One document matched: draft-morton-ippm-2680-bis-04.xml
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<front>
<title abbrev="A One-Way Loss Metric for IPPM">A One-Way Loss Metric for
IPPM</title>
<author fullname="Guy Almes" initials="G." surname="Almes">
<organization>Texas A&M</organization>
<address>
<postal>
<street/>
<city/>
<region/>
<code/>
<country/>
</postal>
<phone/>
<facsimile/>
<email>galmes@tamu.edu</email>
<uri/>
</address>
</author>
<author fullname="Sunil Kalidindi" initials="S." surname="Kalidindi">
<organization>Ixia</organization>
<address>
<postal>
<street/>
<city/>
<region/>
<code/>
<country/>
</postal>
<phone/>
<facsimile/>
<email>skalidindi@ixiacom.com</email>
<uri/>
</address>
</author>
<author fullname="Matt Zekauskas" initials="M." surname="Zekauskas">
<organization>Internet2</organization>
<address>
<postal>
<street/>
<city/>
<region/>
<code/>
<country/>
</postal>
<phone/>
<facsimile/>
<email>matt@internet2.edu</email>
<uri/>
</address>
</author>
<author fullname="Al Morton" initials="A." role="editor" surname="Morton">
<organization>AT&T Labs</organization>
<address>
<postal>
<street>200 Laurel Avenue South</street>
<city>Middletown</city>
<region>NJ</region>
<code>07748</code>
<country>USA</country>
</postal>
<phone>+1 732 420 1571</phone>
<facsimile>+1 732 368 1192</facsimile>
<email>acmorton@att.com</email>
<uri>http://home.comcast.net/~acmacm/</uri>
</address>
</author>
<date day="6" month="October" year="2014"/>
<abstract>
<t>This memo (RFC 2680 bis) defines a metric for one-way loss of packets
across Internet paths. It builds on notions introduced and discussed in
the IPPM Framework document, RFC 2330; the reader is assumed to be
familiar with that document.</t>
</abstract>
<note title="Requirements Language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</note>
</front>
<middle>
<section title="Introduction">
<t>This memo defines a metric for one-way packet loss across Internet
paths. It builds on notions introduced and discussed in the IPPM
Framework document, <xref target="RFC2330"/>; the reader is assumed to
be familiar with that document.</t>
<t>This memo is intended to be parallel in structure to a companion
document for One-way Delay ("A One-way Delay Metric for IPPM") <xref
target="RFC2679"/>; the reader is assumed to be familiar with that
document.</t>
<t>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<xref target="RFC2119"/>.
Although <xref target="RFC2119"/> 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.</t>
<t>The structure of the memo is as follows:</t>
<t>+ A 'singleton' analytic metric, called Type-P-One-way-Packet-Loss,
is introduced to measure a single observation of packet transmission or
loss.</t>
<t>+ Using this singleton metric, a 'sample', called
Type-P-One-way-Packet-Loss-Poisson-Stream, is introduced to measure a
sequence of singleton transmissions and/or losses measured at times
taken from a Poisson process.</t>
<t>+ Using this sample, several 'statistics' of the sample are defined
and discussed.</t>
<t>This progression from singleton to sample to statistics, with clear
separation among them, is important.</t>
<t>Whenever a technical term from the IPPM Framework document is first
used in this memo, it will be tagged with a trailing asterisk. For
example, "term*" indicates that "term" is defined in the Framework.</t>
<section title="Motivation">
<t>Understanding one-way packet loss of Type-P* packets from a source
host* to a destination host is useful for several reasons:</t>
<t>+ Some applications do not perform well (or at all) if end-to-end
loss between hosts is large relative to some threshold value.</t>
<t>+ Excessive packet loss may make it difficult to support certain
real-time applications (where the precise threshold of "excessive"
depends on the application).</t>
<t>+ The larger the value of packet loss, the more difficult it is for
transport-layer protocols to sustain high bandwidths.</t>
<t>+ The sensitivity of real-time applications and of transport-layer
protocols to loss become especially important when very large
delay-bandwidth products must be supported.</t>
<t>The measurement of one-way loss instead of round-trip loss is
motivated by the following factors:</t>
<t>+ In today's Internet, the path from a source to a destination may
be different than the path from the destination back to the source
("asymmetric paths"), such that different sequences of routers are
used for the forward and reverse paths. Therefore round-trip
measurements actually measure the performance of two distinct paths
together. Measuring each path independently highlights the performance
difference between the two paths which may traverse different Internet
service providers, and even radically different types of networks (for
