One document matched: draft-ietf-tsvwg-byte-pkt-congest-04.xml
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<rfc category="info" docName="draft-ietf-tsvwg-byte-pkt-congest-04"
ipr="trust200902" updates="2309">
<front>
<title abbrev="Byte and Packet Congestion Notification">Byte and Packet
Congestion Notification</title>
<author fullname="Bob Briscoe" initials="B." surname="Briscoe">
<organization>BT</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://bobbriscoe.net/</uri>
</address>
</author>
<author fullname="Jukka Manner" initials="J." surname="Manner">
<organization abbrev="Aalto University">Aalto University</organization>
<address>
<postal>
<street>Department of Communications and Networking
(Comnet)</street>
<street>P.O. Box 13000</street>
<code>FIN-00076 Aalto</code>
<country>Finland</country>
</postal>
<phone>+358 9 470 22481</phone>
<email>jukka.manner@tkk.fi</email>
<uri>http://www.netlab.tkk.fi/~jmanner/</uri>
</address>
</author>
<date day="14" month="March" year="2011" />
<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). We give three strong recommendations: (1) packet
size should be taken into account when transports read congestion
indications, (2) packet size should not be taken into account when
network equipment creates congestion signals (marking, dropping), and
therefore (3) the byte-mode packet drop variant of the RED AQM algorithm
that drops fewer small packets should not be used.</t>
</abstract>
</front>
<middle>
<section anchor="pktb_Introduction" title="Introduction">
<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.</t>
<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>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 to measure how congested it is, should the queue
measure its length in bytes or packets?</t>
<t
hangText="Encoding congestion notification into the wire protocol:">When
the congested network resource decides whether to notify the level
of congestion by dropping or marking a particular packet, should its
decision depend on the byte-size of the particular packet being
dropped or marked?</t>
<t
hangText="Decoding congestion notification from the wire protocol:">When
the transport interprets the notification in order to decide how
much to respond to congestion, should it take into account the
byte-size of each missing or marked packet?</t>
</list></t>
<t>Consensus has emerged over the years concerning the first stage:
whether queues are measured in bytes or packets, termed byte-mode queue
measurement or packet-mode queue measurement. This memo records this
consensus in the RFC Series. In summary the choice solely depends on
whether the resource is congested by bytes or packets.</t>
<t>The controversy is mainly around the last two stages: whether to
allow for the size of the specific packet notifying congestion i) when
the network encodes or ii) when the transport decodes the congestion
notification.</t>
<t>Currently, the RFC series is silent on this matter other than a paper
trail of advice referenced from <xref target="RFC2309"></xref>, which
conditionally recommends byte-mode (packet-size dependent) drop <xref
target="pktByteEmail"></xref>. Reducing drop of small packets certainly
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 the problems of one specific transport, as byte-mode variant of RED was
designed to do.</t>
<t>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. It
recommends that (1) packet size should be taken into account when
transports read congestion indications, (2) not when network equipment
writes them.</t>
<t>In particular this means that 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. Fortunately all the RED
implementers who responded to our survey (<xref
target="pktb_Coding_Status_Summary"></xref>) have not followed the
earlier advice to use byte-mode drop, so the consensus this memo argues
for seems to already exist in implementations.</t>
<t>However, at the transport layer, TCP congestion control is a widely
deployed protocol that 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>This memo continues as follows. First it discusses terminology and
scoping. <xref target="pktb_Recommendations"></xref> gives the concrete formal
recommendations, followed by motivating arguments in <xref target="pktb_Motivation"></xref>.
