midboxes that the SYN didn't?
> Again, if that's the case, then you might explain that in the text above
> in the midbox section (1-2 sentences).
>
> Joe
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From fgont@si6networks.com  Fri May 31 03:52:09 2013
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@ISI.EDU>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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Hi, Richard,

For the most part I've not participated in this discussion at all, but
at times have lurked a bit... A small comment here:

On 05/30/2013 04:28 PM, Scheffenegger, Richard wrote:
> 
> My understanding is, that failing the PAWS test should be treated like receiving 
> a segment that is out-of-window. The reason for failing the PAWS test (missing TSopt, 
> or invalid TSval) is secondary, not?
> 
> Thus an ACK for the last in-sequence segment should be sent.
> 
>> 1) MAY/SHOULD/MUST send response for missing timestamp
> 
> This case is not currently covered. As consensus was reached that TSopt should not be allowed to be sent arbitratily for some segments and not for others, I would think a missing timestamp in a segment means, that segment needs to be treated like any other out-of-window segment.

Datapoint: Some OSes accept non-timestamped packets because they've fund
that some implementations (or maybe sites "protected" by some sort of
middlebox) at times stop sending timestamps, or include timstamp options
at their own discretion.

Not that I think that accepting such packets is nice (actually, it
doesn't look "clean") -- but at the end of the day, you need to
interoperate with others.

I suggest that the OpenBSD folks are asked about this one... since I
think they were the ones noting the aforementioned issue.

Cheers, and thank you guys for the hard work on rfc1323bis!
-- 
Fernando Gont
SI6 Networks
e-mail: fgont@si6networks.com
PGP Fingerprint: 6666 31C6 D484 63B2 8FB1 E3C4 AE25 0D55 1D4E 7492





From pasi.sarolahti@iki.fi  Fri May 31 04:28:04 2013
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To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On May 30, 2013, at 7:59 PM, Joe Touch <touch@isi.edu> wrote:
>=20
>> The main point is that the current text only talks about dropping
>> options from SYN segments, but this happens also for data segments, =
as
>> the paper shows. Since we have fairly recent quantified data about =
this,
>> why not include the reference?
>=20
> That's fine, it was the placement issue.
>=20
> However, if your point is that they remove options from data segments =
then you might point that out explicitly in the revised text.
>=20
> Question - is there a reason a midbox would let the option through on =
a SYN and block it on the data segments? Or are you worried about path =
changes whose data segments go through midboxes that the SYN didn't? =
Again, if that's the case, then you might explain that in the text above =
in the midbox section (1-2 sentences).

The paper only speaks about timestamps in data segments, but I didn't =
notice separate analysis on SYN segments, which in retrospect might have =
been interesting to see separately. But yes, in most of the mentioned =
cases probably the option is removed from all segments altogether.

For removing timestamps only from some segments, path changes might be =
one reason, and in earlier mail re-segmenting middleboxes that didn't =
properly support timestamps were mentioned as another possible reason. =
Also TSO is quite common these days, and it is not impossible that some =
TSO implementations did not properly support timestamps, although I =
believe that most of them do. So I don't believe this would be a common =
problem, and can't point out to particular case where this happens, but =
even if it occurred only rarely, the users that are unlucky enough to be =
behind that unfortunate path would constantly suffer quite a performance =
hit, before they learn how to disable timestamps.

I was playing with a new experimental option in our test lab some time =
ago, and there TSO caused the option to disappear from some of the =
segments (those that happened to exceed the native segment size when =
leaving OS), but the option was included in segments that left the OS as =
small enough to fit on wire. Disabling TSO solved the issue. But I also =
checked that timestamps were handled correctly in that case :-)

- Pasi


From michawe@ifi.uio.no  Fri May 31 08:28:04 2013
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] Can't we use newcwv for IW too?
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On May 30, 2013, at 4:05 PM, gorry@erg.abdn.ac.uk wrote:

> See in-lime=85

I even licked it, it's very sour!   :-D

Now you do the same, in lime below:


>> Hi,
>>=20
>> Even reading this 3 times doesn't make me understand it. Could it be =
that
>> we have a misunderstanding and you're thinking of what RFC2140 calls
>> "ensemble sharing", when what I mean is what RFC2140 calls "temporal
>> sharing" ? Or are you just primarily concerned with a situation more
>> complicated than I envisioned?
>>=20
> I was initially thinking of temporal sharing when the final TCB of an
> earlier session is used to initialize parameters of a new TCP =
connection
> to the same host.

ok


>> Let's see. I was only thinking of greedy applications for now. My =
idea was
>> just that, if you have an application that e.g. stops sending and =
starts
>> again, greedily, after 10 seconds, new-cwv has a clear rule on what =
cwnd
>> should be.
>>=20
> All true for a bulk flow: new-cwv would leave the flow with the same =
cwnd
> (likely) as TCP stack without new-cwv. AND in this case cwnd would =
reflect
> the capacity of the network path that was used - validated in the =
language
> of CWV (within a factor of 2 or less).

yes


>> And, if you replace this with an application that, as it stops, also
>> closes the connection and after 10 seconds starts a new connection, =
and
>> then immediately uses it to send, greedily, then there isn't much
>> difference between this case and the case above, and IW could =
probably be
>> set using a similar rule from new-cwv.
>>=20
> We may agree for bulk flows.

