One document matched: draft-ietf-tsvwg-byte-pkt-congest-00.xml
<?xml version="1.0" encoding="US-ASCII"?>
<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
<rfc category="info" docName="draft-ietf-tsvwg-byte-pkt-congest-00"
ipr="full3978">
<?xml-stylesheet type='text/xsl' href='http://xml.resource.org/authoring/rfc2629.xslt' ?>
<!-- Alterations to I-D/RFC boilerplate -->
<?rfc private="" ?>
<!-- Default private="" Produce an internal memo 2.5pp shorter than an I-D or RFC -->
<?rfc topblock="yes" ?>
<!-- Default topblock="yes" put the famous header block on the first page -->
<?rfc footer="" ?>
<!-- Default footer="Expires <date>" override the center footer string -->
<?rfc header="" ?>
<!-- Default header="Internet-Draft" override the leftmost header string -->
<?rfc authorship="yes" ?>
<!-- Default authorship="yes" Render authors' addresses section -->
<?rfc rfcprocack="yes" ?>
<!-- Default rfcprocack="no" add a short sentence acknowledging xml2rfc -->
<?rfc strict="no" ?>
<!-- Default strict="no" Don't check I-D nits -->
<?rfc rfcedstyle="no" ?>
<!-- Default rfcedstyle="yes" attempt to closely follow finer details from the latest observable RFC-Editor style -->
<?rfc colonspace="no" ?>
<!-- Default colonspace="no" put two spaces instead of one after each colon (``:'') in txt or nroff files -->
<?rfc notedraftinprogress="yes" ?>
<!-- Default notedraftinprogress="yes" generates "(work in progress)", as appropriate -->
<?rfc refparent="References" ?>
<!-- Default refparent="References" title of the top-level section containing all references -->
<!-- IETF process -->
<?rfc iprnotified="no" ?>
<!-- Default iprnotified="no" I haven't disclosed existence of IPR to IETF -->
<!-- ToC format -->
<?rfc toc="yes" ?>
<!-- Default toc="no" No Table of Contents -->
<?rfc tocappendix="yes" ?>
<!-- Default tocappendix="yes" control whether the word `Appendix' appears in the table-of-content -->
<?rfc tocdepth="3" ?>
<!-- Default tocdepth="3" if toc is "yes", then this determines the depth of the table-of-contents -->
<?rfc tocindent="yes" ?>
<!-- Default tocindent="yes" if toc is "yes", indent subsections in the table-of-contents -->
<?rfc tocnarrow="yes" ?>
<!-- Default tocnarrow="yes" affects horizontal spacing in the table-of-content -->
<?rfc tocompact="yes" ?>
<!-- Default tocompact="yes" if toc is "yes", then setting this to "no" will make it a little less compact -->
<!-- Cross referencing, footnotes, comments -->
<?rfc symrefs="yes" ?>
<!-- Default symrefs="no" Don't use anchors, but use numbers for refs -->
<?rfc sortrefs="yes"?>
<!-- Default sortrefs="no" Don't sort references into order -->
<?rfc comments="yes" ?>
<!-- Default comments="yes" Don't render comments -->
<?rfc inline="no" ?>
<!-- Default inline="no" if comments is "yes", then render comments inline; otherwise render them in an `Editorial Comments' section -->
<?rfc editing="no" ?>
<!-- Default editing="no" Don't insert editing marks for ease of discussing draft versions -->
<!-- Pagination control -->
<?rfc compact="yes"?>
<!-- Default compact="no" Start sections on new pages -->
<?rfc subcompact="no"?>
<!-- Default subcompact="(as compact setting)" yes/no is not quite as compact as yes/yes -->
<?rfc autobreaks="yes" ?>
<!-- Default autobreaks="yes" avoid widows and orphans (not perfect) -->
<!-- HTML formatting control -->
<?rfc emoticonic="yes" ?>
<!-- Default emoticonic="no" Doesn't prettify HTML format -->
<?rfc background="" ?>
<!-- Default background="" when producing a html file, use this image -->
<?rfc useobject="no" ?>
<!-- Default useobject="no" use <object> not <src> when outputting HTML -->
<?rfc linkmailto="yes" ?>
<!-- Default linkmailto="yes" generate mailto: URL, as appropriate -->
<?rfc docmapping="no" ?>
<!-- Default docmapping="no" use hierarchical tags (e.g., <h1>, <h2>, etc.) for (sub)section titles -->
<?rfc slides="no" ?>
<!-- Default slides="no" when producing a html file, produce multiple files for a slide show -->
<front>
<title abbrev="Byte and Packet Congestion Notification">Byte and Packet
Congestion Notification</title>
<author fullname="Bob Briscoe" initials="B." surname="Briscoe">
<organization>BT & UCL</organization>
<address>
<postal>
<street>B54/77, Adastral Park</street>
<street>Martlesham Heath</street>
<city>Ipswich</city>
<code>IP5 3RE</code>
<country>UK</country>
</postal>
<phone>+44 1473 645196</phone>
<email>bob.briscoe@bt.com</email>
<uri>http://www.cs.ucl.ac.uk/staff/B.Briscoe/</uri>
</address>
</author>
<date day="07" month="August" year="2008" />
<area>Transport</area>
<workgroup>Transport Area Working Group</workgroup>
<keyword>Active queue management (AQM)</keyword>
<keyword>Availability</keyword>
<keyword>Denial of Service</keyword>
<keyword>Quality of Service (QoS)</keyword>
<keyword>Congestion Control</keyword>
<keyword>Fairness</keyword>
<keyword>Incentives</keyword>
<keyword>Protocol</keyword>
<keyword>Architecture layering</keyword>
<abstract>
<t>This memo concerns dropping or marking packets using active queue
management (AQM) such as random early detection (RED) or pre-congestion
notification (PCN). The primary conclusion is that packet size should be
taken into account when transports read congestion indications, not when
network equipment writes them. Reducing drop of small packets has some
tempting advantages: i) it drops less control packets, which tend to be
small and ii) it makes TCP's bit-rate less dependent on packet size.
However, there are ways of addressing these issues at the transport
layer, rather than reverse engineering network forwarding to fix
specific transport problems. Network layer algorithms like the byte-mode
packet drop variant of RED should not be used to drop fewer small
packets, because that creates a perverse incentive for transports to use
tiny segments, consequently also opening up a DoS vulnerability.</t>
</abstract>
</front>
<middle>
<!-- ================================================================ -->
<note title="Relationship to existing RFCs">
<t>To be removed by the RFC Editor on publication (with appropriate
changes to the 'Updates:' header and the RFC Index as appropriate).</t>
<t>This memo intends to update RFC2309, which stated an interim view but
requested that further research was needed on this topic.</t>
</note>
<note title="Changes from Previous Versions">
<t>To be removed by the RFC Editor on publication.</t>
<t>Full incremental diffs between each version are available at
<http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#byte-pkt-congest>
or
<http://tools.ietf.org/wg/tsvwg/draft-ietf-tsvwg-byte-pkt-congest/>
(courtesy of the rfcdiff tool): <list style="hanging">
<t hangText="From briscoe-byte-pkt-mark-02 to ietf-byte-pkt-congest-00 (this version):">
<list>
<t>Added note on relationship to existing RFCs</t>
<t>Posed the question of whether packet-congestion could become
common and deferred it to the IRTF ICCRG. Added ref to the
dual-resource queue (DRQ) proposal.</t>
<t>Changed PCN references from the PCN charter &
architecture to the PCN marking behaviour draft most likely to
imminently become the standards track WG item.</t>
</list>
</t>
<t hangText="From -01 to -02:">
<list>
<t>Abstract reorganised to align with clearer separation of
issue in the memo.</t>
<t>Introduction reorganised with motivating arguments removed to
new <xref target="pktb_Motivation" />.</t>
<t>Clarified avoiding lock-out of large packets is not the main
or only motivation for RED.</t>
<t>Mentioned choice of drop or marking explicitly throughout,
rather than trying to coin a word to mean either.</t>
<t>Generalised the discussion throughout to any packet
forwarding function on any network equipment, not just
routers.</t>
<t>Clarified the last point about why this is a good time to
sort out this issue: because it will be hard / impossible to
design new transports unless we decide whether the network or
the transport is allowing for packet size.</t>
<t>Added statement explaining the horizon of the memo is long
term, but with short term expediency in mind.</t>
<t>Added material on scaling congestion control with packet size
(<xref target="pktb_Scaling" />).</t>
<t>Separated out issue of normalising TCP's bit rate from issue
of preference to control packets (<xref
target="pktb_Small.NE.Control" />).</t>
<t>Divided up Congestion Measurement section for clarity,
including new material on fixed size packet buffers and buffer
carving (<xref target="pktb_Fixed_Buffers" /> & <xref
target="pktb_Network_Bias" />) and on congestion measurement in
wireless link technologies without queues (<xref
target="pktb_Measurement_NoQ" />).</t>
<t>Added section on 'Making Transports Robust against Control
Packet Losses' (<xref
target="pktb_Transport_Robust_Ctrl_Loss" />) with existing &
new material included.</t>
<t>Added tabulated results of vendor survey on byte-mode drop
variant of RED (<xref target="pktb_Tab_RED_Survey" />).</t>
<t />
</list>
</t>
<t hangText="From -00 to -01:">
<list>
<t>Clarified applicability to drop as well as ECN.</t>
<t>Highlighted DoS vulnerability.</t>
<t>Emphasised that drop-tail suffers from similar problems to
byte-mode drop, so only byte-mode drop should be turned off, not
RED itself.</t>
<t>Clarified the original apparent motivations for recommending
byte-mode drop included protecting SYNs and pure ACKs more than
equalising the bit rates of TCPs with different segment sizes.