We then critically
survey the advice given previously in the RFC series and the research
literature (<xref target="pktb_Critique_Advice"></xref>), followed by an
assessment of whether or not this advice has been followed in production
networks (<xref target="pktb_SotA"></xref>). To wrap up, outstanding
issues are discussed that will need resolution both to inform future
protocols designs and to handle legacy (<xref
target="pktb_Issues"></xref>). Then security issues are collected
together in <xref target="pktb_Security_Considerations"></xref> before
conclusions are drawn in <xref target="pktb_Conclusions"></xref>. The
interested reader can find discussion of more detailed issues on the
theme of byte vs. packet in the appendices.</t>
<t>This memo intentionally includes a non-negligible amount of material
on the subject. A busy reader can jump right into <xref
target="pktb_Recommendations"></xref> to read a summary of the
recommendations for the Internet community.</t>
<section anchor="pktb_term" title="Terminology and Scoping">
<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"></xref>.</t>
<!-- Old section 3 below ================================================================
<section anchor="pktb_Congestion_Definition"
title="Working Definition of Congestion Notification">
-->
<t><list style="hanging">
<t hangText="Congestion Notification:">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. E is the instantaneous excess load offered to a
resource that it is either incapable of serving or unwilling to
serve. L is the instantaneous offered load. <vspace
blankLines="1" /> The phrase `unwilling to serve' is added,
because AQM systems (e.g. RED, PCN <xref target="RFC5670"></xref>)
set a virtual limit smaller than the actual limit to the resource,
then notify when this virtual limit is exceeded in order to avoid
congestion of the actual capacity.<vspace blankLines="1" />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 measure of
congestion notification: drop probability (often loosely called
loss 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).</t>
<t hangText="Explicit and Implicit Notification:">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"></xref> or PCN
<xref target="RFC5670"></xref>). 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 hangText="Bit-congestible vs. Packet-congestible:">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.<vspace
blankLines="1" />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, radio power 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"></xref>).<vspace
blankLines="1" />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 [<xref format="counter"
target="RFC6077"></xref>; §3.3], but there is no
guarantee that it will not become common with future technology
trends.<vspace blankLines="1" />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 hangText="Coarser Granularity:">Resources may be congestible at
higher levels of granularity than bits or 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
applicable for congestion notification protocols controlling
per-flow and per-session processing or state.</t>
<t hangText="RED Terminology:">In RED, whether to use packets or
bytes when measuring queues is called respectively packet-mode
queue measurement or byte-mode queue measurement. And whether the
probability of dropping a packet is independent or dependent on
its byte-size is called respectively packet-mode drop or byte-mode
drop. The terms byte-mode and packet-mode should not be used
without specifying whether they apply to queue measurement or to
drop.</t>
</list></t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Recommendations" title="Recommendations">
<section anchor="pktb_Measure_Rec"
title="Recommendation on Queue Measurement">
<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).</t>
<t>If the resource is bit-congestible, the implementation SHOULD
measure the length of the queue in bytes. If the resource is
packet-congestible, the implementation SHOULD measure the length of
the queue 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.</t>
<t>Corollaries:<list style="numbers">
<!--
<t>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>A RED implementation SHOULD use byte mode queue
measurement for measuring the congestion of bit-congestible
resources and packet mode queue measurement for packet-congestible
resources.</t>
<t>"An Admin SHOULD NOT be able to configure the way a queue measures itself,
because wether a queue is bit-congestible or packet-congestible is a property
of the resource." </t>
</list></t>
<t>The recommended approach in less straightforward scenarios, such as
fixed size buffers, and resources without a queue, is discussed in
<xref target="pktb_Measure_Status"></xref>.</t>
</section>
<section anchor="pktb_Notify_Rec"
title="Recommendation on Notifying Congestion">
<t>When notifying congestion, a network device SHOULD treat all packets equally,
regardless of their size. Therefore, the probability that network equipment drops or marks a packet to
notify congestion SHOULD NOT depend on the size of the packet. For instance, to drop any bit with
probability 0.1% it is only necessary to drop every packet with probability 0.1% without regard to
the size of each packet.
