
From henk@ripe.net  Mon Oct  4 01:17:47 2010
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IPPM group,

We will meet in Beijing at IETF79.  If you have any topics to discuss,
please send us a note,

Matt & Henk
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From acmorton@att.com  Wed Oct  6 14:10:31 2010
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Date: Wed, 06 Oct 2010 17:11:32 -0400
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Subject: [ippm] Fwd: I-D Action:draft-morton-ippm-rt-loss-01.txt
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IPPM,

I presented this draft in Maastrict, and would like to get
some more comments on this updated version.

regards,
Al


>From: Internet-Drafts@ietf.org
>To: i-d-announce@ietf.org
>Subject: I-D Action:draft-morton-ippm-rt-loss-01.txt
>Date: Wed,  6 Oct 2010 13:45:03 -0700 (PDT)
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>
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>
>         Title           : Round-trip Loss Metrics
>         Author(s)       : A. Morton
>         Filename        : draft-morton-ippm-rt-loss-01.txt
>         Pages           : 11
>         Date            : 2010-10-06
>
>Many user applications and the transport protocols that make them
>possible require two-way communications.  To address this need, and
>also for system simplicity, round-trip loss measurements are
>frequently conducted in practice.  The Two-Way Active Measurement
>Protocol specified in RFC 5357 establishes a round-trip loss
>measurement capability for the Internet.  However, there is currently
>no metric specified according to the RFC 2330 framework.
>
>This memo proposes/adds round-trip loss to the set of IP Performance
>Metrics (IPPM).
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-morton-ippm-rt-loss-01.txt
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>Below is the data which will enable a MIME compliant mail reader
>implementation to automatically retrieve the ASCII version of the
>Internet-Draft.
>
>
><ftp://ftp.ietf.org/internet-drafts/draft-morton-ippm-rt-loss-01.txt>
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>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


From Barry.Constantine@jdsu.com  Thu Oct  7 06:23:28 2010
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From: Barry Constantine <Barry.Constantine@jdsu.com>
To: Al Morton <acmorton@att.com>, "ippm@ietf.org" <ippm@ietf.org>
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Subject: Re: [ippm] Fwd: I-D Action:draft-morton-ippm-rt-loss-01.txt
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Hi Al,

I reviewed the document and have one comment / question.

Should this draft discuss implications of packet re-ordering to round trip =
loss metrics (in the context of RFC4737)?

Thanks,

Barry

=20

Principal Member of Technical Staff

=20

JDSU Communication Test (formerly Acterna)

Emerging Markets and Technology Research        =20

One Milestone Center Court                             =20

Germantown, MD 20876                                        =20

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(C) 301-325-7069


-----Original Message-----
From: ippm-bounces@ietf.org [mailto:ippm-bounces@ietf.org] On Behalf Of Al =
Morton
Sent: Wednesday, October 06, 2010 5:12 PM
To: ippm@ietf.org
Subject: [ippm] Fwd: I-D Action:draft-morton-ippm-rt-loss-01.txt

IPPM,

I presented this draft in Maastrict, and would like to get
some more comments on this updated version.

regards,
Al


>From: Internet-Drafts@ietf.org
>To: i-d-announce@ietf.org
>Subject: I-D Action:draft-morton-ippm-rt-loss-01.txt
>Date: Wed,  6 Oct 2010 13:45:03 -0700 (PDT)
>X-BeenThere: i-d-announce@ietf.org
>Sender: i-d-announce-bounces@ietf.org
>
>A New Internet-Draft is available from the on-line Internet-Drafts=20
>directories.
>
>         Title           : Round-trip Loss Metrics
>         Author(s)       : A. Morton
>         Filename        : draft-morton-ippm-rt-loss-01.txt
>         Pages           : 11
>         Date            : 2010-10-06
>
>Many user applications and the transport protocols that make them
>possible require two-way communications.  To address this need, and
>also for system simplicity, round-trip loss measurements are
>frequently conducted in practice.  The Two-Way Active Measurement
>Protocol specified in RFC 5357 establishes a round-trip loss
>measurement capability for the Internet.  However, there is currently
>no metric specified according to the RFC 2330 framework.
>
>This memo proposes/adds round-trip loss to the set of IP Performance
>Metrics (IPPM).
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-morton-ippm-rt-loss-01.txt
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>Below is the data which will enable a MIME compliant mail reader
>implementation to automatically retrieve the ASCII version of the
>Internet-Draft.
>
>
><ftp://ftp.ietf.org/internet-drafts/draft-morton-ippm-rt-loss-01.txt>
>_______________________________________________
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>I-D-Announce@ietf.org
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>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt

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

From acmorton@att.com  Thu Oct  7 07:12:42 2010
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Date: Thu, 07 Oct 2010 10:14:00 -0400
To: Barry Constantine <Barry.Constantine@jdsu.com>, "ippm@ietf.org" <ippm@ietf.org>
From: Al Morton <acmorton@att.com>
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Hi Barry,

At 09:23 AM 10/7/2010, Barry Constantine wrote:
>I reviewed the document and have one comment / question.
>
>Should this draft discuss implications of packet re-ordering to 
>round trip loss metrics (in the context of RFC4737)?

I think so, there are certainly implications to the return path
sampling process if the reordering occurs in the forward direction,
and the Src and Dst sequence numbers will not line-up, even if there
is no loss concurrent with the reordering...

Thanks for your review.  I'll try to update this again before the
meeting.  If I've overlooked some implications, let me know...

Would you support this as a chartered WG item?

Al



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<body>
Matt,<br><br>
You asked that I take a look at:<br><br>
<blockquote type=cite class=cite cite=""><h3><b><a name="section-4.5"></a>
4.5.&nbsp; Reordering</b></h3><pre>

&nbsp;&nbsp; Reordering is the fraction of unique packets received for
which the
&nbsp;&nbsp; sequence number of any given packet is less than the highest
sequence
&nbsp;&nbsp; number largest sequence number of any packet previously
received.
</pre></blockquote>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
^^^^^^^^^^^^^^^^^^^^^^^<br>
I think you identified this redundant text during the meeting,
right?&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br><br>
<blockquote type=cite class=cite cite=""><pre>&nbsp;&nbsp; For the
purposes of determining the sequence number of the preceding
&nbsp;&nbsp; packets in this definition, assume sequence numbers starting
with 1,
&nbsp;&nbsp; and an extra packet at the start of the stream of received
packets,
&nbsp;&nbsp; with a sequence number of 0, is considered to be present
(this extra
&nbsp;&nbsp; packet, of course, is not counted for the purposes of
computing the
&nbsp;&nbsp; fraction).
</pre></blockquote><br>
I think your word definition is pretty close, but there should be an<br>
explicit designation of a reordered packet, IMO.&nbsp; Also, the
&quot;unique&quot; aspect<br>
is a fairly subtle way to exclude any duplicate packet from being
reordered.&nbsp; <br>
We could explain this fairly easily.<br><br>
I suggest:<br><br>
4.5. Reordering<br><br>
|&nbsp; Reordering is the fraction of unique packets received for which
the sequence <br>
|&nbsp; number of a packet designated &quot;reordered&quot; is less than
the highest sequence<br>
|&nbsp; number of any packet previously received, and the
&quot;unique&quot; aspect excludes<br>
|&nbsp; any Duplicate packets (as described in section 4.4 above).<br>
&nbsp;&nbsp; For the purposes of determining the sequence number of the
preceding<br>
&nbsp;&nbsp; packets in this definition, assume sequence numbers starting
with 1,<br>
&nbsp;&nbsp; and an extra packet at the start of the stream of received
packets,<br>
&nbsp;&nbsp; with a sequence number of 0, is considered to be present
(this extra<br>
&nbsp;&nbsp; packet, of course, is not counted for the purposes of
computing the<br>
&nbsp;&nbsp; fraction).<br><br>
hope this helps,<br>
Al<br><br>
</body>
</html>


From Anura.Jayasumana@Colostate.edu  Thu Oct  7 13:25:35 2010
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From: Anura Jayasumana <Anura.Jayasumana@Colostate.edu>
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Subject: Re: [ippm] Reordering Def in draft-ietf-ippm-reporting-05
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  Additional information in this topic can be found in  RFC5236 
(http://www.faqs.org/rfcs/rfc5236.html)
as well as at   
http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications

I think 5236 provides a cleaner way to handle this.
-- 
Anura P. Jayasumana <http://www.engr.colostate.edu/%7Eanura>
Professor, Electrical & Computer Engineering
                            and Computer Science
Department of Electrical and Computer Engineering
Colorado State University
Fort Collins, CO 80523
Phone: (970) 491-7855


On 10/7/2010 1:53 PM, Al Morton wrote:
> Matt,
>
> You asked that I take a look at:
>
>>
>>       * 4.5.  Reordering*
>>
>>     Reordering is the fraction of unique packets received for
>> which the
>>     sequence number of any given packet is less than the highest
>> sequence
>>     number largest sequence number of any packet previously
>> received.
>            ^^^^^^^^^^^^^^^^^^^^^^^
> I think you identified this redundant text during the meeting, right?
>
>>     For the
>> purposes of determining the sequence number of the preceding
>>     packets in this definition, assume sequence numbers starting
>> with 1,
>>     and an extra packet at the start of the stream of received
>> packets,
>>     with a sequence number of 0, is considered to be present
>> (this extra
>>     packet, of course, is not counted for the purposes of
>> computing the
>>     fraction).
>
> I think your word definition is pretty close, but there should be an
> explicit designation of a reordered packet, IMO.  Also, the "unique" 
> aspect
> is a fairly subtle way to exclude any duplicate packet from being 
> reordered.
> We could explain this fairly easily.
>
> I suggest:
>
> 4.5. Reordering
>
> |  Reordering is the fraction of unique packets received for which the 
> sequence
> |  number of a packet designated "reordered" is less than the highest 
> sequence
> |  number of any packet previously received, and the "unique" aspect 
> excludes
> |  any Duplicate packets (as described in section 4.4 above).
>    For the purposes of determining the sequence number of the preceding
>    packets in this definition, assume sequence numbers starting with 1,
>    and an extra packet at the start of the stream of received packets,
>    with a sequence number of 0, is considered to be present (this extra
>    packet, of course, is not counted for the purposes of computing the
>    fraction).
>
> hope this helps,
> Al
>
>
> _______________________________________________
> ippm mailing list
> ippm@ietf.org
> https://www.ietf.org/mailman/listinfo/ippm