Removed some conjectured motivations.</t>
<t>Added support for updates to TCP in progress (ackcc &
ecn-syn-ack).</t>
<t>Updated survey results with newly arrived data.</t>
<t>Pulled all recommendations together into the conclusions.</t>
<t>Moved some detailed points into two additional appendices and
a note.</t>
<t>Considerable clarifications throughout.</t>
<t>Updated references</t>
</list>
</t>
</list></t>
</note>
<!-- ================================================================ -->
<note title="Requirements notation">
<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" />.</t>
</note>
<!-- ================================================================ -->
<section anchor="pktb_Introduction" title="Introduction">
<t>When notifying congestion, the problem of how (and whether) to take
packet sizes into account has exercised the minds of researchers and
practitioners for as long as active queue management (AQM) has been
discussed. Indeed, one reason AQM was originally introduced was to
reduce the lock-out effects that small packets can have on large packets
in drop-tail queues. This memo aims to state the principles we should be
using and to come to conclusions on what these principles will mean for
future protocol design, taking into account the deployments we have
already.</t>
<t>Note that the byte vs. packet dilemma concerns congestion
notification irrespective of whether it is signalled implicitly by drop
or using explicit congestion notification (ECN <xref target="RFC3168" />
or PCN <xref target="I-D.eardley-pcn-marking-behaviour" />). Throughout
this document, unless clear from the context, the term marking will be
used to mean notifying congestion explicitly, while congestion
notification will be used to mean notifying congestion either implicitly
by drop or explicitly by marking.</t>
<t>If the load on a resource depends on the rate at which packets
arrive, it is called packet-congestible. If the load depends on the rate
at which bits arrive it is called bit-congestible.</t>
<t>Examples of packet-congestible resources are route look-up engines
and firewalls, because load depends on how many packet headers they have
to process. Examples of bit-congestible resources are transmission
links, and most buffer memory, because the load depends on how many bits
they have to transmit or store. Some machine architectures use fixed
size packet buffers, so buffer memory in these cases is
packet-congestible (see <xref target="pktb_Fixed_Buffers" />).</t>
<t>Note that information is generally processed or transmitted with a
minimum granularity greater than a bit (e.g. octets). The appropriate
granularity for the resource in question SHOULD be used, but for the
sake of brevity we will talk in terms of bytes in this memo.</t>
<t>Resources may be congestible at higher levels of granularity than
packets, for instance stateful firewalls are flow-congestible and
call-servers are session-congestible. This memo focuses on congestion of
connectionless resources, but the same principles may be applied for
congestion notification protocols controlling per-flow and per-session
processing or state.</t>
<t>The byte vs. packet dilemma arises at three stages in the congestion
notification process: <list style="hanging">
<t hangText="Measuring congestion">When the congested resource
decides locally how to measure how congested it is. (Should the
queue be measured in bytes or packets?);</t>
<t
hangText="Coding congestion notification into the wire protocol:">When
the congested resource decides how to notify the level of
congestion. (Should the level of notification depend on the
byte-size of each particular packet carrying the notification?);</t>
<t
hangText="Decoding congestion notification from the wire protocol:">When
the transport interprets the notification. (Should the byte-size of
a missing or marked packet be taken into account?).</t>
</list>In RED, whether to use packets or bytes when measuring queues
is called packet-mode or byte-mode queue measurement. This choice is now
fairly well understood but is included in <xref target="pktb_Measure" />
to document it in the RFC series.</t>
<t>The controversy is mainly around the other two stages: whether to
allow for packet size when the network codes or when the transport
decodes congestion notification. In RED, the variant that reduces drop
probability for packets based on their size in bytes is called byte-mode
drop, while the variant that doesn't is called packet mode drop. Whether
queues are measured in bytes or packets is an orthogonal choice, termed
byte-mode queue measurement or packet-mode queue measurement.</t>
<t>Currently, the RFC series is silent on this matter other than a paper
trail of advice referenced from <xref target="RFC2309" />, which
conditionally recommends byte-mode (packet-size dependent) drop <xref
target="pktByteEmail" />. However, all the implementers who responded to
our survey have not followed this advice. The primary purpose of this
memo is to build a definitive consensus against deliberate preferential
treatment for small packets in AQM algorithms and to record this advice
within the RFC series.</t>
<t>Now is a good time to discuss whether fairness between different
sized packets would best be implemented in the network layer, or at the
transport, for a number of reasons: <list style="numbers">
<t>The packet vs. byte issue requires speedy resolution because the
IETF pre-congestion notification (PCN) working group is about to
standardise the external behaviour of a PCN congestion notification
(AQM) algorithm <xref
target="I-D.eardley-pcn-marking-behaviour" />;</t>
<t><xref target="RFC2309" /> says RED may either take account of
packet size or not when dropping, but gives no recommendation
between the two, referring instead to advice on the performance
implications in an email <xref target="pktByteEmail" />, which
recommends byte-mode drop. Further, just before RFC2309 was issued,
an addendum was added to the archived email that revisited the issue
of packet vs. byte-mode drop in its last para, making the
recommendation less clear-cut;</t>
<t>Without the present memo, the only advice in the RFC series on
packet size bias in AQM algorithms would be a reference to an
archived email in <xref target="RFC2309" /> (including an addendum
at the end of the email to correct the original).</t>
<t>The IRTF Internet Congestion Control Research Group (ICCRG)
recently took on the challenge of building consensus on what common
congestion control support should be required from network
forwarding functions in future <xref
target="I-D.irtf-iccrg-welzl-congestion-control-open-research" />.
The wider Internet community needs to discuss whether the complexity
of adjusting for packet size should be in the network or in
transports;</t>
<t>Given there are many good reasons why larger path max
transmission units (PMTUs) would help solve a number of scaling
issues, we don't want to create any bias against large packets that
is greater than their true cost;</t>
<t>The IETF has started to consider the question of fairness between
flows that use different packet sizes (e.g. in the small-packet
variant of TCP-friendly rate control, TFRC-SP <xref
target="RFC4828" />). Given transports with different packet sizes,
if we don't decide whether the network or the transport should allow
for packet size, it will be hard if not impossible to design any
transport protocol so that its bit-rate relative to other transports
meets design guidelines <xref target="RFC5033" /> (Note however
that, if the concern were fairness between users, rather than
between flows <xref target="Rate_fair_Dis" />, relative rates
between flows would have to come under run-time control rather than
being embedded in protocol designs).</t>
</list></t>
<t>This memo is initially concerned with how we should correctly scale
congestion control functions with packet size for the long term. But it
also recognises that expediency may be necessary to deal with existing
widely deployed protocols that don't live up to the long term goal. It
turns out that the 'correct' variant of RED to deploy seems to be the
one everyone has deployed, and no-one who responded to our survey has
implemented the other variant. However, at the transport layer, TCP
congestion control is a widely deployed protocol that we argue doesn't
scale correctly with packet size. To date this hasn't been a significant
problem because most TCPs have been used with similar packet sizes. But,
as we design new congestion controls, we should build in scaling with
packet size rather than assuming we should follow TCP's example.</t>
<t>Motivating arguments for our advice are given next in <xref
target="pktb_Motivation" />. Then the body of the memo starts from first
principles, defining congestion notification in <xref
target="pktb_Congestion_Definition" /> then determining the correct way
to measure congestion (<xref target="pktb_Measure" />) and to design an
idealised congestion notification protocol (<xref
target="pktb_Ideal_Coding" />). It then surveys the advice given
previously in the RFC series, the research literature and the deployed
legacy (<xref target="pktb_SotA" />) before listing outstanding issues
(<xref target="pktb_Issues" />) that will need resolution both to
achieve the ideal protocol and to handle legacy. After discussing
security considerations (<xref target="pktb_Security_Considerations" />)
strong recommendations for the way forward are given in the conclusions
(<xref target="pktb_Conclusions" />).</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Motivation" title="Motivating Arguments">
<section anchor="pktb_Scaling"
title="Scaling Congestion Control with Packet Size">
<t>There are two ways of interpreting a dropped or marked packet. It
can either be considered as a single loss event or as loss/marking of
the bytes in the packet. Here we try to design a test to see which
approach scales with packet size.</t>
<t>Imagine a bit-congestible link shared by many flows, so that each
busy period tends to cause packets to be lost from different flows.