</t>
<t>
This means
that the Internet's congestion notification protocols (drop, ECN &
PCN) SHOULD NOT take account of packet size when congestion is
notified by network equipment. Allowance for packet size is only
appropriate when the transport responds to congestion (See
Recommendation <xref format="counter"
target="pktb_Respond_Rec"></xref>). 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>Corollaries: <list style="numbers">
<t>AQM algorithms such as RED SHOULD NOT use byte-mode drop, which
deflates RED's drop probability for smaller packet sizes. RED's
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.</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 drop may alter the relative
performance of applications using different packet sizes, so it
would be advisable to establish the implications before turning it
off.<vspace blankLines="1" />NOTE WELL that RED's byte-mode queue
drop is completely orthogonal to byte-mode queue measurement and
should not be confused with it. 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>
</list></t>
<t>The byte mode packet drop variant of RED was recommended in the
past (see <xref target="pktb_Network_Bias"></xref> for how thinking
evolved). However, our survey of 84 vendors across the industry (<xref
target="pktb_SotA"></xref>) has found that none of the 19% who
responded have implemented byte mode drop in RED. Given there appears
to be little, if any, installed base it seems we can deprecate
byte-mode drop in RED with little, if any, incremental deployment
impact.</t>
</section>
<section anchor="pktb_Respond_Rec"
title="Recommendation on Responding to Congestion">
<!--
<t> A transport protocol SHOULD take into account the fraction of bytes that
indicate congestion when determining its sending rate, rather than the
fraction of packets indicating congestion.</t>
-->
<t>When a transport detects that a packet has been lost or congestion
marked, it SHOULD consider the strength of the congestion indication as
proportionate to the size in octets of the missing or marked packet.
</t>
<t>
In other words, when a packet indicates congestion (by being lost or
marked) it can be considered conceptually as if there is a congestion
indication on every octet of the packet, not just one indication per
packet.
</t>
<t>Therefore, instead of network equipment biasing its congestion
notification in
favour of small packets, the IETF transport area should continue its
programme of;<list style="symbols">
<t>updating host-based congestion control protocols to take
account of packet size</t>
<t>making transports less sensitive to losing control packets like
SYNs and pure ACKs.</t>
</list></t>
<t>Corollaries: <list style="numbers">
<t>If two TCPs with different packet sizes are required to run at
equal bit rates under the same path conditions, this SHOULD be
done by altering TCP (<xref target="pktb_Transport_Bias"></xref>),
not network equipment, which would otherwise affect other
transports besides TCP.</t>
<t>If it is desired to improve TCP performance by reducing the
chance that a SYN or a pure ACK will be dropped, this should be
done by modifying TCP (<xref
target="pktb_Transport_Robust_Ctrl_Loss"></xref>), not network
equipment. </t>
</list></t>
</section>
<section anchor="pktb_Respond_Split"
title="Recommendation on Handling Congestion Indications when Splitting or Merging Packets ">
<t>
Packets carrying congestion indications may be split or merged (e.g. at a
transcoder or during fragment reassembly). Splitting and merging only make sense
in the context of ECN, not loss.
</t>
<t>
The general rule to follow is that the number of octets in packets with
congestion indications should be roughly the same before and after merging or
splitting. This is based on the principle used above; that an indication of
congestion on a packet can be considered as an indication of congestion on each
octet of the packet.
</t>
<t>
One can think of a splitting or merging process as if all the incoming
congestion-marked octets increment a counter and all the outgoing marked octets
decrement the same counter. In order to ensure that congestion indications remain
timely, even the smallest positive remainder in the conceptual counter should
trigger the next outgoing packet to be marked (causing the counter to go
negative).
