From matt@advanced.org  Fri Dec  1 11:33:55 2000
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From: "Matthew J Zekauskas" <matt@advanced.org>
To: <agenda@ietf.org>
Cc: <ippm@advanced.org>, "Merike Kaeo" <kaeo@merike.com>,
        "Allison Mankin" <mankin@east.isi.edu>,
        "Will Leland" <wel@research.telcordia.com>
Subject: Agenda for IPPM in San Diego
Date: Fri, 1 Dec 2000 11:25:31 -0500
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Proposed agenda for the IP Performance Metrics WG (ippm)

Tuesday, 12 December 2000, 17:00-18:00
======================================

Introduction of new co-chair & agenda bashing (<= 5 min)

Discuss the new charter and milestones (15 min)

Update on Bulk Transfer Capacity (10 min)
  --Mark Allman
    http://www.ietf.org/internet-drafts/draft-ietf-ippm-btc-framework-03.txt

Update on Network Performance Measurement for Periodic Streams (10 min)
  --Vilho Raisanen
    http://www.ietf.org/internet-drafts/draft-ietf-ippm-npmps-03.txt
    http://www.ietf.org/internet-drafts/draft-raisanen-ippm-npmps-results-00.txt
    (also http://www-nrc.nokia.com/netperf/results.pdf )

One-way delay protocol document discussion (10 min)
  -- Ben Teitelbaum
     http://www.ietf.org/internet-drafts/draft-ietf-ippm-owdp-00.txt

Discussion of any issues with IPDV or loss patterns
  -- speakers TBD, at least the chairs & Phil Chimento.
     http://www.ietf.org/internet-drafts/draft-ietf-ippm-ipdv-05.txt
     http://www.ietf.org/internet-drafts/draft-ietf-ippm-loss-pattern-04.txt


--Matt




From matt@advanced.org  Fri Dec  1 17:29:30 2000
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From: Ben Teitelbaum <ben@internet2.edu>
Organization: Internet2 (UCAID) / Advanced Network & Services
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To: Tamas Varga <Tamas.Varga.II@eth.ericsson.se>
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Subject: Re: I-D ACTION:draft-ietf-ippm-owdp-00.txt
References: <200011242356.SAA00967@ietf.org> <3A2221FD.5FD9E3AA@eth.ericsson.se> <3A240F75.89A51E24@internet2.edu> <3A2626F3.42616444@eth.ericsson.se>
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Tamas Varga wrote:
> 
> Dear Ben,
> 
> Ben Teitelbaum wrote:
> > These are both helpful suggestions. With respect to the first comment, it
> > would seem ideal to identify the Type-P explicitly during  session initiation
> > negotiation. However, since the notion of Type-P has not been formalized, I'm
> > not quite sure how to do include a Type-P identifier. Any suggestions?
> 
> As far as I understood Section 13 and 15 of RFC2330, type-P is a generic
> notation for the packet/service type we want to measure. The problem is
> that it can be substituted by different network, transport & application
> layer protocols and the information required to describe them depends
> on the protocol itself. But this is somewhat easier, since OWDP focuses
> on UDP based test packets only. I think three things may be interesting
> in the IP header: Type-of-Service (TOS), Time-to-time (TTL) and Don't
> Fragment (DF) bit. Filling up TOS with an appropriate DSCP would be useful
> to select packet treatment in a DiffServ network. TTL may limit the scope
> of the measurements. (Anyway, what is if a multicast destination address
> is specified? ). DF can be handy to ensure that the packet was transmitted
> in one piece, otherwise we won't know how many of the fragments were lost.
> 
> I think, setting DSCP is an essential feature. It should be included in
> the Ression-Request message. There is a room nearby the port field.
> The other two feautres may be handy, but it does not necesarily follow
> that we should include them.
> 

Tamas,

Although I like the idea of TTL and DF, I don't actually like the idea of
including DSCP explicitly. As the DiffServ working group has defined DSCPs,
they have only local significance. Instead, I think what we want is the 16-bit
RFC 2836 PHB identification code. This way, if you desire a particular PHB for
your session, you need not know what DSCP in the Session-Sender's local
environment is used to indicate it.

> > With respect to your second comment, I am open to including an option to allow
> > periodic sampling. We thought about supporting arbitrary distributions, but
> > decided to keep things simple by including Poisson only. Adding an option for
> > periodic sampling wouldn't add much complication, so I'm inclided to throw it
> > into the next draft.
> 
> Moreover uniform distribution can be useful, if we choose a random
> value from the (0, 1/Inv-Lambda) interval. Other distributions require
> more parameters, which makes the interpretation of the required fields
> complicated. Anyway, periodic sampling is still reasonable, allocation
> of 1 bit for this purpose is quite easy.

Just curious, in what circumstances is a uniform distribution useful?

> 
> Back to multicasting for a while. As I see, this question is not clarified
> yet in the document. I think it is silently assumed, that unicast test flows
> are used by the measurements. However, the architecture may be multicast
> aware if the Server is able to track Session-Receivers.

Yes, unicast was our silent assumption. As the document now stands, I don't
think it would be particularly difficult to extend it to support multicast. 
We would add an additional Receiver field to each record of the
Retrieve-Session response:

{SeqNo, ReceiverAddr, SendTimestamp, RecvTimestamp}

and then punt on exactly how a Server tracked Session-Receivers. 

-- 

                         ,,,

                        `o-o-  
                          <    Benjamin Teitelbaum
                           -   Advanced Network & Services, Inc.
                           .   Internet2



From matt@advanced.org  Sat Dec  2 06:15:35 2000
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From: "omurhan akdemir" <omurhan@hotmail.com>
To: <ippm@advanced.org>
Subject: Pathchar
Date: Sat, 2 Dec 2000 13:02:16 +0200
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hi

I wonder what the meaning of negative propagation delay values on =
pathchar's output is.
sample output:

2 sdscdmz-fddi.cerf.net (198.17.46.153)
 |    45 Mb/s,   -13 us (2.70 ms)
 3 qualcomm-sdsc-ds3.cerf.net (134.24.47.200)

By the way, is there any way of plotting rtt of each probe of each =
sample versus packet size?

Kind regards
omurhan=20

------=_NextPart_000_0007_01C05C60.18B36640
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	charset="iso-8859-9"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-9">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>hi</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I wonder what the meaning of negative =
propagation=20
delay values on pathchar's output is.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>sample output:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D4>2 sdscdmz-fddi.cerf.net (198.17.46.153)<BR>=20
|&nbsp;&nbsp;&nbsp; 45 Mb/s,&nbsp;&nbsp; -13 us (2.70 ms)<BR> 3=20
qualcomm-sdsc-ds3.cerf.net (134.24.47.200)<BR></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>By the way, is there any way of =
plotting rtt of=20
each probe of each sample versus packet size?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Kind regards</FONT></DIV>
<DIV><FONT face=3DArial =
size=3D2>omurhan</FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0007_01C05C60.18B36640--



From matt@advanced.org  Sat Dec  2 10:47:17 2000
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Subject: Re: Pathchar
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unsubscribe.

Thanks,
Redha Bournas

"omurhan akdemir" <omurhan@hotmail.com> on 12/02/2000 06:02:16 AM

To:   <ippm@advanced.org>
cc:
Subject:  Pathchar




hi

I wonder what the meaning of negative propagation  delay values on
pathchar's output is.
sample output:

2 sdscdmz-fddi.cerf.net (198.17.46.153)
|    45 Mb/s,   -13 us (2.70 ms)
3  qualcomm-sdsc-ds3.cerf.net (134.24.47.200)

By the way, is there any way of plotting rtt of  each probe of each sample
versus packet size?

Kind regards
omurhan




From matt@advanced.org  Mon Dec  4 03:08:37 2000
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Date: Mon, 4 Dec 2000 08:54:45 +0100
From: Mark Santcroos <marks@ripe.net>
To: "omurhan akdemir" <omurhan@hotmail.com>
Cc: <ippm@advanced.org>
Subject: Re: Pathchar
Message-ID: <20001204085444.A2746@ripe.net>
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On Sat, Dec 02, 2000 at 01:02:16PM +0200, omurhan akdemir wrote:
> I wonder what the meaning of negative propagation delay values on
> pathchar's output is.
Hi Omurhan,

Some types of networks are intrinsically difficult for pathchar to
measure.  Two notable examples are switched networks (with multiple
queues at Layer 2) or striped networks.  

Router implementations may very well forward a packet faster than they
can return an ICMP error message in response to a packet.  Because of
this fact, it's possible to see faster response times from longer
partial paths; the result is a seemingly non-sensical, negative
estimate of per-hop round-trip time.

Btw. I suggest you to look at:
http://www.employees.org/~bmah/Software/pchar/

Pchar is an open source re-implementation of Van Jacobson's Pathchar
utility. (It is also where I have shamelessly stole the answer about
negative rrt's from)

Regards,

Mark

-- 
Mark Santcroos			   RIPE Network Coordination Centre

PGP KeyID: 1024/0x3DCBEB8D 
PGP Fingerprint: BB1E D037 F29D 4B40 0B26  F152 795F FCAB 3DCB EB8D



From matt@advanced.org  Mon Dec  4 07:11:39 2000
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From: "Leping HUANG" <lphuang@pipi.iis.u-tokyo.ac.jp>
To: "IPPM Mailing list" <ippm@advanced.org>
Subject: finding bandwidth measurement tool
Date: Mon, 4 Dec 2000 21:09:28 +0900
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Dear Sir/Madam:

I want to do some measurement on my ADSL link, but feel some difficulty to
find a suitable tools to do measurement.
Because My ADSL link only has a private IP address, i can only measure the
uplink using the tools like Pchar.
Obviously, the downlink and uplink has different bandwidth, but i can not
measure it.!!
I think any-ICMP style measurement can not be used to measure my downlink
bandwidth, is it correct?
Are there any tools  can be used to measure downlink bandwidth with/without
receiver side damaon?

thank you.

Leping HUANG
Sezaki Lab
Institute of Industrial Science
University of Tokyo
lphuang@mcl.iis.u-tokyo.ac.jp
office phone/FAX: +81(3)3403-2596







From matt@advanced.org  Tue Dec  5 07:03:48 2000
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From: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
To: Matthew J Zekauskas <matt@advanced.org>
cc: ippm@advanced.org, ben@advanced.org
Subject: Re: Agenda for IPPM in San Diego
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Ben, Matt,

> Proposed agenda for the IP Performance Metrics WG (ippm)
> 
> Tuesday, 12 December 2000, 17:00-18:00
> ======================================

> One-way delay protocol document discussion (10 min)
>   -- Ben Teitelbaum
>      http://www.ietf.org/internet-drafts/draft-ietf-ippm-owdp-00.txt

I read the draft and have 2 major issues (besides a bunch of smaller
comments, which I'll post to the list as soon as I've decoded my own
handwriting in the margin).

1. The protocol assumes a setup mechanism for sessions.  We've always
   used a different approach: tell the sender that it should send 
   something to a target, tell the target that it should listen, but
   don't care if sender or target aren't ready yet.

   This has a practical advantage (it is much easier to implement) and
   a theoretical advantage: one can also easily measure losses over
   broken links (where a setup protocol will fail).  The disadvantage
   is that it is harder to raise alarms when packets don't arrive.

   Should we try to include this method in the protocol?

2. Padding and such.

   In order to avoid compression along the way, we use a random bit
   pattern  to pad the packets.  (If you have a packet with lots of
   zero's, it may be compressed by some intelligent device, and thus
   get a travel time corresponding to a much shorter packet.)
 
   Besides that, we also include much more info in our packets, such
   as NTP (clock) errors, experimental status words, source and 
   destination address, etc, etc.  In short, each packet contains
   everything we want to know about the packet, without having to
   rely on other data at source/target.  

   So, what we do is:
   * take N byte packet
   * Fill it with a random bit pattern.
   * Take a number between 0 and N-l (with l the number of bytes for
     all data).  
   * Put the data starting somewhere between 0 and N-l.
   * Make the 0th byte a pointer to the data.

   What I think we should put in the draft is a mechanism for this, even
   if implementations decide that they don't want to or need to use it.

(Perhaps we should sit down sometime before/after the meeting, since I
have a feeling that we won't be able to resolve these issues in the 10
minutes allocated for this draft).

Henk

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                                               (Kerouac, well before RFC2780).







From matt@advanced.org  Tue Dec  5 09:03:24 2000
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From: Ben Teitelbaum <ben@internet2.edu>
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To: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
Cc: Matthew J Zekauskas <matt@advanced.org>, ippm@advanced.org,
        Stanislav Shalunov <shalunov@internet2.edu>
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Henk,

I'm not sure I understand point 1). With what we have written, a client could
request a session to begin at a future start time; the server can tell the
sender to send and the receiver to listen (the means by which the server tells
them is unspecified); then, if there is 100% packet loss at the time of the
scheduled session, this is captured. Does that work for you?

As for point 2), note that you can always run in Encrypted mode, which
effectively fills the entire packet payload with random bits.

Let's definitely sit down next week. 

Cheers,

"Henk Uijterwaal (RIPE-NCC)" wrote:
> 
> Ben, Matt,
> 
> > Proposed agenda for the IP Performance Metrics WG (ippm)
> >
> > Tuesday, 12 December 2000, 17:00-18:00
> > ======================================
> 
> > One-way delay protocol document discussion (10 min)
> >   -- Ben Teitelbaum
> >      http://www.ietf.org/internet-drafts/draft-ietf-ippm-owdp-00.txt
> 
> I read the draft and have 2 major issues (besides a bunch of smaller
> comments, which I'll post to the list as soon as I've decoded my own
> handwriting in the margin).
> 
> 1. The protocol assumes a setup mechanism for sessions.  We've always
>    used a different approach: tell the sender that it should send
>    something to a target, tell the target that it should listen, but
>    don't care if sender or target aren't ready yet.
> 
>    This has a practical advantage (it is much easier to implement) and
>    a theoretical advantage: one can also easily measure losses over
>    broken links (where a setup protocol will fail).  The disadvantage
>    is that it is harder to raise alarms when packets don't arrive.
> 
>    Should we try to include this method in the protocol?
> 
> 2. Padding and such.
> 
>    In order to avoid compression along the way, we use a random bit
>    pattern  to pad the packets.  (If you have a packet with lots of
>    zero's, it may be compressed by some intelligent device, and thus
>    get a travel time corresponding to a much shorter packet.)
> 
>    Besides that, we also include much more info in our packets, such
>    as NTP (clock) errors, experimental status words, source and
>    destination address, etc, etc.  In short, each packet contains
>    everything we want to know about the packet, without having to
>    rely on other data at source/target.
> 
>    So, what we do is:
>    * take N byte packet
>    * Fill it with a random bit pattern.
>    * Take a number between 0 and N-l (with l the number of bytes for
>      all data).
>    * Put the data starting somewhere between 0 and N-l.
>    * Make the 0th byte a pointer to the data.
> 
>    What I think we should put in the draft is a mechanism for this, even
>    if implementations decide that they don't want to or need to use it.
> 
> (Perhaps we should sit down sometime before/after the meeting, since I
> have a feeling that we won't be able to resolve these issues in the 10
> minutes allocated for this draft).
> 
> Henk
> 
> ------------------------------------------------------------------------------
> Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
> RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
> Singel 258                         Phone: +31.20.535-4414,  Fax -4445
> 1016 AB Amsterdam                   Home: +31.20.4195305
> The Netherlands                   Mobile: +31.6.55861746
> ------------------------------------------------------------------------------
> 
> A man can take a train and never reach his destination.
>                                                (Kerouac, well before RFC2780).

-- 

                         ,,,

                        `o-o-  
                          <    Benjamin Teitelbaum
                           -   Advanced Network & Services, Inc.
                           .   Internet2



From matt@advanced.org  Tue Dec  5 09:31:16 2000
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From: stanislav shalunov <shalunov@internet2.edu>
Date: 05 Dec 2000 09:25:31 -0500
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"Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net> writes:

> 1. The protocol assumes a setup mechanism for sessions.  We've always
>    used a different approach: tell the sender that it should send 
>    something to a target, tell the target that it should listen, but
>    don't care if sender or target aren't ready yet.
> 
>    This has a practical advantage (it is much easier to implement) and
>    a theoretical advantage: one can also easily measure losses over
>    broken links (where a setup protocol will fail).  The disadvantage
>    is that it is harder to raise alarms when packets don't arrive.
> 
>    Should we try to include this method in the protocol?

Henk,

The reason for the setup mechanisms is to allow for a wide range of
possible applications of the protocol.  A central notion is to be able
to run "open" servers--have a one-way ping.

> 2. Padding and such.

The reason padding isn't just supposed to be unspecified pseudo-random
is concern about high-bandwidth covert channels.  Depending on how
much of a concern it is for other people we can take different routes:

* Make it unspecified pseudo-random;

* Take pseudo-random bits from the same source as the one used to
  obtain inter-packet times;

* Use some well-defined source of pseudo-random bits, same for all
  flows.

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

Sex is the mathematics urge sublimated.                 -- M. C. Reed.



From matt@advanced.org  Tue Dec  5 16:39:28 2000
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Comments: In-reply-to Mark Santcroos <marks@ripe.net>
   message dated "Mon, 04 Dec 2000 08:54:45 +0100."
From: bmah@cisco.com (Bruce A. Mah)
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If memory serves me right, Mark Santcroos wrote:
> On Sat, Dec 02, 2000 at 01:02:16PM +0200, omurhan akdemir wrote:
> > I wonder what the meaning of negative propagation delay values on
> > pathchar's output is.

[snip]

> Some types of networks are intrinsically difficult for pathchar to
> measure.  Two notable examples are switched networks (with multiple
> queues at Layer 2) or striped networks.  

These facts are true, but most likely not quite the explanation for 
what the original poster was asking about.

> Router implementations may very well forward a packet faster than they
> can return an ICMP error message in response to a packet.  Because of
> this fact, it's possible to see faster response times from longer
> partial paths; the result is a seemingly non-sensical, negative
> estimate of per-hop round-trip time.

This is more likely to be the case.  Another possibility is that large
(or fluctuating) queue lengths at an interface can interfere with the
linear fit algorithms used by pathchar.

> Btw. I suggest you to look at:
> http://www.employees.org/~bmah/Software/pchar/
> 
> Pchar is an open source re-implementation of Van Jacobson's Pathchar
> utility. (It is also where I have shamelessly stole the answer about
> negative rrt's from)

:-)

Another implementation is clink; while it doesn't run on as many
platforms as pchar does, it does a couple of things that pchar doesn't
do such as kernel-level timestamps.  There's a link to it from pchar's
Web page.

Bruce.





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From matt@advanced.org  Wed Dec  6 19:34:25 2000
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To: ippm@advanced.org
Subject: draft-ietf-ippm-owdp-01
From: stanislav shalunov <shalunov@internet2.edu>
Date: 06 Dec 2000 19:24:35 -0500
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New version of draft-ietf-ippm-owdp is now available at
http://www.internet2.edu/~shalunov/draft-ietf-ippm-owdp-01.txt
(and will be available on the IETF server after accepting drafts is
resumed).

This is the version that will be presented during next week's IETF
meeting.

It fills gaps in version -00, fixes some bugs, and incorporates some
feedback we received from the working group.

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

"The power of accurate observation is commonly called cynicism by
those who have not got it."			-- G. B. Shaw



From matt@advanced.org  Thu Dec  7 17:01:25 2000
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From: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
To: stanislav shalunov <shalunov@internet2.edu>
cc: ippm@advanced.org
Subject: Re: draft-ietf-ippm-owdp-00
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On 5 Dec 2000, stanislav shalunov wrote:

> "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net> writes:
> 
> > 1. The protocol assumes a setup mechanism for sessions.  We've always
> >    used a different approach: tell the sender that it should send 
> >    something to a target, tell the target that it should listen, but
> >    don't care if sender or target aren't ready yet.
> > 
> >    This has a practical advantage (it is much easier to implement) and
> >    a theoretical advantage: one can also easily measure losses over
> >    broken links (where a setup protocol will fail).  The disadvantage
> >    is that it is harder to raise alarms when packets don't arrive.
> > 
> >    Should we try to include this method in the protocol?
> 
> Henk,
> 
> The reason for the setup mechanisms is to allow for a wide range of
> possible applications of the protocol.  A central notion is to be able
> to run "open" servers--have a one-way ping.

We basically have to solve 2 problems here:

a) How to do measurements with 2 different devices.  
   This requires that the packet formats are understandable by both sides.

b) How to set up a measurement relation between source and target.
  
A setup mechanism is one possible method to accomplish (b) but there are
lots of others and I don't think that we should limit ourselves to one
method unless there are convincing technical arguments.  This is something
we should really discuss by voice next week.


> > 2. Padding and such.
> 
> The reason padding isn't just supposed to be unspecified pseudo-random
> is concern about high-bandwidth covert channels.

Please explain.

>  Depending on how
> much of a concern it is for other people we can take different routes:
> 
> * Make it unspecified pseudo-random;

Isn't user data sent over a link random (when viewed by a machine)?

Henk

------------------------------------------------------------------------------
Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
Singel 258                         Phone: +31.20.535-4414,  Fax -4445
1016 AB Amsterdam                   Home: +31.20.4195305
The Netherlands                   Mobile: +31.6.55861746  
------------------------------------------------------------------------------

A man can take a train and never reach his destination.
                                               (Kerouac, well before RFC2780).



From matt@advanced.org  Thu Dec  7 17:30:32 2000
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Subject: Re: draft-ietf-ippm-owdp-00
References: <Pine.BSI.4.05L.10012072241460.11142-100000@kantoor.ripe.net>
From: stanislav shalunov <shalunov@internet2.edu>
Date: 07 Dec 2000 17:25:13 -0500
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"Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net> writes:

> We basically have to solve 2 problems here:
> 
> a) How to do measurements with 2 different devices.  
>    This requires that the packet formats are understandable by both sides.
> 
> b) How to set up a measurement relation between source and target.
>   
> A setup mechanism is one possible method to accomplish (b) but there are
> lots of others and I don't think that we should limit ourselves to one
> method unless there are convincing technical arguments.  This is something
> we should really discuss by voice next week.

Henk,

I think nothing in the draft implies that OWDP-Test sessions have to
be set up with OWDP-Control only.  One can imagine situations when it
is convenient to set the sessions up using different means, and
retrieve the results using yet another method.

Of course, connection parameters have to be somehow transferred
anyway.

Should we explicitly mention the possibility of use of OWDP-Test
without OWDP-Control?  (In fact, if you make Server, Retrieve-Client
and Control-Client the same host in logical roles picture on p.3 of
-01, it collapses into set up without OWDP-Control.)