example, research versus commodity networks, or networks with
asymmetric link capacities, or wireless vs. wireline access).</t>
<t>+ Even when the two paths are symmetric, they may have radically
different performance characteristics due to asymmetric queueing.</t>
<t>+ Performance of an application may depend mostly on the
performance in one direction. For example, a TCP-based communication
may experience reduced throughput if congestion occurs in one
direction of its communication. Trouble shooting may be simplified if
the congested direction of TCP transmission can be identified.</t>
<t>+ In quality-of-service (QoS) enabled networks, provisioning in one
direction may be radically different than provisioning in the reverse
direction, and thus the QoS guarantees differ. Measuring the paths
independently allows the verification of both guarantees.</t>
<t>It is outside the scope of this document to say precisely how loss
metrics would be applied to specific problems.</t>
</section>
<section title="General Issues Regarding Time">
<t>{Comment: the terminology below differs from that defined by ITU-T
documents (e.g., G.810, "Definitions and terminology for
synchronization networks" and I.356, "B-ISDN ATM layer cell transfer
performance"), but is consistent with the IPPM Framework document. In
general, these differences derive from the different backgrounds; the
ITU-T documents historically have a telephony origin, while the
authors of this document (and the Framework) have a computer systems
background. Although the terms defined below have no direct equivalent
in the ITU-T definitions, after our definitions we will provide a
rough mapping. However, note one potential confusion: our definition
of "clock" is the computer operating systems definition denoting a
time-of-day clock, while the ITU-T definition of clock denotes a
frequency reference.}</t>
<t>Whenever a time (i.e., a moment in history) is mentioned here, it
is understood to be measured in seconds (and fractions) relative to
UTC.</t>
<t>As described more fully in the Framework document, there are four
distinct, but related notions of clock uncertainty:</t>
<t>synchronization*</t>
<t>measures the extent to which two clocks agree on what time it is.
For example, the clock on one host might be 5.4 msec ahead of the
clock on a second host. {Comment: A rough ITU-T equivalent is "time
error".}</t>
<t>accuracy*</t>
<t>measures the extent to which a given clock agrees with UTC. For
example, the clock on a host might be 27.1 msec behind UTC. {Comment:
A rough ITU-T equivalent is "time error from UTC".}</t>
<t>resolution*</t>
<t>specification of the smallest unit by which the clock's time is
updated. It gives a lower bound on the clock's uncertainty. For
example, the clock on an old Unix host might tick only once every 10
msec, and thus have a resolution of only 10 msec. {Comment: A very
rough ITU-T equivalent is "sampling period".}</t>
<t>skew*</t>
<t>measures the change of accuracy, or of synchronization, with time.
For example, the clock on a given host might gain 1.3 msec per hour
and thus be 27.1 msec behind UTC at one time and only 25.8 msec an
hour later. In this case, we say that the clock of the given host has
a skew of 1.3 msec per hour relative to UTC, which threatens accuracy.
We might also speak of the skew of one clock relative to another
clock, which threatens synchronization. {Comment: A rough ITU-T
equivalent is "time drift".}</t>
</section>
</section>
<section title="A Singleton Definition for One-way Packet Loss">
<t/>
<section title="Metric Name:">
<t>Type-P-One-way-Packet-Loss</t>
</section>
<section title="Metric Parameters:">
<t>+ Src, the IP address of a host</t>
<t>+ Dst, the IP address of a host</t>
<t>+ T, a time</t>
<t>+ Tmax, a loss threshold waiting time</t>
</section>
<section title="Metric Units:">
<t>The value of a Type-P-One-way-Packet-Loss is either a zero
(signifying successful transmission of the packet) or a one
(signifying loss).</t>
</section>
<section title="Definition:">
<t>>>The *Type-P-One-way-Packet-Loss* from Src to Dst at T is
0<< means that Src sent the first bit of a Type-P packet to Dst
at wire-time* T and that Dst received that packet.</t>
<t>>>The *Type-P-One-way-Packet-Loss* from Src to Dst at T is
1<< means that Src sent the first bit of a type-P packet to Dst
at wire-time T and that Dst did not receive that packet (within the
loss threshold waiting time, Tmax).</t>
</section>
<section title="Discussion:">
<t>Thus, Type-P-One-way-Packet-Loss is 0 exactly when Type-P-One-way-
Delay is a finite value, and it is 1 exactly when Type-P-One-way-
Delay is undefined.</t>
<t>The following issues are likely to come up in practice:</t>
<t>+ A given methodology will have to include a way to distinguish