</t>
</section>
<!--
<section anchor="pktb_Research_Rec" title="Recommended Future Research">
<t>The above conclusions cater for the Internet as it is today with
most resources being primarily bit-congestible. A secondary conclusion
of this memo is that research is needed to determine whether there
might be more packet-congestible resources in the future. Then further
research would be needed to extend the Internet's congestion
notification (drop or ECN) so that it would be able to handle a more
even mix of bit-congestible and packet-congestible resources.</t>
</section>
-->
</section>
<!-- ================================================================ -->
<section anchor="pktb_Motivation" title="Motivating Arguments">
<t>In this section, we evaluate the topic of packet vs. byte based
congestion notifications and motivate the recommendations given in this
document.</t>
<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>
<t>Consider a bit-congestible link shared by many flows
(bit-congestible is the more common case, see <xref
target="pktb_term"></xref>), so that each busy period tends to cause
packets to be lost from different flows. Consider further two sources
that have the same data rate but break the load into large packets in
one application (A) and small packets in the other (B). Of course,
because the load is the same, there will be proportionately more
packets in the small packet flow (B).</t>
<t>If a congestion control scales with packet size 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 (A) or many small packets (B). 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 (B)
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, those with
smaller packets (B) will respond more to the same congestion
excursion. 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"></xref> effectively does), instead of dropped or
marked packets (as TCP does).</t>
</section>
<section anchor="pktb_Layering" title="Transport-Independent Network">
<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.</t>
<t>Even though reducing the drop probability of small packets (e.g.
RED's byte-mode drop) helps ensure TCPs with different packet sizes
will achieve similar bit rates, we argue this correction should be
made to any future transport protocols based on TCP, not to the
network in order to fix one transport, no matter how prominent it is.
Effectively, favouring small packets is reverse engineering of network
equipment around one particular transport protocol (TCP), contrary to
the excellent advice in <xref target="RFC3426"></xref>, which asks
designers to question "Why are you proposing a solution at this layer
of the protocol stack, rather than at another layer?"</t>
<t>RFC2309 refers to an email <xref target="pktByteEmail"></xref> for
advice on how RED should allow for different packet sizes. The email
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 network equipment should not be
specialised for whatever transport is predominant. No matter how
convenient it is, we SHOULD NOT hack the network solely to allow for
omissions from the design of one transport protocol, even if it is as
predominant as TCP.</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><xref target="RFC3426"></xref>. Again, imagine
a scenario where the same bit rate of packets will contribute the same
to bit-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. </t>
<t hangText="Non-malicious transports:">Even if a transport is not
actually malicious, if it finds small packets go faster, over time
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> </t>
<t>
Imagine two unresponsive 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"></xref>
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, RED aims to drop 25x more of the larger packets than the smaller ones.
Thus, for example 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 format="counter" target="RFC4828"></xref>;
§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
750Kbit for the flow of larger packet but 990Kbit for the smaller packets
(but because of rate variation it will be less than this target).</t>
<t>Note that, although the byte-mode drop variant of RED amplifies
small packet attacks, drop-tail queues amplify small packet attacks
even more (see Security Considerations in <xref
target="pktb_Security_Considerations"></xref>). Wherever possible
neither should be used.</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, 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"></xref>).</t>
</section>
<section anchor="pktb_Impl_Efficiency" title="Implementation Efficiency">
<t>Allowing for packet size at the transport rather than in the
network ensures that 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. This
isn't a principled reason in itself, but it is a happy consequence of
the other principled reasons.</t>
</section>
<section anchor="pktb_now" title="Why now?">
<t>Now is a good time to discuss whether fairness between different
sized packets would best be implemented in network equipment, or at
the transport, for a number of reasons: <list style="numbers">
<t>The IETF pre-congestion notification (PCN) working group is
standardising the external behaviour of a PCN congestion
notification (AQM) algorithm <xref target="RFC5670"></xref>;</t>
<t><xref target="RFC2309"></xref> 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"></xref>,
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 paragraph,
making the recommendation less clear-cut. RFC2309 is currently
the only advice in the RFC series on
packet size bias in AQM algorithms;</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="RFC6077"></xref>. 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 audio/video transport (AVT) working group is
standardising how the real-time protocol (RTP) should feedback and
respond to explicit congestion notification (ECN) <xref
target="I-D.ietf-avt-ecn-for-rtp"></xref>.</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"></xref>). 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"></xref> (Note however that, if the concern were
fairness between users, rather than between flows <xref
target="Rate_fair_Dis"></xref>, relative rates between flows would
have to come under run-time control rather than being embedded in
protocol designs).</t>
</list></t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Critique_Advice"
title="A Survey and Critique of Past Advice">
<t>The original 1993 paper on RED <xref target="RED93"></xref> 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"></xref>, the two modes were mentioned implying the
choice between them was a question of performance, referring to a 1997
email <xref target="pktByteEmail"></xref> for advice on tuning. A later
addendum to this email introduced the insight that there are in fact two
orthogonal choices: <list style="symbols">
<t>whether to measure queue length in bytes or packets (<xref
target="pktb_Measure_Status"></xref>)</t>
<t>whether the drop probability of an individual packet should
depend on its own size (<xref
target="pktb_Coding_Status"></xref>).</t>
</list>The rest of this section is structured accordingly.</t>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Measure_Status"
title="Congestion Measurement Advice">
<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 <xref target="pktByteEmail"></xref>.