> On 5 Dec 2000, stanislav shalunov wrote:
> > The reason padding isn't just supposed to be unspecified pseudo-random
> > is concern about high-bandwidth covert channels.
> 
> Please explain.

I had in mind environments where covert channels can be viewed as a
problem (secretive commercial world, the military, etc.).
Low-bandwidth covert channels are of course always available in any
network, but unspecified padding of large packets could be used to
funnel data out of a network at a high rate.

After thinking about your comment and some internal discussion padding
is changed to unspecified pseudo-random with a message to implementers
to make available a switch for zero padding.

Does this address your concern?

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

Sex is the mathematics urge sublimated.                 -- M. C. Reed.



From matt@advanced.org  Thu Dec  7 21:07:24 2000
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Vilho, Glenn, and all,

Your new draft deserves at least one (minor) comment.

Section 4.2.1 gives the Global Metric Parameters,
one of which is: 
+  periodic packet interval incT, a time duration

In practice, it may be difficult for measurement sources
to maintain this periodic sending interval exactly.
I'd like to suggest adding a new parameter for section 
4.2.2 Metrics Reported at MP(Src) to cover the cases
where sending times have some variation from the ideal interval.
Something like:
+ incTerror, offset from the specified incT 
(could be reported on a per packet basis, or as a range
at the end of test).

Note that packetized real-time media streams from users
may also exhibit variation from strict periodic launch times. 
The point is to capture and report any variation
so it doesn't get lost.

Regards,
Al Morton



At 03:47 PM 11/22/00 +0200, vilho.raisanen@nokia.com wrote:
>
>
>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
> Vilho Räisänen                           
> vilho.raisanen@nokia.com                 
> Senior research engineer                 
> Nokia Research Center, Helsinki, Finland 
>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
>
>Dear IPPM list members,
>
>I submitted the latest revised form (-03) of the npmps draft 
>today to internet-drafts@ietf.org; it is also available at
>
>http://www-nrc.nokia.com/netperf/npmps-03.txt
>
>Moreover, following the advice of IPPM chairs, I made an 
>independent draft submission of VoIP media stream emulation 
>measurement results using metrics using npmps-like metrics. 
>The full draft with figures is available from
>
>http://www-nrc.nokia.com/netperf/results.pdf
>
>and a version with good old ASCII from
>
>http://search.ietf.org/internet-drafts/draft-raisanen-ippm-npmps-results-00.
>txt
>
>	Cheers,
>		Vilho
>
>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
> Vilho Räisänen                           
> vilho.raisanen@nokia.com                 
> Senior research engineer                 
> Nokia Research Center, Helsinki, Finland 
>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
>
>
>



From matt@advanced.org  Thu Dec  7 21:07:28 2000
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Date: Thu, 07 Dec 2000 21:01:54 -0500
To: ippm@advanced.org
From: Al Morton <acmorton@att.com>
Subject: Re: Comments on draft-ietf-ippm-ipdv.05
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IPPM folks,

There has been a pause in the discussion on delay variation
since Phil Chimento offered to begin a serious re-write.  
I'd like to offer my support of Mike Pierce's position, that measuring 
a set of delays and determining their variation provides 
information beyond what is available with the current metric
(the difference in delay between consecutive packet pairs).
I've got the impression that Phil will be attempting to include
both these metrics in the text of the new draft 
(is that right, Phil?).

Assuming both these metrics are described in ipdv-06, it may help
to capture the advantages of each, providing guidance to future
readers. Much of this material could be distilled from the recent 
messages. Since my past efforts have been mostly associated with
ITU-T and T1A1, I'll change hats and highlight what I consider to 
be two key advantages of the current ipdv metric:

Ruediger Geib's message reminds us that the pair-wise delay differences
can be calculated without synchronized measurement clocks.  When all the
code and the wires come together (implementation time), it can be very
comforting to know that this measurement will be useful even if others
must be discarded due to clock synchronization problems.

Another point for ipdv is its similarity to the calculation of
interarrival jitter measurement in RTCP reports.  RFC 1889 (RTP)
gives the calculation of interarrival jitter in section 6.3.1, with
a sample implementation in an appendix.  Although there are some 
differences in method (RTCP interarrival jitter uses order of arrival, 
as opposed to sending sequence with ipdv) there should be a favorable 
comparison between a "smoothed jitter" computed using ipdv singletons
and the RTCP reports of jitter in many circumstances (If many packets 
were re-ordered, the results would probably not agree). It seems 
valuable to have a metric that can be related to measurements
made by user's endpoints.

Shortly, the ITU-T will be having discussions on performance parameters for
delay variation.  I'm convinced that both metrics should be included
in the appropriate Recommendations there as well.

Regards,
Al Morton




From matt@advanced.org  Fri Dec  8 10:05:08 2000
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From: Grotefeld Glenn-cecl03 <G.Grotefeld@motorola.com>
To: "'Al Morton'" <acmorton@att.com>, ippm@advanced.org
Subject: RE: FW: draft-ietf-ippm-npmps-03.txt; IPPM-style  measurement res
	ults
Date: Fri, 8 Dec 2000 08:57:00 -0600 
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Al,


One item to be concerned about is that with a MP(Src) that is truly independent of Src, MP (Src)is NOT responsible for sending times, but merely reports on them.  I wanted the flexibility for Src can generate packets that may not follow the "incT" interval, but without any other signaling or messaging that could distort the performance compared to times when Src is NOT being measured.

I am NOT convinced that we should introduce this change. Rather than introducing a new variable, I would prefer to change the description to " +  nominal periodic packet interval incT, a time duration".  This would increase the tie-in to section 4.3, where the general description of the procedure calls incT a "nominal interval".

If we do go forward with a change, I would suggest something other than "incTerror", which suggests incremental terror. 8-)

Glenn Grotefeld

Motorola, Inc.
g.grotefeld@motorola.com
US  847 576-5992
      847 538-7455  FAX


~-----Original Message-----
~From: Al Morton [mailto:acmorton@att.com]
~Sent: Thursday, December 07, 2000 7:59 PM
~To: ippm@advanced.org
~Subject: Re: FW: draft-ietf-ippm-npmps-03.txt; IPPM-style measurement
~results
~
~
~Vilho, Glenn, and all,
~
~Your new draft deserves at least one (minor) comment.
~
~Section 4.2.1 gives the Global Metric Parameters,
~one of which is: 
~+  periodic packet interval incT, a time duration
~
~In practice, it may be difficult for measurement sources
~to maintain this periodic sending interval exactly.
~I'd like to suggest adding a new parameter for section 
~4.2.2 Metrics Reported at MP(Src) to cover the cases
~where sending times have some variation from the ideal interval.
~Something like:
~+ incTerror, offset from the specified incT 
~(could be reported on a per packet basis, or as a range
~at the end of test).
~
~Note that packetized real-time media streams from users
~may also exhibit variation from strict periodic launch times. 
~The point is to capture and report any variation
~so it doesn't get lost.
~
~Regards,
~Al Morton
~
~
~
~At 03:47 PM 11/22/00 +0200, vilho.raisanen@nokia.com wrote:
~>
~>
~>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
~> Vilho Räisänen                           
~> vilho.raisanen@nokia.com                 
~> Senior research engineer                 
~> Nokia Research Center, Helsinki, Finland 
~>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
~>
~>Dear IPPM list members,
~>
~>I submitted the latest revised form (-03) of the npmps draft 
~>today to internet-drafts@ietf.org; it is also available at
~>
~>http://www-nrc.nokia.com/netperf/npmps-03.txt
~>
~>Moreover, following the advice of IPPM chairs, I made an 
~>independent draft submission of VoIP media stream emulation 
~>measurement results using metrics using npmps-like metrics. 
~>The full draft with figures is available from
~>
~>http://www-nrc.nokia.com/netperf/results.pdf
~>
~>and a version with good old ASCII from
~>
~>http://search.ietf.org/internet-drafts/draft-raisanen-ippm-npm
~ps-results-00.
~>txt
~>
~>	Cheers,
~>		Vilho
~>
~>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
~> Vilho Räisänen                           
~> vilho.raisanen@nokia.com                 
~> Senior research engineer                 
~> Nokia Research Center, Helsinki, Finland 
~>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
~>
~>
~>
~



From matt@advanced.org  Sat Dec  9 12:28:23 2000
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From matt@advanced.org  Mon Dec 11 17:29:46 2000
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From: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
To: ippm@advanced.org
Cc: "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu
	 <dromasca@lucent.com>,
        Steve Waldbusser <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: on <draft-ietf-ippm-owdp-00.txt>'s model
Date: Mon, 11 Dec 2000 17:19:01 -0500
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Stanislav,
Benjamin,
Matt,

I have a question regarding the model chosen in the
one way delay measurement protocol.  Basically, the model
chosen is more complete than I had anticipated (not that
this is good or bad).  I copy a figure from your draft
(below).  Your draft covers both the OWDP-Test and OWDP-Control
protocols.

       +----------------+              +------------------+
       | Session-Source |--OWDP-Test-->| Session-Receiver |
       +----------------+              +------------------+
              ^                                ^
              |                                |
              |                                |
              V                                |
       +----------------+<---------------------+
       |     Server     |<------------+
       +----------------+             |
              ^                       |
              |                       |
         OWDP-Control            OWDP-Control
              |                       |
              V                       V
       +----------------+     +-----------------+
       | Control-Client |     | Retrieve-Client |
       +----------------+     +-----------------+


An alternative (I think) is to define the OWDP-Test protocol
and to rely on the work in the RMONMIB WG to handle the
OWDP-Control protocol through SNMP and the current draft
MIBs being discussed therein.  (see figure below).

       +----------------+              +------------------+
       | Session-Source |--OWDP-Test-->| Session-Receiver |
       +----------------+              +------------------+
              ^                                ^
              |                                |
             SNMP                             SNMP
              |                                |
              V                                |
       +----------------+<---------------------+
       |SNMP Server Appl|
       +----------------+            
  

The drafts I refer to are:

<draft-kalbfleisch-sspmmib-01.txt>
<draft-ietf-rmonmib-apm-mib-02.txt>
<draft-ietf-rmonmib-tpm-mib-00.txt>

At a minimum, I would hope that your draft would be constructed
in such a way that it does not exclude the use of the later model
where one uses an SNMP application to configure the test source and sinks
and the measurements are run using the OWDP-Test protocol without
relying on the OWDP-Control protocol.

I would much appreciate hearing your thoughts on this later approach.

Thanks,
Bob

-----------------------------------------------------
Robert G. Cole
AT&T Labs, Network Design and Performance Analysis Dept.
Tel: (410) 939-8732 	FAX: (410) 939-8732 
Pager: (888) 858-PAGE, pin 127113
email: rgcole@att.com	URL: http://chapman.mt.att.com/

330 Saint Johns St., 2nd Floor
Havre de Grace, MD  21078
		USA






From matt@advanced.org  Mon Dec 11 19:30:34 2000
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From: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
To: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
cc: ippm@advanced.org, "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: Re: on <draft-ietf-ippm-owdp-00.txt>'s model
In-Reply-To: <A32A6A6D3178D3119C300090279CB296066C0EB0@njb140po02.ems.att.com>
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Bob,


> An alternative (I think) is to define the OWDP-Test protocol and to
> rely on the work in the RMONMIB WG to handle the OWDP-Control protocol
> through SNMP and the current draft MIBs being discussed therein.  
> (see figure below).
> 
>        +----------------+              +------------------+
>        | Session-Source |--OWDP-Test-->| Session-Receiver |
>        +----------------+              +------------------+
>               ^                                ^
>               |                                |
>              SNMP                             SNMP
>               |                                |
>               V                                |
>        +----------------+<---------------------+
>        |SNMP Server Appl|
>        +----------------+            

In this picture, which process is sitting where and what does SNMP-SA
exactly tell the SS and SR?  

My preference would be: 

SA sits at a control point
SS and SR are measurement nodes

SA tells SS to send measurement packets starting at t=x with some set
            of parameters.
SA tells SR to expect data from SS starting at t=x

SS tells SA what it has sent
SR tells SR what it has received.

SR combines the 2 to report delays

This also works to measure losses (count the number of packets at both
ends) and thus connectivity.

> At a minimum, I would hope that your draft would be constructed in
> such a way that it does not exclude the use of the later model where
> one uses an SNMP application to configure the test source and sinks
> and the measurements are run using the OWDP-Test protocol without
> relying on the OWDP-Control protocol.
> 
> I would much appreciate hearing your thoughts on this later approach.

I think that there is a lot of mileage in this idea.  The feedback that I
get from our customers on our work (http://www.ripe.net/test-traffic), is
that they are interested in eventually integrating delay and loss
measurements in their existing monitoring systems, and not in a
stand-alone device.  If we come up with our own protocol, then I fear that
the next thing we have to do, is to write a wrapper around the OWDP setup
protocol in order to integrate it with SNMP.

Henk

------------------------------------------------------------------------------
Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
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From matt@advanced.org  Mon Dec 11 20:31:09 2000
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To: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
Cc: ippm@advanced.org, "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser <waldbusser@nextbeacon.com>,
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Subject: Re: on <draft-ietf-ippm-owdp-00.txt>'s model
References: <A32A6A6D3178D3119C300090279CB296066C0EB0@njb140po02.ems.att.com>
From: stanislav shalunov <shalunov@internet2.edu>
Date: 11 Dec 2000 20:26:21 -0500
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Lines: 34
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"Cole, Robert G (Bob), ALSVC" <rgcole@att.com> writes:

> Basically, the model chosen is more complete than I had anticipated
> (not that this is good or bad).
[...]
> An alternative (I think) is to define the OWDP-Test protocol
> and to rely on the work in the RMONMIB WG to handle the
> OWDP-Control protocol through SNMP and the current draft
> MIBs being discussed therein.

Robert,

draft-ietf-ippm-owdp-00 defines both OWDP-Control and OWDP-Test, but
it's not anticipated that all hosts that speak OWDP-Test would
necessarily also speak OWDP-Control.

Additionally it should be mentioned that our draft doesn't specify
protocols used on the unlabeled links of the example setup figure (and
doesn't even discuss whether there should be a protocol on this links).
The task of Session-Sender and Session-Receiver configuration could be
accomplished in a number of different ways.

This could be SNMP.  This could be sending shell commands over an SSH
session.  This could be anything.

I suppose we should explicitly mention the possibility of using
OWDP-Test without OWDP-Control, which will be done in the next draft
revision.  This is a prose change only.

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

A large number of installed systems work by fiat.  That is, they work
by being declared to work.                             -- Anatol Holt



From matt@advanced.org  Mon Dec 11 20:50:44 2000
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To: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
Cc: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>, ippm@advanced.org,
        "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: Re: on <draft-ietf-ippm-owdp-00.txt>'s model
References: <Pine.BSI.4.05L.10012120102070.5445-100000@kantoor.ripe.net>
From: stanislav shalunov <shalunov@internet2.edu>
Date: 11 Dec 2000 20:46:49 -0500
In-Reply-To: <Pine.BSI.4.05L.10012120102070.5445-100000@kantoor.ripe.net>
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"Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net> writes:

> If we come up with our own protocol, then I fear that the next thing
> we have to do, is to write a wrapper around the OWDP setup protocol
> in order to integrate it with SNMP.

Henk,

Since OWDP-Test and OWDP-Control are separable, I don't think there's
need to define a wrapper.

On the other hand, having OWDP-Control can be helpful in the scenario
of network monitoring system using SNMP for a single network.  Namely,
you could, e.g., have a single host in your domain that speaks
OWDP-Control and allow (subject to a policy) use of your network
monitoring nodes to send or receive test traffic to/from outside your
domain.

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

"Nuclear war can ruin your whole compile."          -- Karl Lehenbauer



From matt@advanced.org  Mon Dec 11 20:54:14 2000
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From: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
To: stanislav shalunov <shalunov@internet2.edu>
cc: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>, ippm@advanced.org,
        "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: Re: on <draft-ietf-ippm-owdp-00.txt>'s model
In-Reply-To: <87g0juv39e.fsf@cain.internet2.edu>
Message-ID: <Pine.BSI.4.05L.10012120243130.10460-100000@kantoor.ripe.net>
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Stanislav,

> > Basically, the model chosen is more complete than I had anticipated
> > (not that this is good or bad).
> [...]
> > An alternative (I think) is to define the OWDP-Test protocol
> > and to rely on the work in the RMONMIB WG to handle the
> > OWDP-Control protocol through SNMP and the current draft
> > MIBs being discussed therein.

> draft-ietf-ippm-owdp-00 defines both OWDP-Control and OWDP-Test, but
> it's not anticipated that all hosts that speak OWDP-Test would
> necessarily also speak OWDP-Control.
> 
> Additionally it should be mentioned that our draft doesn't specify
> protocols used on the unlabeled links of the example setup figure (and
> doesn't even discuss whether there should be a protocol on this links).
> The task of Session-Sender and Session-Receiver configuration could be
> accomplished in a number of different ways.
> 
> This could be SNMP.  This could be sending shell commands over an SSH
> session.  This could be anything.

If it can be anything, then I believe the draft should simply say "how the
sessions are configured, is not discussed in this draft" instead of
putting 1 specific solution in the draft with a footnote that one can use
something else as well.

Including a specific solution, IMHO, gives the mistaken impression that
this is somehow preferred over other solutions.  If you want to make
OWDP-control a protocol, put it another draft.

Henk

ps. Ben, Matt, Stanislav: are you here and can we meet to discuss this
draft by VOS sometime tomorrow?

------------------------------------------------------------------------------
Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
Singel 258                         Phone: +31.20.535-4414,  Fax -4445
1016 AB Amsterdam                   Home: +31.20.4195305
The Netherlands                   Mobile: +31.6.55861746  
------------------------------------------------------------------------------

A man can take a train and never reach his destination.
                                               (Kerouac, well before RFC2780).




From matt@advanced.org  Mon Dec 11 21:03:44 2000
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Date: Tue, 12 Dec 2000 02:59:32 +0100 (CET)
From: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
To: stanislav shalunov <shalunov@internet2.edu>
cc: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>, ippm@advanced.org,
        "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: Re: on <draft-ietf-ippm-owdp-00.txt>'s model
In-Reply-To: <87d7eyv2ba.fsf@cain.internet2.edu>
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Stanislav,

> > If we come up with our own protocol, then I fear that the next thing
> > we have to do, is to write a wrapper around the OWDP setup protocol
> > in order to integrate it with SNMP.

> Since OWDP-Test and OWDP-Control are separable, I don't think there's
> need to define a wrapper.

If your measurement device uses OWDP-control and mine SNMP, then in order
to set up a measurement session between the two, one will have to
translate OWDP to SNMP (or the other way around).

> On the other hand, having OWDP-Control can be helpful in the scenario
> of network monitoring system using SNMP for a single network.  
> Namely, you could, e.g., have a single host in your domain that speaks
> OWDP-Control and allow (subject to a policy) use of your network
> monitoring nodes to send or receive test traffic to/from outside your
> domain.

I fail to see why we need an extra protocol for this: with SNMP, one can
tell this host just as well to start sending/receiving test-traffic
from/to other sites.  

Henk

------------------------------------------------------------------------------
Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
Singel 258                         Phone: +31.20.535-4414,  Fax -4445
1016 AB Amsterdam                   Home: +31.20.4195305
The Netherlands                   Mobile: +31.6.55861746  
------------------------------------------------------------------------------

A man can take a train and never reach his destination.
                                               (Kerouac, well before RFC2780).




From matt@advanced.org  Tue Dec 12 01:40:36 2000
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Date: Tue, 12 Dec 2000 01:32:16 -0500
To: "Geib, Ruediger" <Ruediger.Geib@telekom.de>, ippm@advanced.org
From: Al Morton <acmorton@att.com>
Subject: RE: Comments on draft-ietf-ippm-ipdv.05
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At 10:16 AM 11/15/00 +0100, Geib, Ruediger wrote:
>Hi IPPMers (and to spare [5] from being deleted ;-)
>
>As "unknown reference" of above draft I'd like to point out my view on
Mike's comments:
<snip>
>
>Finally I'd like to ask Mike to provide access to the following references
quoted by him:
>Y.1540 and Y.1541.
>Distributiuon of these documents to the IPPM WG was promised during the
last meeting. I however never saw any information on the list or on the
archive.
>
> 
Ruediger's request for a copy of Draft ITU-T Rec Y.1541
can't be satisfied by the ITU directly, but since T1A1 
discusses this as well, the 11/2000 version is posted as below:

ftp://ftp.t1.org/pub//t1a1/t1a1.3/0a130560.doc
ftp://ftp.t1.org/pub//t1a1/t1a1.3/0a130560.ps

Note that draft Y.1541 is very much a work-in-progress,
especially some of the values in Table 1, and the 
parameter definition for delay variation.

I'm sure that a near-final version of I.380 -> Y.1540 exists
on T1A1's site, and when I find it I'll post that URL, too.

Al



From matt@advanced.org  Tue Dec 12 11:05:37 2000
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From: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
To: stanislav shalunov <shalunov@internet2.edu>
Cc: ippm@advanced.org, "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser
	 <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: RE: on <draft-ietf-ippm-owdp-00.txt>'s model
Date: Tue, 12 Dec 2000 10:57:30 -0500
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Stanislav,

Thanks.  I agree with what you are saying.  I would just like to see these
options spelled out in an introductory section of the draft.  I would then
also
like to make sure that the source and sink contrl tables in the
kalbfleisch-sspmmib
draft align with the control requirements of the OWDP-test protocol as
spelled out
in your draft.