--------------000602010800060905010706
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    Additional information in this topic can be found in&nbsp; RFC5236
    (<a class="moz-txt-link-freetext" href="http://www.faqs.org/rfcs/rfc5236.html">http://www.faqs.org/rfcs/rfc5236.html</a>)<br>
    as well as at&nbsp;&nbsp;
<a class="moz-txt-link-freetext" href="http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications">http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications</a><br>
    <br>
    I think 5236 provides a cleaner way to handle this.&nbsp; <br>
    <div class="moz-signature"> <span class="moz-txt-tag">--&nbsp;<br>
      </span><a href="http://www.engr.colostate.edu/%7Eanura">Anura P.
        Jayasumana</a>
      <br>
      <font color="#000000">Professor, Electrical &amp; Computer
        Engineering<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
        and Computer Science<br>
        Department of Electrical and Computer Engineering<br>
        Colorado State University<br>
        Fort Collins, CO 80523<br>
        Phone: (970) 491-7855<br>
      </font><br>
    </div>
    <br>
    On 10/7/2010 1:53 PM, Al Morton wrote:
    <blockquote
      cite="mid:201010071953.o97JrOtX031691@alpd052.aldc.att.com"
      type="cite">
      Matt,<br>
      <br>
      You asked that I take a look at:<br>
      <br>
      <blockquote type="cite" class="cite" cite="">
        <h3><b><a moz-do-not-send="true" name="section-4.5"></a>
            4.5.&nbsp; Reordering</b></h3>
        <pre>
&nbsp;&nbsp; Reordering is the fraction of unique packets received for
which the
&nbsp;&nbsp; sequence number of any given packet is less than the highest
sequence
&nbsp;&nbsp; number largest sequence number of any packet previously
received.
</pre>
      </blockquote>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
      ^^^^^^^^^^^^^^^^^^^^^^^<br>
      I think you identified this redundant text during the meeting,
      right?&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>
      <br>
      <blockquote type="cite" class="cite" cite="">
        <pre>&nbsp;&nbsp; For the
purposes of determining the sequence number of the preceding
&nbsp;&nbsp; packets in this definition, assume sequence numbers starting
with 1,
&nbsp;&nbsp; and an extra packet at the start of the stream of received
packets,
&nbsp;&nbsp; with a sequence number of 0, is considered to be present
(this extra
&nbsp;&nbsp; packet, of course, is not counted for the purposes of
computing the
&nbsp;&nbsp; fraction).
</pre>
      </blockquote>
      <br>
      I think your word definition is pretty close, but there should be
      an<br>
      explicit designation of a reordered packet, IMO.&nbsp; Also, the
      "unique" aspect<br>
      is a fairly subtle way to exclude any duplicate packet from being
      reordered.&nbsp; <br>
      We could explain this fairly easily.<br>
      <br>
      I suggest:<br>
      <br>
      4.5. Reordering<br>
      <br>
      |&nbsp; Reordering is the fraction of unique packets received for which
      the sequence <br>
      |&nbsp; number of a packet designated "reordered" is less than
      the highest sequence<br>
      |&nbsp; number of any packet previously received, and the
      "unique" aspect excludes<br>
      |&nbsp; any Duplicate packets (as described in section 4.4 above).<br>
      &nbsp;&nbsp; For the purposes of determining the sequence number of the
      preceding<br>
      &nbsp;&nbsp; packets in this definition, assume sequence numbers starting
      with 1,<br>
      &nbsp;&nbsp; and an extra packet at the start of the stream of received
      packets,<br>
      &nbsp;&nbsp; with a sequence number of 0, is considered to be present
      (this extra<br>
      &nbsp;&nbsp; packet, of course, is not counted for the purposes of
      computing the<br>
      &nbsp;&nbsp; fraction).<br>
      <br>
      hope this helps,<br>
      Al<br>
      <br>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
ippm mailing list
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<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/ippm">https://www.ietf.org/mailman/listinfo/ippm</a>
</pre>
    </blockquote>
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At 04:26 PM 10/7/2010, Anura Jayasumana wrote:
>Additional information in this topic can be found in  RFC5236
             ^^^^^^^^^^^
Since we are doing IETF Standards Track development here, Anura,
an informational publication of alternative ideas is not appropriate.

Al





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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6038

        Title:      Two-Way Active Measurement Protocol (TWAMP) 
                    Reflect Octets and Symmetrical Size Features 
        Author:     A. Morton, L. Ciavattone
        Status:     Standards Track
        Stream:     IETF
        Date:       October 2010
        Mailbox:    acmorton@att.com, 
                    lencia@att.com
        Pages:      18
        Characters: 41273
        Updates:    RFC5357

        I-D Tag:    draft-ietf-ippm-twamp-reflect-octets-09.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6038.txt

This memo describes two closely related features for the core
specification of the Two-Way Active Measurement Protocol (TWAMP): an
optional capability where the responding host returns some of the
command octets or padding octets to the sender, and an optional
sender packet format that ensures equal test packet sizes are used in
both directions.  [STANDARDS-TRACK]

This document is a product of the IP Performance Metrics Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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From lars.eggert@nokia.com  Mon Oct 11 04:17:22 2010
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Subject: [ippm] AD review: draft-ietf-ippm-tcp-throughput-tm-07
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Hi,

summary: Not quite ready. My largest issues are:

	(1) questionable use of RFC2119 language throughout the
	    document (suggestion: remove it)

	(2) big question marks around "traffic management tests"=20
	    (the ID acquired this material between -02 and -03)
	    unclear if this is even in scope of the chartered work
	    and if the approach can evaluate such schemes to any
	    useful degree

	(3) inconsistencies and inaccuracies (see below)

I'm also going to assign a transport directorate reviewer to this =
document (or its next revision.)

Lars



INTRODUCTION, paragraph 6:
>    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
>    "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in =
this
>    document are to be interpreted as described in RFC 2119 [RFC2119].

  There are many instances below where RFC2119 terms are IMO user
  inappropriately. Do you *really* need RFC2119 language in this
  Informational document?


Section 1., paragraph 1:
>    Measuring TCP throughput provides a
>    meaningful measure with respect to the end user experience (and
>    ultimately reach some level of TCP testing interoperability which
>    does not exist today).

  In the last sentence, the parenthesized clause seems to have no
  relation to the rest of the sentence?


Section 1., paragraph 2:
>    Additionally, end-users (business enterprises) seek to conduct
>    repeatable TCP throughput tests between enterprise locations.  =
Since
>    these enterprises rely on the networks of the providers, a common
>    test methodology (and metrics) would be equally beneficial to both
>    parties.

  The abstract says the goal of this draft is for measuring an "an
  end-to-end managed network environment". When end-users start to
  measure paths between locations, their measurements are not normally
  taken from an an end-to-end *managed* network environment. Are we
  focusing this ID on an end-to-end managed network environment, or is
  the intent to support more general end-user measurements? (Whatever it
  is, the ID needs to be consistent.)


Section 1., paragraph 6:
>    This methodology proposes a test which SHOULD be performed in
>    addition to traditional Layer 2/3 type tests, which are conducted =
to
>    verify the integrity of the network before conducting TCP tests.

  I don't think it's appropriate to use an RFC2119 term here.


Section 1.1, paragraph 3:
>    - Customer Provided Equipment (CPE), refers to customer owned

  "customer owned" what?


Section 1.1, paragraph 8:
> *  Bottleneck Bandwidth and Bandwidth are used synonomously in this
>    document.
> ** Most of the time the Bottleneck Bandwidth is in the access portion
>    of the wide area network (CE - PE)

  Footnotes in RFCs don't work well. Suggest to include the comment in
  the list above. Also: Nit: s/synonomously/synonymously/


Section 2., paragraph 0:
>    Note that the NUT may consist of a variety of devices including =
(and
>    NOT limited to): load balancers, proxy servers, WAN acceleration
>    devices.  The detailed topology of the NUT MUST be considered when
>    conducting the TCP throughput tests, but this methodology makes no
>    attempt to characterize TCP performance related to specific network
>    architectures.

  I don't understand what "MUST be considered" means - the paragraph
  leaves it completely open in which way this consideration is supposed
  to happen. (Esp. problematic because you use an RFC2119 term.)=20


Section 2., paragraph 3:
>    - The methodology is not intended to definitively benchmark TCP
>    implementations of one OS to another, although some users MAY find
>    some value in conducting qualitative experiments.

  I don't think it's appropriate to use an RFC2119 term here.


Section 2., paragraph 4:
>    - The methodology is not intended to provide detailed diagnosis
>    of problems within end-points or the network itself as related to
>    non-optimal TCP performance, although a results interpretation
>    section for each test step MAY provide insight into potential
>    issues within the network.

  I don't think it's appropriate to use an RFC2119 term here.


Section 2., paragraph 5:
>    - The methodology does not propose a method to operate permanently
>    with high measurement loads. TCP performance and optimization data =
of
>    operational networks MAY be captured and evaluated by using data of
>    the "TCP Extended Statistics MIB" [RFC4898].

  I don't think it's appropriate to use an RFC2119 term here.