The test compares two identical scenarios with the same applications,
the same numbers of sources and the same load. But the sources break
the load into large packets in one scenario and small packets in the
other. Of course, because the load is the same, there will be
proportionately more packets in the small packet case.</t>
<t>The test of whether a congestion control scales with packet size is
that it should respond in the same way to the same congestion
excursion, irrespective of the size of the packets that the bytes
causing congestion happen to be broken down into.</t>
<t>A bit-congestible queue suffering a congestion excursion has to
drop or mark the same excess bytes whether they are in a few large
packets or many small packets. So for the same congestion excursion,
the same amount of bytes have to be shed to get the load back to its
operating point. But, of course, for smaller packets more packets will
have to be discarded to shed the same bytes.</t>
<t>If all the transports interpret each drop/mark as a single loss
event irrespective of the size of the packet dropped, they will
respond more to the same congestion excursion, failing our test. On
the other hand, if they respond proportionately less when smaller
packets are dropped/marked, overall they will be able to respond the
same to the same congestion excursion.</t>
<t>Therefore, for a congestion control to scale with packet size it
should respond to dropped or marked bytes (as TFRC-SP <xref
target="RFC4828" /> effectively does), not just to dropped or marked
packets irrespective of packet size (as TCP does).</t>
<t>The email <xref target="pktByteEmail" /> referred to by RFC2309
says the question of whether a packet's own size should affect its
drop probability "depends on the dominant end-to-end congestion
control mechanisms". But we argue the network layer should not be
optimised for whatever transport is predominant.</t>
<t>TCP congestion control ensures that flows competing for the same
resource each maintain the same number of segments in flight,
irrespective of segment size. So under similar conditions, flows with
different segment sizes will get different bit rates. But even though
reducing the drop probability of small packets helps ensure TCPs with
different packet sizes will achieve similar bit rates, we argue this
should be achieved in TCP itself, not in the network.</t>
<t>Effectively, favouring small packets is reverse engineering of the
network layer around TCP, contrary to the excellent advice in <xref
target="RFC3426" />, which asks designers to question "Why are you
proposing a solution at this layer of the protocol stack, rather than
at another layer?"</t>
</section>
<section anchor="pktb_Avoiding_Perverse_Incentives"
title="Avoiding Perverse Incentives to (ab)use Smaller Packets">
<t>Increasingly, it is being recognised that a protocol design must
take care not to cause unintended consequences by giving the parties
in the protocol exchange perverse incentives <xref
target="Evol_cc" /><xref target="RFC3426" />. Again, imagine a
scenario where the same bit rate of packets will contribute the same
to congestion of a link irrespective of whether it is sent as fewer
larger packets or more smaller packets. A protocol design that caused
larger packets to be more likely to be dropped than smaller ones would
be dangerous in this case:</t>
<t><list style="hanging">
<t hangText="Malicious transports:">A queue that gives an
advantage to small packets can be used to amplify the force of a
flooding attack. By sending a flood of small packets, the attacker
can get the queue to discard more traffic in large packets,
allowing more attack traffic to get through to cause further
damage. Such a queue allows attack traffic to have a
disproportionately large effect on regular traffic without the
attacker having to do much work. The byte-mode drop variant of RED
amplifies small packet attacks. Drop-tail queues amplify small
packet attacks even more than RED byte-mode drop (see the Security
Considerations section <xref
target="pktb_Security_Considerations" />). Wherever possible
neither should be used.</t>
<t hangText="Normal transports:">Even if a transport is not
malicious, if it finds small packets go faster, it will tend to
act in its own interest and use them. Queues that give advantage
to small packets create an evolutionary pressure for transports to
send at the same bit-rate but break their data stream down into
tiny segments to reduce their drop rate. Encouraging a high volume
of tiny packets might in turn unnecessarily overload a completely
unrelated part of the system, perhaps more limited by
header-processing than bandwidth.</t>
</list>Imagine two flows arrive at a bit-congestible transmission
link each with the same bit rate, say 1Mbps, but one consists of 1500B
and the other 60B packets, which are 25x smaller. Consider a scenario
where gentle RED <xref target="gentle_RED" /> is used, along with the
variant of RED we advise against, i.e. where the RED algorithm is
configured to adjust the drop probability of packets in proportion to
each packet's size (byte mode packet drop). In this case, if RED drops
25% of the larger packets, it will aim to drop 1% of the smaller
packets (but in practice it may drop more as congestion increases
<xref target="RFC4828" />(§B.4)<cref anchor="Note_Variation">The
algorithm of the byte-mode drop variant of RED switches off any bias
towards small packets whenever the smoothed queue length dictates that
the drop probability of large packets should be 100%. In the example
in the Introduction, as the large packet drop probability varies
around 25% the small packet drop probability will vary around 1%, but
with occasional jumps to 100% whenever the instantaneous queue (after
drop) manages to sustain a length above the 100% drop point for longer
than the queue averaging period.</cref>). Even though both flows
arrive with the same bit rate, the bit rate the RED queue aims to pass
to the line will be 750k for the flow of larger packet but 990k for
the smaller packets (but because of rate variation it will be less
than this target). It can be seen that this behaviour reopens the same
denial of service vulnerability that drop tail queues offer to floods
of small packet, though not necessarily as strongly (see <xref
target="pktb_Security_Considerations" />).</t>
</section>
<section anchor="pktb_Small.NE.Control" title="Small != Control">
<t>It is tempting to drop small packets with lower probability to
improve performance, because many control packets are small (TCP SYNs
& ACKs, DNS queries & responses, SIP messages, HTTP GETs, etc)
and dropping fewer control packets considerably improves performance.
However, we must not give control packets preference purely by virtue
of their smallness, otherwise it is too easy for any data source to
get the same preferential treatment simply by sending data in smaller
packets. Again we should not create perverse incentives to favour
small packets rather than to favour control packets, which is what we
intend.</t>
<t>Just because many control packets are small does not mean all small
packets are control packets.</t>
<t>So again, rather than fix these problems in the network layer, we
argue that the transport should be made more robust against losses of
control packets (see 'Making Transports Robust against Control Packet
Losses' in <xref target="pktb_Transport_Robust_Ctrl_Loss" />).</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Congestion_Definition"
title="Working Definition of Congestion Notification">
<t>Rather than aim to achieve what many have tried and failed, this memo
will not try to define congestion. It will give a working definition of
what congestion notification should be taken to mean for this document.
Congestion notification is a changing signal that aims to communicate
the ratio E/L, where E is the instantaneous excess load offered to a
resource that it cannot (or would not) serve and L is the instantaneous
offered load.</t>
<t>The phrase `would not serve' is added, because AQM systems (e.g. RED,
PCN <xref target="I-D.eardley-pcn-marking-behaviour" />) use a virtual
capacity smaller than actual capacity, then notify congestion of this
virtual capacity in order to avoid congestion of the actual
capacity.</t>
<t>Note that the denominator is offered load, not capacity. Therefore
congestion notification is a real number bounded by the range [0,1].
This ties in with the most well-understood form of congestion
notification: drop rate. It also means that congestion has a natural
interpretation as a probability; the probability of offered traffic not
being served (or being marked as at risk of not being served). <xref
target="pktb_CN_Definition" /> describes a further incidental benefit
that arises from using load as the denominator of congestion
notification.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Measure" title="Congestion Measurement">
<section anchor="pktb_Measurement_Q"
title="Congestion Measurement by Queue Length">
<t>Queue length is usually the most correct and simplest way to
measure congestion of a resource. To avoid the pathological effects of
drop tail, an AQM function can then be used to transform queue length
into the probability of dropping or marking a packet (e.g. RED's
piecewise linear function between thresholds). If the resource is
bit-congestible, the length of the queue SHOULD be measured in bytes.
If the resource is packet-congestible, the length of the queue SHOULD
be measured in packets. No other choice makes sense, because the
number of packets waiting in the queue isn't relevant if the resource
gets congested by bytes and vice versa. We discuss the implications on
RED's byte mode and packet mode for measuring queue length in <xref
target="pktb_SotA" />.</t>
<section anchor="pktb_Fixed_Buffers" title="Fixed Size Packet Buffers">
<t>Some, mostly older, queuing hardware sets aside fixed sized
buffers in which to store each packet in the queue. Also, with some
hardware, any fixed sized buffers not completely filled by a packet
are padded when transmitted to the wire. If we imagine a theoretical
forwarding system with both queuing and transmission in fixed,
MTU-sized units, it should clearly be treated as packet-congestible,
because the queue length in packets would be a good model of
congestion of the lower layer link.</t>
<t>If we now imagine a hybrid forwarding system with transmission
delay largely dependent on the byte-size of packets but buffers of
one MTU per packet, it should strictly require a more complex
algorithm to determine the probability of congestion. It should be
treated as two resources in sequence, where the sum of the
byte-sizes of the packets within each packet buffer models
congestion of the line while the length of the queue in packets
models congestion of the queue. Then the probability of congesting
the forwarding buffer would be a conditional
probability—conditional on the previously calculated
probability of congesting the line.</t>
<t>However, in systems that use fixed size buffers, it is unusual
for all the buffers used by an interface to be the same size.