Thanks,
Bob

> -----Original Message-----
> From:	stanislav shalunov [SMTP:shalunov@internet2.edu]
> Sent:	Monday, December 11, 2000 8:26 PM
> To:	Cole, Robert G (Bob), ALSVC
> Cc:	ippm@advanced.org; Carl W. Kalbfleisch; Dan Romascanu; Steve
> Waldbusser; Russell Dietz
> Subject:	Re: on <draft-ietf-ippm-owdp-00.txt>'s model
> 
> "Cole, Robert G (Bob), ALSVC" <rgcole@att.com> writes:
> 
> > Basically, the model chosen is more complete than I had anticipated
> > (not that this is good or bad).
> [...]
> > An alternative (I think) is to define the OWDP-Test protocol
> > and to rely on the work in the RMONMIB WG to handle the
> > OWDP-Control protocol through SNMP and the current draft
> > MIBs being discussed therein.
> 
> Robert,
> 
> draft-ietf-ippm-owdp-00 defines both OWDP-Control and OWDP-Test, but
> it's not anticipated that all hosts that speak OWDP-Test would
> necessarily also speak OWDP-Control.
> 
> Additionally it should be mentioned that our draft doesn't specify
> protocols used on the unlabeled links of the example setup figure (and
> doesn't even discuss whether there should be a protocol on this links).
> The task of Session-Sender and Session-Receiver configuration could be
> accomplished in a number of different ways.
> 
> This could be SNMP.  This could be sending shell commands over an SSH
> session.  This could be anything.
> 
> I suppose we should explicitly mention the possibility of using
> OWDP-Test without OWDP-Control, which will be done in the next draft
> revision.  This is a prose change only.
> 
> -- 
> Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2
> 
> A large number of installed systems work by fiat.  That is, they work
> by being declared to work.                             -- Anatol Holt



From matt@advanced.org  Tue Dec 12 11:06:03 2000
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From: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
To: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
Cc: ippm@advanced.org, "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser
	 <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: RE: on <draft-ietf-ippm-owdp-00.txt>'s model
Date: Tue, 12 Dec 2000 10:54:37 -0500
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Henk,

see my responses below.
thx, Bob

> -----Original Message-----
> From:	Henk Uijterwaal (RIPE-NCC) [SMTP:henk@ripe.net]
> Sent:	Monday, December 11, 2000 7:20 PM
> To:	Cole, Robert G (Bob), ALSVC
> Cc:	ippm@advanced.org; Carl W. Kalbfleisch; Dan Romascanu; Steve
> Waldbusser; Russell Dietz
> Subject:	Re: on <draft-ietf-ippm-owdp-00.txt>'s model
> 
> Bob,
> 
> 
> > An alternative (I think) is to define the OWDP-Test protocol and to
> > rely on the work in the RMONMIB WG to handle the OWDP-Control protocol
> > through SNMP and the current draft MIBs being discussed therein.  
> > (see figure below).
> > 
> >        +----------------+              +------------------+
> >        | Session-Source |--OWDP-Test-->| Session-Receiver |
> >        +----------------+              +------------------+
> >               ^                                ^
> >               |                                |
> >              SNMP                             SNMP
> >               |                                |
> >               V                                |
> >        +----------------+<---------------------+
> >        |SNMP Server Appl|
> >        +----------------+            
> 
> In this picture, which process is sitting where and what does SNMP-SA
> exactly tell the SS and SR?  
> 
> My preference would be: 
> 
> SA sits at a control point
> SS and SR are measurement nodes
> 
> SA tells SS to send measurement packets starting at t=x with some set
>             of parameters.
> SA tells SR to expect data from SS starting at t=x
> 
> SS tells SA what it has sent
> SR tells SR what it has received.
> 
> SR combines the 2 to report delays
	[Cole, Robert G (Bob), ALSVC]  Exactly !  If you look at the
Kalbfleisch sspmmib
	draft, it contains two contrl table; one for source and one for
sink.  The sink table
	tells the sink what to expect and when.  Also, now that you all have
a draft containing the
	OWDP-test protocol, we can align the source and sink table contrl
info with your test protocol
	requirements.

> This also works to measure losses (count the number of packets at both
> ends) and thus connectivity.
> 
> > At a minimum, I would hope that your draft would be constructed in
> > such a way that it does not exclude the use of the later model where
> > one uses an SNMP application to configure the test source and sinks
> > and the measurements are run using the OWDP-Test protocol without
> > relying on the OWDP-Control protocol.
> > 
> > I would much appreciate hearing your thoughts on this later approach.
> 
> I think that there is a lot of mileage in this idea.  The feedback that I
> get from our customers on our work (http://www.ripe.net/test-traffic), is
> that they are interested in eventually integrating delay and loss
> measurements in their existing monitoring systems, and not in a
> stand-alone device.  If we come up with our own protocol, then I fear that
> the next thing we have to do, is to write a wrapper around the OWDP setup
> protocol in order to integrate it with SNMP.
	[Cole, Robert G (Bob), ALSVC]  I don't think we are suggesting any
sort of wrapper, but
	i guess I do not fully understand your concern.  I think the control
aspects of the setup protocol 
	are to be contained in the sspmmib source and sink contrl tables on
the respective
	measurement nodes.

> Henk
> 
> --------------------------------------------------------------------------
> ----
> Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
> RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
> Singel 258                         Phone: +31.20.535-4414,  Fax -4445
> 1016 AB Amsterdam                   Home: +31.20.4195305
> The Netherlands                   Mobile: +31.6.55861746  
> --------------------------------------------------------------------------
> ----
> 
> A man can take a train and never reach his destination.
>                                                (Kerouac, well before
> RFC2780).



From matt@advanced.org  Tue Dec 12 11:53:11 2000
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From: Max Edwards <maxe@marcusevansuk.com>
To: "'ippm@advanced.org'" <ippm@advanced.org>
Subject: eMetrics Conference
Date: Tue, 12 Dec 2000 16:41:34 -0000
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Dear Sir/Madam,

I am producing the UK's first conference on eMetrics which will deal with
monitoring your eBusiness performance. This will include topics such as
aligning IT with corporate eBusiness goals; metrics flexibility for evolving
eBusiness systems; implementing metrics measurement tools for smooth
eBusiness transition and maximising profitability of IT networks and
applications.

I am doing some pre-production research and am interested in finding more
information about the subject area of eBusiness and identifying
gurus/speakers in this area. Would you be interested in an event such as
this or might you be able to assist me in my enquiry?

Thank you for your time and I look forward to hearing from you.

Kind regards,

Max Edwards
> Business Strategy Division
marcus evans
4 Cavendish Square
London W1M 0BX
Tel: (020) 7499 0900
Fax: (020) 7629 3533
email: maxe@marcusevansuk.com

> marcus evans is dedicated to the provision of strategic business
> information for all its customers by understanding what they need, and
> delivering it through premium products and services in the format best
> suited to each customer. To find out more about how you and your
> organisation can access the very best information, please visit us at:
> www.marcusevans.com
> 
> 
> 
> 



From matt@advanced.org  Tue Dec 12 18:06:12 2000
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From: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
To: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>,
        stanislav shalunov
	 <shalunov@internet2.edu>
Cc: ippm@advanced.org, "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
        Steve Waldbusser
	 <waldbusser@nextbeacon.com>,
        Russell Dietz <rsdietz@apptitude.com>
Subject: RE: on <draft-ietf-ippm-owdp-00.txt>'s model
Date: Tue, 12 Dec 2000 17:56:21 -0500
MIME-Version: 1.0
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Henk,

> -----Original Message-----
> From:	Henk Uijterwaal (RIPE-NCC) [SMTP:henk@ripe.net]
> Sent:	Monday, December 11, 2000 9:00 PM
> To:	stanislav shalunov
> Cc:	Cole, Robert G (Bob), ALSVC; ippm@advanced.org; Carl W. Kalbfleisch;
> Dan Romascanu; Steve Waldbusser; Russell Dietz
> Subject:	Re: on <draft-ietf-ippm-owdp-00.txt>'s model
> 
> Stanislav,
> 
> 
> I fail to see why we need an extra protocol for this: with SNMP, one can
> tell this host just as well to start sending/receiving test-traffic
> from/to other sites.  
	[Cole, Robert G (Bob), ALSVC]  Will there be cases where the snmp
mgmr and the
	OWDP source are within the same AS and the sink is outside, in which
case it may be
	necessary to have a control protocol between the source and sink to
configure the sink?

	Bob

> Henk
> 
> --------------------------------------------------------------------------
> ----
> Henk Uijterwaal                    Email: henk.uijterwaal@ripe.net
> RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
> Singel 258                         Phone: +31.20.535-4414,  Fax -4445
> 1016 AB Amsterdam                   Home: +31.20.4195305
> The Netherlands                   Mobile: +31.6.55861746  
> --------------------------------------------------------------------------
> ----
> 
> A man can take a train and never reach his destination.
>                                                (Kerouac, well before
> RFC2780).
> 



From matt@advanced.org  Tue Dec 12 23:14:37 2000
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From: vilho.raisanen@nokia.com
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To: ippm@advanced.org
Cc: matt@advanced.org, kaeo@merike.com
Subject: npmps-04
Date: Wed, 13 Dec 2000 06:06:53 +0200
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_000_01C064BA.208F49D0
Content-Type: text/plain;
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Hello,

please find attached my (intended) slides, the results document, as =
well as
the v-04 of the npmps draft. Lacking my slides, I forgot the IP version =
from
the list of additions to the -03 version of the metric. Moreover, I had =
read
Glenn's answer to Al sloppily, which I apologize. I now realize that =
Glenn
was not in favour of including error of incT into the actual metric.

After some thought, I came to the conclusion that I support Glenn's =
proposal
of calling incT a nominal duration of the inter-packet interval and
mentioning the possiblility of error in discussion later. If process
scheduling or some other reason causes marked variations in incT, this
should be treated as a deviation from the norm rather than a part of a
normal measurement arrangement.

The only changes in v-04 to v-03 are as follows:

- changed definition of incT at p. 8 into

   +  incT, nominal duration of inter-packet interval

- added text on incT error into Sec. 4.6:

   +  Error due to variation of incT

- For clarity, I added a mention of packet loss as the second sentence =
to to
Sec 4.4, which now reads

   The sample metric thus defined is intended to probe the delays and
   the delay variation as experienced by multimedia streams of
   an application. Due to the definition of the metric, also packet =
loss
   status of packets is recorded. The delay is assumed to be measured =
at
   transport layer level. Since a range of packet sizes and nominal=20
   interval between packets is used, the method probes only a specific=20
   time scale of network QoS variations.

For those intested in a computer repair story with a human interest =
tint,
I'll attach one at the end of the message. For the rest, just read the =
docs
& the slides and you'll do fine.

	BR,
	   Vilho

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%=20
 Vilho R=E4is=E4nen                          =20
 vilho.raisanen@nokia.com                =20
 Senior research engineer                =20
 Nokia Research Center, Helsinki, Finland=20
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%

 * * * end of real business * * *

~~ How to boot up a stubborn computer - a small tale of fatherly wisdom =
~~

At boot, my laptop reported that it was the modem driver that caused =
the
crash. From previous experience, I knew that the order in which device
drivers are loaded sometimes makes a difference, at least in some =
operating
systems. Armed with this piece of what I thought of as profound =
knowledge, I
booted NT with and without my PCMCIA NIC, without CD-ROM drive and
experimenting with all combinations. I even tried the VGA mode. To no =
avail.

At this point I remembered my father's tale of a dysfunctional old =
radio. He
told me having opened the casing, and - lacking training in electronics =
-
could not do very much more. So he just blew some dust off the valves =
and
closed it again. Surprise, surprise: now it worked.

Lacking better ideas, I did just the same: I located the modem, opened =
a
tiny cover and blew some imaginary dust off the printed board. =
Reassembly,
boot - succcess.


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Network Working Group                                        V. =
Raisanen
INTERNET-DRAFT                                                     =
Nokia
Expiration Date:  June 2001                                 G. =
Grotefeld
                                                                =
Motorola
                                                           December =
2000


	    Network performance measurement for periodic streams
                       <draft-ietf-ippm-npmps-04.txt>


1. Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six =
months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at               =
=20
   http://www.ietf.org/ietf/1id-abstracts.txt                           =
=20

   The list of Internet-Draft shadow directories can be accessed at     =
=20
   http://www.ietf.org/shadow.html

   This memo provides information for the Internet community. This
   memo does not specify an Internet standard of any
   kind. Distribution of this memo is unlimited.


2. Abstract

   This document describes a sample metric suitable for application-
   level IP network transport measurement for periodic streams, such as
   VoIP or streaming multimedia over IP. In this document,  the reader
   is assumed to be familiar with the terminology of the  Framework for
   IP Performance Metrics RFC 2330 [1]. This document is parallel to
   A One-way Delay Metric for IPPM RFC 2679 [2]. Although this document
   is based on the delay metrics, other characteristics can be measured
   with this approach, too. For example, packet loss rate, reordering /
   out-of sequence, and successive delay variation are all additional
   metrics which can be built from this baseline set of measurements.




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3. Introduction

   This document discusses concepts relevant to  application-level
   performance measurements of an IP network. The original driver for
   this work is Quality of Service of interactive periodic streams such
   as multimedia conference over IP, but the idea of application-level
   measurementJune have a wider scope. In the following, interactive
   multimedia traffic is used as an example to illustrate the concept.

   A constant bit-rate (CBR), or nearly CBR, streaming (hereinafter
   called periodic) multimedia bit streamJune be simulated by
   transmitting uniformly sized packets (or mostly uniformly sized
   packets) at regular intervals through the network to be evaluated.
   The "mostly uniformly sized packets"June be found in applications
   thatJune use smaller packets during a portion of the stream (e.g.
   digitally coded voice during silence periods). As noted in the
   framework document [1], a sample metric  using regularly spaced
   singleton tests has some limitations when considered from a
   general measurement point of view: only part of the network
   performance spectrum is sampled. However, from the point of view of
   application-level performance, this is actually good news as
   explained below.

   IP delivery service measurements have been discussed within the
   International Telecommunications Union (ITU). A framework for IP
   service level measurements (with references to the framework for IP
   performance [1]) that is intended to be suitable for service =
planning
   has been approved as I.380 [3]. The emphasis in the ITU
   recommendation is on passive measurements, though not explicitly
   forbidding active measurements. The present contribution proposes a
   method that is usable both for service planning and end-user testing
   purposes, and is based on active measurements.

3.1 Terminology

   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 [4].
   Although RFC 2119 was written with protocols in mind, the key words
   are used in this document for similar reasons.  They are used to
   ensure the results of measurements from two different =
implementations
   are comparable, and to note instances when an implementation could
   perturb the network.







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3.2 Considerations related to delay

   For interactive multimedia sessions, end-to-end delay is an
   important factor. Too large a delay reduces the quality of the
   multimedia session as perceived by the participants. One approach =
for
   managing end-to-end delays on an Internet path involving
   heterogeneous link layer technologies is to use per-domain delay
   quotas (e.g. 50 ms for a particular IP domain). The 50 ms would
   then be included into a calculation of an end-to-end delay bound. A
   practical implementation of such as scheme ought to address issues
   like possibility of asymmetric delays in a route in different
   directions, and sensitivity of an application to delay variations in
   a given domain. There are several alternatives as to which kind of
   derivative delay metric one ought to use in managing end-to-end QoS.
   This question, although very interesting, is not within the scope of
   this draft and is not discussed further here.

   In the following, a methodology and metric are presented for
   measuring media stream transport QoS in an IP domain. The
   measurement resultsJune be used in derivative metrics such as
   average and maximum delays. A metric is presented that is a standard
   way for performing a measurement irrespective of the possible QoS
   mechanism utilized in the core network. As an example, for a QoS
   mechanism without hard guarantees, measurementsJune be used to
   ascertain that the "best" class gets the service that has been
   promised for the traffic class in question. Moreover, an operator
   could study the quality of a cheap, low-guarantee service
   implemented using possible slack bandwidth in other classes. Such
   measurements could be made either in studying the feasibility of a
   new service, or on a regular basis.

   The present draft seeks to formalize the measurements in such a way
   that interoperable results are achieved.

3.3 Protocol level issues

   The version of the Internet Protocol used in the measurement affects
   (at least) packet sizes, and should be reported.  =20

   Fig.1 illustrates measurements on multiple protocol levels that
   are relevant to this draft. The major focus of the present draft
   is on transport quality evaluation from application point of
   view. However, to properly account for quality effects of, e.g.,
   operating system and codec on packet voice, it is beneficial to be
   able to measure quality at IP level [5]. Link layer monitoring
   provides a way of accounting for link layer characteristics such
   as bit error rates.



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     ---------------
     | application |
     ---------------
     |  transport  | <--
     ---------------
     |   network   | <--
     ---------------
     |    link     | <--
     ---------------
     |   physical  |
     ---------------=20

   Fig. 1: Different possibilities for performing measurements: a
   protocol view. Above, "application" refers to all layers above
   L4 and is not used in the OSI sense.

   In general, the results of measurementsJune be influenced by
   individual application requirements/responses related to the
   following issues:

   +  Lost packets: ApplicationsJune have varying tolerance to lost=20
      packets.  Another consideration is the distribution of lost
      packets (i.e. random or bursty).
   +  Long delays: Many applications will consider packets delayed=20
      longer than a certain value to be equivalent to lost packets
      (i.e. real time applications).
   +  Duplicate packets: Some applicationsJune be perturbed if=20
      duplicate packets are received.
   +  Out-of-sequence: Some applicationsJune be perturbed if packets
      are received out of sequence. ThisJune be in addition to the
      possibility of exceeding the "long" delay threshold as a result
      of being out of sequence. An out-of-sequence packet outcome
      occurs when a single IP packet received at a DST measurement
      point has a sequence number  higher than that which is
      expected, and therefore, the packet is OOS due to re-ordering.=20
   +  Corrupt packet header: Most applications will probably treat a
      packet with a corrupt header as equivalent to a lost packet.
   +  Corrupt packet payload: Some applications (e.g. digital voice=20
      codecs)June accept corrupt packet payload.  In some cases, the
      packet payloadJune contain application specific forward error
      correction (FEC) that can compensate for some level of=20
      corruption.
   +  Spurious packet: DstJune receive spurious packets (i.e. packets
      that are not sent by the Src as part of the metric).  Many
      applicationsJune be perturbed by spurious packets.

   Depending, e.g., on the observed protocol level, some issues listed
   aboveJune be indistinguishable from others by the application, it
  June be important to preserve the distinction for the operators of
   Src, Dst, and/or the intermediate network(s).

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   Because of the possible errors listed above, in most cases it is=20
   recommended to use a packet identifier for each packet generated at
   Src. Identifiers for the metric sampleJune be those used by the
   underlying transport layer (e.g. RTP sequence number) or the same=20
   identifiers used by an application if the application to be modeled=20
   by the metric uses an identifier. The possibility of identifier=20
   roll-over (reuse if intentional) during a metric collected over
   a "long" (application dependent) time should be observed.

   If the application does not use an identifier, itJune still be=20
   useful to add identifiers to the packets in the metric sample to=20
   help identify possible anomalies such as out of sequence packets.
   This would be most useful in the case where the application=20
   expects to receive packets in sequence, but has no capability to
   identify the sequence of packets received at Dst.

3.4 Application-level measurement

   In what follows, a metric is proposed for application-level network
   performance measurement. In effect, the metric is an emulation of
   periodic multimedia stream performance. The justification for using
   realistic application metrics in the measurement:

   +  The results of the measurement are automatically relevant to the
      performance as perceived by the application in question.
   +  All the packets in the measurement contribute to accuracy of the
      estimation of performance variation at timescale that is
      important to the multimedia application (packetization
      interval).
   +  Effects of elastic traffic (TCP) on measurement packets are
      different for a sustained stream than for single packets during
      overloading situations as discussed in [3].

3.5 Measurement types

   Delay measurements can be one-way [2,3], paired one-way, or=20
   round-trip [6]. Accordingly, the measurementsJune be performed
   either with synchronized or unsynchronized Src/Dst host clocks.
   Different possibilities are listed below.

   The reference measurement setup for all measurement types is
   shown in Fig. 2.








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     ----------------< IP >--------------------
     |          |                  |          |
   -------   -------           --------    --------
   | Src |   | MP  |           | MP   |    | Dst  |
   -------   |(Src)|           |(Dst) |    --------
             -------           --------   =20

   Fig. 2: Example setup for the metric usage.

   An example of the use of the metric is a setup with a source host
   (Src), a destination host (Dst), and corresponding measurement
   points (MP(Src) and MP(Dst)) as shown in Figure 2. Separate =
equipment
   for measurement pointsJune be used if having Src and/or Dst conduct=20
   the measurementJune significantly affect the delay performance to be
   measured. MP(Src)should be placed/measured close to the egress point =

   of packets from Src. MP(Dst) should be placed/measure close to=20
   the ingress point of packets for Dst. "Close" is defined as a
   distance sufficiently small so that application-level performance
   characteristics measured (such as delay) can be expected to follow=20
   the corresponding performance characteristic between Src and Dst to=20
   an adequate accuracy. Basic principle here is that measurement
   results between MP(Src) and MP(Dst) should be the same as for a
   measurement between Src and Dst, within the general error margin
   target of the measurement (e.g., < 1 ms; number of lost packets is
   the same). If this is not possible, the difference between MP-MP
   measurement and Src-Dst measurement should preferably be systematic.
=20
   The test setup just described fulfills two important criteria:
   1) Test is made with realistic stream metrics, emulating - for =
example -
   a full-duplex Voice over IP (VoIP) call.
   2) Either one-way or round-trip characteristicsJune be obtained.

   It is also possible to have intermediate measurement points between=20
   MP(Src) and MP(Dst), but that is beyond the scope of this document.

3.5.1 One way measurement

   In the interests of specifying metrics that are as generally usable
   as possible, application-level measurements based on one-way delays
   are used in the example metrics. The implication of =
application-level
   measurement for bi-directional applications such as interactive
   multimedia conferencing is discussed below.

   Performing a single one-way measurement only yields information on
   network behavior in one direction. Moreover, the stream at the
   network transport level does not emulate accurately a full-duplex
   multimedia connection.



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3.5.2 Paired one way measurement

   Paired one way delay refers to two multimedia streams: Src to Dst=20
   and Dst to Src for the same Src and Dst. By way of example, for
   some applications, the delay performance of each one way path is
   more important than the  round trip delay. This is the case for
   delay-limited signals such as VoIP. Possible reasons for the
   difference between one-way delays is different routing of streams
   from Src to Dst vs. Dst to Src.

   For example, a paired one way measurementJune show that Src to Dst
   has an average delay of 30ms while Dst to Src has an average delay
   of 120ms. To a round trip delay measurement, this example would
   look like an average of 150ms delay.  Without the knowledge of the
   asymmetry, we might miss a problem that the application at either
   endJune have with delays averaging more than 100ms.

   Moreover, paired one way delay measurement emulates a full-duplex
   VoIP call more accurately than a single one-way measurement only.

3.5.3 Round trip measurement

   From the point of view of periodic multimedia streams,
   round-trip measurements have two advantages: they avoid the need of
   host clock synchronization  and they allow for a simulation of
   full-duplex connections. The former aspect means that a measurement
   is easily performed, since no special equipment or NTP setup is
   needed. The latter property means that measurement streams are
   transmitted in both directions. Thus, the measurement provides
   information on quality of service as experienced by appropriate
   application.