Section 2., paragraph 9:
>    - Provide a practical test approach that specifies well understood,
>    end-user configurable TCP parameters such as TCP Window size, MSS
>    (Maximum Segment Size), number of connections, and how these affect
>    the outcome of TCP performance over a network.

  The window size for a flow is not typically end-user configurable.
  Neither is the MSS (because PMTUD/Nagle is in use.)=20


Section 2., paragraph 10:
>    - Provide specific test conditions (link speed, RTT, TCP Window =
size,
>    etc.) and maximum achievable TCP throughput under TCP Equilibrium
>    conditions.

  Again, window size is not typically a configurable OS parameter.


Section 2., paragraph 12:
>    - In test situations where the RECOMMENDED procedure does not yield
>    the maximum achievable TCP throughput result, this methodology
>    provides some possible areas within the end host or network that
>    SHOULD be considered for investigation (although again, this
>    methodology is not intended to provide a detailed diagnosis of =
these
>    issues).

  I don't think it's appropriate to use an RFC2119 term here.


Section 2.1, paragraph 5:
>    The following diagram depicts these phases.

  The diagram does not show a "retransmission phase".


Section 2.1, paragraph 7:
>    This TCP methodology provides guidelines to measure the equilibrium
>    throughput which refers to the maximum sustained rate obtained by
>    congestion avoidance before packet loss conditions occur (which MAY
>    cause the state change from congestion avoidance to a =
retransmission
>    phase). All maximum achievable throughputs specified in Section 3 =
are
>    with respect to this equilibrium state.

  I don't think it's appropriate to use an RFC2119 term here.


Section 2.2, paragraph 2:
>    The first metric is the TCP Transfer Time, which is simply the
>    measured time it takes to transfer a block of data across
>    simultaneous TCP connections.  The concept is useful when
>    benchmarking traffic management techniques, where multiple
>    connections MAY be REQUIRED.

  I don't think it's appropriate to use RFC2119 terms here. (And I have
  no idea what "MAY be REQUIRED" is supposed to indicate.)


Section 2.2, paragraph 7:
>    Table 2.2: Link Speed, RTT, TCP Throughput, Ideal TCP Transfer time

  Please give all link speeds in Mb/s, otherwise folks can't check the
  math. Please also give a formula that shows how exactly you computed
  these numbers (esp. since there are errors or inaccuracies, see
  below.) You really need to double-check the calculations here.


Section 2.2, paragraph 8:
>    Link                 Maximum Achievable     Ideal TCP Transfer time
>    Speed     RTT (ms)   TCP Throughput(Mbps)   Time in seconds
>    =
--------------------------------------------------------------------
>     T1          20              1.17                  684.93
>     T1          50              1.40                  570.61
>     T1         100              1.40                  570.61

  Why is the performance the same for 50 and 100 ms RTT?


Section 2.2, paragraph 9:
>     T3          10             42.05                   19.03
>     T3          15             42.05                   19.03
>     T3          25             41.52                   18.82

  Why is the transfer time faster for 25ms than for 15ms?


Section 2.2, paragraph 10:
>     T3(ATM)     10             36.50                   21.92
>     T3(ATM)     15             36.23                   22.14
>     T3(ATM)     25             36.27                   22.05

  Why is the transfer time faster for 25ms than for 15ms?


Section 2.2, paragraph 12:
>     *   Calculation is based on File Size in Bytes X 8 / TCP =
Throughput.
>     **  TCP Throughput is derived from Table 3.3.

  I see no asterisks in Table 2.2 - where do these remarks belong?


Section 2.2, paragraph 20:
>    And the third metric is the Buffer Delay Percentage, which =
represents
>    the increase in RTT during a TCP throughput test from the inherent
>    network RTT (baseline RTT).  The baseline RTT is the round-trip =
time
>    inherent to the network path under non-congested conditions.

  Note that the RTT across some paths is a factor of the load, esp. on
  paths that include wide-area radio links. Also, is the baseline RTT
  the shortest measured TCP RTT sample?


Section 2.2, paragraph 21:
>    The Buffer Delay Percentage is defined as:
>               Average RTT during Transfer - Baseline RTT
>               ------------------------------------------ x 100
>                              Baseline RTT

  How do you measure the average RTT during the transfer? Do you want to
  average over all TCP RTT samples? If so, how do you get at these? Do
  you do out-of-band measurements?


Section 2.2, paragraph 22:
>    As an example, the baseline RTT for the network path is 25 msec.
>    During the course of a TCP transfer, the average RTT across the
>    entire transfer increased to 32 msec.  In this example, the Buffer
>    Delay Percentage WOULD be calculated as:

  OK, so by now I am convinced that this ID should simply refrain from
  using any RFC2119 terms. (Which, by the way, do not include "WOULD".)


Section 1., paragraph 0:
>    1. Identify the Path MTU.  Packetization Layer Path MTU Discovery
>    or PLPMTUD, [RFC4821], MUST be conducted to verify the maximum
>    network path MTU.  Conducting PLPMTUD establishes the upper limit =
for
>    the MSS to be used in subsequent steps.

  You cannot run RFC4821 in isolation - it it a technique to be used
  when implementing a transport protocol. The Linux stack AFAIK does
  implement it, but many other stacks do not. The best you can do is
  identify the PMTU for a flow while or after that flow is being
  transmitted. (Because due to ECMP it may be five-tuple dependent.)


Section 3., paragraph 0:
>    3. TCP Connection Throughput Tests.  With baseline measurements
>    of Round Trip Time and bottleneck bandwidth, a series of single and
>    multiple TCP connection throughput tests SHOULD be conducted to
>    baseline the network performance expectations.

  Which exact series should be tested here? (Even unclear after reading
  Section 3.3).


Section 4., paragraph 0:

>    4. Traffic Management Tests.  Various traffic management and =
queuing
>    techniques SHOULD be tested in this step, using multiple TCP
>    connections.  Multiple connection testing SHOULD verify that the
>    network is configured properly for traffic shaping versus policing,
>    various queuing implementations, and RED.

  Even less precise than the previous paragraph. You really can't say
  "SHOULD" and then wave your hands on what exactly is to be done.
  (Even unclear after reading Section 3.4.)

Section 4., paragraph 4:
>    - Most importantly, the TCP test host must be capable of generating
>    and receiving stateful TCP test traffic at the full link speed of =
the

  What do you mean by "stateful"?


Section 3.1., paragraph 0:
> 3.1. Determine Network Path MTU

  It seems that you intend to describe a mechanism that runs RFC4821
  before you run the actual throughput test. That's not really how
  stacks work; RFC4821 (if it's implemented at all) is integrated into
  the stack, because the PMTUD is flow-specific thanks to ECMP. *If* you
  want to describe a protocol wrapper for running RFC4821 in a
  standalone manner, you need to fully specify that wrapper (packet
  formats, state machine, etc.) instead of just outlining its operation.


Section 3.2.1, paragraph 3:
>    During the actual sustained TCP throughput tests, RTT MUST be
>    measured along with TCP throughput. Buffer delay effects can be
>    isolated if RTT is concurrently measured.

  Out-of-band RTT measurements won't necessarily measure the RTT
  increase seen by the TCP stream, although they will measure the RTT
  increase sen by competing traffic (that happens to have chosen the
  same five-tuple as the OOB  RTT measurement). What is the intent here?
  (In-band measurements are difficult on uninstrumented TCP stacks.)


Section 3.2.1, paragraph 7:
>   - ICMP Pings MAY also be adequate to provide round trip time
>    estimations.  Some limitations of ICMP Ping MAY include msec
>    resolution and whether the network elements respond to pings (or
>    block them).

  ICMP is often rate-limited and segregated into different queues, so
  it's not as reliable and accurate as in-band measurements.


Section 3.3.1, paragraph 0:
> 3.3.1 Calculate Ideal TCP Window Size

  Why does the "ideal" TCP window size even matter? First, most stacks
  do auto-tuning these days. Second, as long as the window is larger
  than what you need for the BDP, you should be fine. Isn't it enough to
  require that windows should be large enough?


Section 3.3.1, paragraph 6:
>    Table 3.3: Link Speed, RTT and calculated BDP, TCP Throughput

  See my comments about Table 2.2.


Section 3.3.4, paragraph 2:
>    - Network congestion causing packet loss which MAY be inferred from
>    a poor TCP Efficiency metric (100% =3D no loss)

  "No loss" for TCP means that you're receive window limited. A TCP flow
  that will always show some losses, otherwise it isn't fully using the
  path.


Section 3.3.4, paragraph 3:
>    - Network congestion causing an increase in RTT which MAY be =
inferred
>    from the Buffer Delay metric (0% =3D no increase in RTT over =
baseline)

  Unless you're running a delay-based congestion controller, it will be
  really rare to see delay go up but no loss occurring.


Section 3.4.1, paragraph 3:
>    Traffic shaping is generally configured for TCP data services and
>    can provide improved TCP performance since the retransmissions are
>    reduced, which in turn optimizes TCP throughput for the given
>    available bandwidth.

  It's not clear at all to me that traffic shaping will improve
  performance. If you simply buffer data past the allocated rate, TCP
  will not slow down until you drop, plus then you may have introduced a
  long delay-spike in the payload stream data.


Section 3.4.1.1, paragraph 1:
>    By plotting the throughput achieved by each TCP connection, the =
fair
>    sharing of the bandwidth is generally very obvious when traffic
>    shaping is properly configured for the bottleneck interface.  For =
the
>    previous example of 5 connections sharing 500 Mbps, each connection
>    would consume ~100 Mbps with a smooth variation.  If traffic =
policing
>    was present on the bottleneck interface, the bandwidth sharing MAY
>    not be fair and the resulting throughput plot MAY reveal "spikey"
>    throughput consumption of the competing TCP connections (due to the
>    retransmissions).