Typically pools of different sized buffers are provided (Cisco uses
the term 'buffer carving' for the process of dividing up memory into
these pools <xref target="IOSArch" />). Usually, if the pool of
small buffers is exhausted, arriving small packets can borrow space
in the pool of large buffers, but not vice versa. However, it is
easier to work out what should be done if we temporarily set aside
the possibility of such borrowing. Then, with fixed pools of buffers
for different sized packets and no borrowing, the size of each pool
and the current queue length in each pool would both be measured in
packets. So an AQM algorithm would have to maintain the queue length
for each pool, and judge whether to drop/mark a packet of a
particular size by looking at the pool for packets of that size and
using the length (in packets) of its queue.</t>
<t>We now return to the issue we temporarily set aside: small
packets borrowing space in larger buffers. In this case, the only
difference is that the pools for smaller packets have a maximum
queue size that includes all the pools for larger packets. And every
time a packet takes a larger buffer, the current queue size has to
be incremented for all queues in the pools of buffers less than or
equal to the buffer size used.</t>
<t>We will return to borrowing of fixed sized buffers when we
discuss biasing the drop/marking probability of a specific packet
because of its size in <xref target="pktb_Network_Bias" />. But here
we can give a simple summary of the present discussion on how to
measure the length of queues of fixed buffers: no matter how
complicated the scheme is, ultimately any fixed buffer system will
need to measure its queue length in packets not bytes.</t>
</section>
</section>
<section anchor="pktb_Measurement_NoQ"
title="Congestion Measurement without a Queue">
<t>AQM algorithms are nearly always described assuming there is a
queue for a congested resource and the algorithm can use the queue
length to determine the probability that it will drop or mark each
packet. But not all congested resources lead to queues. For instance,
wireless spectrum is bit-congestible (for a given coding scheme),
because interference increases with the rate at which bits are
transmitted. But wireless link protocols do not always maintain a
queue that depends on spectrum interference. Similarly, power limited
resources are also usually bit-congestible if energy is primarily
required for transmission rather than header processing, but it is
rare for a link protocol to build a queue as it approaches maximum
power.</t>
<t>However, AQM algorithms don't require a queue in order to work. For
instance spectrum congestion can be modelled by signal quality using
target bit-energy-to-noise-density ratio. And, to model radio power
exhaustion, transmission power levels can be measured and compared to
the maximum power available. <xref target="ECNFixedWireless" />
proposes a practical and theoretically sound way to combine congestion
notification for different bit-congestible resources at different
layers along an end to end path, whether wireless or wired, and
whether with or without queues.</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Ideal_Coding" title="Idealised Wire Protocol Coding">
<t>We will start by inventing an idealised congestion notification
protocol before discussing how to make it practical. The idealised
protocol is shown to be correct using examples in <xref
target="pktb_Scenarios" />.</t>
<t>Congestion notification involves the congested resource coding a
congestion notification signal into the packet stream and the transports
decoding it. The idealised protocol uses two different (imaginary)
fields in each datagram to signal congestion: one for byte congestion
and one for packet congestion.</t>
<t>We are not saying two ECN fields will be needed (and we are not
saying that somehow a resource should be able to drop a packet in one of
two different ways so that the transport can distinguish which sort of
drop it was!). These two congestion notification channels are just a
conceptual device. They allow us to defer having to decide whether to
distinguish between byte and packet congestion when the network resource
codes the signal or when the transport decodes it.</t>
<t>However, although this idealised mechanism isn't intended for
implementation, we do want to emphasise that we may need to find a way
to implement it, because it could become necessary to somehow
distinguish between bit and packet congestion <xref target="RFC3714" />.
Currently a design goal of network processing equipment such as routers
and firewalls is to keep packet processing uncongested even under worst
case bit rates with minimum packet sizes. Therefore, packet-congestion
is currently rare, but there is no guarantee that it will not become
common with future technology trends.</t>
<t>The idealised wire protocol is given below. It accounts for packet
sizes at the transport layer, not in the network, and then only in the
case of bit-congestible resources. This avoids the perverse incentive to
send smaller packets and the DoS vulnerability that would otherwise
result if the network were to bias towards them (see the motivating
argument about avoiding perverse incentives in <xref
target="pktb_Avoiding_Perverse_Incentives" />). Incidentally, it also
ensures neither the network nor the transport needs to do a multiply
operation—multiplication by packet size is effectively achieved as
a repeated add when the transport adds to its count of marked bytes as
each congestion event is fed to it: <list style="symbols">
<t>A packet-congestible resource trying to code congestion level p_p
into a packet stream should mark the idealised `packet congestion'
field in each packet with probability p_p irrespective of the
packet's size. The transport should then take a packet with the
packet congestion field marked to mean just one mark, irrespective
of the packet size.</t>
<t>A bit-congestible resource trying to code time-varying
byte-congestion level p_b into a packet stream should mark the `byte
congestion' field in each packet with probability p_b, again
irrespective of the packet's size. Unlike before, the transport
should take a packet with the byte congestion field marked to count
as a mark on each byte in the packet.</t>
</list></t>
<t>The worked examples in <xref target="pktb_Scenarios" /> show that
transports can extract sufficient and correct congestion notification
from these protocols for cases when two flows with different packet
sizes have matching bit rates or matching packet rates. Examples are
also given that mix these two flows into one to show that a flow with
mixed packet sizes would still be able to extract sufficient and correct
information.</t>
<t>Sufficient and correct congestion information means that there is
sufficient information for the two different types of transport
requirements: <list style="hanging">
<t hangText="Ratio-based:">Established transport congestion controls
like TCP's <xref target="RFC2581" /> aim to achieve equal segment
rates per RTT through the same bottleneck—TCP friendliness
<xref target="RFC3448" />. They work with the ratio of dropped to
delivered segments (or marked to unmarked segments in the case of
ECN). The example scenarios show that these ratio-based transports
are effectively the same whether counting in bytes or packets,
because the units cancel out. (Incidentally, this is why TCP's bit
rate is still proportional to packet size even when byte-counting is
used, as recommended for TCP in <xref
target="I-D.ietf-tcpm-rfc2581bis" />, mainly for orthogonal security
reasons.)</t>
<t hangText="Absolute-target-based:">Other congestion controls
proposed in the research community aim to limit the volume of
congestion caused to a constant weight parameter. <xref
target="MulTCP" /><xref target="WindowPropFair" /> are examples of
weighted proportionally fair transports designed for cost-fair
environments <xref target="Rate_fair_Dis" />. In this case, the
transport requires a count (not a ratio) of dropped/marked bytes in
the bit-congestible case and of dropped/marked packets in the packet
congestible case.</t>
</list></t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_SotA" title="The State of the Art">
<t>The original 1993 paper on RED <xref target="RED93" /> proposed two
options for the RED active queue management algorithm: packet mode and
byte mode. Packet mode measured the queue length in packets and dropped
(or marked) individual packets with a probability independent of their
size. Byte mode measured the queue length in bytes and marked an
individual packet with probability in proportion to its size (relative
to the maximum packet size). In the paper's outline of further work, it
was stated that no recommendation had been made on whether the queue
size should be measured in bytes or packets, but noted that the
difference could be significant.</t>
<t>When RED was recommended for general deployment in 1998 <xref
target="RFC2309" />, the two modes were mentioned implying the choice
between them was a question of performance, referring to a 1997 email
<xref target="pktByteEmail" /> for advice on tuning. This email
clarified that there were in fact two orthogonal choices: whether to
measure queue length in bytes or packets (<xref
target="pktb_Measure_Status" /> below) and whether the drop probability
of an individual packet should depend on its own size (<xref
target="pktb_Coding_Status" /> below).</t>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Measure_Status"
title="Congestion Measurement: Status">
<t>The choice of which metric to use to measure queue length was left
open in RFC2309. It is now well understood that queues for
bit-congestible resources should be measured in bytes, and queues for
packet-congestible resources should be measured in packets (see <xref
target="pktb_Measure" />).</t>
<t>Where buffers are not configured or legacy buffers cannot be
configured to the above guideline, we don't have to make allowances
for such legacy in future protocol design. If a bit-congestible buffer
is measured in packets, the operator will have set the thresholds
mindful of a typical mix of packets sizes. Any AQM algorithm on such a
buffer will be oversensitive to high proportions of small packets,
e.g. a DoS attack, and undersensitive to high proportions of large
packets. But an operator can safely keep such a legacy buffer because
any undersensitivity during unusual traffic mixes cannot lead to
congestion collapse given the buffer will eventually revert to tail
drop, discarding proportionately more large packets.</t>
<t>Some modern queue implementations give a choice for setting RED's
thresholds in byte-mode or packet-mode. This may merely be an
administrator-interface preference, not altering how the queue itself
is measured but on some hardware it does actually change the way it
measures its queue. Whether a resource is bit-congestible or
packet-congestible is a property of the resource, so an admin SHOULD
NOT ever need to, or be able to, configure the way a queue measures
itself.</t>
<t>We believe the question of whether to measure queues in bytes or
packets is fairly well understood these days. The only outstanding
issues concern how to measure congestion when the queue is bit
congestible but the resource is packet congestible or vice versa (see
<xref target="pktb_Measure" />). But there is no controversy over what
should be done. It's just you have to be an expert in probability to
work out what should be done and, even if you have, it's not always
easy to find a practical algorithm to implement it.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Coding_Status" title="Congestion Coding: Status">
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Network_Bias" title="Network Bias when Encoding">
<t>The previously mentioned email <xref target="pktByteEmail" />
referred to by <xref target="RFC2309" /> said that the choice over
whether a packet's own size should affect its drop probability
"depends on the dominant end-to-end congestion control mechanisms".