   The downsides of round-trip measurement are the need for more
   bandwidth than an one-way test and more complex accounting of
   packet loss. Moreover, the stream that is returning towards the
   original senderJune be more bursty than the one on the first "leg" =
of
   the round-trip journey. The last issue, however, means in practice
   that returning stream experiences worse QoS than the other one, and
   the performance estimates thus obtained are pessimistic ones. The
   possibility of asymmetric routing and queuing must be taken into
   account during analysis of the results.

   Please note that with suitable arrangements, round-trip measurements
  June be performed using paired one way measurements.






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4 Sample metric for multimedia stream simulation

   The sample metric presented here is similar to the sample metric
   Type-P-One-way-Delay-Poisson-Stream presented in [2]. "Singletons", =
as
   defined in [1] and [2] are not directly used in this document =
because
   certain key results (such as duplicate or out of sequence packets)
   cannot be identified in the context of a singleton, but only as part
   of a sample.

4.1 Metric name
  =20
   Type-P-One-way-Delay-Periodic-Stream

4.2 Metric parameters

4.2.1 Global metric parameters

   These parameters are applicable to the metrics collected in the=20
   following sections (4.2.2, 4.2.3, and 4.2.4).

   +  Src, the IP address of a host
   +  Dst, the IP address of a host
   +  IPV, the IP version (IPv4/IPv6) used in the measurement
   +  T0, a time, for starting to generate packets and taking=20
         measurements for a sample
   +  Tf, a time, greater than T0, for stopping generation of packets
         for a sample
   +  incT, nominal duration of inter-packet interval
   +  packet size p(j), the number of bytes in each packet of Type-P of =

         size j
   +  dTloss, a time interval, used for determining if a packet should
         be considered lost
   +  Tcons, a time interval [optional]

      While a number of applications will use one packet size (j =3D =
1),
      other applicationsJune use packets of different sizes (j > 1).=20
      Especially in cases of congestion, itJune be useful to have=20
      packets smaller than the maximum or predominant size of packets
      in the periodic stream.

4.2.2 Metrics collected at MP(Src)

   +  Tstamp(Src)[i], for each packet [i], the time of the packet as
      measured at MP(Src)
   +  PktID [i], for each packet [i], an identification number for the
      the packet sent from Src to Dst.




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   +  PktSiTy [i], for each packet [i], the packet size and/or type.
      Some applicationsJune use packets of different size, either=20
      because of application requirements or in response to IP=20
      performance experienced.

4.2.3 Metrics collected at MP (Dst)

   +  dTstop, a time interval, used to add to time Tf to determine when =
to
         stop collecting metrics for a sample
   +  Tstamp(Dst)[i], for each packet [i], the time of the packet as
      measured at MP(Dst)
   +  PktID [i], for each packet [i], an identification number for the
      the packet received at Dst from Src.
   +  PktSiTy [i], for each packet [i], the packet size and/or type.
      Some applicationsJune use packets of different size, either=20
      because of application requirements or in response to IP=20
      performance experienced.
   +  PktStatus [i], for each packet [i], the status of the packet
      received.  Possible status includes: OK, packet header corrupt,
      packet payload corrupt, spurious, duplicate, out-of-sequence.

4.2.4 Metrics resulting when metrics collected at MP(Src) and MP(Dst)
      are merged

   These parameters are only available as a complete set when the=20
   parameters from the preceding sections (4.2.1, 4.2.2, and 4.2.3 are=20
   combined. =20

   +  Tstamp(Src)[i], for each packet [i], the time of the packet as
      measured at MP(Src).  This entryJune be blank or noted as N/A
      for spurious packets received at MP(Dst)
   +  Tstamp(Dst)[i], for each packet [i], the time of the packet as
      measured at MP(Dst).  This entryJune be blank or noted as N/A
      for packets not received at MP(Dst), received with corrupt=20
      packet headers, or for duplicate packets received at MP(Dst).
   +  PktID [i], for each packet [i], an identification number for the
      the packet received.  This identification numberJune be corrupted
      for certain packets received at MP (Dst).
   +  PktSiTy [i], for each packet [i], the packet size and/or type.
   +  PktStatus [i], for each packet [i], the status of the packet=20
      received.  Possible status includes: OK, packet header corrupt,
      packet payload corrupt, spurious, duplicate, out-of-sequence.









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   +  Delay [i], for each packet [i], the time interval Tstamp(Dst)[i] =
-=20
      Tstamp(Src)[i].  For the following conditions, it will not be=20
      possible to be able to compute delay:
         Spurious: There will be no Tstamp(Src)[i] time
         Not received: There will be no Tstamp (Dst) [i]
         Corrupt packet header: There will be no Tstamp (Dst) [i]
         Duplicate:  Only the first non-corrupt copy of the packet=20
         received at  Dst should have Delay [i] computed.
   +  SDV[i] [optional] , for each packet [i] except the first one:=20
         momentary delay variation between successive packets, i.e., =
the=20
         time interval Delay[i] - Delay [i-1].  SDV[i]June be negative, =

         zero, or positive. Delay for both packets i and i+1 must be=20
         calculable according to the definition above or SDV[i] is=20
         undefined.

4.3 High level description of the procedure to collect a sample

   Beginning on or after time T0, Type-P packets are generated=20
   by Src and sent to Dst until time Tf is reached with a nominal=20
   interval between the first bit of successive packets of incT as=20
   measured at MP(Src).  incTJune be nominal due to a number of =
reasons:
   variation in packet generation at Src, clock issues (see section =
4.6),
   etc.

   MP(Src) records the following information only for packets with=20
   timestamps between and including T0 and Tf: timestamp,=20
   packet identifier, and packet size/type of each packet sent from Src =

   to Dst that is part of the sample. =20

   MP (Dst) records the following information only for packets with=20
   time stamps between T0 and (Tf+ dTstop): timestamp, packet =
identifier,=20
   packet size/type, and received status of each packet received from=20
   Src at Dst that is part of the sample.  Optionally, at a time Tf +=20
   Tcons, the data from MP(Src) and MP(Dst) are consolidated to derive=20
   the results of the sample metric.=20

   To prevent stopping data collection too soon, dTcons should be =
greater
   than or equal to dTstop.  Conversely, to keep data collection=20
   reasonably efficient, dTstop should be some reasonable time interval =

   (seconds/minutes/hours), even if dTloss is infinite or extremely =
long.









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4.4 Discussion

   The sample metric thus defined is intended to probe the delays and
   the delay variation as experienced by multimedia streams of
   an application. Due to the definition of the metric, also packet =
loss
   status of packets is recorded. The delay is assumed to be measured =
at
   transport layer level. Since a range of packet sizes and nominal=20
   interval between packets is used, the method probes only a specific=20
   time scale of network QoS variations.

   There are a number of factors that should be taken into account when =

   collecting a sample metric of Type-P-One-way-Delay-Periodic-Stream.

   +  T0 and (Tf + dTloss) should specify a long enough time interval =
to=20
      represent a reasonable use of the application under test (e.g. do =

      not provide only a 100 ms time interval for a phone call)

   +  T0 and (Tf + dTloss) should specify a time interval that is not=20
      excessively long compared to the usage of the application under =
test
      (e.g. do not provide a one week continuous phone call)

   +  The nominal interval between packets (incT) and the packet =
size(s)
      (p(j)) should not define an equivalent bit rate that is in excess =

      of the capacity of the egress port of Src, the ingress port of =
Dst,
      or the carrying capacity of the intervening network(s). ThereJune
      be exceptional cases to test the response of the application to
      overload conditions in the transport networks, but these cases=20
      should be strictly controlled.

   +  Real delay values will be positive.  Therefore, it does not make
      sense to report a negative value as a real delay.  However, an
      individual zero or negative delay value might be useful as part =
of
      a stream when trying to discover a distribution of the delay =
values
      of a stream.

   +  Depending on measurement topology, delay valuesJune be as low as
      100 usec to 10 msec, whereby itJune be important for Src and Dst =
to
      synchronize very closely.  GPS systems afford one way to achieve
      synchronization to within several 10s of usec.  Ordinary =
application
      of NTPJune allow synchronization to within several msec, but this
      depends on the stability and symmetry of delay properties among =
those
      NTP agents used, and this delay is what we are trying to measure. =
A
      combination of some GPS-based NTP servers and a conservatively
      designed and deployed set of other NTP servers should yield good
      results, but this is yet to be tested.

   +  Reordering of packets is best discussed in terms of the entire
      set of measurement packets received, i.e. should be addressed in
      Sec. 4.9.1.

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   +  A given methodology will have to include a way to determine
      whether packet was lost or whether delay is merely very large =
(and
      the packet is yet to arrive at Dst). The global metric parameter
      dTloss defines a time interval such that delays larger than =
dTloss
      are interpreted as losses.=20
      {Comment: Note that, for many applications of these metrics, the
      harm in treating a large delay as infinite might be zero or very
      small.  A TCP data packet, for example, that arrives only after
      several multiples of the RTTJune as well have been lost.}

4.5 Additional Methodology Aspects

   As with other Type-P-* metrics, the detailed methodology will depend
   on the Type-P (e.g., protocol number, UDP/TCP port number, size,
   precedence).

4.6 Errors and uncertainties

   The description of any specific measurement method should include an
   accounting and analysis of various sources of error or uncertainty.
   The Framework document [1] provides general guidance on this point,=20
   but we note here the following specifics related to delay metrics:

   +  Error due to variation of incT

   +  Errors or uncertainties due to uncertainties in the clocks of the
      MP(Src) and MP(Dst) measurement points.

   +  Errors or uncertainties due to the difference between 'wire time'
      and 'host time'.

4.6.1. Errors or uncertainties related to Clocks

   The uncertainty in a measurement of one-way delay is related, in
   part, to uncertainties in the clocks of MP(Src) and MP(Dst). In
   the following, we refer to the clock used to measure when the packet
   was measured at MP(Src) as the MP(Src) clock and we refer to the=20
   clock used to measure when the packet was received at MP(Dst) as the
   MP(Dst) clock.  Alluding to the notions of synchronization, =
accuracy,
   resolution, and skew, we note the following:

   +  Any error in the synchronization between the MP(Src) clock and=20
      the MP(Dst) clock will contribute to error in the delay
      measurement.  We say that the MP(Src) clock and the MP(Dst)
      clock have a synchronization error of Tsynch if the MP(Src) clock
      is Tsynch ahead of the MP(Dst) clock.  Thus, if we know the
      value of Tsynch exactly, we could correct for clock
      synchronization by adding Tsynch to the uncorrected value of
      Tstamp(Dst)[i] - Tstamp(Src) [i].

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   +  The accuracy of a clock is important only in identifying the time
      at which a given delay was measured.  Accuracy, per se, has no
      importance to the accuracy of the measurement of delay.  When
      computing delays, we are interested only in the differences
      between clock values, not the values themselves.

   +  The resolution of a clock adds to uncertainty about any time
      measured with it.  Thus, if the MP(Src) clock has a resolution of
      10 msec, then this adds 10 msec of uncertainty to any time value
      measured with it.  We will denote the resolution of the source
      clock and the MP(Dst) clock as ResMP(Src) and ResMP(Dst),
      respectively.
   +  The skew of a clock is not so much an additional issue as it is a
      realization of the fact that Tsynch is itself a function of time.
      Thus, if we attempt to measure or to bound Tsynch, this needs to
      be done periodically.  Over some periods of time, this function
      can be approximated as a linear function plus some higher order
      terms; in these cases, one option is to use knowledge of the
      linear component to correct the clock.  Using this correction, =
the
      residual Tsynch is made smaller, but remains a source of
      uncertainty that must be accounted for.  We use the function
      Esynch(t) to denote an upper bound on the uncertainty in
      synchronization.  Thus, |Tsynch(t)| <=3D Esynch(t).

   Taking these items together, we note that naive computation=20
   Tstamp(Dst)[i] - Tstamp(Src) [i] will be off by Tsynch(t) +/-=20
   (ResMP(SRc) + ResMP(Dst)).  Using the notion of Esynch(t), we note =20
   that these clock-related problems introduce a total uncertainty of=20
   Esynch(t)+ Rsource + Rdest.  This estimate of total clock-related=20
   uncertainty should be included in the error/uncertainty analysis of=20
   any measurement implementation.

4.6.2. Errors or uncertainties related to Wire-time vs Host-time

   As we have defined one-way periodic delay, we would like to measure=20
   the time between when a packet is measured and time-stamped at
   MP(Src) and when it arrives and is time-stamped at MP(Dst) and we
   refer to these as "wire times."  If the timings are themselves
   performed by software on Src and Dst, however, then this software =
can
   only directly measure the time between when Src generates the packet
   just prior to sending the test packet and when Dst has started to
   process the packet after having received the test packet, and we =
refer
   to these two points as "host times".

   To the extent that the difference between wire time and host time is
   accurately known, this knowledge can be used to correct for wire =
time
   measurements and the corrected value more accurately estimates the
   desired (host time) metric.


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   To the extent, however, that the difference between wire time and
   host time is uncertain, this uncertainty must be accounted for in an
   analysis of a given measurement method.  We denote by Hsource an
   upper bound on the uncertainty in the difference between wire time
   of MP(Src) and host time on the Src host, and similarly define Hdest
   for the difference between the host time on the Dst host and the =
wire
   time of MP(Dst).  We then note that these problems introduce a total
   uncertainty of Hsource+Hdest.  This estimate of total wire-vs-host
   uncertainty should be included in the error/uncertainty analysis of=20
   any measurement implementation.

4.6.3. Calibration

   Generally, the measured values can be decomposed as follows:

      measured value =3D true value + systematic error + random error

   If the systematic error (the constant bias in measured values) can =
be
   determined, it can be compensated for in the reported results.

      reported value =3D measured value - systematic error

   therefore

      reported value =3D true value + random error

   The goal of calibration is to determine the systematic and random
   error generated by the instruments themselves in as much detail as
   possible.  At a minimum, a bound ("e") should be found such that the
   reported value is in the range (true value - e) to (true value + e)
   at least 95 percent of the time.  We call "e" the calibration error
   for the measurements.  It represents the degree to which the values
   produced by the measurement instrument are repeatable; that is, how
   closely an actual delay of 30 ms is reported as 30 ms.  {Comment: 95
   percent was chosen due to reasons discussed in [2], briefly
   summarized as (1) some confidence level is desirable to be able to
   remove outliers, which will be found in measuring any physical
   property; (2) a particular confidence level should be specified so
   that the results of independent implementations can be compared.}

   From the discussion in the previous two sections, the error in
   measurements could be bounded by determining all the individual
   uncertainties, and adding them together to form

       Esynch(t) + ResMP(Src) + ResMP(Dst) + Hsource + Hdest.





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   However, reasonable bounds on both the clock-related uncertainty
   captured by the first three terms and the host-related uncertainty
   captured by the last two terms should be possible by careful design
   techniques and calibrating the instruments using a known, isolated,
   network in a lab.

   For example, the clock-related uncertainties are greatly reduced
   through the use of a GPS time source.  The sum of Esynch(t) +=20
   ResMP(Src) + ResMP(Dst) is small, and is also bounded for the=20
   duration of the measurement because of the global time source.

   The host-related uncertainties, Hsource + Hdest, could be bounded by
   connecting two instruments back-to-back with a high-speed serial =
link
   or isolated LAN segment.  In this case, repeated measurements are
   measuring the same one-way delay.

   If the test packets are small, such a network connection has a
   minimal delay thatJune be approximated by zero.  The measured delay
   therefore contains only systematic and random error in the
   instrumentation.  The "average value" of repeated measurements is =
the
   systematic error, and the variation is the random error.

   One way to compute the systematic error, and the random error to a
   95% confidence is to repeat the experiment many times - at least
   hundreds of tests.  The systematic error would then be the median.
   The random error could then be found by removing the systematic =
error
   from the measured values.  The 95% confidence interval would be the
   range from the 2.5th percentile to the 97.5th percentile of these
   deviations from the true value.  The calibration error "e" could =
then
   be taken to be the largest absolute value of these two numbers, plus
   the clock-related uncertainty.  {Comment: as described, this bound =
is
   relatively loose since the uncertainties are added, and the absolute
   value of the largest deviation is used.  As long as the resulting
   value is not a significant fraction of the measured values, it is a
   reasonable bound.  If the resulting value is a significant fraction
   of the measured values, then more exact methods will be needed to
   compute the calibration error.}

   Note that random error is a function of measurement load.  For
   example, if many paths will be measured by one instrument, this =
might
   increase interrupts, process scheduling, and disk I/O (for example,
   recording the measurements), all of whichJune increase the random
   error in measured singletons.  Therefore, in addition to minimal =
load
   measurements to find the systematic error, calibration measurements
   should be performed with the same measurement load that the
   instruments will see in the field.




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   We wish to reiterate that this statistical treatment refers to the
   calibration of the instrument; it is used to "calibrate the meter
   stick" and say how well the meter stick reflects reality.

4.7 Reporting the metric

   The calibration and context in which the metric is measured MUST be
   carefully considered, and SHOULD always be reported along with =
metric
   results.  We now present five items to consider: the Type-P of test
   packets, the threshold of delay equivalent to loss, error
   calibration, the path traversed by the test packets, and background=20
   conditions at Src, Dst, and the intervening networks during a =
sample.
   This list is not exhaustive; any additional information that could =
be=20
   useful in interpreting applications of the metrics should also be=20
   reported.

4.7.1. Type-P

   As noted in the Framework document [1], the value of the metricJune
   depend on the type of IP packets used to make the measurement, or
   "type-P".  The value of Type-P-One-way-Periodic-Delay could change=20
   if the protocol (UDP or TCP), port number, size, or arrangement for=20
   special treatment (e.g., IP precedence or RSVP) changes.  The exact=20
   Type-P used to make the measurements MUST be accurately reported.

4.7.2. Threshold for delay equivalent to loss

   In addition, the threshold for delay equivalent to loss (or=20
   methodology to determine this threshold) MUST be reported.

4.7.3. Calibration results

   +  If the systematic error can be determined, it SHOULD be removed
      from the measured values.

   +  You SHOULD also report the calibration error, e, such that the
      true value is the reported value plus or minus e, with 95%
      confidence (see the last section.)

   +  If possible, the conditions under which a test packet with finite
      delay is reported as lost due to resource exhaustion on the
      measurement instrument SHOULD be reported.








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4.7.4. Path

   The path traversed by the packets SHOULD be reported, if possible.
   In general it is impractical to know the precise path a given packet
   takes through the network.  The precise pathJune be known for
   certain Type-P packets on short or stable paths. If Type-P includes
   the record route (or loose-source route) option in the IP header,
   and the path is short enough, and all routers* on the path support
   record (or loose-source) route, then the path will be precisely
   recorded.

   ThisJune be impractical because the route must be short enough,
   many routers do not support (or are not configured for) record =
route,
   and use of this feature would often artificially worsen the
   performance observed by removing the packet from common-case
   processing.  However, partial information is still valuable context.
   For example, if a host can choose between two links* (and hence two
   separate routes from Src to Dst), then the initial link used is
   valuable context.  {Comment: For example, with Merit's NetNow setup,
   a Src on one NAP can reach a Dst on another NAP by either of several
   different backbone networks.}

4.7.5 Background conditions

   In many cases, the results of a sampleJune be influenced by =
conditions
   at Src, Dst, and/or any intervening networks.  Some things thatJune=20
   affect the results of a sample include:  traffic levels and/or =
bursts=20
   during the sample, link and/or host failures, etc.  Information =
about
   the background conditionsJune only be available by non-Internet =
means
   (e.g. phone calls, television) andJune only become available days =
after
   samples are taken.

4.8 Single sample vs. a "sample of samples"

   Because this metric represents a periodic stream as one sample, =
there=20
  June be value in running multiple tests using this metric to collect
   a "sample of samples".  For example, itJune be more appropriate to=20
   test 1,000 two-minute VoIP calls rather than a single 2,000 minute
   VoIP call.  When considering collection of a sample of samples, =
issues
   like the interval between samples (e.g. Poisson vs. periodic, time =
of
   day/day of week), composition of samples (e.g. equal (Tf-T0 =
duration,
   different packet sizes), and network considerations (e.g. run =
different
   samples over different intervening link-host combinations) should be
   taken into account.  For items like the interval between samples,
   the pattern of use of the application being measured should be=20
   considered.




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4.9 Statistics based on Type-P-One-way-Delay-Periodic-Stream

4.9.1 Statistics calculable from one sample

   As a metric based on a sample representative of certain=20
   applications, some general purpose statistics (e.g. median and=20
   percentile)June be less applicable than ways to characterize the=20
   range of delay values recorded during the sample metrics.

   Example, a sample metric generates 100 packets as measured at =
MP(Src)
   with the following measurements at MP(Dst)

     +  80 packets received with delay [i] <=3D 20 ms
     +   8 packets received with delay [i] > 20 ms
     +   5 packets received with corrupt packet headers
     +   4 packets from MP(Src) with no matching packet recorded
           at MP(Dst) (effectively lost)
     +   3 packets received with corrupt packet payload and
           and delay [i] <=3D 20 ms
     +   2 packets that duplicate one of the 80 packets received=20
           correctly in the first line

   For this example, packets are considered acceptable if they are
   received with less than or equal to 20ms delays and without corrupt
   packet headers or packet payload.  In this case, the percentage
   of acceptable packets is 80/100 =3D 80%.

   For a different application which will accept packets with corrupt
   packet payload and no delay bound (so long as the packet is =
received),=20
   the percentage of acceptable packets is (80+8+3)/100 =3D 91%.

4.9.2 Statistics calculable from multiple samples

   For computing statistics, a "sample of samples" series of=20
   measurementsJune be performed. As discussed in section 4.8, under=20
   these conditions, general purpose statistics (e.g. median, =
percentile,
   etc.)June be more relevant as a more statistically significant
   number of packets are used.


5. Security Considerations

5.1 Denial of Service Attacks

   This metric generates a periodic stream of packets from one host =
(Src)
   to another host (Dst) through intervening networks.  This metric
   could be abused for denial of service attacks directed at Dst and/or
   the intervening network(s).


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   Administrators of Src, Dst, and the intervening network(s) should=20
   establish bilateral or multi-lateral agreements regarding the =
timing,
   size, and frequency of collection of sample metrics.  Use of this
   metric in excess the terms agreed between the participantsJune BE=20
   cause for immediate rejection or discard of packets or other=20
   escalation procedures defined between the affected parties.

5.2 User data confidentiality

   This metric generates packets for a sample metric, rather than
   taking samples based on user data.  Thus, this metric does not=20
   threaten user data confidentiality.