  The spikes aren't due to retransmissions, they're due to loss
  probabilities not being fully equal, so that some flows end up with a
  larger capacity share then others for a while.


Section 3.4.2, paragraph 1:
>    Random Early Discard techniques are specifically targeted to =
provide
>    congestion avoidance for TCP traffic.  Before the network element
>    queue "fills" and enters the tail drop state, RED drops packets at
>    configurable queue depth thresholds.  This action causes TCP
>    connections to back-off which helps to prevent tail drop, which in
>    turn helps to prevent global TCP synchronization.

  But mostly, it keeps the queues (=3D delay) short compared to FIFO.


Section 3.4.2, paragraph 2:

>    Again, rate limited interfaces can benefit greatly from RED based
>    techniques.  Without RED, TCP is generally not able to achieve the
>    full bandwidth of the bottleneck interface.  With RED enabled, TCP
>    congestion avoidance throttles the connections on the higher speed
>    interface (i.e. LAN) and can reach equilibrium with the bottleneck
>    bandwidth (achieving closer to full throughput).

  I think this paragraph is overstating things. Yes, RED reduces the
  see-saw and can improve capacity utilization. But TCP can get within a
  few percent of link utilization even with FIFO (depends on the path
  parameters).


Section 3.4.2, paragraph 3:
>    The ability to detect proper RED configuration is more easily
>    diagnosed when conducting a multiple TCP connection test.  Multiple
>    TCP connections provide the multiple bursty sources that emulate =
the
>    real-world conditions for which RED was intended.

  Multiple parallel bulk transfers to not provide multiple bursty
  sources. If you want burstiness, you need to start and stop many
  shorter flows according to some distribution.


Section 3.4.2.1, paragraph 1:
>    The default queuing technique for most network devices is FIFO =
based.
>    Without RED, the FIFO based queue will cause excessive loss to all =
of
>    the TCP connections and in the worst case global TCP =
synchronization.

  But only in the very artificial case of a few perfectly synchronized
  bulk transfers. With more statmux, there is little issue.



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From Barry.Constantine@jdsu.com  Mon Oct 11 05:12:19 2010
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From: Barry Constantine <Barry.Constantine@jdsu.com>
To: Lars Eggert <lars.eggert@nokia.com>, "ippm@ietf.org WG" <ippm@ietf.org>
Date: Mon, 11 Oct 2010 05:11:18 -0700
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Hi Lars,

Thank you for the thorough review. We will address all technical comments a=
nd revise or clarify accordingly.  I think we can get a draft-08 version ou=
t by the end of this week.

Regarding the use of RFC2119 language in this document, some comments were =
made to do so and we added this to the draft-07.

These comments may have been off-list and I'd have to dig around, but I see=
m to recall the intent was to clarify what was essential (must do) and opti=
onal (should do) and that is how the RFC2119 language came into play.

Just to be clear, do Informational works generally (or always) refrain from=
 using RFC2119 language?

Thanks,

Barry


-----Original Message-----
From: ippm-bounces@ietf.org [mailto:ippm-bounces@ietf.org] On Behalf Of Lar=
s Eggert
Sent: Monday, October 11, 2010 7:18 AM
To: ippm@ietf.org WG
Subject: [ippm] AD review: draft-ietf-ippm-tcp-throughput-tm-07

Hi,

summary: Not quite ready. My largest issues are:

        (1) questionable use of RFC2119 language throughout the
            document (suggestion: remove it)

        (2) big question marks around "traffic management tests"
            (the ID acquired this material between -02 and -03)
            unclear if this is even in scope of the chartered work
            and if the approach can evaluate such schemes to any
            useful degree

        (3) inconsistencies and inaccuracies (see below)

I'm also going to assign a transport directorate reviewer to this document =
(or its next revision.)

Lars



INTRODUCTION, paragraph 6:
>    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
>    "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
>    document are to be interpreted as described in RFC 2119 [RFC2119].

  There are many instances below where RFC2119 terms are IMO user
  inappropriately. Do you *really* need RFC2119 language in this
  Informational document?


Section 1., paragraph 1:
>    Measuring TCP throughput provides a
>    meaningful measure with respect to the end user experience (and
>    ultimately reach some level of TCP testing interoperability which
>    does not exist today).

  In the last sentence, the parenthesized clause seems to have no
  relation to the rest of the sentence?


Section 1., paragraph 2:
>    Additionally, end-users (business enterprises) seek to conduct
>    repeatable TCP throughput tests between enterprise locations.  Since
>    these enterprises rely on the networks of the providers, a common
>    test methodology (and metrics) would be equally beneficial to both
>    parties.

  The abstract says the goal of this draft is for measuring an "an
  end-to-end managed network environment". When end-users start to
  measure paths between locations, their measurements are not normally
  taken from an an end-to-end *managed* network environment. Are we
  focusing this ID on an end-to-end managed network environment, or is
  the intent to support more general end-user measurements? (Whatever it
  is, the ID needs to be consistent.)


Section 1., paragraph 6:
>    This methodology proposes a test which SHOULD be performed in
>    addition to traditional Layer 2/3 type tests, which are conducted to
>    verify the integrity of the network before conducting TCP tests.

  I don't think it's appropriate to use an RFC2119 term here.


Section 1.1, paragraph 3:
>    - Customer Provided Equipment (CPE), refers to customer owned

  "customer owned" what?


Section 1.1, paragraph 8:
> *  Bottleneck Bandwidth and Bandwidth are used synonomously in this
>    document.
> ** Most of the time the Bottleneck Bandwidth is in the access portion
>    of the wide area network (CE - PE)

  Footnotes in RFCs don't work well. Suggest to include the comment in
  the list above. Also: Nit: s/synonomously/synonymously/


Section 2., paragraph 0:
>    Note that the NUT may consist of a variety of devices including (and
>    NOT limited to): load balancers, proxy servers, WAN acceleration
>    devices.  The detailed topology of the NUT MUST be considered when
>    conducting the TCP throughput tests, but this methodology makes no
>    attempt to characterize TCP performance related to specific network
>    architectures.

  I don't understand what "MUST be considered" means - the paragraph
  leaves it completely open in which way this consideration is supposed
  to happen. (Esp. problematic because you use an RFC2119 term.)


Section 2., paragraph 3:
>    - The methodology is not intended to definitively benchmark TCP
>    implementations of one OS to another, although some users MAY find
>    some value in conducting qualitative experiments.

  I don't think it's appropriate to use an RFC2119 term here.


Section 2., paragraph 4:
>    - The methodology is not intended to provide detailed diagnosis
>    of problems within end-points or the network itself as related to
>    non-optimal TCP performance, although a results interpretation
>    section for each test step MAY provide insight into potential
>    issues within the network.

  I don't think it's appropriate to use an RFC2119 term here.


Section 2., paragraph 5:
>    - The methodology does not propose a method to operate permanently
>    with high measurement loads. TCP performance and optimization data of
>    operational networks MAY be captured and evaluated by using data of
>    the "TCP Extended Statistics MIB" [RFC4898].

  I don't think it's appropriate to use an RFC2119 term here.


Section 2., paragraph 9:
>    - Provide a practical test approach that specifies well understood,
>    end-user configurable TCP parameters such as TCP Window size, MSS
>    (Maximum Segment Size), number of connections, and how these affect
>    the outcome of TCP performance over a network.

  The window size for a flow is not typically end-user configurable.
  Neither is the MSS (because PMTUD/Nagle is in use.)


Section 2., paragraph 10:
>    - Provide specific test conditions (link speed, RTT, TCP Window size,
>    etc.) and maximum achievable TCP throughput under TCP Equilibrium
>    conditions.

  Again, window size is not typically a configurable OS parameter.


Section 2., paragraph 12:
>    - In test situations where the RECOMMENDED procedure does not yield
>    the maximum achievable TCP throughput result, this methodology
>    provides some possible areas within the end host or network that
>    SHOULD be considered for investigation (although again, this
>    methodology is not intended to provide a detailed diagnosis of these
>    issues).

  I don't think it's appropriate to use an RFC2119 term here.


Section 2.1, paragraph 5:
>    The following diagram depicts these phases.

  The diagram does not show a "retransmission phase".


Section 2.1, paragraph 7:
>    This TCP methodology provides guidelines to measure the equilibrium
>    throughput which refers to the maximum sustained rate obtained by
>    congestion avoidance before packet loss conditions occur (which MAY
>    cause the state change from congestion avoidance to a retransmission
>    phase). All maximum achievable throughputs specified in Section 3 are
>    with respect to this equilibrium state.

  I don't think it's appropriate to use an RFC2119 term here.


Section 2.2, paragraph 2:
>    The first metric is the TCP Transfer Time, which is simply the
>    measured time it takes to transfer a block of data across
>    simultaneous TCP connections.  The concept is useful when
>    benchmarking traffic management techniques, where multiple
>    connections MAY be REQUIRED.

  I don't think it's appropriate to use RFC2119 terms here. (And I have
  no idea what "MAY be REQUIRED" is supposed to indicate.)


Section 2.2, paragraph 7:
>    Table 2.2: Link Speed, RTT, TCP Throughput, Ideal TCP Transfer time

  Please give all link speeds in Mb/s, otherwise folks can't check the
  math. Please also give a formula that shows how exactly you computed
  these numbers (esp. since there are errors or inaccuracies, see
  below.) You really need to double-check the calculations here.


Section 2.2, paragraph 8:
>    Link                 Maximum Achievable     Ideal TCP Transfer time
>    Speed     RTT (ms)   TCP Throughput(Mbps)   Time in seconds
>    --------------------------------------------------------------------
>     T1          20              1.17                  684.93
>     T1          50              1.40                  570.61
>     T1         100              1.40                  570.61

  Why is the performance the same for 50 and 100 ms RTT?