between a packet loss and a very large (but finite) delay. As noted by
Mahdavi and Paxson <xref target="RFC2678"/>, simple upper bounds (such
as the 255 seconds theoretical upper bound on the lifetimes of IP
packets <xref target="RFC0791"/>) could be used, but good engineering,
including an understanding of packet lifetimes, will be needed in
practice. {Comment: Note that, for many applications of these metrics,
there may be no harm in treating a large delay as packet loss. An
audio playback packet, for example, that arrives only after the
playback point may as well have been lost. See section 4.1.1 of <xref
target="RFC6703"/> for examination of unusual packet delays and
application performance estimation.}</t>
<t>+ If the packet arrives, but is corrupted, then it is counted as
lost. {Comment: one is tempted to count the packet as received since
corruption and packet loss are related but distinct phenomena. If the
IP header is corrupted, however, one cannot be sure about the source
or destination IP addresses and is thus on shaky grounds about knowing
that the corrupted received packet corresponds to a given sent test
packet. Similarly, if other parts of the packet needed by the
methodology to know that the corrupted received packet corresponds to
a given sent test packet, then such a packet would have to be counted
as lost. Counting these packets as lost but packet with corruption in
other parts of the packet as not lost would be inconsistent.}</t>
<t>+ If the packet is duplicated along the path (or paths) so that
multiple non-corrupt copies arrive at the destination, then the packet
is counted as received.</t>
<t>+ If the packet is fragmented and if, for whatever reason,
reassembly does not occur, then the packet will be deemed lost.</t>
</section>
<section title="Methodologies:">
<t>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).</t>
<t>Generally, for a given Type-P, one possible methodology would
proceed as follows:</t>
<t>+ Arrange that Src and Dst have clocks that are synchronized with
each other. The degree of synchronization is a parameter of the
methodology, and depends on the threshold used to determine loss (see
below).</t>
<t>+ At the Src host, select Src and Dst IP addresses, and form a test
packet of Type-P with these addresses.</t>
<t>+ At the Dst host, arrange to receive the packet.</t>
<t>+ At the Src host, place a timestamp in the prepared Type-P packet,
and send it towards Dst (ideally minimizing time before sending).</t>
<t>+ If the packet arrives within a reasonable period of time, the
one-way packet-loss is taken to be zero (and take a timestamp as soon
as possible upon the receipt of the packet).</t>
<t>+ If the packet fails to arrive within a reasonable period of time,
Tmax, the one-way packet-loss is taken to be one. Note that the
threshold of "reasonable" here is a parameter of the metric.</t>
<t>{Comment: The definition of reasonable is intentionally vague, and
is intended to indicate a value "Th" so large that any value in the
closed interval [Th-delta, Th+delta] is an equivalent threshold for
loss. Here, delta encompasses all error in clock synchronization and
timestamp acquisition and assignment along the measured path. If there
is a single value, Tmax, after which the packet must be counted as
lost, then we reintroduce the need for a degree of clock
synchronization similar to that needed for one-way delay, and
virtually all practical measurement systems combine methods for delay
and loss. Therefore, if a measure of packet loss parameterized by a
specific non-huge "reasonable" time-out value is needed, one can
always measure one-way delay and see what percentage of packets from a
given stream exceed a given time-out value. This point is examined in
detail in <xref target="RFC6703"/>, including analysis preferences to
assign undefined delay to packets that fail to arrive with the
difficulties emerging from the informal "infinite delay" assignment,
and an estimation of an upper bound on waiting time for packets in
transit. Further, enforcing a specific constant waiting time on stored
singletons of one-way delay is compliant with this specification and
may allow the results to serve more than one reporting audience.}</t>
<t>Issues such as the packet format, the means by which Dst knows when
to expect the test packet, and the means by which Src and Dst are
synchronized are outside the scope of this document. {Comment: We plan
to document elsewhere our own work in describing such more detailed
implementation techniques and we encourage others to as well.}</t>
</section>
<section title="Errors and Uncertainties:">
<t>The description of any specific measurement method should include
an accounting and analysis of various sources of error or uncertainty.