5.3 Interference with the metric

   ItJune be possible to identify that a certain packet or stream of=20
   packets are part of a sample metric. With that knowledge at Dst=20
   and/or the intervening networks, it is possible to change the=20
   processing of the packets (e.g. increasing or decreasing delay)=20
   thatJune distort the measured performance.  ItJune also be=20
   possible to generate additional packets that appear to be part of=20
   the sample metric. These additional packets are likely to perturb=20
   the results of the sample measurement.

   To discourage the kind of interference mentioned above, packet
   interference checks, such as cryptographic hash,June be used.


6. Acknowledgements

   The authors wish to thank the chairs of the IPPM WG for comments
   that have made the present draft clearer and more focused. Howard
   Stanislevic and Al Morton ahave presented useful comments and
   questions. The authors have also built on the substantial
   foundations laid by the authors of the framework for IP
   performance [1].














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7. References
         =20
   [1] V.Paxson, G.Almes, J.Mahdavi, and M.Mathis: Framework for IP     =
 =20
       Performance Metrics, IETF RFC 2330,June 1998.
   [2] G.Almes, S.Kalidindi, and M.Zekauskas: A one-way delay metric
       for IPPM, IETF RFC 2679, September 1999.
   [3] International Telecommunications Union recommendation I.380,
       February 1999.
   [4] S. Bradner: Key words for use in RFCs to Indicate Requirement
       Levels, RFC 2119, March 1997.
   [5] ETSI TIPHON document TS-101329-5 (to be published in July).
   [6] G.Almes, S.Kalidindi, and M.Zekauskas: A round-trip delay
       metric for IPPM, IETF RFC 2681.


8. Authors' contact information

   Vilho Raisanen <Vilho.Raisanen@nokia.com>
   P.O. Box 407
   Communication Systems Laboratory
   Nokia Research Center
   FIN-00045 Nokia Group
   Finland
   Phone +358 9 4376 1
   Fax. +358 9 4376 6852

   Glenn Grotefeld <g.grotefeld@motorola.com>
   Motorola, Inc.
   1303 E. Algonquin Road
   4th Floor
   Schaumburg, IL 60196
   USA
   Phone  +1 847 576-5992
   Fax    +1 847 538-7455
















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To: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
Cc: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>, ippm@advanced.org,
        "Carl W. Kalbfleisch" <cwk@verio.net>,
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Subject: Re: on <draft-ietf-ippm-owdp-00.txt>'s model
References: <A32A6A6D3178D3119C300090279CB296066FB984@njb140po02.ems.att.com>
From: stanislav shalunov <shalunov@internet2.edu>
Date: 13 Dec 2000 02:17:55 -0500
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"Cole, Robert G (Bob), ALSVC" <rgcole@att.com> writes:

> Will there be cases where the snmp mgmr and the OWDP source are
> within the same AS and the sink is outside, in which case it may be
> necessary to have a control protocol between the source and sink to
> configure the sink?

I think at this point it's agreed that OWDP-Control should stay,
and that language should be added to the draft to make clear the fact
that it's not the only way to set up OWDP-Test sessions.

The remaining question is whether OWDP-Control and OWDP-Test should be
in different drafts, or in the same draft.

I would say that OWDP-Control should only be made a separate draft if
it's clear how to make it applicable to other types of measurement
traffic.  (The next question, of course, being what set of other
types we're talking about.)

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

"Hey!  Who took the cork off my lunch?!"               -- W. C. Fields



From matt@advanced.org  Wed Dec 13 13:10:11 2000
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Date: Wed, 13 Dec 2000 12:52:19 -0500
To: ippm@advanced.org
From: Al Morton <acmorton@att.com>
Subject: Re: npmps-04
Cc: matt@advanced.org, kaeo@merike.com
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Vilho, Glenn, all,

I agree with your proposed resolution, to indicate that
incT is a nominal value and list this deviation in the error section.
(I was thinking about it as error when I suggested "incTerror".)
I'm not sure if the lone bullet item is sufficient (maybe
a short subsection will help explain it, extracted from e-mails).

At least, the suggestion below may be helpful.
Thanks for addressing this,
Al

<from npmps-04>
4.6 Errors and uncertainties

   The description of any specific measurement method should include an
   accounting and analysis of various sources of error or uncertainty.
   The Framework document [1] provides general guidance on this point, 
   but we note here the following specifics related to 
>>>_periodic_streams_and_<<<< delay metrics:

   +  Error due to variation of incT
   +  Errors or uncertainties due to uncertainties in the clocks of the
      MP(Src) and MP(Dst) measurement points.
   +  others...

At 06:06 AM 12/13/00 +0200, vilho.raisanen@nokia.com wrote:
>Hello,
>
>please find attached my (intended) slides, the results document, as well as
>the v-04 of the npmps draft. Lacking my slides, I forgot the IP version from
>the list of additions to the -03 version of the metric. Moreover, I had read
>Glenn's answer to Al sloppily, which I apologize. I now realize that Glenn
>was not in favour of including error of incT into the actual metric.
>
>After some thought, I came to the conclusion that I support Glenn's proposal
>of calling incT a nominal duration of the inter-packet interval and
>mentioning the possiblility of error in discussion later. If process
>scheduling or some other reason causes marked variations in incT, this
>should be treated as a deviation from the norm rather than a part of a
>normal measurement arrangement.
>
>The only changes in v-04 to v-03 are as follows:
>
>- changed definition of incT at p. 8 into
>
>   +  incT, nominal duration of inter-packet interval
>
>- added text on incT error into Sec. 4.6:
>
>   +  Error due to variation of incT
>
>- For clarity, I added a mention of packet loss as the second sentence to to
>Sec 4.4, which now reads
>
>   The sample metric thus defined is intended to probe the delays and
>   the delay variation as experienced by multimedia streams of
>   an application. Due to the definition of the metric, also packet loss
>   status of packets is recorded. The delay is assumed to be measured at
>   transport layer level. Since a range of packet sizes and nominal 
>   interval between packets is used, the method probes only a specific 
>   time scale of network QoS variations.
>
>For those intested in a computer repair story with a human interest tint,
>I'll attach one at the end of the message. For the rest, just read the docs
>& the slides and you'll do fine.
>
>	BR,
>	   Vilho
>
>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
> Vilho Räisänen                           
> vilho.raisanen@nokia.com                 
> Senior research engineer                 
> Nokia Research Center, Helsinki, Finland 
>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
>
> * * * end of real business * * *
>
>~~ How to boot up a stubborn computer - a small tale of fatherly wisdom ~~
>
>At boot, my laptop reported that it was the modem driver that caused the
>crash. From previous experience, I knew that the order in which device
>drivers are loaded sometimes makes a difference, at least in some operating
>systems. Armed with this piece of what I thought of as profound knowledge, I
>booted NT with and without my PCMCIA NIC, without CD-ROM drive and
>experimenting with all combinations. I even tried the VGA mode. To no avail.
>
>At this point I remembered my father's tale of a dysfunctional old radio. He
>told me having opened the casing, and - lacking training in electronics -
>could not do very much more. So he just blew some dust off the valves and
>closed it again. Surprise, surprise: now it worked.
>
>Lacking better ideas, I did just the same: I located the modem, opened a
>tiny cover and blew some imaginary dust off the printed board. Reassembly,
>boot - succcess.
>
>
>Attachment Converted: "C:\Utilities\Eudora\Attach\npmps-res.ppt"
>
>Attachment Converted: "C:\Utilities\Eudora\Attach\1kalvo.ppt"
>
>Attachment Converted: "C:\Utilities\Eudora\Attach\results.pdf"
>
>Attachment Converted:
"C:\Utilities\Eudora\Attach\draft-ietf-ippm-npmps-04.txt"
>



From matt@advanced.org  Wed Dec 13 14:01:57 2000
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From: "Henk Uijterwaal (RIPE-NCC)" <henk@ripe.net>
To: "Cole, Robert G (Bob), ALSVC" <rgcole@att.com>
cc: stanislav shalunov <shalunov@internet2.edu>, ippm@advanced.org,
        "Carl W. Kalbfleisch" <cwk@verio.net>,
        Dan Romascanu <dromasca@lucent.com>,
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Subject: RE: on <draft-ietf-ippm-owdp-00.txt>'s model
In-Reply-To: <A32A6A6D3178D3119C300090279CB296066FB984@njb140po02.ems.att.com>
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Bob,

> 	[Cole, Robert G (Bob), ALSVC] Will there be cases where the snmp
> mgmr and the
> 	OWDP source are within the same AS and the sink is outside, in which
> case it may be
> 	necessary to have a control protocol between the source and sink to
> configure the sink?

Yes, in order to keep this generic, I think we should assume that control,
source and sink are in 3 different AS'ses.  

Henk


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RIPE Network Coordination Centre     WWW: http://www.ripe.net/home/henk
Singel 258                         Phone: +31.20.535-4414,  Fax -4445
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From matt@advanced.org  Wed Dec 13 19:58:27 2000
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From: Sue Moon <Sbmoon@sprintlabs.com>
To: "'ippm@advanced.org'" <ippm@advanced.org>
Subject: Protocol - OWDP?
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I must be asking a very naive question,
but couldn't answer it myself.
Why do we work on the protocol to run measurement
experiments in IPPM working group, whose focus, I believe,
is on performance metrics?
-Sue



From matt@advanced.org  Wed Dec 13 20:15:55 2000
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cc: "'ippm@advanced.org'" <ippm@advanced.org>
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On Wed, 13 Dec 2000, Sue Moon wrote:

> Why do we work on the protocol to run measurement experiments in IPPM
> working group, whose focus, I believe, is on performance metrics?

Making the implementations interoperable is a good thing, as it allows you
to measure between sites equiped with measurement hardware from different
implementations.  The people implementing the metrics follow the IPPM WG,
so one might as well discuss the practical details of making the metrics
interoperable there.  

Henk

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From matt@advanced.org  Thu Dec 14 00:25:37 2000
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To: "Sue Moon" <Sbmoon@sprintlabs.com>, <ippm@advanced.org>
Cc: "Matt Zekauskas" <matt@advanced.org>, "Merike Kaeo" <kaeo@merike.com>
Subject: RE: Protocol - OWDP?
Date: Thu, 14 Dec 2000 00:25:05 -0500
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To add to what Henk said, people within the working group thought
that making interoperable implementations possible would be a good
thing, implementors were already in the group, and it wasn't being
done elsewhere.   However, it is strictly out of the current charter,
but is allowed by the proposed charter (and included in milestones).

A revised charter based on comments here and in the working group
meeting yesterday will be posted within the next few days to the
list.  If you feel that some of the activities are not appropriate
(or if you have other activities that you feel we should be addressing),
please comment on the revised charter.

--Matt

> -----Original Message-----
> From: Sue Moon [mailto:Sbmoon@sprintlabs.com]
> Sent: Wednesday, December 13, 2000 7:51 PM
> To: 'ippm@advanced.org'
> Subject: Protocol - OWDP?
> 
> 
> I must be asking a very naive question,
> but couldn't answer it myself.
> Why do we work on the protocol to run measurement
> experiments in IPPM working group, whose focus, I believe,
> is on performance metrics?
> -Sue
> 



From matt@advanced.org  Thu Dec 14 00:54:41 2000
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Cc: "Matt Zekauskas" <matt@advanced.org>, <lcitkuse@telcordia.com>,
        "William G. Couch" <wcouch@telcordia.com>
Subject: Recent updates on IP QoS metrics in ITU-T SG 13
Date: Thu, 14 Dec 2000 00:56:37 -0500
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Lubomir Citkusev (copied on this mail) created a short presentation
on the U.S. position on IP QoS classes in Y.1541 into the recent
ITU-T SG 13 meeting and the results of the meeting on this topic.
However, due to the limited time available, we could not fit his
presentation into the schedule.

The presentation itself has been pdf'ified and put on the ippm web
server.

http://www.advanced.org/IPPM/Meetings/ietf49/ITU-2000-12-12-Citkusev-IPPM_QoS_Dec00.pdf

The three documents it references are available on the web site as well.
Temporary Document 4-40
  http://www.advanced.org/IPPM/Meetings/ietf49/ITU-2000-12-12-4-40.pdf
Temporary Document 4-41
  http://www.advanced.org/IPPM/Meetings/ietf49/ITU-2000-12-12-4-41.pdf
Temporary Document 4-44
  http://www.advanced.org/IPPM/Meetings/ietf49/ITU-2000-12-12-4-44.pdf

If anyone has a comment based on our "IETF" perspective, please comment
either here, or directly to Lubomir.  If you do comment directly, I would
appreciate a CC to myself.

--Matt



From matt@advanced.org  Thu Dec 14 12:33:19 2000
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Hi--

I had a few comments about the latest incarnation of the OWDP draft.  
Hope these will be helpful, or at least entertaining:

1.  In section 4.1, some mention is made of the "modes octet".  It looks to 
me, however, like the modes field in the connection setup message is 
really 16 bits, or two octets.  Thus, the "unused" space is really 14 
octets, not 15.

2.  As you know, TCP is a bytestream protocol, and thus doesn't
guarantee atomic delivery of messages (with message boundaries
preserved).  This means that either messages need to be fixed sized, or
there needs to be some way of determining the end of each message.
However, it's not clear, for example, how long a Request-Session (4.3)
message is...how long is the zero padding field at the end?

3.  What is an implementation supposed to do in the case of partial 
delivery of a control message (i.e. sender failure)?  I didn't see this 
specified.

4.  A number of fields are multi-byte quantities.  I didn't see any 
discussion of byte-ordering.  I can probably assume you meant 
network byte-order, but this really should be mentioned at least 
once...something to the effect of, "All multi-octet quantities are 
represented in network byte order."

5.  A philosophical statement:  I've become a real fan over the years of
control protocols that are line-oriented, and use ASCII text for their
messages.  It makes for greater ease of understanding and debugging.
Examples of protocols like this are SMTP, NNTP, FTP, and HTTP. Have you
thought about this?

	5a.  As a side note, I think that if either end of the protocol 
	becomes desynchronized (like due to a faulty implementation),
	it'll be rather difficult to get resynchronized with respect to
	where the message boundaries are.

6.  Back to the connection setup in 4.1, I note that there isn't a mode 
*bit* (you used the term "mode value") for unencrypted transfers.  Was 
this an oversight?  Also I note that according to the protocol as 
defined, it's possible for the client to respond with a contradictory 
set of mode bits...presumably the server should drop the connection or 
do some similarly drastic thing if this happens.

7.  The server's second message in connection setup (does it have a
name?) has a field called "Yes/No".  I suggest a more descriptive name. 
It also might be useful to have a way for a server to communicate to a 
client why an authentication was rejected.  (This gets a lot easier if 
you buy into #5 above.)

OK, not quite as organized as I'd like and certainly not a comprehensive
analysis.  Just a few thoughts I had while reading through this draft
over the past couple days.

Bruce.




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From matt@advanced.org  Thu Dec 14 12:33:40 2000
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Subject: Re: draft-ietf-ippm-owdp-01.txt comments
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From: stanislav shalunov <shalunov@internet2.edu>
Date: 14 Dec 2000 04:43:21 -0500
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bmah@cisco.com (Bruce A. Mah) writes:

> I had a few comments about the latest incarnation of the OWDP draft.  
> Hope these will be helpful, or at least entertaining:

Bruce,

Thank you very much for high-quality comments!

I'll incorporate fixes before -02 version, which will reflect IPPM
session discussion, goes out.

Ben, Matt,

I would make server drop connection after a possible timeout in (3)
and do nothing about (5a) [rationale below]; we can discuss it on the
plane if you can think of something smarter to do.

> 1.  In section 4.1, some mention is made of the "modes octet".  It looks to 
> me, however, like the modes field in the connection setup message is 
> really 16 bits, or two octets.  Thus, the "unused" space is really 14 
> octets, not 15.

Thanks for noticing it.  Initially, I made Modes one octet wide, then
decided to add more space for possible future expansion.  Someone
suggested that 13 spare modes might not be enough, so I'll increase it
even further (unused space is there anyway).

> 2.  As you know, TCP is a bytestream protocol, and thus doesn't
> guarantee atomic delivery of messages (with message boundaries
> preserved).  This means that either messages need to be fixed sized, or
> there needs to be some way of determining the end of each message.

The intent was to make it possible to discover the size of a message
after reading (and, possibly, decrypting) first 16 octets.  Each
message type has a specific size.

> However, it's not clear, for example, how long a Request-Session (4.3)
> message is...how long is the zero padding field at the end?

Those three lines attempt to say it's 12 octets.  I guess we should
say it explicitly everywhere.

> 3.  What is an implementation supposed to do in the case of partial 
> delivery of a control message (i.e. sender failure)?  I didn't see this 
> specified.

We need to specify a timeout, I suppose.  Any suggestions of good,
currently fashionable values?

> 4.  A number of fields are multi-byte quantities.  I didn't see any 
> discussion of byte-ordering.  I can probably assume you meant 
> network byte-order, but this really should be mentioned at least 
> once...something to the effect of, "All multi-octet quantities are 
> represented in network byte order."

OK.

> 5.  A philosophical statement:  I've become a real fan over the years of
> control protocols that are line-oriented, and use ASCII text for their
> messages.  It makes for greater ease of understanding and debugging.
> Examples of protocols like this are SMTP, NNTP, FTP, and HTTP. Have you
> thought about this?

Yes.  The trade-off was: easy way to authenticate messages and a way
to encrypt them in a mode already standartized by IETF or be
line-oriented.  A decision was made after some internal discussion to
go with the former option.  Do you feel the advantages of easy
understanding and debugging outweight CBC mode properties?

Additional advantage of simple binary protocol is that it's slightly
easier to code.  (Of course, making reading tcpdump -x output harder
at the same time.)

> 	5a.  As a side note, I think that if either end of the protocol 
> 	becomes desynchronized (like due to a faulty implementation),
> 	it'll be rather difficult to get resynchronized with respect to
> 	where the message boundaries are.

Yes...  I don't see how we could easily fix it.  On the other hand,
there are only four commands, each having a fixed size.  Each of the
commands is mandatory and will be used even in simple scenarios.
I would expect an implementation that desynchronizes sometimes to
desynchronize during almost every session (and the bug would thus be
noticed and fixed).

It's not clear to me what would one do if desynchronization happens,
and one later manages to resynchronize.  State would not be known...
(Do we say "Stop-Sessions" or "Retrieve-Session"?)

> 6.  Back to the connection setup in 4.1, I note that there isn't a mode 
> *bit* (you used the term "mode value") for unencrypted transfers.  Was 
> this an oversight? 

In unauthenticated mode nothing at all is encrypted.  Or did you mean
something else?

> Also I note that according to the protocol as defined, it's possible
> for the client to respond with a contradictory set of mode
> bits...presumably the server should drop the connection or do some
> similarly drastic thing if this happens.

The client actually MUST respond with no more than one bit set in Mode
field.  This needs to be clarified in the draft (oversight).

> 7.  The server's second message in connection setup (does it have a
> name?) has a field called "Yes/No".  I suggest a more descriptive
> name.

"Accept" was used in similar situations elsewhere.  Sounds better?

> It also might be useful to have a way for a server to communicate to a 
> client why an authentication was rejected.  (This gets a lot easier if 
> you buy into #5 above.)

A negative response is any non-zero value, here and elsewhere.  We
intended to define a set of error codes once the draft is more mature.
Would that be sufficient?

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

A large number of installed systems work by fiat.  That is, they work
by being declared to work.                             -- Anatol Holt



From matt@advanced.org  Thu Dec 14 12:33:44 2000
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From: bmah@cisco.com (Bruce A. Mah)
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If memory serves me right, stanislav shalunov wrote:

> I'll incorporate fixes before -02 version, which will reflect IPPM
> session discussion, goes out.

I forgot, for some reason, to CC the ippm list.  If you think it's 
useful, feel free to forward my comments there.

[snip]

> > 2.  As you know, TCP is a bytestream protocol, and thus doesn't
> > guarantee atomic delivery of messages (with message boundaries
> > preserved).  This means that either messages need to be fixed sized, or
> > there needs to be some way of determining the end of each message.
> 
> The intent was to make it possible to discover the size of a message
> after reading (and, possibly, decrypting) first 16 octets.  Each
> message type has a specific size.

OK, sounds good.  Upon reading the draft a little more closely, I now 
realize that the encryption parameters are meant to apply to subsequent 
OWDP-Control messages, where I originally thought they applied to the 
OWDP-Test packets.  This raises another question in my mind...what 
about just using (for example) SSL or some other more established 
encryption layer to encrypt the entire contents of the TCP connection?  
That should simplify both the protocol and its implementation at least 
a little bit (code reuse and all that).

> > However, it's not clear, for example, how long a Request-Session (4.3)
> > message is...how long is the zero padding field at the end?
> 
> Those three lines attempt to say it's 12 octets.  I guess we should
> say it explicitly everywhere.

I generally assume when I'm reading these things that every diagram is
"not to scale".  :-)

> > 3.  What is an implementation supposed to do in the case of partial 
> > delivery of a control message (i.e. sender failure)?  I didn't see this 
> > specified.
> 
> We need to specify a timeout, I suppose.  Any suggestions of good,
> currently fashionable values?

Hmmm.  The only values that come to mind are those that sendmail 
uses for SMTP conversations.  They're highly configurable but I think 
they're on the order of minutes.  I can't remember if SMTP specifies 
exact values or not.

> > 5.  A philosophical statement:  I've become a real fan over the years of
> > control protocols that are line-oriented, and use ASCII text for their
> > messages.  It makes for greater ease of understanding and debugging.
> > Examples of protocols like this are SMTP, NNTP, FTP, and HTTP. Have you
> > thought about this?
> 
> Yes.  The trade-off was: easy way to authenticate messages and a way
> to encrypt them in a mode already standartized by IETF or be
> line-oriented.  A decision was made after some internal discussion to
> go with the former option.  Do you feel the advantages of easy
> understanding and debugging outweight CBC mode properties?

No, not necessarily.  (IETF standardized AES in CBC mode?  I missed 
that part, I guess.  I have to go look up exactly what cipher block 
chaining means now.)

> Additional advantage of simple binary protocol is that it's slightly
> easier to code.  (Of course, making reading tcpdump -x output harder
> at the same time.)

Hmmm.  I'll reiterate my preference for ASCII text, line-oriented
protocols, and stop there.  :-)  (Wait, in the cases where you're
encrypting, isn't tcpdump -x going to be real hard to read anyways?)