Section 2.2, paragraph 9:
>     T3          10             42.05                   19.03
>     T3          15             42.05                   19.03
>     T3          25             41.52                   18.82

  Why is the transfer time faster for 25ms than for 15ms?


Section 2.2, paragraph 10:
>     T3(ATM)     10             36.50                   21.92
>     T3(ATM)     15             36.23                   22.14
>     T3(ATM)     25             36.27                   22.05

  Why is the transfer time faster for 25ms than for 15ms?


Section 2.2, paragraph 12:
>     *   Calculation is based on File Size in Bytes X 8 / TCP Throughput.
>     **  TCP Throughput is derived from Table 3.3.

  I see no asterisks in Table 2.2 - where do these remarks belong?


Section 2.2, paragraph 20:
>    And the third metric is the Buffer Delay Percentage, which represents
>    the increase in RTT during a TCP throughput test from the inherent
>    network RTT (baseline RTT).  The baseline RTT is the round-trip time
>    inherent to the network path under non-congested conditions.

  Note that the RTT across some paths is a factor of the load, esp. on
  paths that include wide-area radio links. Also, is the baseline RTT
  the shortest measured TCP RTT sample?


Section 2.2, paragraph 21:
>    The Buffer Delay Percentage is defined as:
>               Average RTT during Transfer - Baseline RTT
>               ------------------------------------------ x 100
>                              Baseline RTT

  How do you measure the average RTT during the transfer? Do you want to
  average over all TCP RTT samples? If so, how do you get at these? Do
  you do out-of-band measurements?


Section 2.2, paragraph 22:
>    As an example, the baseline RTT for the network path is 25 msec.
>    During the course of a TCP transfer, the average RTT across the
>    entire transfer increased to 32 msec.  In this example, the Buffer
>    Delay Percentage WOULD be calculated as:

  OK, so by now I am convinced that this ID should simply refrain from
  using any RFC2119 terms. (Which, by the way, do not include "WOULD".)


Section 1., paragraph 0:
>    1. Identify the Path MTU.  Packetization Layer Path MTU Discovery
>    or PLPMTUD, [RFC4821], MUST be conducted to verify the maximum
>    network path MTU.  Conducting PLPMTUD establishes the upper limit for
>    the MSS to be used in subsequent steps.

  You cannot run RFC4821 in isolation - it it a technique to be used
  when implementing a transport protocol. The Linux stack AFAIK does
  implement it, but many other stacks do not. The best you can do is
  identify the PMTU for a flow while or after that flow is being
  transmitted. (Because due to ECMP it may be five-tuple dependent.)


Section 3., paragraph 0:
>    3. TCP Connection Throughput Tests.  With baseline measurements
>    of Round Trip Time and bottleneck bandwidth, a series of single and
>    multiple TCP connection throughput tests SHOULD be conducted to
>    baseline the network performance expectations.

  Which exact series should be tested here? (Even unclear after reading
  Section 3.3).


Section 4., paragraph 0:

>    4. Traffic Management Tests.  Various traffic management and queuing
>    techniques SHOULD be tested in this step, using multiple TCP
>    connections.  Multiple connection testing SHOULD verify that the
>    network is configured properly for traffic shaping versus policing,
>    various queuing implementations, and RED.

  Even less precise than the previous paragraph. You really can't say
  "SHOULD" and then wave your hands on what exactly is to be done.
  (Even unclear after reading Section 3.4.)

Section 4., paragraph 4:
>    - Most importantly, the TCP test host must be capable of generating
>    and receiving stateful TCP test traffic at the full link speed of the

  What do you mean by "stateful"?


Section 3.1., paragraph 0:
> 3.1. Determine Network Path MTU

  It seems that you intend to describe a mechanism that runs RFC4821
  before you run the actual throughput test. That's not really how
  stacks work; RFC4821 (if it's implemented at all) is integrated into
  the stack, because the PMTUD is flow-specific thanks to ECMP. *If* you
  want to describe a protocol wrapper for running RFC4821 in a
  standalone manner, you need to fully specify that wrapper (packet
  formats, state machine, etc.) instead of just outlining its operation.


Section 3.2.1, paragraph 3:
>    During the actual sustained TCP throughput tests, RTT MUST be
>    measured along with TCP throughput. Buffer delay effects can be
>    isolated if RTT is concurrently measured.

  Out-of-band RTT measurements won't necessarily measure the RTT
  increase seen by the TCP stream, although they will measure the RTT
  increase sen by competing traffic (that happens to have chosen the
  same five-tuple as the OOB  RTT measurement). What is the intent here?
  (In-band measurements are difficult on uninstrumented TCP stacks.)


Section 3.2.1, paragraph 7:
>   - ICMP Pings MAY also be adequate to provide round trip time
>    estimations.  Some limitations of ICMP Ping MAY include msec
>    resolution and whether the network elements respond to pings (or
>    block them).

  ICMP is often rate-limited and segregated into different queues, so
  it's not as reliable and accurate as in-band measurements.


Section 3.3.1, paragraph 0:
> 3.3.1 Calculate Ideal TCP Window Size

  Why does the "ideal" TCP window size even matter? First, most stacks
  do auto-tuning these days. Second, as long as the window is larger
  than what you need for the BDP, you should be fine. Isn't it enough to
  require that windows should be large enough?


Section 3.3.1, paragraph 6:
>    Table 3.3: Link Speed, RTT and calculated BDP, TCP Throughput

  See my comments about Table 2.2.


Section 3.3.4, paragraph 2:
>    - Network congestion causing packet loss which MAY be inferred from
>    a poor TCP Efficiency metric (100% =3D no loss)

  "No loss" for TCP means that you're receive window limited. A TCP flow
  that will always show some losses, otherwise it isn't fully using the
  path.


Section 3.3.4, paragraph 3:
>    - Network congestion causing an increase in RTT which MAY be inferred
>    from the Buffer Delay metric (0% =3D no increase in RTT over baseline)

  Unless you're running a delay-based congestion controller, it will be
  really rare to see delay go up but no loss occurring.


Section 3.4.1, paragraph 3:
>    Traffic shaping is generally configured for TCP data services and
>    can provide improved TCP performance since the retransmissions are
>    reduced, which in turn optimizes TCP throughput for the given
>    available bandwidth.

  It's not clear at all to me that traffic shaping will improve
  performance. If you simply buffer data past the allocated rate, TCP
  will not slow down until you drop, plus then you may have introduced a
  long delay-spike in the payload stream data.


Section 3.4.1.1, paragraph 1:
>    By plotting the throughput achieved by each TCP connection, the fair
>    sharing of the bandwidth is generally very obvious when traffic
>    shaping is properly configured for the bottleneck interface.  For the
>    previous example of 5 connections sharing 500 Mbps, each connection
>    would consume ~100 Mbps with a smooth variation.  If traffic policing
>    was present on the bottleneck interface, the bandwidth sharing MAY
>    not be fair and the resulting throughput plot MAY reveal "spikey"
>    throughput consumption of the competing TCP connections (due to the
>    retransmissions).

  The spikes aren't due to retransmissions, they're due to loss
  probabilities not being fully equal, so that some flows end up with a
  larger capacity share then others for a while.


Section 3.4.2, paragraph 1:
>    Random Early Discard techniques are specifically targeted to provide
>    congestion avoidance for TCP traffic.  Before the network element
>    queue "fills" and enters the tail drop state, RED drops packets at
>    configurable queue depth thresholds.  This action causes TCP
>    connections to back-off which helps to prevent tail drop, which in
>    turn helps to prevent global TCP synchronization.

  But mostly, it keeps the queues (=3D delay) short compared to FIFO.


Section 3.4.2, paragraph 2:

>    Again, rate limited interfaces can benefit greatly from RED based
>    techniques.  Without RED, TCP is generally not able to achieve the
>    full bandwidth of the bottleneck interface.  With RED enabled, TCP
>    congestion avoidance throttles the connections on the higher speed
>    interface (i.e. LAN) and can reach equilibrium with the bottleneck
>    bandwidth (achieving closer to full throughput).

  I think this paragraph is overstating things. Yes, RED reduces the
  see-saw and can improve capacity utilization. But TCP can get within a
  few percent of link utilization even with FIFO (depends on the path
  parameters).


Section 3.4.2, paragraph 3:
>    The ability to detect proper RED configuration is more easily
>    diagnosed when conducting a multiple TCP connection test.  Multiple
>    TCP connections provide the multiple bursty sources that emulate the
>    real-world conditions for which RED was intended.

  Multiple parallel bulk transfers to not provide multiple bursty
  sources. If you want burstiness, you need to start and stop many
  shorter flows according to some distribution.


Section 3.4.2.1, paragraph 1:
>    The default queuing technique for most network devices is FIFO based.
>    Without RED, the FIFO based queue will cause excessive loss to all of
>    the TCP connections and in the worst case global TCP synchronization.

  But only in the very artificial case of a few perfectly synchronized
  bulk transfers. With more statmux, there is little issue.



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PMOL WG,
cc IPPM and BMWG WGs,

This message begins a PMOL WG Last Call on the following draft:

        Title:   Guidelines for Considering New Performance Metric Development
	Author(s)       : A. Clark, B. Claise
	Filename        : draft-ietf-pmol-metrics-framework-05.txt
	Pages           : 21
	Date            : 2010-10-10
	
This memo describes a framework and guidelines for the development of
performance metrics that are beyond the scope of existing working
group charters in the IETF.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-pmol-metrics-framework-05.txt

There has been discussion of this draft for over 3 years
on the PMOL list and at meetings, both formal and informal.
The previous WGLC closed on January 2, 2009.
The Authors believe all comments to date have been addressed.

Please weigh-in on whether or not this Draft should be forwarded
to the Area Directors for publication as a Standards Track RFC.
Send your comments to the PMOL list or the co-chairs.