The Framework document provides general guidance on this point.</t>
<t>For loss, there are three sources of error:</t>
<t>+ Synchronization between clocks on Src and Dst.</t>
<t>+ The packet-loss threshold (which is related to the
synchronization between clocks).</t>
<t>+ Resource limits in the network interface or software on the
receiving instrument.</t>
<t>The first two sources are interrelated and could result in a test
packet with finite delay being reported as lost. Type-P-One-way-
Packet-Loss is 1 if the test packet does not arrive, or if it does
arrive and the difference between Src timestamp and Dst timestamp is
greater than the "reasonable period of time", or loss threshold. If
the clocks are not sufficiently synchronized, the loss threshold may
not be "reasonable" - the packet may take much less time to arrive
than its Src timestamp indicates. Similarly, if the loss threshold is
set too low, then many packets may be counted as lost. The loss
threshold must be high enough, and the clocks synchronized well enough
so that a packet that arrives is rarely counted as lost. (See the
discussions in the previous two sections.)</t>
<t>Since the sensitivity of packet loss measurement alone to lack of
clock synchronization is less than for delay, we refer the reader to
the treatment of synchronization errors in the One-way Delay metric
<xref target="RFC2330"/> for more details.</t>
<t>The last source of error, resource limits, cause the packet to be
dropped by the measurement instrument, and counted as lost when in
fact the network delivered the packet in reasonable time.</t>
<t>The measurement instruments should be calibrated such that the loss
threshold is reasonable for application of the metrics and the clocks
are synchronized enough so the loss threshold remains reasonable.</t>
<t>In addition, the instruments should be checked to ensure the that
the possibility a packet arrives at the network interface, but is lost
due to congestion on the interface or to other resource exhaustion
(e.g., buffers) on the instrument is low.</t>
</section>
<section title="Reporting the metric:">
<t>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 four items to consider: Type-P of the test
packets, the loss threshold, instrument calibration, and the path
traversed by the test packets. This list is not exhaustive; any
additional information that could be useful in interpreting
applications of the metrics should also be reported (see <xref
target="RFC6703"/> for extensive discussion of reporting
considerations for different audiences).</t>
<section title="Type-P">
<t>As noted in the Framework document <xref target="RFC2330"/>, 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-Delay
could change if the protocol (UDP or TCP), port number, size, or
arrangement for special treatment (e.g., IP precedence or RSVP)
changes. The exact Type-P used to make the measurements MUST be
accurately reported.</t>
</section>
<section title="Loss Threshold">
<t>The threshold, Tmax, (or methodology to distinguish) between a
large finite delay and loss MUST be reported.</t>
</section>
<section title="Calibration Results">
<t>The degree of synchronization between the Src and Dst clocks MUST
be reported. If possible, possibility that a test packet that
arrives at the Dst network interface is reported as lost due to
resource exhaustion on Dst SHOULD be reported.</t>
</section>
<section title="Path">
<t>Finally, the path traversed by the packet 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 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 is 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.}</t>
</section>
</section>
</section>
<section title="A Definition for Samples of One-way Packet Loss">
<t>Given the singleton metric Type-P-One-way-Packet-Loss, we now define
one particular sample of such singletons. The idea of the sample is to
select a particular binding of the parameters Src, Dst, and Type- P,
then define a sample of values of parameter T. The means for defining
the values of T is to select a beginning time T0, a final time Tf, and
an average rate lambda, then define a pseudo-random Poisson process of
rate lambda, whose values fall between T0 and Tf. The time interval
between successive values of T will then average 1/lambda.</t>
<t>Note that Poisson sampling is only one way of defining a sample.