[<xref target="pktb_Motivation" /> argues against this approach,
citing the excellent advice in RFC3246.] The referenced email went
on to argue that drop probability should depend on the size of the
packet being considered for drop if the resource is bit-congestible,
but not if it is packet-congestible, but advised that most scarce
resources in the Internet were currently bit-congestible. The
argument continued that if packet drops were inflated by packet size
(byte-mode dropping), "a flow's fraction of the packet drops is then
a good indication of that flow's fraction of the link bandwidth in
bits per second". This was consistent with a referenced policing
mechanism being worked on at the time for detecting unusually high
bandwidth flows, eventually published in 1999 <xref
target="pBox" />. [The problem could have been solved by making the
policing mechanism count the volume of bytes randomly dropped, not
the number of packets.]</t>
<t>A few months before RFC2309 was published, an addendum was added
to the above archived email referenced from the RFC, in which the
final paragraph seemed to partially retract what had previously been
said. It clarified that the question of whether the probability of
dropping/marking a packet should depend on its size was not related
to whether the resource itself was bit congestible, but a completely
orthogonal question. However the only example given had the queue
measured in packets but packet drop depended on the byte-size of the
packet in question. No example was given the other way round.</t>
<t>In 2000, Cnodder et al <xref target="REDbyte" /> pointed out that
there was an error in the part of the original 1993 RED algorithm
that aimed to distribute drops uniformly, because it didn't
correctly take into account the adjustment for packet size. They
recommended an algorithm called RED_4 to fix this. But they also
recommended a further change, RED_5, to adjust drop rate dependent
on the square of relative packet size. This was indeed consistent
with one stated motivation behind RED's byte mode drop—that we
should reverse engineer the network to improve the performance of
dominant end-to-end congestion control mechanisms.</t>
<t>By 2003, a further change had been made to the adjustment for
packet size, this time in the RED algorithm of the ns2 simulator.
Instead of taking each packet's size relative to a `maximum packet
size' it was taken relative to a `mean packet size', intended to be
a static value representative of the `typical' packet size on the
link. We have not been able to find a justification for this change
in the literature, however Eddy and Allman conducted experiments
<xref target="REDbias" /> that assessed how sensitive RED was to
this parameter, amongst other things. No-one seems to have pointed
out that this changed algorithm can often lead to drop probabilities
of greater than 1 [which should ring alarm bells hinting that
there's a mistake in the theory somewhere]. On 10-Nov-2004, this
variant of byte-mode packet drop was made the default in the ns2
simulator.</t>
<t>The byte-mode drop variant of RED is, of course, not the only
possible bias towards small packets in queueing algorithms. We have
already mentioned that tail-drop queues naturally tend to lock-out
large packets once they are full. But also queues with fixed sized
buffers reduce the probability that small packets will be dropped if
(and only if) they allow small packets to borrow buffers from the
pools for larger packets. As was explained in <xref
target="pktb_Fixed_Buffers" /> on fixed size buffer carving,
borrowing effectively makes the maximum queue size for small packets
greater than that for large packets, because more buffers can be
used by small packets while less will fit large packets.</t>
<t>However, in itself, the bias towards small packets caused by
buffer borrowing is perfectly correct. Lower drop probability for
small packets is legitimate in buffer borrowing schemes, because
small packets genuinely congest the machine's buffer memory less
than large packets, given they can fit in more spaces. The bias
towards small packets is not artificially added (as it is in RED's
byte-mode drop algorithm), it merely reflects the reality of the way
fixed buffer memory gets congested. Incidentally, the bias towards
small packets from buffer borrowing is nothing like as large as that
of RED's byte-mode drop.</t>
<t>Nonetheless, fixed-buffer memory with tail drop is still prone to
lock-out large packets, purely because of the tail-drop aspect. So a
good AQM algorithm like RED with packet-mode drop should be used
with fixed buffer memories where possible. If RED is too complicated
to implement with multiple fixed buffer pools, the minimum necessary
to prevent large packet lock-out is to ensure smaller packets never
use the last available buffer in any of the pools for larger
packets.</t>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Transport_Bias"
title="Transport Bias when Decoding">
<t>The above proposals to alter the network layer to give a bias
towards smaller packets have largely carried on outside the IETF
process (unless one counts a reference in an informational RFC to an
archived email!). Whereas, within the IETF, there are many different
proposals to alter transport protocols to achieve the same goals,
i.e. either to make the flow bit-rate take account of packet size,
or to protect control packets from loss. This memo argues that
altering transport protocols is the more principled approach.</t>
<t>A recently approved experimental RFC adapts its transport layer
protocol to take account of packet sizes relative to typical TCP
packet sizes. This proposes a new small-packet variant of
TCP-friendly rate control <xref target="RFC3448" /> called TFRC-SP
<xref target="RFC4828" />. Essentially, it proposes a rate equation
that inflates the flow rate by the ratio of a typical TCP segment
size (1500B including TCP header) over the actual segment size <xref
target="PktSizeEquCC" />. (There are also other important
differences of detail relative to TFRC, such as using virtual
packets <xref target="CCvarPktSize" /> to avoid responding to
multiple losses per round trip and using a minimum inter-packet
interval.)</t>
<t>Section 4.5.1 of this TFRC-SP spec discusses the implications of
operating in an environment where queues have been configured to
drop smaller packets with proportionately lower probability than
larger ones. But it only discusses TCP operating in such an
environment, only mentioning TFRC-SP briefly when discussing how to
define fairness with TCP. And it only discusses the byte-mode
dropping version of RED as it was before Cnodder et al pointed out
it didn't sufficiently bias towards small packets to make TCP
independent of packet size.</t>
<t>So the TFRC-SP spec doesn't address the issue of which of the
network or the transport <spanx style="emph">should</spanx> handle
fairness between different packet sizes. In its Appendix B.4 it
discusses the possibility of both TFRC-SP and some network buffers
duplicating each other's attempts to deliberately bias towards small
packets. But the discussion is not conclusive, instead reporting
simulations of many of the possibilities in order to assess
performance but not recommending any particular course of
action.</t>
<t>The paper originally proposing TFRC with virtual packets
(VP-TFRC) <xref target="CCvarPktSize" /> proposed that there should
perhaps be two variants to cater for the different variants of RED.
However, as the TFRC-SP authors point out, there is no way for a
transport to know whether some queues on its path have deployed RED
with byte-mode packet drop (except if an exhaustive survey found
that no-one has deployed it!—see <xref
target="pktb_Coding_Status_Summary" />). Incidentally, VP-TFRC also
proposed that byte-mode RED dropping should really square the packet
size compensation factor (like that of RED_5, but apparently unaware
of it).</t>
<t>Pre-congestion notification <xref
target="I-D.eardley-pcn-marking-behaviour" /> is a proposal to use a
virtual queue for AQM marking for packets within one Diffserv class
in order to give early warning prior to any real queuing. The
proposed PCN marking algorithms have been designed not to take
account of packet size when forwarding through queues. Instead the
general principle has been to take account of the sizes of marked
packets when monitoring the fraction of marking at the edge of the
network.</t>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Transport_Robust_Ctrl_Loss"
title="Making Transports Robust against Control Packet Losses">
<t>Recently, two drafts have proposed changes to TCP that make it
more robust against losing small control packets <xref
target="I-D.ietf-tcpm-ecnsyn" /> <xref
target="I-D.floyd-tcpm-ackcc" />. In both cases they note that the
case for these TCP changes would be weaker if RED were biased
against dropping small packets. We argue here that these two
proposals are a safer and more principled way to achieve TCP
performance improvements than reverse engineering RED to benefit
TCP.</t>
<t>Although no proposals exist as far as we know, it would also be
possible and perfectly valid to make control packets robust against
drop by explicitly requesting a lower drop probability using their
Diffserv code point <xref target="RFC2474" /> to request a
scheduling class with lower drop.</t>
<t>The re-ECN protocol proposal <xref target="Re-TCP" /> is designed
so that transports can be made more robust against losing control
packets. It gives queues an incentive to optionally give preference
against drop to packets with the 'feedback not established'
codepoint in the proposed 'extended ECN' field. Senders have
incentives to use this codepoint sparingly, but they can use it on
control packets to reduce their chance of being dropped. For
instance, the proposed modification to TCP for re-ECN uses this
codepoint on the SYN and SYN-ACK.</t>
<t>Although not brought to the IETF, a simple proposal from Wischik
<xref target="DupTCP" /> suggests that the first three packets of
every TCP flow should be routinely duplicated after a short delay.