The Last Call will be open till November 8, 2010.

thanks for your review and comments,
Al Morton
co-chair, PMOL WG


From Anura.Jayasumana@colostate.edu  Mon Oct 11 05:55:16 2010
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I was just  pointing to  more accurate  and  useful ways of defining and 
characterizing reordering  (---  RFC 5236  and
some peer reviewed literature that  thoroughly compare the pros and cons 
of different metrics for reordering
at   
http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications 
-- -).

There is no suggestion that  5236 be used in the  development nor  do I 
have any interest in  getting anything   to the "standard track".. 
The fact that  people have voluntarily  used RD and its subsequent 
derivatives in their work speaks for itself.

However, I do find the statement "alternative ideas  ..  not 
appropriate" quite amusing -  irrespective of  their source.

Wish you good luck in your Standard Track development   - irrespective 
of what ideas you  wish to include and exclude.


Best!
-- 
Anura P. Jayasumana <http://www.engr.colostate.edu/%7Eanura>
Professor, Electrical & Computer Engineering
                           and Computer Science
Department of Electrical and Computer Engineering
Colorado State University
Fort Collins, CO 80523



Al Morton wrote:
> At 04:26 PM 10/7/2010, Anura Jayasumana wrote:
>> Additional information in this topic can be found in  RFC5236
>             ^^^^^^^^^^^
> Since we are doing IETF Standards Track development here, Anura,
> an informational publication of alternative ideas is not appropriate.
>
> Al
>
>> Additional information in this topic can be found in  RFC5236 
>> (http://www.faqs.org/rfcs/rfc5236.html)
>> as well as at   
>> http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications
>>
>> I think 5236 provides a cleaner way to handle this. 
>> -- 
>> Anura P. Jayasumana <http://www.engr.colostate.edu/%7Eanura>
>> Professor, Electrical & Computer Engineering
>>                            and Computer Science
>> Department of Electrical and Computer Engineering
>> Colorado State University
>> Fort Collins, CO 80523
>> Phone: (970) 491-7855
>>
>>
>> On 10/7/2010 1:53 PM, Al Morton wrote:
>>> Matt,
>>>
>>> You asked that I take a look at:
>>>
>>>>
>>>>       * 4.5.  Reordering*
>>>>
>>>>    Reordering is the fraction of unique packets received for
>>>> which the
>>>>    sequence number of any given packet is less than the highest
>>>> sequence
>>>>    number largest sequence number of any packet previously
>>>> received.
>>>>         
>>>            ^^^^^^^^^^^^^^^^^^^^^^^
>>> I think you identified this redundant text during the meeting, 
>>> right?         
>>>
>>>>    For the
>>>> purposes of determining the sequence number of the preceding
>>>>    packets in this definition, assume sequence numbers starting
>>>> with 1,
>>>>    and an extra packet at the start of the stream of received
>>>> packets,
>>>>    with a sequence number of 0, is considered to be present
>>>> (this extra
>>>>    packet, of course, is not counted for the purposes of
>>>> computing the
>>>>    fraction).
>>>>         
>>>
>>> I think your word definition is pretty close, but there should be an
>>> explicit designation of a reordered packet, IMO.  Also, the "unique" 
>>> aspect
>>> is a fairly subtle way to exclude any duplicate packet from being 
>>> reordered. 
>>> We could explain this fairly easily.
>>>
>>> I suggest:
>>>
>>> 4.5. Reordering
>>>
>>> |  Reordering is the fraction of unique packets received for which 
>>> the sequence
>>> |  number of a packet designated "reordered" is less than the 
>>> highest sequence
>>> |  number of any packet previously received, and the "unique" aspect 
>>> excludes
>>> |  any Duplicate packets (as described in section 4.4 above).
>>>    For the purposes of determining the sequence number of the preceding
>>>    packets in this definition, assume sequence numbers starting with 1,
>>>    and an extra packet at the start of the stream of received packets,
>>>    with a sequence number of 0, is considered to be present (this extra
>>>    packet, of course, is not counted for the purposes of computing the
>>>    fraction).
>>>
>>> hope this helps,
>>> Al
>>>
>>>
>>> _______________________________________________
>>> ippm mailing list
>>> ippm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/ippm
>>>       
>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
  <title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
I was just&nbsp; pointing to&nbsp; more accurate&nbsp; and&nbsp; useful ways of defining
and characterizing reordering&nbsp; (---&nbsp; RFC 5236&nbsp; and <br>
some peer reviewed literature that&nbsp; thoroughly compare the pros and
cons of different metrics for reordering <br>
at&nbsp;&nbsp;
<a class="moz-txt-link-freetext"
 href="http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications">http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications</a>
-- -).<br>
<br>
There is no suggestion that&nbsp; 5236 be used in the&nbsp; development nor&nbsp; do I
have any interest in&nbsp; getting anything &nbsp; to the "standard track"..&nbsp; <br>
The fact that&nbsp; people have voluntarily&nbsp; used RD and its subsequent
derivatives in their work speaks for itself.<br>
<br>
However, I do find the statement "alternative ideas&nbsp; ..&nbsp; not
appropriate" quite amusing -&nbsp; irrespective of&nbsp; their source.<br>
<br>
Wish you good luck in your Standard Track development&nbsp;&nbsp; - irrespective
of what ideas you&nbsp; wish to include and exclude.<br>
<br>
<br>
Best!<br>
<span class="moz-txt-tag">--&nbsp;<br>
</span><a href="http://www.engr.colostate.edu/%7Eanura">Anura P.
Jayasumana</a> <br>
<font color="#000000">Professor, Electrical &amp; Computer Engineering<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and Computer Science<br>
Department of Electrical and Computer Engineering<br>
Colorado State University<br>
Fort Collins, CO 80523</font><br>
<br>
<br>
<br>
Al Morton wrote:
<blockquote cite="mid:201010081206.o98C65wf002259@alpd052.aldc.att.com"
 type="cite">At 04:26 PM 10/7/2010, Anura Jayasumana wrote:
  <br>
  <blockquote type="cite">Additional information in this topic can be
found in&nbsp; RFC5236
    <br>
  </blockquote>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^^^^^^^
  <br>
Since we are doing IETF Standards Track development here, Anura,
  <br>
an informational publication of alternative ideas is not appropriate.
  <br>
  <br>
Al
  <br>
  <br>
  <blockquote type="cite"> Additional information in this topic can be
found in&nbsp; RFC5236 (<a class="moz-txt-link-freetext"
 href="http://www.faqs.org/rfcs/rfc5236.html">http://www.faqs.org/rfcs/rfc5236.html</a>)<br>
as well as at&nbsp;&nbsp;
    <a class="moz-txt-link-freetext"
 href="http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications">http://www.cnrl.colostate.edu/Projects/PacketReordering/packetreordering.html#Publications</a><br>
    <br>
I think 5236 provides a cleaner way to handle this.&nbsp; <br>
    <div class="moz-signature"> <span class="moz-txt-tag">--&nbsp;<br>
    </span><a href="http://www.engr.colostate.edu/%7Eanura">Anura P.
Jayasumana</a> <br>
    <font color="#000000">Professor, Electrical &amp; Computer
Engineering<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and Computer Science<br>
Department of Electrical and Computer Engineering<br>
Colorado State University<br>
Fort Collins, CO 80523<br>
Phone: (970) 491-7855<br>
    </font><br>
    </div>
    <br>
On 10/7/2010 1:53 PM, Al Morton wrote:
    <blockquote
 cite="mid:201010071953.o97JrOtX031691@alpd052.aldc.att.com" type="cite">
Matt,<br>
      <br>
You asked that I take a look at:<br>
      <br>
      <blockquote type="cite" class="cite" cite="">
        <h3><b><a moz-do-not-send="true" name="section-4.5"></a> 4.5.&nbsp;
Reordering</b></h3>
        <pre>&nbsp;&nbsp; Reordering is the fraction of unique packets received for
which the
&nbsp;&nbsp; sequence number of any given packet is less than the highest
sequence
&nbsp;&nbsp; number largest sequence number of any packet previously
received.
        </pre>
      </blockquote>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^^^^^^^^^^^^^^^^^^^<br>
I think you identified this redundant text during the meeting,
right?&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>
      <br>
      <blockquote type="cite" class="cite" cite="">
        <pre>&nbsp;&nbsp; For the
purposes of determining the sequence number of the preceding
&nbsp;&nbsp; packets in this definition, assume sequence numbers starting
with 1,
&nbsp;&nbsp; and an extra packet at the start of the stream of received
packets,
&nbsp;&nbsp; with a sequence number of 0, is considered to be present
(this extra
&nbsp;&nbsp; packet, of course, is not counted for the purposes of
computing the
&nbsp;&nbsp; fraction).
        </pre>
      </blockquote>
      <br>
I think your word definition is pretty close, but there should be an<br>
explicit designation of a reordered packet, IMO.&nbsp; Also, the "unique"
aspect<br>
is a fairly subtle way to exclude any duplicate packet from being
reordered.&nbsp; <br>
We could explain this fairly easily.<br>
      <br>
I suggest:<br>
      <br>
4.5. Reordering<br>
      <br>
|&nbsp; Reordering is the fraction of unique packets received for which the
sequence <br>
|&nbsp; number of a packet designated "reordered" is less than the highest
sequence<br>
|&nbsp; number of any packet previously received, and the "unique" aspect
excludes<br>
|&nbsp; any Duplicate packets (as described in section 4.4 above).<br>
&nbsp;&nbsp; For the purposes of determining the sequence number of the preceding<br>
&nbsp;&nbsp; packets in this definition, assume sequence numbers starting with 1,<br>
&nbsp;&nbsp; and an extra packet at the start of the stream of received packets,<br>
&nbsp;&nbsp; with a sequence number of 0, is considered to be present (this extra<br>
&nbsp;&nbsp; packet, of course, is not counted for the purposes of computing the<br>
&nbsp;&nbsp; fraction).<br>
      <br>
hope this helps,<br>
Al<br>
      <br>
      <pre wrap=""><fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
ippm mailing list
<a class="moz-txt-link-abbreviated" href="mailto:ippm@ietf.org">ippm@ietf.org</a>
<a class="moz-txt-link-freetext"
 href="https://www.ietf.org/mailman/listinfo/ippm">https://www.ietf.org/mailman/listinfo/ippm</a>
      </pre>
    </blockquote>
  </blockquote>
  <br>
</blockquote>
</body>
</html>

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To: ippm@ietf.org, pmol@ietf.org
From: Al Morton <acmorton@att.com>
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Subject: [ippm] Fwd: [bmwg] WGLC: draft-ietf-bmwg-reset-02
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IPPM, PMOL,

Some folks may also find this draft & methods
useful or interesting and wish to comment.