</t>
<!-- (see <xref
target="pktb_Measure" />).</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>NOTE: Congestion in some legacy bit-congestible buffers is only
measured in packets not bytes. In such cases, the operator has to 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. However, there is no need to make
allowances for the possibility of such legacy in future protocol
design. This is safe 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>
<section anchor="pktb_Fixed_Buffers" title="Fixed Size Packet Buffers">
<t>Although the question of whether to measure queues in bytes or
packets is fairly well understood these days, measuring congestion
is not straightforward when the resource is bit congestible but the
queue is packet congestible or vice versa. This section outlines the
approach to take. There is no controversy over what should be done,
you just need to be expert in probability to work it out. And, even
if you know what should be done, it's not always easy to find a
practical algorithm to implement it.</t>
<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>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"></xref>). 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"></xref>. But
here we can give a at least one simple rule for 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 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>Nonetheless, AQM algorithms do not 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"></xref> 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_Coding_Status"
title="Congestion Notification Advice">
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Network_Bias" title="Network Bias when Encoding">
<t>The previously mentioned email <xref
target="pktByteEmail"></xref> referred to by <xref
target="RFC2309"></xref> advised that most scarce resources in the
Internet were bit-congestible, which is still believed to be true
(<xref target="pktb_term"></xref>). But it went on to give advice we
now disagree with. It said 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. 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"></xref>. However, the problem could and should 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"></xref> 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 implied 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. But it is not consistent with the present
recommendations of <xref target="pktb_Recommendations"></xref>.</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 in the
literature for this change, however Eddy and Allman conducted
experiments <xref target="REDbias"></xref> 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).</t>
<t>On 10-Nov-2004, this variant of byte-mode packet drop was made
the default in the ns2 simulator. None of the responses to our
admittedly limited survey of implementers (<xref
target="pktb_SotA"></xref>) found any variant of byte-mode drop had
been implemented. Therefore any conclusions based on ns2 simulations
that use RED without disabling byte-mode drop are likely to be
highly questionable.</t>
<t>The byte-mode drop variant of RED is, of course, not the only
possible bias towards small packets in queueing systems. 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"></xref> 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>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 equipment to 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"></xref> called
TFRC-SP <xref target="RFC4828"></xref>. 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"></xref>. (There are also
other important differences of detail relative to TFRC, such as
using virtual packets <xref target="CCvarPktSize"></xref> 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"></xref> 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"></xref>). Incidentally, VP-TFRC
also proposed that byte-mode RED dropping should really square the
packet size compensation factor (like that of Cnodder's RED_5, but
apparently unaware of it).</t>
<t>Pre-congestion notification <xref target="RFC5670"></xref> 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, as recommended here.</t>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Transport_Robust_Ctrl_Loss"
title="Making Transports Robust against Control Packet Losses">
<t>Recently, two RFCs have defined changes to TCP that make it more
robust against losing small control packets <xref
target="RFC5562"></xref> <xref target="RFC5690"></xref>. In both
cases they note that the case for these two 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"></xref> to request a
scheduling class with lower drop.</t>
<!--{ToDo: If ConEx defines optional preferential drop,
add its protocol definition to the Diffserv ref above}-->
<t>Although not brought to the IETF, a simple proposal from Wischik
<xref target="DupTCP"></xref> 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 Notification: Summary of Conflicting Advice">
<?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"></xref> aims to summarise the
potential effects of all the advice from different sources. 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="RFC5681"></xref>
and TFRC <xref target="RFC3448"></xref> on the top row with TFRC-SP