It shows that this would greatly improve the chances of short flows
completing quickly, but it would hardly increase traffic levels on
the Internet, because Internet bytes have always been concentrated
in the large flows. It further shows that the performance of many
typical applications depends on completion of long serial chains of
short messages. It argues that, given most of the value people get
from the Internet is concentrated within short flows, this simple
expedient would greatly increase the value of the best efforts
Internet at minimal cost.</t>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Coding_Status_Summary"
title="Congestion Coding: Summary of Status">
<?rfc needLines="6" ?>
<texttable anchor="pktb_Tab_TFRC-SP"
title="Dependence of flow bit-rate per RTT on packet size s and drop rate p when network and/or transport bias towards small packets to varying degrees">
<ttcol align="right">transport cc</ttcol>
<ttcol align="center">RED_1 (packet mode drop)</ttcol>
<ttcol align="center">RED_4 (linear byte mode drop)</ttcol>
<ttcol align="center">RED_5 (square byte mode drop)</ttcol>
<c>TCP or TFRC</c>
<c>s/sqrt(p)</c>
<c>sqrt(s/p)</c>
<c>1/sqrt(p)</c>
<c>TFRC-SP</c>
<c>1/sqrt(p)</c>
<c>1/sqrt(sp)</c>
<c>1/(s.sqrt(p))</c>
</texttable>
<t><xref target="pktb_Tab_TFRC-SP" /> aims to summarise the
positions we may now be in. Each column shows a different possible
AQM behaviour in different queues in the network, using the
terminology of Cnodder et al outlined earlier (RED_1 is basic RED
with packet-mode drop). Each row shows a different transport
behaviour: TCP <xref target="RFC2581" /> and TFRC <xref
target="RFC3448" /> on the top row with TFRC-SP <xref
target="RFC4828" /> below. Suppressing all inessential details the
table shows that independence from packet size should either be
achievable by not altering the TCP transport in a RED_5 network, or
using the small packet TFRC-SP transport in a network without any
byte-mode dropping RED (top right and bottom left). Top left is the
`do nothing' scenario, while bottom right is the `do-both' scenario
in which bit-rate would become far too biased towards small packets.
Of course, if any form of byte-mode dropping RED has been deployed
on a selection of congested queues, each path will present a
different hybrid scenario to its transport.</t>
<t>Whatever, we can see that the linear byte-mode drop column in the
middle considerably complicates the Internet. It's a half-way house
that doesn't bias enough towards small packets even if one believes
the network should be doing the biasing. We argue below that <spanx
style="emph">all</spanx> network layer bias towards small packets
should be turned off—if indeed any equipment vendors have
implemented it—leaving packet size bias solely as the preserve
of the transport layer (solely the leftmost, packet-mode drop
column).</t>
<t>A survey has been conducted of 84 vendors to assess how widely
drop probability based on packet size has been implemented in RED.
Prior to the survey, an individual approach to Cisco received
confirmation that, having checked the code-base for each of the
product ranges, Cisco has not implemented any discrimination based
on packet size in any AQM algorithm in any of its products. Also an
individual approach to Alcatel-Lucent drew a confirmation that it
was very likely that none of their products contained RED code that
implemented any packet-size bias.</t>
<t>Turning to our more formal survey (<xref
target="pktb_Tab_RED_Survey" />), about 19% of those surveyed have
replied so far, giving a sample size of 16. Although we do not have
permission to identify the respondents, we can say that those that
have responded include most of the larger vendors, covering a large
fraction of the market. They range across the large network
equipment vendors at L3 & L2, firewall vendors, wireless
equipment vendors, as well as large software businesses with a small
selection of networking products. So far, all those who have
responded have confirmed that they have not implemented the variant
of RED with drop dependent on packet size (2 are fairly sure they
haven't but need to check more thoroughly).</t>
<texttable anchor="pktb_Tab_RED_Survey"
title="Vendor Survey on byte-mode drop variant of RED (lower drop probability for small packets)">
<preamble />
<ttcol align="right">Response</ttcol>
<ttcol align="right">No. of vendors</ttcol>
<ttcol align="right">%age of vendors</ttcol>
<c>Not implemented</c>
<c>14</c>
<c>17%</c>
<c>Not implemented (probably)</c>
<c>2</c>
<c>2%</c>
<c>Implemented</c>
<c>0</c>
<c>0%</c>
<c>No response</c>
<c>68</c>
<c>81%</c>
<c>Total companies/orgs surveyed</c>
<c>84</c>
<c>100%</c>
<postamble />
</texttable>
<t>Where reasons have been given, the extra complexity of packet
bias code has been most prevalent, though one vendor had a more
principled reason for avoiding it—similar to the argument of
this document. We have established that Linux does not implement RED
with packet size drop bias, although we have not investigated a
wider range of open source code.</t>
<t>Finally, we repeat that RED's byte mode drop is not the only way
to bias towards small packets—tail-drop tends to lock-out
large packets very effectively. Our survey was of vendor
implementations, so we cannot be certain about operator deployment.
But we believe many queues in the Internet are still tail-drop. My
own company (BT) has widely deployed RED, but there are bound to be
many tail-drop queues, particularly in access network equipment and
on middleboxes like firewalls, where RED is not always available.
Routers using a memory architecture based on fixed size buffers with
borrowing may also still be prevalent in the Internet. As explained
in <xref target="pktb_Network_Bias" />, these also provide a
marginal (but legitimate) bias towards small packets. So even though
RED byte-mode drop is not prevalent, it is likely there is still
some bias towards small packets in the Internet due to tail drop and
fixed buffer borrowing.</t>
</section>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Issues" title="Outstanding Issues and Next Steps">
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-World" title="Bit-congestible World">
<t>For a connectionless network with nearly all resources being
bit-congestible we believe the recommended position is now unarguably
clear—that the network should not make allowance for packet
sizes and the transport should. This leaves two outstanding issues:
<list style="symbols">
<t>How to handle any legacy of AQM with byte-mode drop already
deployed;</t>
<t>The need to start a programme to update transport congestion
control protocol standards to take account of packet size.</t>
</list></t>
<t>The sample of returns from our vendor survey <xref
target="pktb_Coding_Status_Summary" /> suggest that byte-mode packet
drop seems not to be implemented at all let alone deployed, or if it
is, it is likely to be very sparse. Therefore, we do not really need a
migration strategy from all but nothing to nothing.</t>
<t>A programme of standards updates to take account of packet size in
transport congestion control protocols has started with TFRC-SP <xref
target="RFC4828" />, while weighted TCPs implemented in the research
community <xref target="WindowPropFair" /> could form the basis of a
future change to TCP congestion control <xref target="RFC2581" />
itself.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-Pkt-World"
title="Bit- & Packet-congestible World">
<t>Nonetheless, a connectionless network with both bit-congestible and
packet-congestible resources is a different matter. If we believe we
should allow for this possibility in the future, this space contains a
truly open research issue.</t>
<t>The idealised wire protocol coding described in <xref
target="pktb_Ideal_Coding" /> requires at least two flags for
congestion of bit-congestible and packet-congestible resources. This
hides a fundamental problem—much more fundamental than whether
we can magically create header space for yet another ECN flag in IPv4,
or whether it would work while being deployed incrementally. A
congestion notification protocol must survive a transition from low
levels of congestion to high. Marking two states is feasible with
explicit marking, but much harder if packets are dropped. Also, it
will not always be cost-effective to implement AQM at every low level
resource, so drop will often have to suffice. Distinguishing drop from
delivery naturally provides just one congestion flag—it is hard
to drop a packet in two ways that are distinguishable remotely. This
is a similar problem to that of distinguishing wireless transmission
losses from congestive losses.</t>
<t>We should also note that, strictly, packet-congestible resources
are actually cycle-congestible because load also depends on the
complexity of each look-up and whether the pattern of arrivals is
amenable to caching or not. Further, this reminds us that any solution
must not require a forwarding engine to use excessive processor cycles
in order to decide how to say it has no spare processor cycles.</t>
<t>Recently, the dual resource queue (DRQ) proposal <xref
target="DRQ" /> has been made on the premise that, as network
processors become more cost effective, per packet operations will
become more complex (irrespective of whether more function in the
network layer is desirable). Consequently the premise is that CPU
congestion will become more common. DRQ is a proposed modification to
the RED algorithm that folds both bit congestion and packet congestion
into one signal (either loss or ECN).</t>
<t>The problem of signalling packet processing congestion is not
pressing, as most Internet resources are designed to be
bit-congestible before packet processing starts to congest. However,
the IRTF Internet congestion control research group (ICCRG) has set
itself the task of reaching consensus on generic forwarding mechanisms
that are necessary and sufficient to support the Internet's future
congestion control requirements (the first challenge in <xref
target="I-D.irtf-iccrg-welzl-congestion-control-open-research" />).