Al

>Date: Mon, 11 Oct 2010 08:39:49 -0400
>To: bmwg@ietf.org
>From: Al Morton <acmorton@att.com>
>Subject: [bmwg] WGLC: draft-ietf-bmwg-reset-02
>X-BeenThere: bmwg@ietf.org
>
>BMWG,
>
>This message begins the Second WG Last call on the draft:
>
>         Title           : Device Reset Characterization
>         Author(s)       : R. Asati, et al.
>         Filename        : draft-ietf-bmwg-reset-02.txt
>         Pages           : 18
>         Date            : 2010-10-10
>
>A URL for this draft is:
>http://tools.ietf.org/html/draft-ietf-bmwg-reset-02
>
>The Last Call will end on November 8, 2010.
>
>Since this is the second WGLC (1st closed July 21, 2010),
>I ask folks to consider items where they commented earlier,
>and make sure that the resolutions are satisfactory.
>And it's not too late to read the draft for the first time.
>Almost everyone of us has conducted this sort of testing and
>should have an opinion on the methods.
>
>Please weigh-in on whether or not this Internet-Draft
>should be given to the Area Directors and IESG for consideration and
>publication as an Informational RFC.  Send your comments
>to this list and/or acmorton@att.com.
>
>Al
>bmwg chair
>
>_______________________________________________
>bmwg mailing list
>bmwg@ietf.org
>https://www.ietf.org/mailman/listinfo/bmwg


From Barry.Constantine@jdsu.com  Mon Oct 11 15:16:16 2010
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To: Al Morton <acmorton@att.com>, "ippm@ietf.org" <ippm@ietf.org>
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Hi Al,

I would support this work, not interested in directly contributing, and wou=
ld periodically provide comments.

Thanks,

Barry



-----Original Message-----
From: Al Morton [mailto:acmorton@att.com]=20
Sent: Thursday, October 07, 2010 10:14 AM
To: Barry Constantine; ippm@ietf.org
Subject: RE: [ippm] Fwd: I-D Action:draft-morton-ippm-rt-loss-01.txt

Hi Barry,

At 09:23 AM 10/7/2010, Barry Constantine wrote:
>I reviewed the document and have one comment / question.
>
>Should this draft discuss implications of packet re-ordering to=20
>round trip loss metrics (in the context of RFC4737)?

I think so, there are certainly implications to the return path
sampling process if the reordering occurs in the forward direction,
and the Src and Dst sequence numbers will not line-up, even if there
is no loss concurrent with the reordering...

Thanks for your review.  I'll try to update this again before the
meeting.  If I've overlooked some implications, let me know...

Would you support this as a chartered WG item?

Al



From yaakov_s@rad.com  Tue Oct 12 00:16:42 2010
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From: Yaakov Stein <yaakov_s@rad.com>
To: Al Morton <acmorton@att.com>
Date: Tue, 12 Oct 2010 09:16:55 +0200
Thread-Topic: [PMOL] Repeated WGLC on draft-ietf-pmol-metrics-framework-05
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I think there are important ideas here.

Three minor comments:

1) The acronym PM is used by the OAM community for "Performance Management"
(in contrast to FM which stands for "Fault Management").
Re-use for "Performance Metrics" in a related field
(especially considering the weak intercommunications between the communitie=
s)
could lead to confusion.

2) I understand that the entity/directorate will be reviewing documents
that define performance metrics. Who determines which documents need to be =
reviewed ?
For example, what about an ID from some WG in transport or security that in=
 passing
defines a measured parameter ? (Some directorates look at every document th=
at makes it to the IESG)

3) I am not a strong supporter of meta-documents -
I much prefer RFCs with meaningful technical content to those that dictate =
internal processes.
I see this document as a "Tao" for the PMOL and IPPM communities.

I remember that 2418 mentions the concept of directorate,
and found a list of directorates here - http://www.ietf.org/iesg/directorat=
e/=20
but do not recall RFCs for the security, transport, ... directorates,
except for the quoted 5706.

Since I was not involved in the discussions in PMOL, I would like to know i=
f alternative mechanisms
for making this information available (e.g., a Performance Metric wiki)
have been discussed.=20

Y(J)S


-----Original Message-----
From: pmol-bounces@ietf.org [mailto:pmol-bounces@ietf.org] On Behalf Of Al =
Morton
Sent: Monday, October 11, 2010 14:28
To: pmol@ietf.org
Cc: bmwg@ietf.org; ippm@ietf.org
Subject: [PMOL] Repeated WGLC on draft-ietf-pmol-metrics-framework-05

PMOL WG,
cc IPPM and BMWG WGs,

This message begins a PMOL WG Last Call on the following draft:

        Title:   Guidelines for Considering New Performance Metric Developm=
ent
	Author(s)       : A. Clark, B. Claise
	Filename        : draft-ietf-pmol-metrics-framework-05.txt
	Pages           : 21
	Date            : 2010-10-10
=09
This memo describes a framework and guidelines for the development of
performance metrics that are beyond the scope of existing working
group charters in the IETF.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-pmol-metrics-framework-05.tx=
t

There has been discussion of this draft for over 3 years
on the PMOL list and at meetings, both formal and informal.
The previous WGLC closed on January 2, 2009.
The Authors believe all comments to date have been addressed.

Please weigh-in on whether or not this Draft should be forwarded
to the Area Directors for publication as a Standards Track RFC.
Send your comments to the PMOL list or the co-chairs.

The Last Call will be open till November 8, 2010.

thanks for your review and comments,
Al Morton
co-chair, PMOL WG

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

From henk@ripe.net  Wed Oct 13 05:02:55 2010
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IPPM group,


On 06/09/2010 10:34, Henk Uijterwaal wrote:
> Please see below.  In Maastricht, we discussed this topic without reaching
> a conclusion.  I did promise to take the discussion to the list though.
> 
> So, this is call for comments about the registry and it's future.  In
> particular, I'd like to hear from people who are using the current registry,
> who have developed a registry themselves for a particular application,
> or who think that a different registry would be useful for them.
> 
> Note that comments made on the list so-far or at the Maastricht meeting
> have been noted.  Please respond by October 15, so we can discuss
> further steps by the Beijing meeting.

There have been no comments on this, if you do have an opinion please
speak up in the next day or two.

Henk

-- 
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I confirm today what I denied yesterday.            Anonymous Politician.

From Xiangsong.Cui@huawei.com  Sun Oct 17 20:25:08 2010
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Date: Mon, 18 Oct 2010 11:26:23 +0800
From: Xiangsong Cui <Xiangsong.Cui@huawei.com>
To: 'IETF IPPM WG' <ippm@ietf.org>
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Subject: [ippm] Solicit feedback about network capacity draft, thanks in advance!
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Hi,

As you know, we had a long discussion on RFC5136, but we didn't get enough input for next step, http://www.ietf.org/mail-archive/web/ippm/current/msg02445.html,
so I submit this draft, the draft may be accessed by http://tools.ietf.org/id/draft-cui-ippm-rfc5136bis-00.txt.

Firstly, I would like to acknowledge Phil, Joseph and everyone who had made contribution to RFC5136, which is the basis of the new draft.

And the main modifications in this draft consist of following points:
1, Clarification on definition of node, link, path, etc.
2, Clarification on IP-layer link capacity, path capacity, etc.
3, Router capacity and usage.

Another point is that the informative reference

   [PDM]      Dovrolis, C., Ramanathan, P., and D. Moore, "Packet
              Dispersion Techniques and a Capacity Estimation
              Methodology", IEEE/ACM Transactions on Networking 12(6):
              963-977, December 2004.

is removed in the draft, because this paper assumes a path definition which violates the path definition in RFC2330, while RFC2330 is a normative reference.

This paper reads "Consider a network path as a sequence of firstcome first-served (FCFS) store-and-forward links that transfer packets from the sender to the receiver ." While in RFC2330, path is not the concatenation of links, but the sequence of node and link. 

Additionally, this paper clearly states: 
"Network operators commonly use tools such as MRTG [16]
to monitor the utilization of their links with information obtained
from the router management software. These techniques
are based on counters maintained by routers, and they are normally
*accurate*. Their drawback, however, is that they can only
be performed with router access, and such access is usually limited
to the corresponding network administrators. Instead, in
this paper, we focus on an /end-to-end bandwidth monitoring/
approach that requires cooperation of only the *path end-points*.
Even though end-to-end approaches are typically not as accurate
as router-based measurements, they often are the only feasible
approach for monitoring a path that crosses several networks."

We are discussing metrics but not monitoring approach here, and I personally think when we are defining metrics we should prefer accuracy to feasibility, even though some metrics are theoretical number.


Well, now I would like to know your thought on this item, your feedback is highly appreciated!