<xref target="RFC4828"></xref> below.</t>
<t>Let us assume that the goal is for the bit-rate of a flow to be
independent of packet size. Suppressing all inessential details, the
table shows that this should either be achievable by not altering
the TCP transport in a RED_5 network, or using the small packet
TFRC-SP transport (or similar) 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
subset of queues that congest, each path through the network 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. <xref
target="pktb_Recommendations"></xref> recommends that <spanx
style="emph">all</spanx> bias in network equipment 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>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_SotA" title="RED Implementation Status">
<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"></xref>), 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 were fairly sure they
had not but needed to check more thoroughly). 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>
<texttable anchor="pktb_Tab_RED_Survey"
title="Vendor Survey on byte-mode drop variant of RED (lower drop probability for small packets)">
<preamble></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></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. </t>
<t>Finally, we repeat that RED's byte mode drop SHOULD be disabled,
but active queue management such as RED SHOULD be enabled wherever
possible if we are to eradicate bias towards small
packets—without any AQM at all, tail-drop tends to lock-out
large packets very effectively. </t>
<t>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. The company of one of the co-authors
(BT) has widely deployed RED, but many tail-drop queues are there
are bound to still exist, particularly in access network equipment
and on middleboxes like firewalls, where RED is not always
available.</t>
<t>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"></xref>, 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"></xref> 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"></xref>, while weighted TCPs implemented in the
research community <xref target="WindowPropFair"></xref> could form
the basis of a future change to TCP congestion control <xref
target="RFC5681"></xref> itself.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-Pkt-World"
title="Bit- & Packet-congestible World">
<t>Nonetheless, the position is much less clear-cut if the Internet
becomes populated by a more even mix of both packet-congestible and
bit-congestible resources. If we believe we should allow for this
possibility in the future, this space contains a truly open research
issue.</t>
<t>We develop the concept of an idealised congestion notification
protocol that supports both bit-congestible and packet-congestible
resources in <xref target="pktb_Ideal"></xref>. This congestion
notification 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.
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>This problem would not be solved even if ECN were universally
deployed. 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. </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"></xref> 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 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 (see <xref
target="pktb_term"></xref>). 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="RFC6077"></xref>). 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="RFC6077"></xref>).</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"></xref> 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>This memo strongly recommends that the size of an individual packet
that is dropped or marked should only be taken into account when a
transport reads this as a congestion indication, not when network
equipment writes it. The memo therefore strongly deprecates using RED's
byte-mode of packet drop in network equipment. </t>
<t>Whether network equipment should measure the length of a queue by
counting bytes or counting packets is a different question to whether it
should take into account the size of each packet being dropped or
marked. The answer depends on whether the network resource is congested
respectively by bytes or by packets. This means that RED's byte-mode
queue measurement will often be appropriate even though byte-mode drop
is strongly deprecated.</t>
<t>At the transport layer the IETF should continue updating congestion
control protocols to take account of the size of each packet that
indicates congestion. Also the IETF should continue to make transports
less sensitive to losing control packets like SYNs, pure ACKs and DNS
exchanges. Although many control packets happen to be small, the
alternative of network equipment favouring all small packets would be
dangerous. That would create perverse incentives to split data transfers
into smaller packets.</t>
<t>The memo develops these recommendations from principled arguments
concerning scaling, layering, incentives, inherent efficiency, security
and policability. But it also addresses practical issues such as
specific buffer architectures and incremental deployment. Indeed a
limited survey of RED implementations is included, which shows there
appears to be little, if any, installed base of RED's byte-mode drop.