Poisson has the advantage of limiting bias, but other methods of
sampling will be appropriate for different situations. For example, a
truncated Poisson distribution may be needed to avoid reactive network
state changes during intervals of inactivity, see section 4.6 of <xref
target="RFC7321"/>. Sometimes, the goal is sampling with a known bias,
and <xref target="RFC3432"/> describes a method for periodic sampling
with random start times.</t>
<section title="Metric Name:">
<t>Type-P-One-way-Packet-Loss-Poisson-Stream</t>
</section>
<section title="Metric Parameters:">
<t>+ Src, the IP address of a host</t>
<t>+ Dst, the IP address of a host</t>
<t>+ T0, a time</t>
<t>+ Tf, a time</t>
<t>+ Tmax, a loss threshold waiting time</t>
<t>+ lambda, a rate in reciprocal seconds</t>
</section>
<section title="Metric Units:">
<t>A sequence of pairs; the elements of each pair are:</t>
<t>+ T, a time, and</t>
<t>+ L, either a zero or a one</t>
<t>The values of T in the sequence are monotonic increasing. Note that
T would be a valid parameter to Type-P-One-way-Packet-Loss, and that L
would be a valid value of Type-P-One-way-Packet-Loss.</t>
</section>
<section title="Definition:">
<t>Given T0, Tf, and lambda, we compute a pseudo-random Poisson
process beginning at or before T0, with average arrival rate lambda,
and ending at or after Tf. Those time values greater than or equal to
T0 and less than or equal to Tf are then selected. At each of the
times in this process, we obtain the value of
Type-P-One-way-Packet-Loss at this time. The value of the sample is
the sequence made up of the resulting <time, loss> pairs. If
there are no such pairs, the sequence is of length zero and the sample
is said to be empty.</t>
</section>
<section title="Discussion:">
<t>The reader should be familiar with the in-depth discussion of
Poisson sampling in the Framework document <xref target="RFC2330"/>,
which includes methods to compute and verify the pseudo-random Poisson
process.</t>
<t>We specifically do not constrain the value of lambda, except to
note the extremes. If the rate is too large, then the measurement
traffic will perturb the network, and itself cause congestion. If the
rate is too small, then you might not capture interesting network
behavior. {Comment: We expect to document our experiences with, and
suggestions for, lambda elsewhere, culminating in a "best current
practices" document.}</t>
<t>Since a pseudo-random number sequence is employed, the sequence of
times, and hence the value of the sample, is not fully specified.
Pseudo-random number generators of good quality will be needed to
achieve the desired qualities.</t>
<t>The sample is defined in terms of a Poisson process both to avoid
the effects of self-synchronization and also capture a sample that is
statistically as unbiased as possible. The Poisson process is used to
schedule the loss measurements. The test packets will generally not
arrive at Dst according to a Poisson distribution, since they are
influenced by the network. Time-slotted links described in <xref
target="RFC7321"/> can greatly modify the sample characteristics.</t>
<t>{Comment: there is, of course, no claim that real Internet traffic
arrives according to a Poisson arrival process.</t>
<t>It is important to note that, in contrast to this metric, loss
rates observed by transport connections do not reflect unbiased
samples. For example, TCP transmissions both (1) occur in bursts,
which can induce loss due to the burst volume that would not otherwise
have been observed, and (2) adapt their transmission rate in an
attempt to minimize the loss rate observed by the connection.}</t>
<t>All the singleton Type-P-One-way-Packet-Loss metrics in the
sequence will have the same values of Src, Dst, and Type-P.</t>
<t>Note also that, given one sample that runs from T0 to Tf, and given
new time values T0' and Tf' such that T0 <= T0' <= Tf' <= Tf,
the subsequence of the given sample whose time values fall between T0'
and Tf' are also a valid Type-P-One-way-Packet-Loss-Poisson-Stream
sample.</t>
</section>
<section title="Methodologies:">
<t>The methodologies follow directly from:</t>
<t>+ the selection of specific times, using the specified Poisson
arrival process, and</t>
<t>+ the methodologies discussion already given for the singleton
Type- P-One-way-Packet-Loss metric.</t>
<t>Care must be given to correctly handle out-of-order arrival of test
packets; it is possible that the Src could send one test packet at
TS[i], then send a second one (later) at TS[i+1], while the Dst could
receive the second test packet at TR[i+1], and then receive the first
one (later) at TR[i]. Metrics for reordering may be found in <xref
target="RFC4737"/>.</t>
</section>
<section title="Errors and Uncertainties:">
<t>In addition to sources of errors and uncertainties associated with
methods employed to measure the singleton values that make up the
sample, care must be given to analyze the accuracy of the Poisson
arrival process of the wire-times of the sending of the test packets.