Best regards,
Xiangsong


-----Original Message-----
From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] 
Sent: Saturday, October 16, 2010 9:32 PM
To: Xiangsong.Cui@huawei.com
Subject: New Version Notification for draft-cui-ippm-rfc5136bis-00


A new version of I-D, draft-cui-ippm-rfc5136bis-00.txt has been successfully submitted by Xiangsong Cui and posted to the IETF repository.

Filename:	 draft-cui-ippm-rfc5136bis
Revision:	 00
Title:		 Defining Network Capacity
Creation_date:	 2010-10-16
WG ID:		 Independent Submission
Number_of_pages: 28

Abstract:
This document defines the metric of network capacity, including link
capacity aspect, router capacity aspect and path capacity aspect.
RFC5136 has defined link capacity and path capacity, where the router
impact is implicitly considered in link capacity.  However, in this
document, router capacity is considerred as a separate factor of path
capacity, no longer a factor of link capacity.  This document
explicitly describes the router capacity and its impact to network
capacity, e.g. how to evaluate path capacity.

This document is derived from RFC5136 and obsoletes RFC5136.
                                                                                  


The IETF Secretariat.



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IPPM group,

Please see below, if there is anything you'd like to discuss, let us know.

Matt & Henk

- - -



AGENDA IPPM Meeting @ IETF79
============================

(Monday, 9:00 - 11:30)

0.   5'  Administrativia
         Scribes: ##
         (Minutes, Scribe, Blue Sheet, Note Well, ...)

1.   5'  Status of drafts not discussed today

Ongoing work
============
2.   ...

Possible new work
=================

3.  20'  Comments on RFC5136/Xiangsong Cui
         draft-cui-ippm-rfc5136bis-00

4.  30'  Metrics registry and the way forward.

5.       ...

N.       AOB
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IPPM,

We've put together an improved version of the metricstest draft.

>Filename:       draft-ietf-ippm-metrictest
>Revision:       01
>Title:          IPPM standard advancement testing
>Creation_date:  2010-10-24
>WG ID:          ippm
>Number_of_pages: 39
>
>Abstract:
>This document specifies tests to determine if multiple independent
>instantiations of a performance metric RFC have implemented the
>specifications in the same way.  This is the performance metric
>equivalent of interoperability, required to advance RFCs along the
>standards track.  Results from different implementations of metric
>RFCs will be collected under the same underlying network conditions
>and compared using state of the art statistical methods.  The goal is
>an evaluation of the metric RFC itself, whether its definitions are
>clear and unambiguous to implementors and therefore a candidate for
>advancement on the IETF standards track.
>

Here's a list of the recent changes:

    Changes from WG -00 to WG -01 draft

    o  Discussion on merits and requirements of a distributed lab test
       using only local load generators.

    o  Proposal of metrics suitable for tests using the proposed
       measurement configuration.

    o  Hint on delay caused by software based L2TPv3 implementation.

    o  Added an appendix with a test configuration allowing remote tests
       comparing different implementations accross the network.

    o  Proposal for maximum error of "equivalence", based on performance
       comparison of identical implementations.  This may be useful for
       both ADK and non-ADK comparisons.

regards,
Ruediger and Al


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the IP Performance Metrics Working Group of the IETF.


	Title           : IPPM standard advancement testing
	Author(s)       : R. Geib, et al.
	Filename        : draft-ietf-ippm-metrictest-01.txt
	Pages           : 39
	Date            : 2010-10-24

This document specifies tests to determine if multiple independent
instantiations of a performance metric RFC have implemented the
specifications in the same way.  This is the performance metric
equivalent of interoperability, required to advance RFCs along the
standards track.  Results from different implementations of metric
RFCs will be collected under the same underlying network conditions
and compared using state of the art statistical methods.  The goal is
an evaluation of the metric RFC itself, whether its definitions are
clear and unambiguous to implementors and therefore a candidate for
advancement on the IETF standards track.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-ippm-metrictest-01.txt

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the IP Performance Metrics Working Group of the IETF.


	Title           : Reporting Metrics: Different Points of View
	Author(s)       : A. Morton, et al.
	Filename        : draft-ietf-ippm-reporting-metrics-04.txt
	Pages           : 25
	Date            : 2010-10-25

Consumers of IP network performance metrics have many different uses
in mind.  The memo provides "long-term" reporting considerations
(e.g, days, weeks or months, as opposed to 10 seconds), based on
analysis of the two key audience points-of-view.  It describes how
the audience categories affect the selection of metric parameters and
options when seeking info that serves their needs.

A URL for this Internet-Draft is:
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IPPM Group,

Please see below.  Comments welcome, if you are presenting, then please
send us your slides (preferably the evening before the meeting).

Matt & Henk

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

AGENDA IPPM Meeting @ IETF79
============================

Draft, Oct 26.

(Monday, 9:00 - 11:30)

0.   5'  Administrativia
         Scribes: É.
         (Minutes, Scribe, Blue Sheet, Note Well, ...)

1.   5'  Status of drafts not discussed today

Ongoing work
============
2.  20'  IPPM standard advancement testing (Al Morton)
         draft-ietf-ippm-metrictest-02

3.  10'  Long term reporting draft (Al Morton)
         draft-ietf-ippm-reporting-metrics-04.txt

4.  10'  Liaison statement from ITU-T Study Group 12. (Al Morton)

Possible new work
=================

5.  30'  Comments on RFC5136 (Xiangsong Cui)
         draft-cui-ippm-rfc5136bis-00

6.  30'  Metrics registry and the way forward.
         * Introduction (Henk Uijterwaal)
         * draft-morton-ippm-rfc4148-obsolete (Al Morton)
         * draft-stephan-ippm-registry-ext (Emile Stephan)
         * Discussion

7.  15'  Round Trip Loss Metrics (Al Morton)
         draft-morton-ippm-rt-loss-01

8.       AOB
         Comments on draft-morton-bmwg-imix-genome-00.txt

-- 
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I confirm today what I denied yesterday.            Anonymous Politician.

From Barry.Constantine@jdsu.com  Wed Oct 27 12:37:29 2010
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From: Barry Constantine <Barry.Constantine@jdsu.com>
To: "ippm@ietf.org" <ippm@ietf.org>
Date: Wed, 27 Oct 2010 12:37:07 -0700
Thread-Topic: [ippm] I-D Action:draft-ietf-ippm-reporting-metrics-04.txt
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Subject: Re: [ippm] I-D Action:draft-ietf-ippm-reporting-metrics-04.txt
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Hello,

I had some time to review this work and found it interesting.  Explaining t=
he relationship of PDV from an RFC5481/3393 to Y.1540 is very useful.  Also=
, applying the ITU concept of availability is a good idea.

Below are my comments:

4.1.2.  It is difficult to specify what an application's threshold for dela=
y is.  Applications have many unknown behaviors and loss / delay is certain=
ly one of them.  I think the intent of this paragraph is good, but not sure=
 if it is practical.

4.2 Last sentence.  Could not agree more with this point; satellite communi=
cations is a classic example where error correction is used and it is highl=
y desirable to know which packets were truly dropped versus ones that could=
 not be corrected.

5.1.1 Last paragraph.  I had a hard time understanding the concept of the l=
ast sentence and how it pertained to this section's theme.

5.2 For long term monitoring, the 95th percentile is considered one of key =
value.  It is mentioned in this section, but may be useful to expand on a l=
ittle as was done for median (and why 95th percentile is useful versus maxi=
mum).

7.3, section 2.  There was a placeholder for equilibrium.  Feel free to cop=
y from the TCP framework draft if useful:

"the equilibrium throughput refers to the maximum sustained rate obtained b=
y congestion avoidance before packet loss conditions occur (which may cause=
 the state change from congestion avoidance to a retransmission
phase)."

"the equilibrium throughput reflects the maximum rate achieved by a TCP con=
nection within the congestion avoidance phase on an end-end network path."=
=20

7.4 last bullet.  We defined a couple of useful metrics that you may want t=
o consider adding here (TCP Efficiency and Buffer Delay %) also in the TCP =
framework draft.

7.5 Variability, last two bullets. Plotting retransmissions and RTT during =
the course of a BTC are excellent ways to understand causes of variability.


I think this is a good work and I hope these comments are useful.

Thanks,

Barry




-----Original Message-----
From: ippm-bounces@ietf.org [mailto:ippm-bounces@ietf.org] On Behalf Of Int=
ernet-Drafts@ietf.org
Sent: Monday, October 25, 2010 6:30 PM
To: i-d-announce@ietf.org
Cc: ippm@ietf.org
Subject: [ippm] I-D Action:draft-ietf-ippm-reporting-metrics-04.txt

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
This draft is a work item of the IP Performance Metrics Working Group of th=
e IETF.


	Title           : Reporting Metrics: Different Points of View
	Author(s)       : A. Morton, et al.
	Filename        : draft-ietf-ippm-reporting-metrics-04.txt
	Pages           : 25
	Date            : 2010-10-25

Consumers of IP network performance metrics have many different uses
in mind.  The memo provides "long-term" reporting considerations
(e.g, days, weeks or months, as opposed to 10 seconds), based on
analysis of the two key audience points-of-view.  It describes how
the audience categories affect the selection of metric parameters and
options when seeking info that serves their needs.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-ippm-reporting-metrics-04.tx=
t

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

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Internet-Draft.

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To: Barry Constantine <Barry.Constantine@jdsu.com>, "ippm@ietf.org" <ippm@ietf.org>
From: Al Morton <acmorton@att.com>
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Subject: Re: [ippm] I-D Action:draft-ietf-ippm-reporting-metrics-04.txt
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At 03:37 PM 10/27/2010, Barry Constantine wrote:
>...
>I think this is a good work and I hope these comments are useful.

Thanks for your review, Barry, especially your comments in the
new section 7 (Restricted Capacity).  I'll reply with follow-up
on the list when time permits,

Al