Therefore it can be deprecated with little, if any, incremental
deployment complications.</t>
<t>The recommendations have been developed on the well-founded basis
that most Internet resources are bit-congestible not packet-congestible.
We need to know the likelihood that this assumption will prevail longer
term and, if it might not, what protocol changes will be needed to cater
for a mix of the two. These questions have been delegated to the
IRTF.</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 Philip Eardley, Toby
Moncaster, Arnaud Jacquet and Mirja Kuehlewind as well as helpful explanations of
different hardware approaches from Larry Dunn and Fred Baker. We are
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>
<t>Bob Briscoe and Jukka Manner are partly funded by Trilogy, a research
project (ICT- 216372) supported by the European Community under its
Seventh Framework Programme. The views expressed here are those of the
authors only.</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.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="localref.Briscoe07.Rate_fair_Dis" ?>
<?rfc include="reference.RFC.5681" ?>
<!-- <?rfc include="reference.RFC.3465" ?> -->
<!-- <?rfc include="localref.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'?>
<!--
<reference anchor="I-D.irtf-iccrg-welzl">
<front>
<title>Open Research Issues in Internet Congestion Control</title>
<author fullname="Michael Welzl" initials="M" surname="Welzl">
<organization></organization>
</author>
<author fullname="Michael Scharf" initials="M" surname="Scharf">
<organization></organization>
</author>
<author fullname="Bob Briscoe" initials="B" surname="Briscoe">
<organization></organization>
</author>
<author fullname="Dimitri Papadimitriou" initials="D"
surname="Papadimitriou">
<organization></organization>
</author>
<date day="2" month="September" year="2010" />
<abstract>
<t>This document describes some of the open problems in Internet
congestion control that are known today. This includes several new
challenges that are becoming important as the network grows, as
well as some issues that have been known for many years. These
challenges are generally considered to be open research topics
that may require more study or application of innovative
techniques before Internet- scale solutions can be confidently
engineered and deployed. This document represents the work and the
consensus of the ICCRG.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="draft-irtf-iccrg-welzl-congestion-control-open-research-08" />
<format target="http://www.ietf.org/internet-drafts/draft-irtf-iccrg-welzl-congestion-control-open-research-08.txt"
type="TXT" />
</reference>
-->
<!--
<reference anchor="I-D.conex-concepts-uses">
<front>
<title>ConEx Concepts and Use Cases</title>
<author fullname="Bob Briscoe" initials="B" surname="Briscoe">
<organization></organization>
</author>
<author fullname="Richard Woundy" initials="R" surname="Woundy">
<organization></organization>
</author>
<author fullname="Toby Moncaster" initials="T" surname="Moncaster">
<organization></organization>
</author>
<author fullname="John Leslie" initials="J" surname="Leslie">
<organization></organization>
</author>
<date day="12" month="July" year="2010" />
<abstract>
<t>Internet Service Providers (ISPs) are facing problems where
localized congestion prevents full utilization of the path between
sender and receiver at today's "broadband" speeds. ISPs desire to
control this congestion, which often appears to be caused by a
small number of users consuming a large amount of bandwidth.