It seems like it!


>> More below:
>>=20
>> On 30. mai 2013, at 08:30, gorry@erg.abdn.ac.uk wrote:
>>=20
>>>=20
>>> So here are some thoughts - if we were talking about "bulk" flows =
where
>>> the cwnd reflects the network capacity to a specific destination, =
then
>>> sharing of the cwnd is something that is being done by many stacks. =
In
>>=20
>> Until I sent my first email about this, I wasn't aware of that (but I =
got
>> a private message telling me, straight away). I knew that some parts =
of
>> RFC2140 are implemented, but I didn't expect cwnd to be reused beyond =
the
>> end of a connection to set the next IW. Anyway, as Joe said, RFC2140 =
is
>> informational, and I don't think it has clear rules about how long =
cwnd
>> could be kept etc., so what I'm suggesting is to make a =
recommendation to
>> let new-cwv consider a connection that terminates the same way as it
>> considers a connection that enters the non-validated phase, and then
>> allows re-using the cwnd in a following connection by noticing that =
the
>> non-validated phase ends. If a very similar reasonable behavior is =
already
>> out there, then I guess all I'm saying is "legalise it!"  :-)
>>=20
> *IF* the flow was validated when it closed, then using new-cwv rules =
for
> decay of the value with time may be useful - this could control the =
size
> of the contribution to the temporal pool for new flows, decaying with
> time. i.e. the older the close was, the fewer the bytes you allow to =
be
> reused in an initial window for a new flow.

Yes!


> If the values are not right in new-cwv, then I'd welcome that we can
> discuss that=85

Yes, below


>>> reality the cwnd that is "shared" is often higher than that =
sustained on
>>> the path, since this is inflated each RTT (the extra between 1 MSS =
and
>>> cwnd).
>>=20
>> I don't get this; sorry if it should be obvious. We're talking about =
the
>> time between the end of connection 1 and the beginning of connection =
2,
>> surely this cached value isn't then updated every RTT when there's =
nothing
>> going on? And what do you mean with "the extra between 1 MSS and =
cwnd" ?
>>=20
> This is a small point that we have not discussed with respect to the =
final
> value of cwnd in a closed flow.

Got it. Now I understand what you meant -


> cwnd reflects the estimate of what we think is safe for the next RTT,
> rather than what was actually found to be safe from ACKs returned in =
the
> last RTT (a fundamental difference between pipeACK and cwnd). I was =
simply
> suggesting that we shouldn't restart at cwnd but 1/2 cwnd?

Yes, why not. I hadn't really thought of this case yet.


>>> Now, if we have a flow that does not fully utilise cwnd (variously
>>> called
>>> thin,variable rate, bursty, intermittent, ...) then the cwnd can be =
much
>>> higher than the that sustained on the path. RFC 2861, attempted to =
keep
>>> the cwnd close to the actual used capacity, but this restricts the
>>> dynamics o an application and adds latency to applications that are
>>> bursty. Hence if we desire to enable these applications we should =
not
>>> aggressively trim the cwnd to what was used.
>>>=20
>>> New-cwv tries to avoid unecessary cwnd growth, but proposes to avoid
>>> forcing a low cwnd on a bursty application. Instead it creates a new
>>> phase
>>> (the non-validated phase) where cwnd exceeds the actual path =
capacity
>>> used, and where the cwnd growth is capped, but importantly the
>>> congestion
>>> response is changed to reduce more I think this helps motivate
>>> persistent
>>> use of TCP.
>>>=20
>>> Now, my question is what cwnd value should be "shared" when the flow =
is
>>> in
>>> the  non-validated phase? - If a flow was to share non-valiudated =
cwnd,
>>> and all flows implemented the new-cwv algorithm, then the flow that
>>> received the "extra" cwnd would be non-validated, and that would be =
OK.
>>> If
>>> it were a normal TCP flow, then I think this inflates cwnd in an
>>> unreasonable way.
>>=20
>> This last paragraph gives me the impression that you're thinking of
>> ensemble sharing, not temporal sharing. I'm not suggesting giving =
away the
>> cwnd of a flow while it's running (even if it's taking a break). Yes =
that
>> could be reasonable as you sketch it (only for new-cwv-supporting =
flows,
>> to make flows go from non-validated to validated), but it's just a =
more
>> complicated beast than what I was thinking about.
>>=20
> It applies also to temporal sharing.
>=20
> Here is the issue... when an app uses TCP as currently defined, it can
> send as much or as little as it chooses. The cwnd reflects an upper =
limit
> to the maximum the app can send on one RTT. For bulk flows the effect =
of
> sharing is clear (above).
>=20
> For a flow that has been sending less than cwnd, it would not be
> reasonable to say that the non-validated cwnd value of a closed flow =
could
> be used by a new flow that did not start in a non-validated phase. If =
this
> happens, the flow could have "far too much credit".