Therefore, rather than not giving this problem any thought at all,
just because it is hard and currently hypothetical, we defer the
question of whether packet congestion might become common and what to
do if it does to the IRTF (the 'Small Packets' challenge in <xref
target="I-D.irtf-iccrg-welzl-congestion-control-open-research" />).</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Security_Considerations"
title="Security Considerations">
<t>This draft recommends that queues do not bias drop probability
towards small packets as this creates a perverse incentive for
transports to break down their flows into tiny segments. One of the
benefits of implementing AQM was meant to be to remove this perverse
incentive that drop-tail queues gave to small packets. Of course, if
transports really want to make the greatest gains, they don't have to
respond to congestion anyway. But we don't want applications that are
trying to behave to discover that they can go faster by using smaller
packets.</t>
<t>In practice, transports cannot all be trusted to respond to
congestion. So another reason for recommending that queues do not bias
drop probability towards small packets is to avoid the vulnerability to
small packet DDoS attacks that would otherwise result. One of the
benefits of implementing AQM was meant to be to remove drop-tail's DoS
vulnerability to small packets, so we shouldn't add it back again.</t>
<t>If most queues implemented AQM with byte-mode drop, the resulting
network would amplify the potency of a small packet DDoS attack. At the
first queue the stream of packets would push aside a greater proportion
of large packets, so more of the small packets would survive to attack
the next queue. Thus a flood of small packets would continue on towards
the destination, pushing regular traffic with large packets out of the
way in one queue after the next, but suffering much less drop
itself.</t>
<t><xref target="pktb_Policing_Congestion_Response" /> explains why the
ability of networks to police the response of <spanx
style="emph">any</spanx> transport to congestion depends on
bit-congestible network resources only doing packet-mode not byte-mode
drop. In summary, it says that making drop probability depend on the
size of the packets that bits happen to be divided into simply
encourages the bits to be divided into smaller packets. Byte-mode drop
would therefore irreversibly complicate any attempt to fix the
Internet's incentive structures.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Conclusions" title="Conclusions">
<t>The strong conclusion is that AQM algorithms such as RED SHOULD NOT
use byte-mode drop. More generally, the Internet's congestion
notification protocols (drop, ECN & PCN) SHOULD take account of
packet size when the notification is read by the transport layer, NOT
when it is written by the network layer. This approach offers sufficient
and correct congestion information for all known and future transport
protocols and also ensures no perverse incentives are created that would
encourage transports to use inappropriately small packet sizes.</t>
<t>The alternative of deflating RED's drop probability for smaller
packet sizes (byte-mode drop) has no enduring advantages. It is more
complex, it creates the perverse incentive to fragment segments into
tiny pieces and it reopens the vulnerability to floods of small-packets
that drop-tail queues suffered from and AQM was designed to remove.
Byte-mode drop is a change to the network layer that makes allowance for
an omission from the design of TCP, effectively reverse engineering the
network layer to contrive to make two TCPs with different packet sizes
run at equal bit rates (rather than packet rates) under the same path
conditions. It also improves TCP performance by reducing the chance that
a SYN or a pure ACK will be dropped, because they are small. But we
SHOULD NOT hack the network layer to improve or fix certain transport
protocols. No matter how predominant a transport protocol is (even if
it's TCP), trying to correct for its failings by biasing towards small
packets in the network layer creates a perverse incentive to break down
all flows from all transports into tiny segments.</t>
<t>So far, our survey of 84 vendors across the industry has drawn
responses from about 19%, none of whom have implemented the byte mode
packet drop variant of RED. Given there appears to be little, if any,
installed base recommending removal of byte-mode drop from RED is
possibly only a paper exercise with few, if any, incremental deployment
issues.</t>
<t>If a vendor has implemented byte-mode drop, and an operator has
turned it on, it is strongly RECOMMENDED that it SHOULD be turned off.
Note that RED as a whole SHOULD NOT be turned off, as without it, a drop
tail queue also biases against large packets. But note also that turning
off byte-mode may alter the relative performance of applications using
different packet sizes, so it would be advisable to establish the
implications before turning it off.</t>
<t>Instead, the IETF transport area should continue its programme of
updating congestion control protocols to take account of packet size and
to make transports less sensitive to losing control packets like SYNs
and pure ACKS.</t>
<t>NOTE WELL that RED's byte-mode queue measurement is fine, being
completely orthogonal to byte-mode drop. If a RED implementation has a
byte-mode but does not specify what sort of byte-mode, it is most
probably byte-mode queue measurement, which is fine. However, if in
doubt, the vendor should be consulted.</t>
<t>The above conclusions cater for the Internet as it is today with
most, if not all, resources being primarily bit-congestible. A secondary
conclusion of this memo is that we may see more packet-congestible
resources in the future, so research may be needed to extend the
Internet's congestion notification (drop or ECN) so that it can handle a
mix of bit-congestible and packet-congestible resources.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Acknowledgements" title="Acknowledgements">
<t>Thank you to Sally Floyd, who gave extensive and useful review
comments. Also thanks for the reviews from Toby Moncaster and Arnaud
Jacquet. I am grateful to Bruce Davie and his colleagues for providing a
timely and efficient survey of RED implementation in Cisco's product
range. Also grateful thanks to Toby Moncaster, Will Dormann, John
Regnault, Simon Carter and Stefaan De Cnodder who further helped survey
the current status of RED implementation and deployment and, finally,
thanks to the anonymous individuals who responded.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Comments_Solicited" title="Comments Solicited">
<t>Comments and questions are encouraged and very welcome. They can be
addressed to the IETF Transport Area working group mailing list
<tsvwg@ietf.org>, and/or to the authors.</t>
</section>
</middle>
<back>
<!-- ================================================================ -->
<references title="Normative References">
<?rfc include="reference.RFC.2119" ?>
<?rfc include="reference.RFC.2309" ?>
<?rfc include="reference.RFC.3168" ?>
<?rfc include="reference.RFC.3426" ?>
<?rfc include='reference.RFC.5033'?>
</references>
<references title="Informative References">
<?rfc include="localref.Floyd93.RED" ?>
<?rfc include="localref.Floyd97.REDPktByteEmail" ?>
<?rfc include="localref.Floyd99.Penalty_box" ?>
<?rfc include="localref.Crowcroft98.MulTCP" ?>
<?rfc include="localref.Gibbens99.Evol_cc" ?>
<?rfc include="localref.Elloumi00.REDbyte" ?>
<?rfc include="localref.Vasallo00.PktSizeEquCC" ?>
<?rfc include="localref.Siris02a.Window_ECN" ?>
<?rfc include="localref.Siris02.RscCtrlCDMA" ?>
<?rfc include="reference.RFC.2474" ?>
<?rfc include="reference.RFC.2581" ?>
<?rfc include="reference.RFC.3714" ?>
<?rfc include="reference.RFC.3448" ?>
<?rfc include='reference.RFC.4828'?>
<?rfc include="localref.Eddy03.REDbias" ?>
<?rfc include="localref.Widmer04.CCvarPktSize" ?>
<?rfc include="localref.I-D.briscoe-tsvwg-re-ecn-tcp" ?>
<?rfc include="reference.I-D.eardley-pcn-marking-behaviour" ?>
<?rfc include="localref.Briscoe07.Rate_fair_Dis" ?>
<?rfc include="reference.I-D.ietf-tcpm-rfc2581bis" ?>
<?rfc include="reference.I-D.falk-xcp-spec" ?>
<?rfc include="reference.RFC.4782" ?>
<?rfc include='localref.Floyd00.gentle_RED'?>
<?rfc include='reference.I-D.ietf-tcpm-ecnsyn'?>
<?rfc include='reference.I-D.floyd-tcpm-ackcc'?>
<?rfc include='localref.Wischik07.ShortMsgs'?>
<?rfc include='localref.Shin08.DRQ'?>
<?rfc include='localref.Bolla00.Cisco_IOS_Arch'?>
<?rfc include='reference.I-D.irtf-iccrg-welzl-congestion-control-open-research'?>
</references>
<!-- ================================================================ -->
<section anchor="pktb_Scenarios" title="Example Scenarios">
<!--{ToDo: Tabulate these subsections}-->
<!-- ________________________________________________________________ -->
<section anchor="pktb_Notation" title="Notation">
<t>To prove the two sets of assertions in the idealised wire protocol
(<xref target="pktb_Ideal_Coding"></xref>) are true, we will compare
two flows with different packet sizes, s_1 and s_2 [bit/pkt], to make
sure their transports each see the correct congestion notification.
Initially, within each flow we will take all packets as having equal
sizes, but later we will generalise to flows within which packet sizes
vary. A flow's bit rate, x [bit/s], is related to its packet rate, u
[pkt/s], by <list style="empty">
<t>x(t) = s.u(t).</t>
</list></t>
<t>We will consider a 2x2 matrix of four scenarios:</t>
<?rfc needLines="6" ?>
<texttable anchor="pktb_Tab_Scenarios">
<ttcol align="right">resource type and congestion level</ttcol>
<ttcol align="center">A) Equal bit rates</ttcol>
<ttcol align="center">B) Equal pkt rates</ttcol>
<c>i) bit-congestible, p_b</c>
<c>(Ai)</c>
<c>(Bi)</c>
<c>ii) pkt-congestible, p_p</c>
<c>(Aii)</c>
<c>(Bii)</c>
</texttable>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Ai"
title="Bit-congestible resource, equal bit rates (Ai)">
<t>Starting with the bit-congestible scenario, for two flows to
maintain equal bit rates (Ai) the ratio of the packet rates must be
the inverse of the ratio of packet sizes: u_2/u_1 = s_1/s_2. So, for
instance, a flow of 60B packets would have to send 25x more packets to
achieve the same bit rate as a flow of 1500B packets. If a congested
resource marks proportion p_b of packets irrespective of size, the
ratio of marked packets received by each transport will still be the
same as the ratio of their packet rates, p_b.u_2/p_b.u_1 = s_1/s_2. So
of the 25x more 60B packets sent, 25x more will be marked than in the
1500B packet flow, but 25x more won't be marked too.</t>
<t>In this scenario, the resource is bit-congestible, so it always
uses our idealised bit-congestion field when it marks packets.