Problems with this process could be caused by several things,
including problems with the pseudo-random number techniques used to
generate the Poisson arrival process. The Framework document shows how
to use the Anderson-Darling test verify the accuracy of the Poisson
process over small time frames. {Comment: The goal is to ensure that
the test packets are sent "close enough" to a Poisson schedule, and
avoid periodic behavior.}</t>
</section>
<section title="Reporting the metric:">
<t>The calibration and context for the underlying singletons MUST be
reported along with the stream. (See "Reporting the metric" for
Type-P-One-way-Packet-Loss.)</t>
</section>
</section>
<section title="Some Statistics Definitions for One-way Packet Loss">
<t>Given the sample metric Type-P-One-way-Packet-Loss-Poisson-Stream, we
now offer several statistics of that sample. These statistics are
offered mostly to be illustrative of what could be done. See <xref
target="RFC6703"/> for additional discussion of statistics that are
relevant to different audiences.</t>
<section title="Type-P-One-way-Packet Loss-Average">
<t>Given a Type-P-One-way-Packet-Loss-Poisson-Stream, the average of
all the L values in the Stream. In addition, the
Type-P-One-way-Packet- Loss-Average is undefined if the sample is
empty.</t>
<t>Example: suppose we take a sample and the results are:</t>
<t>Stream1 = <</t>
<t><T1, 0></t>
<t><T2, 0></t>
<t><T3, 1></t>
<t><T4, 0></t>
<t><T5, 0></t>
<t>></t>
<t>Then the average would be 0.2.</t>
<t>Note that, since healthy Internet paths should be operating at loss
rates below 1% (particularly if high delay-bandwidth products are to
be sustained), the sample sizes needed might be larger than one would
like. Thus, for example, if one wants to discriminate between various
fractions of 1% over one-minute periods, then several hundred samples
per minute might be needed. This would result in larger values of
lambda than one would ordinarily want.</t>
<t>Note that although the loss threshold should be set such that any
errors in loss are not significant, if the possibility that a packet
which arrived is counted as lost due to resource exhaustion is
significant compared to the loss rate of interest,
Type-P-One-way-Packet-Loss-Average will be meaningless.</t>
</section>
</section>
<section anchor="Security" title="Security Considerations">
<t>Conducting Internet measurements raises both security and privacy
concerns. This memo does not specify an implementation of the metrics,
so it does not directly affect the security of the Internet nor of
applications which run on the Internet. However, implementations of
these metrics must be mindful of security and privacy concerns.</t>
<t>There are two types of security concerns: potential harm caused by
the measurements, and potential harm to the measurements. The
measurements could cause harm because they are active, and inject
packets into the network. The measurement parameters MUST be carefully
selected so that the measurements inject trivial amounts of additional
traffic into the networks they measure. If they inject "too much"
traffic, they can skew the results of the measurement, and in extreme
cases cause congestion and denial of service.</t>
<t>The measurements themselves could be harmed by routers giving
measurement traffic a different priority than "normal" traffic, or by an
attacker injecting artificial measurement traffic. If routers can
recognize measurement traffic and treat it separately, the measurements
will not reflect actual user traffic. If an attacker injects artificial
traffic that is accepted as legitimate, the loss rate will be
artificially lowered. Therefore, the measurement methodologies SHOULD
include appropriate techniques to reduce the probability measurement
traffic can be distinguished from "normal" traffic. Authentication
techniques, such as digital signatures, may be used where appropriate to
guard against injected traffic attacks.</t>
<t>The privacy concerns of network measurement are limited by the active
measurements described in this memo. Unlike passive measurements, there
can be no release of existing user data.</t>
</section>
<section title="Acknowledgements">
<t>Thanks are due to Matt Mathis for encouraging this work and for
calling attention on so many occasions to the significance of packet
loss.</t>
<t>Thanks are due also to Vern Paxson for his valuable comments on early
drafts, and to Garry Couch and Will Leland for several useful
suggestions.</t>
</section>
<section title="RFC 2680 bis">
<t>The text above constitutes RFC 2680 bis proposed for advancement on
the IETF Standards Track.</t>
<t><xref target="RFC7290"/> provides the test plan and results
supporting <xref target="RFC2680"/> advancement along the standards
track, according to the process in <xref target="RFC6576"/>. The
conclusions of <xref target="RFC7290"/> list four minor modifications
for inclusion:</t>
<t><list style="numbers">
<t>Section 6.2.3 of <xref target="RFC7290"/> asserts that the
assumption of post-processing to enforce a constant waiting time