> > 	5a.  As a side note, I think that if either end of the protocol 
> > 	becomes desynchronized (like due to a faulty implementation),
> > 	it'll be rather difficult to get resynchronized with respect to
> > 	where the message boundaries are.
> 
> Yes...  I don't see how we could easily fix it.  On the other hand,
> there are only four commands, each having a fixed size.  Each of the
> commands is mandatory and will be used even in simple scenarios.
> I would expect an implementation that desynchronizes sometimes to
> desynchronize during almost every session (and the bug would thus be
> noticed and fixed).

OK, true.  It's not like this is the FTP control protocol, which has 
$DIETY knows how many commands, all with variable sizes.

> It's not clear to me what would one do if desynchronization happens,
> and one later manages to resynchronize.  State would not be known...
> (Do we say "Stop-Sessions" or "Retrieve-Session"?)

I'm not sure what the right answer for this is, either.  This is not a 
huge problem, it just affects the ability of the protocol to be robust 
in the face of failures (note that the TCP checksum mechanism isn't 
100% reliable).  On the other hand, for scenarios where encryption is 
used, it's likely that any integrity checks in that layer would catch 
problems like this and deal with them in some way.

> > 6.  Back to the connection setup in 4.1, I note that there isn't a mode 
> > *bit* (you used the term "mode value") for unencrypted transfers.  Was 
> > this an oversight? 
> 
> In unauthenticated mode nothing at all is encrypted.  Or did you mean
> something else?

What I meant was that there's a bit that means "will accept 
unauthenticated", a bit that means "will accept authenticated", and a 
bit that means "will accept encrypted".  My thinking is colored by 
using PGP for many years; in that system, encryption and authentication 
are orthogonal as you probably know.  So I was thinking that there 
should be a bit that means "will accept unencrypted", for completeness. 
It sounds from your comment that this is not necessary.

[snip]

> > 7.  The server's second message in connection setup (does it have a
> > name?) has a field called "Yes/No".  I suggest a more descriptive
> > name.
> 
> "Accept" was used in similar situations elsewhere.  Sounds better?

Much better.  :-)

> > It also might be useful to have a way for a server to communicate to a 
> > client why an authentication was rejected.  (This gets a lot easier if 
> > you buy into #5 above.)
> 
> A negative response is any non-zero value, here and elsewhere.  We
> intended to define a set of error codes once the draft is more mature.
> Would that be sufficient?

Yeah, that sounds good.  You also have those 14 unused bytes to play 
with; the server can just blat some ASCII text into those bytes.

Cheers,

Bruce.






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To: stanislav shalunov <shalunov@internet2.edu>,
        "Bruce A. Mah" <bmah@cisco.com>
Cc: Matthew Zekauskas <matt@advanced.org>
Subject: Re: draft-ietf-ippm-owdp-01.txt comments
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Bruce,

Let me add my thanks to you for your careful reading and thoughtful
suggestions. 

Stas,

I agreed with all your responses to Bruce, with the possible exception of your
response to issue 2 (finding message boundaries in the TCP byte stream). Let
me make sure I'm understanding. I think what you are saying is that we don't
need a length field or some sort of message delimiter, because:

  1) Each message type is of fixed size;

  2) Each command begins with a "Command-Type" octet;

  3) Command response messages, though lacking a "Command-Type" octet happen
in lock-step with the commands;

Is this correct? I think a state machine diagram might help make the
lock-step-ness easier to see. I'll try to draw one.

PS: Matt and I don't see you on the plane and have concluded that you missed
the flight; hope you are OK and find your way to Ann Arbor OK.

stanislav shalunov wrote:
> 
> bmah@cisco.com (Bruce A. Mah) writes:
> 
> > I had a few comments about the latest incarnation of the OWDP draft.
> > Hope these will be helpful, or at least entertaining:
> 
> Bruce,
> 
> Thank you very much for high-quality comments!
> 
> I'll incorporate fixes before -02 version, which will reflect IPPM
> session discussion, goes out.
> 
> Ben, Matt,
> 
> I would make server drop connection after a possible timeout in (3)
> and do nothing about (5a) [rationale below]; we can discuss it on the
> plane if you can think of something smarter to do.
> 
> > 1.  In section 4.1, some mention is made of the "modes octet".  It looks to
> > me, however, like the modes field in the connection setup message is
> > really 16 bits, or two octets.  Thus, the "unused" space is really 14
> > octets, not 15.
> 
> Thanks for noticing it.  Initially, I made Modes one octet wide, then
> decided to add more space for possible future expansion.  Someone
> suggested that 13 spare modes might not be enough, so I'll increase it
> even further (unused space is there anyway).
> 
> > 2.  As you know, TCP is a bytestream protocol, and thus doesn't
> > guarantee atomic delivery of messages (with message boundaries
> > preserved).  This means that either messages need to be fixed sized, or
> > there needs to be some way of determining the end of each message.
> 
> The intent was to make it possible to discover the size of a message
> after reading (and, possibly, decrypting) first 16 octets.  Each
> message type has a specific size.
> 
> > However, it's not clear, for example, how long a Request-Session (4.3)
> > message is...how long is the zero padding field at the end?
> 
> Those three lines attempt to say it's 12 octets.  I guess we should
> say it explicitly everywhere.
> 
> > 3.  What is an implementation supposed to do in the case of partial
> > delivery of a control message (i.e. sender failure)?  I didn't see this
> > specified.
> 
> We need to specify a timeout, I suppose.  Any suggestions of good,
> currently fashionable values?
> 
> > 4.  A number of fields are multi-byte quantities.  I didn't see any
> > discussion of byte-ordering.  I can probably assume you meant
> > network byte-order, but this really should be mentioned at least
> > once...something to the effect of, "All multi-octet quantities are
> > represented in network byte order."
> 
> OK.
> 
> > 5.  A philosophical statement:  I've become a real fan over the years of
> > control protocols that are line-oriented, and use ASCII text for their
> > messages.  It makes for greater ease of understanding and debugging.
> > Examples of protocols like this are SMTP, NNTP, FTP, and HTTP. Have you
> > thought about this?
> 
> Yes.  The trade-off was: easy way to authenticate messages and a way
> to encrypt them in a mode already standartized by IETF or be
> line-oriented.  A decision was made after some internal discussion to
> go with the former option.  Do you feel the advantages of easy
> understanding and debugging outweight CBC mode properties?
> 
> Additional advantage of simple binary protocol is that it's slightly
> easier to code.  (Of course, making reading tcpdump -x output harder
> at the same time.)
> 
> >       5a.  As a side note, I think that if either end of the protocol
> >       becomes desynchronized (like due to a faulty implementation),
> >       it'll be rather difficult to get resynchronized with respect to
> >       where the message boundaries are.
> 
> Yes...  I don't see how we could easily fix it.  On the other hand,
> there are only four commands, each having a fixed size.  Each of the
> commands is mandatory and will be used even in simple scenarios.
> I would expect an implementation that desynchronizes sometimes to
> desynchronize during almost every session (and the bug would thus be
> noticed and fixed).
> 
> It's not clear to me what would one do if desynchronization happens,
> and one later manages to resynchronize.  State would not be known...
> (Do we say "Stop-Sessions" or "Retrieve-Session"?)
> 
> > 6.  Back to the connection setup in 4.1, I note that there isn't a mode
> > *bit* (you used the term "mode value") for unencrypted transfers.  Was
> > this an oversight?
> 
> In unauthenticated mode nothing at all is encrypted.  Or did you mean
> something else?
> 
> > Also I note that according to the protocol as defined, it's possible
> > for the client to respond with a contradictory set of mode
> > bits...presumably the server should drop the connection or do some
> > similarly drastic thing if this happens.
> 
> The client actually MUST respond with no more than one bit set in Mode
> field.  This needs to be clarified in the draft (oversight).
> 
> > 7.  The server's second message in connection setup (does it have a
> > name?) has a field called "Yes/No".  I suggest a more descriptive
> > name.
> 
> "Accept" was used in similar situations elsewhere.  Sounds better?
> 
> > It also might be useful to have a way for a server to communicate to a
> > client why an authentication was rejected.  (This gets a lot easier if
> > you buy into #5 above.)
> 
> A negative response is any non-zero value, here and elsewhere.  We
> intended to define a set of error codes once the draft is more mature.
> Would that be sufficient?
> 
> --
> Stanislav Shalunov <shalunov@internet2.edu>     Internet Engineer, Internet2
> 
> A large number of installed systems work by fiat.  That is, they work
> by being declared to work.                             -- Anatol Holt

-- 

                         ,,,

                        `o-o-  
                          <    Benjamin Teitelbaum
                           -   Advanced Network & Services, Inc.
                           .   Internet2





From matt@advanced.org  Fri Dec 15 08:07:51 2000
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To: "Bruce A. Mah" <bmah@cisco.com>
Cc: ippm@advanced.org
Subject: Re: draft-ietf-ippm-owdp-01.txt comments
References: <200012140530.eBE5UNw59574@bmah-freebsd-0.cisco.com> <871yvbxrra.fsf@cain.internet2.edu> <200012141702.eBEH2im63987@bmah-freebsd-0.cisco.com>
From: stanislav shalunov <shalunov@internet2.edu>
Date: 14 Dec 2000 14:58:50 -0500
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bmah@cisco.com (Bruce A. Mah) writes:

> Upon reading the draft a little more closely, I now realize that the
> encryption parameters are meant to apply to subsequent OWDP-Control
> messages, where I originally thought they applied to the OWDP-Test
> packets.

Encryption parameters are applied to both OWDP-Control and OWDP-Test.

> This raises another question in my mind...what about just using (for
> example) SSL or some other more established encryption layer to
> encrypt the entire contents of the TCP connection?  That should
> simplify both the protocol and its implementation at least a little
> bit (code reuse and all that).

Several reasons:

* OWDP-Test needs a way to encrypt/authenticate messages anyway.  Once
  you have your own encryption in OWDP-Test, why not use it in
  OWDP-Control and depend on just one external library (rather than
  two: AES + SSL).

* SSL doesn't authenticate individual messages.  If one highjacks its
  TCP connection and modifies some bits, all subsequent bits will
  decypher to garbage, but there's no clear way to check for this.
  With knowledge of message boundaries of underlying protocol one can
  authenticate each message.

* SSL is complex and solves slightly different problem.
  OWDP-Control's mechanisms are light-weight.  You'd have about the
  same amount of security-related code in both cases (possibly decrypt
  after read, possibly encrypt before write), but AES library would be
  two orders of magnitude smaller than available SSL libraries.

> I generally assume when I'm reading these things that every diagram is
> "not to scale".  :-)

Explicitly specified size of all long fields everywhere.

> Hmmm.  The only values that come to mind are those that sendmail 
> uses for SMTP conversations.  They're highly configurable but I think 
> they're on the order of minutes.  I can't remember if SMTP specifies 
> exact values or not.

Specified 30 minutes for now.  It's more than what most TCPs would
take to timeout on the other end (if the problem is with network
connection).

> IETF standardized AES in CBC mode?

Not AES, but CBC itself is explained in an IPSec RFC.  (I'm
disconnected right now, can't provide reference.)

> (Wait, in the cases where you're encrypting, isn't tcpdump -x going
> to be real hard to read anyways?)

That's right.  Additional reason why it doesn't matter much if it's
block- or line-oriented.  And, since block-oriented protocol has these
other advantages...

> What I meant was that there's a bit that means "will accept 
> unauthenticated", a bit that means "will accept authenticated", and a 
> bit that means "will accept encrypted".  My thinking is colored by 
> using PGP for many years; in that system, encryption and authentication 
> are orthogonal as you probably know.  So I was thinking that there 
> should be a bit that means "will accept unencrypted", for completeness. 
> It sounds from your comment that this is not necessary.

Authenticating in some way and then having a cleartext connection
doesn't help against active man-in-the-middle attacks.  Since AES
encrypting is very cheap, I don't see a reason to have such a mode.

> Yeah, that sounds good.  You also have those 14 unused bytes to play 
> with; the server can just blat some ASCII text into those bytes.

I thought about it.  Until now, there wasn't anything in the protocol
that was intended for humans (everything is for computers).  I feel
that having descriptive error codes would be more convenient for
appliances and automated tests.  If there's a place to put a human
explanation, the implementer could be pushed towards using a more
generic error code plus a verbal description of the error condition.

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

"Computer Science is no more about computers than astronomy is about
telescopes."					-- E. W. Dijkstra



From matt@advanced.org  Fri Dec 15 16:30:22 2000
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From: Grotefeld Glenn-cecl03 <G.Grotefeld@motorola.com>
To: "'vilho.raisanen@nokia.com'" <vilho.raisanen@nokia.com>, ippm@advanced.org
Cc: matt@advanced.org, kaeo@merike.com
Subject: RE: npmps-04
Date: Fri, 15 Dec 2000 15:25:13 -0600
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Vilho,

It appears that your text REPLACE command played a trick on you.  In a number of places (I won't say every case), where you intended to update the expiration from "May 2001" to "June 2001", it also replaced " may" in the text with "June".

I leave it to our chairs to inform us whether we should be using the official IETF file identifiers in a version that is circulated directly rather than sent to the Internet drafts sited and then posted and announced.  Do we need to go to -05 for the next version?