Therefore the transport should count marked bytes not packets. But it
doesn't actually matter for ratio-based transports like TCP (<xref
target="pktb_Ideal_Coding"></xref>). The ratio of marked to unmarked
bytes seen by each flow will be p_b, as will the ratio of marked to
unmarked packets. Because they are ratios, the units cancel out.</t>
<t>If a flow sent an inconsistent mixture of packet sizes, we have
said it should count the ratio of marked and unmarked bytes not
packets in order to correctly decode the level of congestion. But
actually, if all it is trying to do is decode p_b, it still doesn't
matter. For instance, imagine the two equal bit rate flows were
actually one flow at twice the bit rate sending a mixture of one 1500B
packet for every thirty 60B packets. 25x more small packets will be
marked and 25x more will be unmarked. The transport can still
calculate p_b whether it uses bytes or packets for the ratio. In
general, for any algorithm which works on a ratio of marks to
non-marks, either bytes or packets can be counted interchangeably,
because the choice cancels out in the ratio calculation.</t>
<t>However, where an absolute target rather than relative volume of
congestion caused is important (<xref
target="pktb_Ideal_Coding"></xref>), as it is for congestion
accountability <xref target="Rate_fair_Dis"></xref>, the transport
must count marked bytes not packets, in this bit-congestible case.
Aside from the goal of congestion accountability, this is how the bit
rate of a transport can be made independent of packet size; by
ensuring the rate of congestion caused is kept to a constant weight
<xref target="WindowPropFair"></xref>, rather than merely responding
to the ratio of marked and unmarked bytes.</t>
<t>Note the unit of byte-congestion volume is the byte.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bi"
title="Bit-congestible resource, equal packet rates (Bi)">
<t>If two flows send different packet sizes but at the same packet
rate, their bit rates will be in the same ratio as their packet sizes,
x_2/x_1 = s_2/s_1. For instance, a flow sending 1500B packets at the
same packet rate as another sending 60B packets will be sending at 25x
greater bit rate. In this case, if a congested resource marks
proportion p_b of packets irrespective of size, the ratio of packets
received with the byte-congestion field marked by each transport will
be the same, p_b.u_2/p_b.u_1 = 1.</t>
<t>Because the byte-congestion field is marked, the transport should
count marked bytes not packets. But because each flow sends
consistently sized packets it still doesn't matter for ratio-based
transports. The ratio of marked to unmarked bytes seen by each flow
will be p_b, as will the ratio of marked to unmarked packets.
Therefore, if the congestion control algorithm is only concerned with
the ratio of marked to unmarked packets (as is TCP), both flows will
be able to decode p_b correctly whether they count packets or
bytes.</t>
<t>But if the absolute volume of congestion is important, e.g. for
congestion accountability, the transport must count marked bytes not
packets. Then the lower bit rate flow using smaller packets will
rightly be perceived as causing less byte-congestion even though its
packet rate is the same.</t>
<t>If the two flows are mixed into one, of bit rate x1+x2, with equal
packet rates of each size packet, the ratio p_b will still be
measurable by counting the ratio of marked to unmarked bytes (or
packets because the ratio cancels out the units). However, if the
absolute volume of congestion is required, the transport must count
the sum of congestion marked bytes, which indeed gives a correct
measure of the rate of byte-congestion p_b(x_1 + x_2) caused by the
combined bit rate.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Aii"
title="Pkt-congestible resource, equal bit rates (Aii)">
<t>Moving to the case of packet-congestible resources, we now take two
flows that send different packet sizes at the same bit rate, but this
time the pkt-congestion field is marked by the resource with
probability p_p. As in scenario Ai with the same bit rates but a
bit-congestible resource, the flow with smaller packets will have a
higher packet rate, so more packets will be both marked and unmarked,
but in the same proportion.</t>
<t>This time, the transport should only count marks without taking
into account packet sizes. Transports will get the same result, p_p,
by decoding the ratio of marked to unmarked packets in either
flow.</t>
<t>If one flow imitates the two flows but merged together, the bit
rate will double with more small packets than large. The ratio of
marked to unmarked packets will still be p_p. But if the absolute
number of pkt-congestion marked packets is counted it will accumulate
at the combined packet rate times the marking probability,
p_p(u_1+u_2), 26x faster than packet congestion accumulates in the
single 1500B packet flow of our example, as required.</t>
<t>But if the transport is interested in the absolute number of packet
congestion, it should just count how many marked packets arrive. For
instance, a flow sending 60B packets will see 25x more marked packets
than one sending 1500B packets at the same bit rate, because it is
sending more packets through a packet-congestible resource.</t>
<t>Note the unit of packet congestion is packets.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bii"
title="Pkt-congestible resource, equal packet rates (Bii)">
<t>Finally, if two flows with the same packet rate, pass through a
packet-congestible resource, they will both suffer the same proportion
of marking, p_p, irrespective of their packet sizes. On detecting that
the pkt-congestion field is marked, the transport should count
packets, and it will be able to extract the ratio p_p of marked to
unmarked packets from both flows, irrespective of packet sizes.</t>
<t>Even if the transport is monitoring the absolute amount of packets
congestion over a period, still it will see the same amount of packet
congestion from either flow.</t>
<t>And if the two equal packet rates of different size packets are
mixed together in one flow, the packet rate will double, so the
absolute volume of packet-congestion will accumulate at twice the rate
of either flow, 2p_p.u_1 = p_p(u_1+u_2).</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_CN_Definition"
title="Congestion Notification Definition: Further Justification">
<t>In <xref target="pktb_Congestion_Definition"></xref> on the
definition of congestion notification, load not capacity was used as the
denominator. This also has a subtle significance in the related debate
over the design of new transport protocols—typical new protocol
designs (e.g. in XCP <xref target="I-D.falk-xcp-spec"></xref> &
Quickstart <xref target="RFC4782"></xref>) expect the sending transport
to communicate its desired flow rate to the network and network elements
to progressively subtract from this so that the achievable flow rate
emerges at the receiving transport.</t>
<t>Congestion notification with total load in the denominator can serve
a similar purpose (though in retrospect not in advance like XCP &
QuickStart). Congestion notification is a dimensionless fraction but
each source can extract necessary rate information from it because it
already knows what its own rate is. Even though congestion notification
doesn't communicate a rate explicitly, from each source's point of view
congestion notification represents the fraction of the rate it was
sending a round trip ago that couldn't (or wouldn't) be served by
available resources. After they were sent, all these fractions of each
source's offered load added up to the aggregate fraction of offered load
seen by the congested resource. So, the source can also know the total
excess rate by multiplying total load by congestion level. Therefore
congestion notification, as one scale-free dimensionless fraction,
implicitly communicates the instantaneous excess flow rate, albeit a RTT
ago.</t>
</section>
<section anchor="pktb_Policing_Congestion_Response"
title="Byte-mode Drop Complicates Policing Congestion Response">
<t>This appendix explains why the ability of networks to police the
response of <spanx style="emph">any</spanx> transport to congestion
depends on bit-congestible network resources only doing packet-mode not
byte-mode drop.</t>
<t>To be able to police a transport's response to congestion when
fairness can only be judged over time and over all an individual's
flows, the policer has to have an integrated view of all the congestion
an individual (not just one flow) has caused due to all traffic entering
the Internet from that individual. This is termed congestion
accountability.</t>
<t>But with byte-mode drop, one dropped or marked packet is not
necessarily equivalent to another unless you know the MTU that caused it
to be dropped/marked. To have an integrated view of a user, we believe
congestion policing has to be located at an individual's attachment
point to the Internet <xref target="Re-TCP"></xref>. But from there it
cannot know the MTU of each remote queue that caused each drop/mark.
Therefore it cannot take an integrated approach to policing all the
responses to congestion of all the transports of one individual.
Therefore it cannot police anything.</t>
<t>The security/incentive argument <spanx style="emph">for</spanx>
packet-mode drop is similar. Firstly, confining RED to packet-mode drop
would not preclude bottleneck policing approaches such as <xref
target="pBox"></xref> as it seems likely they could work just as well by
monitoring the volume of dropped bytes rather than packets. Secondly
packet-mode dropping/marking naturally allows the congestion
notification of packets to be globally meaningful without relying on MTU
information held elsewhere.</t>
<t>Because we recommend that a dropped/marked packet should be taken to
mean that all the bytes in the packet are dropped/marked, a policer can
remain robust against bits being re-divided into different size packets
or across different size flows <xref target="Rate_fair_Dis"></xref>.
Therefore policing would work naturally with just simple packet-mode
drop in RED.</t>
<t>In summary, making drop probability depend on the size of the packets
that bits happen to be divided into simply encourages the bits to be
divided into smaller packets. Byte-mode drop would therefore
irreversibly complicate any attempt to fix the Internet's incentive
structures.</t>
</section>
</back>
</rfc>| PAFTECH AB 2003-2026 | 2026-04-23 06:43:49 |