threshold is compliant, and that the text of the RFC should be
revised slightly to include this point (see the last list item of
section 2.6, above).</t>
<t>Section 6.5 of <xref target="RFC7290"/> indicates that
Type-P-One-way-Packet-Loss-Average statistic is more commonly called
Packet Loss Ratio, so it is re-named in RFC2680bis (this small
discrepancy does not affect candidacy for advancement) (see section
4.1, above).</t>
<t>The IETF has reached consensus on guidance for reporting metrics
in <xref target="RFC6703"/>, and this memo should be referenced in
RFC2680bis to incorporate recent experience where appropriate (see
the last list item of section 2.6, section 2.8, and section 4
above).</t>
<t>There are currently two errata with status "Verified" and "Held
for document update" for <xref target="RFC2680"/>, and it appears
these minor revisions should be incorporated in RFC2680bis (see
section 1 and section 2.7).</t>
</list>A number of updates to the <xref target="RFC2680"/> text have
been implemented in the text, to reference key IPPM RFCs that were
approved after <xref target="RFC2680"/> (see sections 3 and 3.6, above),
and to address comments on the IPPM mailing list describing current
conditions and experience.</t>
<t><list style="numbers">
<t>Near the end of section 1.1, update of a network example using
ATM and clarification of TCP's affect on queue occupation and
importance of one-way delay measurement.</t>
<t>Clarification of the definition of "resolution" in section
1.2.</t>
<t>Explicit inclusion of the maximum waiting time input parameter in
sections 2.2, 2.4, and 3.2, reflecting recognition of this parameter
in more recent RFCs and ITU-T Recommendation Y.1540.</t>
<t>Addition of reference to RFC6703 in the discussion of packet life
time and application timeouts in section 2.5.</t>
<t>Added parenthetical guidance on minimizing interval between
timestamp placement to send time or reception time in section 2.6.
Also, the text now recognizes the timestamp acquisition process and
that practical systems measure both delay and loss (thus require the
max waiting time parameter).</t>
<t>Added reference to RFC 3432 Periodic sampling alongside Poisson
sampling in section 3, and also noting that a truncated Poisson
distribution may be needed with modern networks as described in the
IPPM Framework update, RFC7312.</t>
<t>Recognition that Time-slotted links described in [RFC7321] can
greatly modify the sample characteristics, in section 3.5.</t>
<t>Add reference to RFC 4737 Reordering metric in the related
discussion of section 3.6, Methodologies.</t>
<t/>
</list>Section 5.4.4 of <xref target="RFC6390"/> suggests a common
template for performance metrics partially derived from previous IPPM
and BMWG RFCs, but also contains some new items. All of the <xref
target="RFC6390"/> Normative points are covered, but not quite in the
same section names or orientation. Several of the Informative points are
covered. Maintaining the familiar outline of IPPM literature has both
value and minimizes unnecessary differences between this revised RFC and
current/future IPPM RFCs.</t>
<t/>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This memo makes no requests of IANA.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>Special thanks are due to Vern Paxson of Lawrence Berkeley Labs for
his helpful comments on issues of clock uncertainty and statistics.
Thanks also to Garry Couch, Will Leland, Andy Scherrer, Sean Shapira,
and Roland Wittig for several useful suggestions.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.0791"?>
<?rfc include='reference.RFC.2119'?>
<?rfc include='reference.RFC.2026'?>
<?rfc include='reference.RFC.2330'?>
<?rfc include='reference.RFC.2678'?>
<?rfc include='reference.RFC.2679'?>
<?rfc include='reference.RFC.2680'?>
<?rfc include='reference.RFC.3432'?>
<?rfc include='reference.RFC.4656'?>
<?rfc include='reference.RFC.6049'?>
<?rfc include='reference.RFC.5835'?>
<?rfc include='reference.RFC.5357'?>
<?rfc include='reference.RFC.5657'?>
<?rfc include='reference.RFC.6576'?>
<?rfc include='reference.RFC.6703'?>
<?rfc include='reference.RFC.7321'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.6390'?>
<?rfc include='reference.I-D.ietf-ippm-testplan-rfc2680'?>
<?rfc include='reference.RFC.3931'?>
<?rfc include='reference.RFC.4737'?>
<?rfc include='reference.RFC.7290'?>
<reference anchor="ADK">
<front>
<title>K-sample Anderson-Darling Tests of fit, for continuous and
discrete cases</title>
<author fullname="Fred Scholz" initials="F.W." surname="Scholz">
<!-- fullname="F.W. Scholz" -->
<organization abbrev="Boeing">Boeing Computer
Services</organization>
</author>
<author initials="M.A." surname="Stephens">
<!-- fullname="M.A. Stephens" -->
<organization>Simon Fraser University</organization>
</author>
<date month="May" year="1986"/>
</front>
<seriesInfo name="University of Washington, Technical Report"
value="No. 81"/>
</reference>
</references>
</back>
</rfc>
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