Yes.


> Of course if the new flow also used new-cwv and restored other TCB =
state
> it would be the same as if the same TCP flow had been used =
persistently.
> That would be nice.

Not sure I get that. The good news is, it's the only paragraph in your =
email that I can't follow, and it's much shorter than last time  :-)

Cheers,
Michael


From touch@isi.edu  Fri May 31 10:25:58 2013
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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On 5/31/2013 3:51 AM, Fernando Gont wrote:
> Hi, Richard,
>
> For the most part I've not participated in this discussion at all, but
> at times have lurked a bit... A small comment here:
>
> On 05/30/2013 04:28 PM, Scheffenegger, Richard wrote:
>>
>> My understanding is, that failing the PAWS test should be treated like receiving
>> a segment that is out-of-window. The reason for failing the PAWS test (missing TSopt,
>> or invalid TSval) is secondary, not?
>>
>> Thus an ACK for the last in-sequence segment should be sent.
>>
>>> 1) MAY/SHOULD/MUST send response for missing timestamp
>>
>> This case is not currently covered. As consensus was reached that TSopt should not be allowed to be sent arbitratily for some segments and not for others, I would think a missing timestamp in a segment means, that segment needs to be treated like any other out-of-window segment.
>
> Datapoint: Some OSes accept non-timestamped packets because they've fund
> that some implementations (or maybe sites "protected" by some sort of
> middlebox) at times stop sending timestamps, or include timstamp options
> at their own discretion.
>
> Not that I think that accepting such packets is nice (actually, it
> doesn't look "clean") -- but at the end of the day, you need to
> interoperate with others.

Interoperation has two interpretations:

- accept packets under any circumstances

- accept packets that conform to the requirements of the connection

It's critical to note that accepting non-TSopt packets in a connection 
that has successfully negotiated TSopt effectively defeats PAWS. There's 
no such thing as "just for this packet".

So the key question is:
	
	- should a connection follow the constraints of its
	negotiation, or should being "interoperable" trump
	the entire existence of the PAWS feature?

I don't think so. I.e., being "interoperable" here means "being incorrect".

Joe

From shep@xplot.org  Fri May 31 11:47:59 2013
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From: Tim Shepard <shep@alum.mit.edu>
To: Joe Touch <touch@isi.edu>
In-reply-to: Your message of Fri, 31 May 2013 10:24:59 -0700. <51A8DCEB.2090401@isi.edu> 
Date: Fri, 31 May 2013 14:47:36 -0400
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Cc: Fernando Gont <fgont@si6networks.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-1323bis-13.txt
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If a TCP connection has succesfully negotiated timestamp option on,
and a segment shows up with no timestamp option, I can think of only
two reasons for this:

	1.   The other TCP is buggy in some way.  (Or maybe *this*
             TCP implementation is buggy in some way and corrupted
	     the TCB block variable that remembers whether we've got
	     timestamp options on or off.  In any case, there is a
	     bug.)
	
	2.   The segment that arrived is from some old TCP connection,
	     probably to or from some previous user of a reused IP
	     address.  

	     (The MSL is a pure myth.  There's no way to guarantee any
	     upper bound for how long some packet may be delayed in
	     transit.  15+ years ago I did a trivially easy
	     demonstration that delayed several ping packets for over
	     an hour (over lunch), with no Internet routers involved,
	     on a LAN using hardware that was in widespread use at the
	     time.  E.g. Someone trips over a cable which slightly
	     dislodges it, then hours or days later, someone later
	     notices and pushes it back in.)


So, you can either accept the packet (A) or drop the packet (B).

    1. (A)     hmm... other end is buggy, *maybe* accepting the packet
	       would be the right thing.

    1. (B)     other end is buggy, dropping the packet *might* cause a
	       TCP connection to get stuck that otherwise would not
	       have gotten stuck.  (Hard to say... the other end is buggy!)


    2. (A)     data corruption (if the packet is otherwise acceptable)

    2. (B)     clearly the right thing to do.


With no reliable way to distinguish case 1 from case 2 at the
receiver, my gut is to prefer option B and live with the possible
downside of  the  1. (B)  situation.

	     
I also think letting the connection get stuck in the 1. (B) situation
has an upside... it might draw attention to the bug and cause the bug
to get fixed.  

... as long as there is no ambiguity on which of the two TCP
implementations is buggy.  (If we have this problem, then we have a
much bigger problem.)


			-Tim Shepard
			 shep@alum.mit.edu