Glenn Grotefeld

Motorola, Inc.
g.grotefeld@motorola.com
US  847 576-5992
      847 538-7455  FAX


~-----Original Message-----
~From: vilho.raisanen@nokia.com [mailto:vilho.raisanen@nokia.com]
~Sent: Tuesday, December 12, 2000 10:07 PM
~To: ippm@advanced.org
~Cc: matt@advanced.org; kaeo@merike.com
~Subject: npmps-04
~
~
~Hello,
~
~please find attached my (intended) slides, the results 
~document, as well as
~the v-04 of the npmps draft. Lacking my slides, I forgot the 
~IP version from
~the list of additions to the -03 version of the metric. 
~Moreover, I had read
~Glenn's answer to Al sloppily, which I apologize. I now 
~realize that Glenn
~was not in favour of including error of incT into the actual metric.
~
~After some thought, I came to the conclusion that I support 
~Glenn's proposal
~of calling incT a nominal duration of the inter-packet interval and
~mentioning the possiblility of error in discussion later. If process
~scheduling or some other reason causes marked variations in incT, this
~should be treated as a deviation from the norm rather than a part of a
~normal measurement arrangement.
~
~The only changes in v-04 to v-03 are as follows:
~
~- changed definition of incT at p. 8 into
~
~   +  incT, nominal duration of inter-packet interval
~
~- added text on incT error into Sec. 4.6:
~
~   +  Error due to variation of incT
~
~- For clarity, I added a mention of packet loss as the second 
~sentence to to
~Sec 4.4, which now reads
~
~   The sample metric thus defined is intended to probe the delays and
~   the delay variation as experienced by multimedia streams of
~   an application. Due to the definition of the metric, also 
~packet loss
~   status of packets is recorded. The delay is assumed to be 
~measured at
~   transport layer level. Since a range of packet sizes and nominal 
~   interval between packets is used, the method probes only a specific 
~   time scale of network QoS variations.
~
~For those intested in a computer repair story with a human 
~interest tint,
~I'll attach one at the end of the message. For the rest, just 
~read the docs
~& the slides and you'll do fine.
~
~	BR,
~	   Vilho
~
~%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
~ Vilho Räisänen                           
~ vilho.raisanen@nokia.com                 
~ Senior research engineer                 
~ Nokia Research Center, Helsinki, Finland 
~%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
~
~ * * * end of real business * * *
~
~~~ How to boot up a stubborn computer - a small tale of 
~fatherly wisdom ~~
~
~At boot, my laptop reported that it was the modem driver that 
~caused the
~crash. From previous experience, I knew that the order in which device
~drivers are loaded sometimes makes a difference, at least in 
~some operating
~systems. Armed with this piece of what I thought of as 
~profound knowledge, I
~booted NT with and without my PCMCIA NIC, without CD-ROM drive and
~experimenting with all combinations. I even tried the VGA 
~mode. To no avail.
~
~At this point I remembered my father's tale of a dysfunctional 
~old radio. He
~told me having opened the casing, and - lacking training in 
~electronics -
~could not do very much more. So he just blew some dust off the 
~valves and
~closed it again. Surprise, surprise: now it worked.
~
~Lacking better ideas, I did just the same: I located the 
~modem, opened a
~tiny cover and blew some imaginary dust off the printed board. 
~Reassembly,
~boot - succcess.
~
~



From matt@advanced.org  Mon Dec 18 16:19:43 2000
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To: ippm@advanced.org
Subject: draft-ietf-ippm-owdp: should it be split?
From: stanislav shalunov <shalunov@internet2.edu>
Date: 18 Dec 2000 16:08:52 -0500
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There was a suggestion made during physical IPPM WG meeting that the
draft be split into two: one describing OWDP-Control, the other one
describing OWDP-Test.

What is everybody's opinion on this suggestion?

NOTE:  Split or not split, you *can* use OWDP-Test without OWDP-Control.

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

A fanatic is one who can't change his mind and won't change the
subject.                                   -- Winston Churchill



From matt@advanced.org  Mon Dec 18 17:06:56 2000
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To: ippm@advanced.org
Subject: draft-ietf-ippm-owdp: arbitrary packets?
From: stanislav shalunov <shalunov@internet2.edu>
Date: 18 Dec 2000 17:01:15 -0500
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There was a suggestion made during physical IPPM WG meeting that the
protocol should be modified in such a way as to allow specification of
arbitrary packet format, along with an indication of where within
these packets the actual timestamp is located.

This would address Type-P issue in a rather general way.  (But not
completely, since one can imagine situations where advance
reservations via a bandwidth broker or RSVP would change the results
of a measurement.)

This suggestion seems appealing, however, there are certain
problematic areas that need to be addressed if we want to implement
it.

* Many OWDP-Test speakers will be Unix or other general purpose
  computers.  These will inevitably have higher losses when listening
  to raw network traffic.  Raw sockets will induce higher loss rate
  than one would have with UDP measurements.

* How would one distinguish between a situation where packets aren't
  supposed to be routed to the receiver and situation of 100% loss?

* Carefully crafted packets could cause disruption to some link-layer
  protocols.  How would an implementer decide what to allow so as not
  to increase risks?

* Suppose an identity of an authenticated user becomes compromised.
  Now the attacker could use that to run TCP sessions to rlogin port
  of machines around servers that trust this user to perform
  measurements (or, less drastically, to send spam from that network).
  Ability to perform measurements is transformed into ability to
  generate arbitrary traffic on behalf of all the senders an
  OWDP-Control server controls.

* How do we specify criteria for listening to traffic?  Standartizing
  tcpdump syntax is hardly an option, but what else do we have?

Thoughts?  Should we just stick with UDP plus possibly a few options
for things like PHB/DSCP, TTL, DF, etc.?

-- 
Stanislav Shalunov <shalunov@internet2.edu>	Internet Engineer, Internet2

Beware of Programmers who carry screwdrivers.    -- Leonard Brandwein



From matt@advanced.org  Tue Dec 19 15:18:07 2000
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To: "'matt@advanced.org'" <matt@advanced.org>,
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Antigen for Exchange found Navidad.exe infected with W32/Navidad virus.
The file is currently Deleted.  The message, "Revision to loss pattern
draft: Update to author contact info", was
sent from Ing. IDJ van der Ven  and was discovered in IMC Queues\Inbound
located at Netcom Systems/NETCOMSYSTEMS/EXCHANGE1.

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From matt@advanced.org  Tue Dec 19 15:35:10 2000
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From: "Matthew J Zekauskas" <matt@advanced.org>
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Subject: [ADMIN] virus warnings from today
Date: Tue, 19 Dec 2000 15:36:47 -0500
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These were stuck in some long-dormant sendmail queue.  Sendmail was
restarted... and they popped out.  As far as I know, there has been
no new propogation.

--Matt



From matt@advanced.org  Wed Dec 20 07:29:33 2000
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the IP Performance Metrics Working Group of the IETF.

	Title		: A One-way Delay Measurement Protocol
	Author(s)	: S. Shalunov, B. Teitelbaum, M. Zekauskas
	Filename	: draft-ietf-ippm-owdp-01.txt
	Pages		: 20
	Date		: 19-Dec-00
	
The IETF IP Performance Metrics (IPPM) working group has proposed
draft standard metrics for one-way packet delay [RFC2679] and loss
[RFC 2680] across Internet paths.  Although there are now several
measurement platforms that implement collection of these metrics
[SURVEYOR], [RIPE], there is to date no standard that would permit
initiation of test streams or exchange of packets to collect
singleton metrics in an interoperable manner.

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From matt@advanced.org  Thu Dec 21 07:13:29 2000
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the IP Performance Metrics Working Group of the IETF.

	Title		: A Framework for Defining Empirical Bulk Transfer 
                          Capacity Metrics
	Author(s)	: M. Mathis, M. Allman
	Filename	: draft-ietf-ippm-btc-framework-04.txt
	Pages		: 9
	Date		: 20-Dec-00
	
Bulk Transport Capacity (BTC) is a measure of a network's ability to
transfer significant quantities of data with a single
congestion-aware transport connection (e.g., TCP).  The intuitive
definition of BTC is the expected long term average data rate (bits
per second) of a single ideal TCP implementation over the path in
question.  However, there are many congestion control algorithms
(and hence transport implementations) permitted by IETF standards.
This diversity in transport algorithms creates a difficulty for
standardizing BTC metrics because the allowed diversity is
sufficient to lead to situations where different implementations
will yield non-comparable measures -- and potentially fail the
formal tests for being a metric.
This document defines a framework for standardizing multiple BTC
metrics that parallel the permitted transport diversity.  Two
approaches are used.  First, each BTC metric must be much more
tightly specified than the typical IETF protocol.  Pseudo-code or
reference implementations are expected to be the norm.  Second, each
BTC methodology is expected to collect some ancillary metrics which
are potentially useful to support analytical models of BTC.

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From matt@advanced.org  Fri Dec 22 05:52:20 2000
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Dear IPPMers,

please find attached the version -04bis of the npmps draft. Thanks to =
Al &
Glenn for comments. The following changes to -04 have been made:

- I got rid of the the spurious month names due to Emacs' =
case-insensitive
search & replace algorithm. Indeed, these JUNE have caused some =
confusion.
- I implemented Al's refinement into 4.6. and added some more text into =
incT
error bullet pt. The section now reads

   The description of any specific measurement method should include an
   accounting and analysis of various sources of error or uncertainty.
   The Framework document [1] provides general guidance on this point,=20
   but we note here the following specifics related to periodic
   streams and delay metrics:

   +  Error due to variation of incT. The reasons for this can be e.g.
      uneven process scheduling, possibly due to CPU load.
   +  Errors or uncertainties due to uncertainties in the clocks of the
      MP(Src) and MP(Dst) measurement points.
   +  Errors or uncertainties due to the difference between 'wire time'
      and 'host time'.

Earlier during the e-mail thread, Al commented that also real sources =
of
periodic streams may (and do) exhibit variation in transmission times. =
This
is true, but incorporation of intentional deviations from periodic =
behaviour
into the actual measurement metrics would result in a quite complex
construct. Basically, a (public domain) source should be cited for
definitive research results on "typical" variations due to O/S or
application. This should then be mapped into a corresponding metrics
description and so forth.

Thus, I find it a bit difficult to justify adding a mention of real
application behaviour in this context. In my view, the goal should be =
that
incT is (for practical purposes) negligible.

Season's greetings,

	Vilho

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%=20
 Vilho R=E4is=E4nen                          =20
 vilho.raisanen@nokia.com                =20
 Senior research engineer                =20
 Nokia Research Center, Helsinki, Finland=20
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%


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Network Working Group                                        V. =
Raisanen
INTERNET-DRAFT                                                     =
Nokia
Expiration Date:  June 2001                                 G. =
Grotefeld
                                                                =
Motorola
                                                           December =
2000


	    Network performance measurement for periodic streams
                       <draft-ietf-ippm-npmps-04.txt>


1. Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six =
months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at               =
=20
   http://www.ietf.org/ietf/1id-abstracts.txt                           =
=20

   The list of Internet-Draft shadow directories can be accessed at     =
=20
   http://www.ietf.org/shadow.html

   This memo provides information for the Internet community. This
   memo does not specify an Internet standard of any
   kind. Distribution of this memo is unlimited.


2. Abstract

   This document describes a sample metric suitable for application-
   level IP network transport measurement for periodic streams, such as
   VoIP or streaming multimedia over IP. In this document,  the reader
   is assumed to be familiar with the terminology of the  Framework for
   IP Performance Metrics RFC 2330 [1]. This document is parallel to
   A One-way Delay Metric for IPPM RFC 2679 [2]. Although this document
   is based on the delay metrics, other characteristics can be measured
   with this approach, too. For example, packet loss rate, reordering /
   out-of sequence, and successive delay variation are all additional
   metrics which can be built from this baseline set of measurements.




Raisanen, Grotefeld           expires     June  2001              [Page =
1]=0C
Internet Draft         <draft-ietf-ippm-npmps-04.txt>       December =
2000


3. Introduction

   This document discusses concepts relevant to  application-level
   performance measurements of an IP network. The original driver for
   this work is Quality of Service of interactive periodic streams such
   as multimedia conference over IP, but the idea of application-level
   measurement may have a wider scope. In the following, interactive
   multimedia traffic is used as an example to illustrate the concept.

   A constant bit-rate (CBR), or nearly CBR, streaming (hereinafter
   called periodic) multimedia bit stream may be simulated by
   transmitting uniformly sized packets (or mostly uniformly sized
   packets) at regular intervals through the network to be evaluated.
   The "mostly uniformly sized packets" may be found in applications
   that may use smaller packets during a portion of the stream (e.g.
   digitally coded voice during silence periods). As noted in the
   framework document [1], a sample metric  using regularly spaced
   singleton tests has some limitations when considered from a
   general measurement point of view: only part of the network
   performance spectrum is sampled. However, from the point of view of
   application-level performance, this is actually good news as
   explained below.

   IP delivery service measurements have been discussed within the
   International Telecommunications Union (ITU). A framework for IP
   service level measurements (with references to the framework for IP
   performance [1]) that is intended to be suitable for service =
planning
   has been approved as I.380 [3]. The emphasis in the ITU
   recommendation is on passive measurements, though not explicitly
   forbidding active measurements. The present contribution proposes a
   method that is usable both for service planning and end-user testing
   purposes, and is based on active measurements.

3.1 Terminology

   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 [4].
   Although RFC 2119 was written with protocols in mind, the key words
   are used in this document for similar reasons.  They are used to
   ensure the results of measurements from two different =
implementations
   are comparable, and to note instances when an implementation could
   perturb the network.







Raisanen, Grotefeld           expires     June  2001              [Page =
2]=0C
Internet Draft         <draft-ietf-ippm-npmps-04.txt>       December =
2000


3.2 Considerations related to delay

   For interactive multimedia sessions, end-to-end delay is an
   important factor. Too large a delay reduces the quality of the
   multimedia session as perceived by the participants. One approach =
for
   managing end-to-end delays on an Internet path involving
   heterogeneous link layer technologies is to use per-domain delay
   quotas (e.g. 50 ms for a particular IP domain). The 50 ms would
   then be included into a calculation of an end-to-end delay bound. A
   practical implementation of such as scheme ought to address issues
   like possibility of asymmetric delays in a route in different
   directions, and sensitivity of an application to delay variations in
   a given domain. There are several alternatives as to which kind of
   derivative delay metric one ought to use in managing end-to-end QoS.
   This question, although very interesting, is not within the scope of
   this draft and is not discussed further here.

   In the following, a methodology and metric are presented for
   measuring media stream transport QoS in an IP domain. The
   measurement results may be used in derivative metrics such as
   average and maximum delays. A metric is presented that is a standard
   way for performing a measurement irrespective of the possible QoS
   mechanism utilized in the core network. As an example, for a QoS
   mechanism without hard guarantees, measurements may be used to
   ascertain that the "best" class gets the service that has been
   promised for the traffic class in question. Moreover, an operator
   could study the quality of a cheap, low-guarantee service
   implemented using possible slack bandwidth in other classes. Such
   measurements could be made either in studying the feasibility of a
   new service, or on a regular basis.

   The present draft seeks to formalize the measurements in such a way
   that interoperable results are achieved.

3.3 Protocol level issues

   The version of the Internet Protocol used in the measurement affects
   (at least) packet sizes, and should be reported.  =20

   Fig.1 illustrates measurements on multiple protocol levels that
   are relevant to this draft. The major focus of the present draft
   is on transport quality evaluation from application point of
   view. However, to properly account for quality effects of, e.g.,
   operating system and codec on packet voice, it is beneficial to be
   able to measure quality at IP level [5]. Link layer monitoring
   provides a way of accounting for link layer characteristics such
   as bit error rates.



Raisanen, Grotefeld           expires     June  2001              [Page =
3]=0C
Internet Draft         <draft-ietf-ippm-npmps-04.txt>       December =
2000

     ---------------
     | application |
     ---------------
     |  transport  | <--
     ---------------
     |   network   | <--
     ---------------
     |    link     | <--
     ---------------
     |   physical  |
     ---------------=20

   Fig. 1: Different possibilities for performing measurements: a
   protocol view. Above, "application" refers to all layers above
   L4 and is not used in the OSI sense.

   In general, the results of measurements may be influenced by
   individual application requirements/responses related to the
   following issues:

   +  Lost packets: Applications may have varying tolerance to lost=20
      packets.  Another consideration is the distribution of lost
      packets (i.e. random or bursty).
   +  Long delays: Many applications will consider packets delayed=20
      longer than a certain value to be equivalent to lost packets
      (i.e. real time applications).
   +  Duplicate packets: Some applications may be perturbed if=20
      duplicate packets are received.
   +  Out-of-sequence: Some applications may be perturbed if packets
      are received out of sequence. This may be in addition to the
      possibility of exceeding the "long" delay threshold as a result
      of being out of sequence. An out-of-sequence packet outcome
      occurs when a single IP packet received at a DST measurement
      point has a sequence number  higher than that which is
      expected, and therefore, the packet is OOS due to re-ordering.=20
   +  Corrupt packet header: Most applications will probably treat a
      packet with a corrupt header as equivalent to a lost packet.
   +  Corrupt packet payload: Some applications (e.g. digital voice=20
      codecs) may accept corrupt packet payload.  In some cases, the
      packet payload may contain application specific forward error
      correction (FEC) that can compensate for some level of=20
      corruption.
   +  Spurious packet: Dst may receive spurious packets (i.e. packets
      that are not sent by the Src as part of the metric).  Many
      applications may be perturbed by spurious packets.

   Depending, e.g., on the observed protocol level, some issues listed
   above may be indistinguishable from others by the application, it
   may be important to preserve the distinction for the operators of
   Src, Dst, and/or the intermediate network(s).

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   Because of the possible errors listed above, in most cases it is=20
   recommended to use a packet identifier for each packet generated at
   Src. Identifiers for the metric sample may be those used by the
   underlying transport layer (e.g. RTP sequence number) or the same=20
   identifiers used by an application if the application to be modeled=20
   by the metric uses an identifier. The possibility of identifier=20
   roll-over (reuse if intentional) during a metric collected over
   a "long" (application dependent) time should be observed.

   If the application does not use an identifier, it may still be=20
   useful to add identifiers to the packets in the metric sample to=20
   help identify possible anomalies such as out of sequence packets.
   This would be most useful in the case where the application=20
   expects to receive packets in sequence, but has no capability to
   identify the sequence of packets received at Dst.

3.4 Application-level measurement

   In what follows, a metric is proposed for application-level network
   performance measurement. In effect, the metric is an emulation of
   periodic multimedia stream performance. The justification for using
   realistic application metrics in the measurement:

   +  The results of the measurement are automatically relevant to the
      performance as perceived by the application in question.
   +  All the packets in the measurement contribute to accuracy of the
      estimation of performance variation at timescale that is
      important to the multimedia application (packetization
      interval).
   +  Effects of elastic traffic (TCP) on measurement packets are
      different for a sustained stream than for single packets during
      overloading situations as discussed in [3].

3.5 Measurement types

   Delay measurements can be one-way [2,3], paired one-way, or=20
   round-trip [6]. Accordingly, the measurements may be performed
   either with synchronized or unsynchronized Src/Dst host clocks.
   Different possibilities are listed below.

   The reference measurement setup for all measurement types is
   shown in Fig. 2.








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     ----------------< IP >--------------------
     |          |                  |          |
   -------   -------           --------    --------
   | Src |   | MP  |           | MP   |    | Dst  |
   -------   |(Src)|           |(Dst) |    --------
             -------           --------   =20

   Fig. 2: Example setup for the metric usage.

   An example of the use of the metric is a setup with a source host
   (Src), a destination host (Dst), and corresponding measurement
   points (MP(Src) and MP(Dst)) as shown in Figure 2. Separate =
equipment
   for measurement points may be used if having Src and/or Dst conduct=20
   the measurement may significantly affect the delay performance to be
   measured. MP(Src)should be placed/measured close to the egress point =

   of packets from Src. MP(Dst) should be placed/measure close to=20
   the ingress point of packets for Dst. "Close" is defined as a
   distance sufficiently small so that application-level performance
   characteristics measured (such as delay) can be expected to follow=20
   the corresponding performance characteristic between Src and Dst to=20
   an adequate accuracy. Basic principle here is that measurement
   results between MP(Src) and MP(Dst) should be the same as for a
   measurement between Src and Dst, within the general error margin
   target of the measurement (e.g., < 1 ms; number of lost packets is
   the same). If this is not possible, the difference between MP-MP
   measurement and Src-Dst measurement should preferably be systematic.
=20
   The test setup just described fulfills two important criteria:
   1) Test is made with realistic stream metrics, emulating - for =
example -
   a full-duplex Voice over IP (VoIP) call.
   2) Either one-way or round-trip characteristics may be obtained.

   It is also possible to have intermediate measurement points between=20
   MP(Src) and MP(Dst), but that is beyond the scope of this document.

3.5.1 One way measurement

   In the interests of specifying metrics that are as generally usable
   as possible, application-level measurements based on one-way delays
   are used in the example metrics. The implication of =
application-level
   measurement for bi-directional applications such as interactive
   multimedia conferencing is discussed below.

   Performing a single one-way measurement only yields information on
   network behavior in one direction. Moreover, the stream at the
   network transport level does not emulate accurately a full-duplex
   multimedia connection.



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3.5.2 Paired one way measurement

   Paired one way delay refers to two multimedia streams: Src to Dst=20
   and Dst to Src for the same Src and Dst. By way of example, for
   some applications, the delay performance of each one way path is
   more important than the  round trip delay. This is the case for
   delay-limited signals such as VoIP. Possible reasons for the
   difference between one-way delays is different routing of streams
   from Src to Dst vs. Dst to Src.

   For example, a paired one way measurement may show that Src to Dst
   has an average delay of 30ms while Dst to Src has an average delay
   of 120ms. To a round trip delay measurement, this example would
   look like an average of 150ms delay.  Without the knowledge of the
   asymmetry, we might miss a problem that the application at either
   end may have with delays averaging more than 100ms.

   Moreover, paired one way delay measurement emulates a full-duplex
   VoIP call more accurately than a single one-way measurement only.

3.5.3 Round trip measurement

   From the point of view of periodic multimedia streams,
   round-trip measurements have two advantages: they avoid the need of
   host clock synchronization  and they allow for a simulation of
   full-duplex connections. The former aspect means that a measurement
   is easily performed, since no special equipment or NTP setup is
   needed. The latter property means that measurement streams are
   transmitted in both directions. Thus, the measurement provides
   information on quality of service as experienced by appropriate
   application.

   The downsides of round-trip measurement are the need for more
   bandwidth than an one-way test and more complex accounting of
   packet loss. Moreover, the stream that is returning towards the
   original sender may be more bursty than the one on the first "leg" =
of
   the round-trip journey. The last issue, however, means in practice
   that returning stream experiences worse QoS than the other one, and
   the performance estimates thus obtained are pessimistic ones. The
   possibility of asymmetric routing and queuing must be taken into
   account during analysis of the results.

   Please note that with suitable arrangements, round-trip measurements
   may be performed using paired one way measurements.






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4 Sample metric for multimedia stream simulation

   The sample metric presented here is similar to the sample metric
   Type-P-One-way-Delay-Poisson-Stream presented in [2]. "Singletons", =
as
   defined in [1] and [2] are not directly used in this document =
because
   certain key results (such as duplicate or out of sequence packets)
   cannot be identified in the context of a singleton, but only as part
   of a sample.

4.1 Metric name
  =20
   Type-P-One-way-Delay-Periodic-Stream

4.2 Metric parameters

4.2.1 Global metric parameters

   These parameters are applicable to the metrics collected in the=20
   following sections (4.2.2, 4.2.3, and 4.2.4).

   +  Src, the IP address of a host
   +  Dst, the IP address of a host
   +  IPV, the IP version (IPv4/IPv6) used in the measurement
   +  T0, a time, for starting to generate packets and taking=20
         measurements for a sample
   +  Tf, a time, greater than T0, for stopping generation of packets
         for a sample
   +  incT, nominal duration of inter-packet interval
   +  packet size p(j), the number of bytes in each packet of Type-P of =

         size j
   +  dTloss, a time interval, used for determining if a packet should
         be considered lost
   +  Tcons, a time interval [optional]

      While a number of applications will use one packet size (j =3D =
1),
      other applications may use packets of different sizes (j > 1).=20
      Especially in cases of congestion, it may be useful to have=20
      packets smaller than the maximum or predominant size of packets
      in the periodic stream.

4.2.2 Metrics collected at MP(Src)

   +  Tstamp(Src)[i], for each packet [i], the time of the packet as
      measured at MP(Src)
   +  PktID [i], for each packet [i], an identification number for the
      the packet sent from Src to Dst.




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   +  PktSiTy [i], for each packet [i], the packet size and/or type.
      Some applications may use packets of different size, either=20
      because of application requirements or in response to IP=20
      performance experienced.

4.2.3 Metrics collected at MP (Dst)

   +  dTstop, a time interval, used to add to time Tf to determine when =
to
         stop collecting metrics for a sample
   +  Tstamp(Dst)[i], for each packet [i], the time of the packet as
      measured at MP(Dst)
   +  PktID [i], for each packet [i], an identification number for the
      the packet received at Dst from Src.
   +  PktSiTy [i], for each packet [i], the packet size and/or type.
      Some applications may use packets of different size, either=20
      because of application requirements or in response to IP=20
      performance experienced.
   +  PktStatus [i], for each packet [i], the status of the packet
      received.  Possible status includes: OK, packet header corrupt,
      packet payload corrupt, spurious, duplicate, out-of-sequence.

4.2.4 Metrics resulting when metrics collected at MP(Src) and MP(Dst)
      are merged

   These parameters are only available as a complete set when the=20
   parameters from the preceding sections (4.2.1, 4.2.2, and 4.2.3 are=20
   combined. =20

   +  Tstamp(Src)[i], for each packet [i], the time of the packet as
      measured at MP(Src).  This entry may be blank or noted as N/A
      for spurious packets received at MP(Dst)
   +  Tstamp(Dst)[i], for each packet [i], the time of the packet as
      measured at MP(Dst).  This entry may be blank or noted as N/A
      for packets not received at MP(Dst), received with corrupt=20
      packet headers, or for duplicate packets received at MP(Dst).
   +  PktID [i], for each packet [i], an identification number for the
      the packet received.  This identification number may be corrupted
      for certain packets received at MP (Dst).
   +  PktSiTy [i], for each packet [i], the packet size and/or type.
   +  PktStatus [i], for each packet [i], the status of the packet=20
      received.  Possible status includes: OK, packet header corrupt,
      packet payload corrupt, spurious, duplicate, out-of-sequence.









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   +  Delay [i], for each packet [i], the time interval Tstamp(Dst)[i] =
-=20
      Tstamp(Src)[i].  For the following conditions, it will not be=20
      possible to be able to compute delay:
         Spurious: There will be no Tstamp(Src)[i] time
         Not received: There will be no Tstamp (Dst) [i]
         Corrupt packet header: There will be no Tstamp (Dst) [i]
         Duplicate:  Only the first non-corrupt copy of the packet=20
         received at  Dst should have Delay [i] computed.
   +  SDV[i] [optional] , for each packet [i] except the first one:=20
         momentary delay variation between successive packets, i.e., =
the=20
         time interval Delay[i] - Delay [i-1].  SDV[i] may be negative, =

         zero, or positive. Delay for both packets i and i+1 must be=20
         calculable according to the definition above or SDV[i] is=20
         undefined.

4.3 High level description of the procedure to collect a sample

   Beginning on or after time T0, Type-P packets are generated=20
   by Src and sent to Dst until time Tf is reached with a nominal=20
   interval between the first bit of successive packets of incT as=20
   measured at MP(Src).  incT may be nominal due to a number of =
reasons:
   variation in packet generation at Src, clock issues (see section =
4.6),
   etc.

   MP(Src) records the following information only for packets with=20
   timestamps between and including T0 and Tf: timestamp,=20
   packet identifier, and packet size/type of each packet sent from Src =

   to Dst that is part of the sample. =20

   MP (Dst) records the following information only for packets with=20
   time stamps between T0 and (Tf+ dTstop): timestamp, packet =
identifier,=20
   packet size/type, and received status of each packet received from=20
   Src at Dst that is part of the sample.  Optionally, at a time Tf +=20
   Tcons, the data from MP(Src) and MP(Dst) are consolidated to derive=20
   the results of the sample metric.=20

   To prevent stopping data collection too soon, dTcons should be =
greater
   than or equal to dTstop.  Conversely, to keep data collection=20
   reasonably efficient, dTstop should be some reasonable time interval =

   (seconds/minutes/hours), even if dTloss is infinite or extremely =
long.









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4.4 Discussion

   The sample metric thus defined is intended to probe the delays and
   the delay variation as experienced by multimedia streams of
   an application. Due to the definition of the metric, also packet =
loss
   status of packets is recorded. The delay is assumed to be measured =
at
   transport layer level. Since a range of packet sizes and nominal=20
   interval between packets is used, the method probes only a specific=20
   time scale of network QoS variations.

   There are a number of factors that should be taken into account when =

   collecting a sample metric of Type-P-One-way-Delay-Periodic-Stream.

   +  T0 and (Tf + dTloss) should specify a long enough time interval =
to=20
      represent a reasonable use of the application under test (e.g. do =

      not provide only a 100 ms time interval for a phone call)

   +  T0 and (Tf + dTloss) should specify a time interval that is not=20
      excessively long compared to the usage of the application under =
test
      (e.g. do not provide a one week continuous phone call)

   +  The nominal interval between packets (incT) and the packet =
size(s)
      (p(j)) should not define an equivalent bit rate that is in excess =

      of the capacity of the egress port of Src, the ingress port of =
Dst,
      or the carrying capacity of the intervening network(s). There may
      be exceptional cases to test the response of the application to
      overload conditions in the transport networks, but these cases=20
      should be strictly controlled.

   +  Real delay values will be positive.  Therefore, it does not make
      sense to report a negative value as a real delay.  However, an
      individual zero or negative delay value might be useful as part =
of
      a stream when trying to discover a distribution of the delay =
values
      of a stream.

   +  Depending on measurement topology, delay values may be as low as
      100 usec to 10 msec, whereby it may be important for Src and Dst =
to
      synchronize very closely.  GPS systems afford one way to achieve
      synchronization to within several 10s of usec.  Ordinary =
application
      of NTP may allow synchronization to within several msec, but this
      depends on the stability and symmetry of delay properties among =
those
      NTP agents used, and this delay is what we are trying to measure. =
A
      combination of some GPS-based NTP servers and a conservatively
      designed and deployed set of other NTP servers should yield good
      results, but this is yet to be tested.

   +  Reordering of packets is best discussed in terms of the entire
      set of measurement packets received, i.e. should be addressed in
      Sec. 4.9.1.

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   +  A given methodology will have to include a way to determine
      whether packet was lost or whether delay is merely very large =
(and
      the packet is yet to arrive at Dst). The global metric parameter
      dTloss defines a time interval such that delays larger than =
dTloss
      are interpreted as losses.=20
      {Comment: Note that, for many applications of these metrics, the
      harm in treating a large delay as infinite might be zero or very
      small.  A TCP data packet, for example, that arrives only after
      several multiples of the RTT may as well have been lost.}

4.5 Additional Methodology Aspects

   As with other Type-P-* metrics, the detailed methodology will depend
   on the Type-P (e.g., protocol number, UDP/TCP port number, size,
   precedence).

4.6 Errors and uncertainties

   The description of any specific measurement method should include an
   accounting and analysis of various sources of error or uncertainty.
   The Framework document [1] provides general guidance on this point,=20
   but we note here the following specifics related to periodic
   streams and delay metrics:

   +  Error due to variation of incT. The reasons for this can be e.g.
      uneven process scheduling, possibly due to CPU load.
   +  Errors or uncertainties due to uncertainties in the clocks of the
      MP(Src) and MP(Dst) measurement points.
   +  Errors or uncertainties due to the difference between 'wire time'
      and 'host time'.

4.6.1. Errors or uncertainties related to Clocks

   The uncertainty in a measurement of one-way delay is related, in
   part, to uncertainties in the clocks of MP(Src) and MP(Dst). In
   the following, we refer to the clock used to measure when the packet
   was measured at MP(Src) as the MP(Src) clock and we refer to the=20
   clock used to measure when the packet was received at MP(Dst) as the
   MP(Dst) clock.  Alluding to the notions of synchronization, =
accuracy,
   resolution, and skew, we note the following:

   +  Any error in the synchronization between the MP(Src) clock and=20
      the MP(Dst) clock will contribute to error in the delay
      measurement.  We say that the MP(Src) clock and the MP(Dst)
      clock have a synchronization error of Tsynch if the MP(Src) clock
      is Tsynch ahead of the MP(Dst) clock.  Thus, if we know the
      value of Tsynch exactly, we could correct for clock
      synchronization by adding Tsynch to the uncorrected value of
      Tstamp(Dst)[i] - Tstamp(Src) [i].

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   +  The accuracy of a clock is important only in identifying the time
      at which a given delay was measured.  Accuracy, per se, has no
      importance to the accuracy of the measurement of delay.  When
      computing delays, we are interested only in the differences
      between clock values, not the values themselves.

   +  The resolution of a clock adds to uncertainty about any time
      measured with it.  Thus, if the MP(Src) clock has a resolution of
      10 msec, then this adds 10 msec of uncertainty to any time value
      measured with it.  We will denote the resolution of the source
      clock and the MP(Dst) clock as ResMP(Src) and ResMP(Dst),
      respectively.
   +  The skew of a clock is not so much an additional issue as it is a
      realization of the fact that Tsynch is itself a function of time.
      Thus, if we attempt to measure or to bound Tsynch, this needs to
      be done periodically.  Over some periods of time, this function
      can be approximated as a linear function plus some higher order
      terms; in these cases, one option is to use knowledge of the
      linear component to correct the clock.  Using this correction, =
the
      residual Tsynch is made smaller, but remains a source of
      uncertainty that must be accounted for.  We use the function
      Esynch(t) to denote an upper bound on the uncertainty in
      synchronization.  Thus, |Tsynch(t)| <=3D Esynch(t).

   Taking these items together, we note that naive computation=20
   Tstamp(Dst)[i] - Tstamp(Src) [i] will be off by Tsynch(t) +/-=20
   (ResMP(SRc) + ResMP(Dst)).  Using the notion of Esynch(t), we note =20
   that these clock-related problems introduce a total uncertainty of=20
   Esynch(t)+ Rsource + Rdest.  This estimate of total clock-related=20
   uncertainty should be included in the error/uncertainty analysis of=20
   any measurement implementation.

4.6.2. Errors or uncertainties related to Wire-time vs Host-time

   As we have defined one-way periodic delay, we would like to measure=20
   the time between when a packet is measured and time-stamped at
   MP(Src) and when it arrives and is time-stamped at MP(Dst) and we
   refer to these as "wire times."  If the timings are themselves
   performed by software on Src and Dst, however, then this software =
can
   only directly measure the time between when Src generates the packet
   just prior to sending the test packet and when Dst has started to
   process the packet after having received the test packet, and we =
refer
   to these two points as "host times".

   To the extent that the difference between wire time and host time is
   accurately known, this knowledge can be used to correct for wire =
time
   measurements and the corrected value more accurately estimates the
   desired (host time) metric.


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   To the extent, however, that the difference between wire time and
   host time is uncertain, this uncertainty must be accounted for in an
   analysis of a given measurement method.  We denote by Hsource an
   upper bound on the uncertainty in the difference between wire time
   of MP(Src) and host time on the Src host, and similarly define Hdest
   for the difference between the host time on the Dst host and the =
wire
   time of MP(Dst).  We then note that these problems introduce a total
   uncertainty of Hsource+Hdest.  This estimate of total wire-vs-host
   uncertainty should be included in the error/uncertainty analysis of=20
   any measurement implementation.

4.6.3. Calibration

   Generally, the measured values can be decomposed as follows:

      measured value =3D true value + systematic error + random error

   If the systematic error (the constant bias in measured values) can =
be
   determined, it can be compensated for in the reported results.

      reported value =3D measured value - systematic error

   therefore

      reported value =3D true value + random error

   The goal of calibration is to determine the systematic and random
   error generated by the instruments themselves in as much detail as
   possible.  At a minimum, a bound ("e") should be found such that the
   reported value is in the range (true value - e) to (true value + e)
   at least 95 percent of the time.  We call "e" the calibration error
   for the measurements.  It represents the degree to which the values
   produced by the measurement instrument are repeatable; that is, how
   closely an actual delay of 30 ms is reported as 30 ms.  {Comment: 95
   percent was chosen due to reasons discussed in [2], briefly
   summarized as (1) some confidence level is desirable to be able to
   remove outliers, which will be found in measuring any physical
   property; (2) a particular confidence level should be specified so
   that the results of independent implementations can be compared.}

   From the discussion in the previous two sections, the error in
   measurements could be bounded by determining all the individual
   uncertainties, and adding them together to form

       Esynch(t) + ResMP(Src) + ResMP(Dst) + Hsource + Hdest.





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   However, reasonable bounds on both the clock-related uncertainty
   captured by the first three terms and the host-related uncertainty
   captured by the last two terms should be possible by careful design
   techniques and calibrating the instruments using a known, isolated,
   network in a lab.

   For example, the clock-related uncertainties are greatly reduced
   through the use of a GPS time source.  The sum of Esynch(t) +=20
   ResMP(Src) + ResMP(Dst) is small, and is also bounded for the=20
   duration of the measurement because of the global time source.

   The host-related uncertainties, Hsource + Hdest, could be bounded by
   connecting two instruments back-to-back with a high-speed serial =
link
   or isolated LAN segment.  In this case, repeated measurements are
   measuring the same one-way delay.

   If the test packets are small, such a network connection has a
   minimal delay that may be approximated by zero.  The measured delay
   therefore contains only systematic and random error in the
   instrumentation.  The "average value" of repeated measurements is =
the
   systematic error, and the variation is the random error.

   One way to compute the systematic error, and the random error to a
   95% confidence is to repeat the experiment many times - at least
   hundreds of tests.  The systematic error would then be the median.
   The random error could then be found by removing the systematic =
error
   from the measured values.  The 95% confidence interval would be the
   range from the 2.5th percentile to the 97.5th percentile of these
   deviations from the true value.  The calibration error "e" could =
then
   be taken to be the largest absolute value of these two numbers, plus
   the clock-related uncertainty.  {Comment: as described, this bound =
is
   relatively loose since the uncertainties are added, and the absolute
   value of the largest deviation is used.  As long as the resulting
   value is not a significant fraction of the measured values, it is a
   reasonable bound.  If the resulting value is a significant fraction
   of the measured values, then more exact methods will be needed to
   compute the calibration error.}

   Note that random error is a function of measurement load.  For
   example, if many paths will be measured by one instrument, this =
might
   increase interrupts, process scheduling, and disk I/O (for example,
   recording the measurements), all of which may increase the random
   error in measured singletons.  Therefore, in addition to minimal =
load
   measurements to find the systematic error, calibration measurements
   should be performed with the same measurement load that the
   instruments will see in the field.




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   We wish to reiterate that this statistical treatment refers to the
   calibration of the instrument; it is used to "calibrate the meter
   stick" and say how well the meter stick reflects reality.

4.7 Reporting the metric

   The calibration and context in which the metric is measured MUST be
   carefully considered, and SHOULD always be reported along with =
metric
   results.  We now present five items to consider: the Type-P of test
   packets, the threshold of delay equivalent to loss, error
   calibration, the path traversed by the test packets, and background=20
   conditions at Src, Dst, and the intervening networks during a =
sample.
   This list is not exhaustive; any additional information that could =
be=20
   useful in interpreting applications of the metrics should also be=20
   reported.

4.7.1. Type-P

   As noted in the Framework document [1], the value of the metric may
   depend on the type of IP packets used to make the measurement, or
   "type-P".  The value of Type-P-One-way-Periodic-Delay could change=20
   if the protocol (UDP or TCP), port number, size, or arrangement for=20
   special treatment (e.g., IP precedence or RSVP) changes.  The exact=20
   Type-P used to make the measurements MUST be accurately reported.

4.7.2. Threshold for delay equivalent to loss

   In addition, the threshold for delay equivalent to loss (or=20
   methodology to determine this threshold) MUST be reported.

4.7.3. Calibration results

   +  If the systematic error can be determined, it SHOULD be removed
      from the measured values.

   +  You SHOULD also report the calibration error, e, such that the
      true value is the reported value plus or minus e, with 95%
      confidence (see the last section.)

   +  If possible, the conditions under which a test packet with finite
      delay is reported as lost due to resource exhaustion on the
      measurement instrument SHOULD be reported.








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2000


4.7.4. Path

   The path traversed by the packets SHOULD be reported, if possible.
   In general it is impractical to know the precise path a given packet
   takes through the network.  The precise path may be known for
   certain Type-P packets on short or stable paths. If Type-P includes
   the record route (or loose-source route) option in the IP header,
   and the path is short enough, and all routers* on the path support
   record (or loose-source) route, then the path will be precisely
   recorded.

   This may be impractical because the route must be short enough,
   many routers do not support (or are not configured for) record =
route,
   and use of this feature would often artificially worsen the
   performance observed by removing the packet from common-case
   processing.  However, partial information is still valuable context.
   For example, if a host can choose between two links* (and hence two
   separate routes from Src to Dst), then the initial link used is
   valuable context.  {Comment: For example, with Merit's NetNow setup,
   a Src on one NAP can reach a Dst on another NAP by either of several
   different backbone networks.}

4.7.5 Background conditions

   In many cases, the results of a sample may be influenced by =
conditions
   at Src, Dst, and/or any intervening networks.  Some things that may=20
   affect the results of a sample include:  traffic levels and/or =
bursts=20
   during the sample, link and/or host failures, etc.  Information =
about
   the background conditions may only be available by non-Internet =
means
   (e.g. phone calls, television) and may only become available days =
after
   samples are taken.

4.8 Single sample vs. a "sample of samples"

   Because this metric represents a periodic stream as one sample, =
there=20
   may be value in running multiple tests using this metric to collect
   a "sample of samples".  For example, it may be more appropriate to=20
   test 1,000 two-minute VoIP calls rather than a single 2,000 minute
   VoIP call.  When considering collection of a sample of samples, =
issues
   like the interval between samples (e.g. Poisson vs. periodic, time =
of
   day/day of week), composition of samples (e.g. equal (Tf-T0 =
duration,
   different packet sizes), and network considerations (e.g. run =
different
   samples over different intervening link-host combinations) should be
   taken into account.  For items like the interval between samples,
   the pattern of use of the application being measured should be=20
   considered.




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17]=0C
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2000


4.9 Statistics based on Type-P-One-way-Delay-Periodic-Stream

4.9.1 Statistics calculable from one sample

   As a metric based on a sample representative of certain=20
   applications, some general purpose statistics (e.g. median and=20
   percentile) may be less applicable than ways to characterize the=20
   range of delay values recorded during the sample metrics.

   Example, a sample metric generates 100 packets as measured at =
MP(Src)
   with the following measurements at MP(Dst)

     +  80 packets received with delay [i] <=3D 20 ms
     +   8 packets received with delay [i] > 20 ms
     +   5 packets received with corrupt packet headers
     +   4 packets from MP(Src) with no matching packet recorded
           at MP(Dst) (effectively lost)
     +   3 packets received with corrupt packet payload and
           and delay [i] <=3D 20 ms
     +   2 packets that duplicate one of the 80 packets received=20
           correctly in the first line

   For this example, packets are considered acceptable if they are
   received with less than or equal to 20ms delays and without corrupt
   packet headers or packet payload.  In this case, the percentage
   of acceptable packets is 80/100 =3D 80%.

   For a different application which will accept packets with corrupt
   packet payload and no delay bound (so long as the packet is =
received),=20
   the percentage of acceptable packets is (80+8+3)/100 =3D 91%.

4.9.2 Statistics calculable from multiple samples

   For computing statistics, a "sample of samples" series of=20
   measurements may be performed. As discussed in section 4.8, under=20
   these conditions, general purpose statistics (e.g. median, =
percentile,
   etc.) may be more relevant as a more statistically significant
   number of packets are used.


5. Security Considerations

5.1 Denial of Service Attacks

   This metric generates a periodic stream of packets from one host =
(Src)
   to another host (Dst) through intervening networks.  This metric
   could be abused for denial of service attacks directed at Dst and/or
   the intervening network(s).


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18]=0C
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2000


   Administrators of Src, Dst, and the intervening network(s) should=20
   establish bilateral or multi-lateral agreements regarding the =
timing,
   size, and frequency of collection of sample metrics.  Use of this
   metric in excess the terms agreed between the participants may BE=20
   cause for immediate rejection or discard of packets or other=20
   escalation procedures defined between the affected parties.

5.2 User data confidentiality

   This metric generates packets for a sample metric, rather than
   taking samples based on user data.  Thus, this metric does not=20
   threaten user data confidentiality.

5.3 Interference with the metric

   It may be possible to identify that a certain packet or stream of=20
   packets are part of a sample metric. With that knowledge at Dst=20
   and/or the intervening networks, it is possible to change the=20
   processing of the packets (e.g. increasing or decreasing delay)=20
   that may distort the measured performance.  It may also be=20
   possible to generate additional packets that appear to be part of=20
   the sample metric. These additional packets are likely to perturb=20
   the results of the sample measurement.

   To discourage the kind of interference mentioned above, packet
   interference checks, such as cryptographic hash, may be used.


6. Acknowledgements

   The authors wish to thank the chairs of the IPPM WG for comments
   that have made the present draft clearer and more focused. Howard
   Stanislevic and Al Morton ahave presented useful comments and
   questions. The authors have also built on the substantial
   foundations laid by the authors of the framework for IP
   performance [1].














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19]=0C
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2000


7. References
         =20
   [1] V.Paxson, G.Almes, J.Mahdavi, and M.Mathis: Framework for IP     =
 =20
       Performance Metrics, IETF RFC 2330, May 1998.
   [2] G.Almes, S.Kalidindi, and M.Zekauskas: A one-way delay metric
       for IPPM, IETF RFC 2679, September 1999.
   [3] International Telecommunications Union recommendation I.380,
       February 1999.
   [4] S. Bradner: Key words for use in RFCs to Indicate Requirement
       Levels, RFC 2119, March 1997.
   [5] ETSI TIPHON document TS-101329-5 (to be published in July).
   [6] G.Almes, S.Kalidindi, and M.Zekauskas: A round-trip delay
       metric for IPPM, IETF RFC 2681.


8. Authors' contact information

   Vilho Raisanen <Vilho.Raisanen@nokia.com>
   P.O. Box 407
   Communication Systems Laboratory
   Nokia Research Center
   FIN-00045 Nokia Group
   Finland
   Phone +358 9 4376 1
   Fax. +358 9 4376 6852

   Glenn Grotefeld <g.grotefeld@motorola.com>
   Motorola, Inc.
   1303 E. Algonquin Road
   4th Floor
   Schaumburg, IL 60196
   USA
   Phone  +1 847 576-5992
   Fax    +1 847 538-7455
















                           EXPIRES    June  2001
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From matt@advanced.org  Wed Dec 27 10:31:42 2000
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From: Grotefeld Glenn-cecl03 <G.Grotefeld@motorola.com>
To: "'vilho.raisanen@nokia.com'" <vilho.raisanen@nokia.com>, ippm@advanced.org
Cc: acmorton@att.com, matt@advanced.org, kaeo@merike.com
Subject: RE: npmps-04bis
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Vilho,

Thank you for the update.  Unfortunately, I got the to end of your note and found this:

"Thus, I find it a bit difficult to justify adding a mention of real
application behaviour in this context. In my view, the goal should be that
incT is (for practical purposes) negligible."


It appears that something may have been left out of this sentence, possibly from the earlier discussion about errors in incT.  For Voice over IP (VoIP)and interactive multi-media applications, incT is nominally 20ms with a desired one-way delay on the order of 100ms.  On this scale, I would consider incT non-negligible.

Glenn Grotefeld

Motorola, Inc.
g.grotefeld@motorola.com
US  847 576-5992
      847 538-7455  FAX


~-----Original Message-----
~From: vilho.raisanen@nokia.com [mailto:vilho.raisanen@nokia.com]
~Sent: Friday, December 22, 2000 4:42 AM
~To: ippm@advanced.org
~Cc: acmorton@att.com; G.Grotefeld; matt@advanced.org; kaeo@merike.com;
~vilho.raisanen@nokia.com
~Subject: npmps-04bis
~
~
~Dear IPPMers,
~
~please find attached the version -04bis of the npmps draft. 
~Thanks to Al &
~Glenn for comments. The following changes to -04 have been made:
~
~- I got rid of the the spurious month names due to Emacs' 
~case-insensitive
~search & replace algorithm. Indeed, these JUNE have caused 
~some confusion.
~- I implemented Al's refinement into 4.6. and added some more 
~text into incT
~error bullet pt. The section now reads
~
~   The description of any specific measurement method should include an
~   accounting and analysis of various sources of error or uncertainty.
~   The Framework document [1] provides general guidance on this point, 
~   but we note here the following specifics related to periodic
~   streams and delay metrics:
~
~   +  Error due to variation of incT. The reasons for this can be e.g.
~      uneven process scheduling, possibly due to CPU load.
~   +  Errors or uncertainties due to uncertainties in the clocks of the
~      MP(Src) and MP(Dst) measurement points.
~   +  Errors or uncertainties due to the difference between 'wire time'
~      and 'host time'.
~
~Earlier during the e-mail thread, Al commented that also real 
~sources of
~periodic streams may (and do) exhibit variation in 
~transmission times. This
~is true, but incorporation of intentional deviations from 
~periodic behaviour
~into the actual measurement metrics would result in a quite complex
~construct. Basically, a (public domain) source should be cited for
~definitive research results on "typical" variations due to O/S or
~application. This should then be mapped into a corresponding metrics
~description and so forth.
~
~Thus, I find it a bit difficult to justify adding a mention of real
~application behaviour in this context. In my view, the goal 
~should be that
~incT is (for practical purposes) negligible.
~
~Season's greetings,
~
~	Vilho
~
~%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
~ Vilho Räisänen                           
~ vilho.raisanen@nokia.com                 
~ Senior research engineer                 
~ Nokia Research Center, Helsinki, Finland 
~%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
~
~



From matt@advanced.org  Thu Dec 28 04:36:25 2000
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My intention was indeed to say that the _error_ in incT is preferably
negligible in a realistic measurement setup.

This correction applies to my explanation, and does not - in my
understanding - require any changes to the draft.

	BR,
		Vilho

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
 Vilho Räisänen                           
 vilho.raisanen@nokia.com                 
 Senior research engineer                 
 Nokia Research Center, Helsinki, Finland 
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%

> -----Original Message-----
> From: EXT Grotefeld Glenn-cecl03 [mailto:G.Grotefeld@motorola.com]
> Sent: 27. December 2000 17:13
> To: 'vilho.raisanen@nokia.com'; ippm@advanced.org
> Cc: acmorton@att.com; matt@advanced.org; kaeo@merike.com
> Subject: RE: npmps-04bis
> 
> 
> Vilho,
> 
> Thank you for the update.  Unfortunately, I got the to end of 
> your note and found this:
> 
> "Thus, I find it a bit difficult to justify adding a mention of real
> application behaviour in this context. In my view, the goal 
> should be that
> incT is (for practical purposes) negligible."
> 
> 
> It appears that something may have been left out of this 
> sentence, possibly from the earlier discussion about errors 
> in incT.  For Voice over IP (VoIP)and interactive multi-media 
> applications, incT is nominally 20ms with a desired one-way 
> delay on the order of 100ms.  On this scale, I would consider 
> incT non-negligible.
> 
> Glenn Grotefeld
> 
> Motorola, Inc.
> g.grotefeld@motorola.com
> US  847 576-5992
>       847 538-7455  FAX
> 
> 
> ~-----Original Message-----
> ~From: vilho.raisanen@nokia.com [mailto:vilho.raisanen@nokia.com]
> ~Sent: Friday, December 22, 2000 4:42 AM
> ~To: ippm@advanced.org
> ~Cc: acmorton@att.com; G.Grotefeld; matt@advanced.org; 
> kaeo@merike.com;
> ~vilho.raisanen@nokia.com
> ~Subject: npmps-04bis
> ~
> ~
> ~Dear IPPMers,
> ~
> ~please find attached the version -04bis of the npmps draft. 
> ~Thanks to Al &
> ~Glenn for comments. The following changes to -04 have been made:
> ~
> ~- I got rid of the the spurious month names due to Emacs' 
> ~case-insensitive
> ~search & replace algorithm. Indeed, these JUNE have caused 
> ~some confusion.
> ~- I implemented Al's refinement into 4.6. and added some more 
> ~text into incT
> ~error bullet pt. The section now reads
> ~
> ~   The description of any specific measurement method should 
> include an
> ~   accounting and analysis of various sources of error or 
> uncertainty.
> ~   The Framework document [1] provides general guidance on 
> this point, 
> ~   but we note here the following specifics related to periodic
> ~   streams and delay metrics:
> ~
> ~   +  Error due to variation of incT. The reasons for this 
> can be e.g.
> ~      uneven process scheduling, possibly due to CPU load.
> ~   +  Errors or uncertainties due to uncertainties in the 
> clocks of the
> ~      MP(Src) and MP(Dst) measurement points.
> ~   +  Errors or uncertainties due to the difference between 
> 'wire time'
> ~      and 'host time'.
> ~
> ~Earlier during the e-mail thread, Al commented that also real 
> ~sources of
> ~periodic streams may (and do) exhibit variation in 
> ~transmission times. This
> ~is true, but incorporation of intentional deviations from 
> ~periodic behaviour
> ~into the actual measurement metrics would result in a quite complex
> ~construct. Basically, a (public domain) source should be cited for
> ~definitive research results on "typical" variations due to O/S or
> ~application. This should then be mapped into a corresponding metrics
> ~description and so forth.
> ~
> ~Thus, I find it a bit difficult to justify adding a mention of real
> ~application behaviour in this context. In my view, the goal 
> ~should be that
> ~incT is (for practical purposes) negligible.
> ~
> ~Season's greetings,
> ~
> ~	Vilho
> ~
> ~%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% 
> ~ Vilho Räisänen                           
> ~ vilho.raisanen@nokia.com                 
> ~ Senior research engineer                 
> ~ Nokia Research Center, Helsinki, Finland 
> ~%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
> ~
> ~
> 