Building out more capacity along all of the path to handle this
congestion can be expensive and may not result in improvements for
all users so network operators have sought other ways to manage
congestion. The current mechanisms all suffer from difficulty
measuring the congestion (as distinguished from the total
traffic). The ConEx Working Group is designing a mechanism to make
congestion along any path visible at the Internet Layer. This
document describes example cases where this mechanism would be
useful.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="draft-moncaster-conex-concepts-uses-01" />
<format target="http://www.ietf.org/internet-drafts/draft-moncaster-conex-concepts-uses-01.txt"
type="TXT" />
</reference>
-->
<?rfc include='reference.RFC.5670'?>
<?rfc include='reference.RFC.6077'?>
<?rfc include='reference.RFC.5562'?>
<?rfc include='reference.RFC.5690'?>
<?rfc include='reference.I-D.ietf-avt-ecn-for-rtp'?>
<?rfc include='reference.I-D.ietf-conex-concepts-uses'?>
</references>
<!-- ================================================================ -->
<!-- ================================================================ -->
<!--
<section anchor="pktb_CN_Definition"
title="Congestion Notification Definition: Further Justification">
<t>In <xref target="pktb_term"></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="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.</t>
</section>
-->
<!-- Old Section 5 ============================================ -->
<section anchor="pktb_Ideal" title="Idealised Wire Protocol">
<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 later in this
appendix.</t>
<section anchor="pktb_Ideal_Coding" title="Protocol Coding">
<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"></xref>. Currently, packet-congestion is not the
common case, 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"></xref>): <list
style="numbers">
<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"></xref> 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="RFC5681"></xref> aim to achieve
equal segment rates per RTT through the same bottleneck—TCP
friendliness <xref target="RFC3448"></xref>. 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="RFC5681"></xref>, 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><xref target="WindowPropFair"></xref> are
examples of weighted proportionally fair transports designed for
cost-fair environments <xref target="Rate_fair_Dis"></xref>. 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_Scenarios" title="Example Scenarios">
<!--{ToDo: Tabulate these subsections}-->
<!-- ________________________________________________________________ -->
<section anchor="pktb_Notation" title="Notation">
<t>To prove our idealised wire protocol (<xref
target="pktb_Ideal_Coding"></xref>) is correct, 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 a packet.</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>
<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 a byte-mode drop algorithm has to depend on the local MTU of the
line - an algorithm needs to use some concept of a 'normal' packet size.
Therefore, one dropped or marked packet is not necessarily equivalent to
another unless you know the MTU at the queue where it was
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="I-D.ietf-conex-concepts-uses"></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>
<section anchor="changelog" 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 -03 to -04:"><list style="symbols">
<t>Reordered Sections 2 and 3, and some clarifications here and there based on
feedback from Colin Perkins and Mirja Kuehlewind.
</t>
</list></t>
<t hangText="From -02 to -03 (this version)"><list style="symbols">
<t>Structural changes: <list style="symbols">
<t>Split off text at end of "Scaling Congestion Control with
Packet Size" into new section "Transport-Independent
Network"</t>
<t>Shifted "Recommendations" straight after "Motivating
Arguments" and added "Conclusions" at end to reinforce
Recommendations</t>
<t>Added more internal structure to Recommendations, so that
recommendations specific to RED or to TCP are just
corollaries of a more general recommendation, rather than
being listed as a separate recommendation.</t>
<t>Renamed "State of the Art" as "Critical Survey of
Existing Advice" and retitled a number of subsections with
more descriptive titles.</t>
<t>Split end of "Congestion Coding: Summary of Status" into
a new subsection called "RED Implementation Status".</t>
<t>Removed text that had been in the Appendix "Congestion
Notification Definition: Further Justification".</t>
</list></t>
<t>Reordered the intro text a little.</t>
<t>Made it clearer when advice being reported is deprecated and
when it is not.</t>
<t>Described AQM as in network equipment, rather than saying "at
the network layer" (to side-step controversy over whether
functions like AQM are in the transport layer but in network
equipment).</t>
<t>Minor improvements to clarity throughout</t>
</list></t>
<t hangText="From -01 to -02:"><list style="symbols">
<t>Restructured the whole document for (hopefully) easier
reading and clarity. The concrete recommendation, in RFC2119
language, is now in <xref target="pktb_Conclusions"></xref>.</t>
</list></t>
<t hangText="From -00 to -01:"><list style="symbols">
<t>Minor clarifications throughout and updated references</t>
</list></t>
<t
hangText="From briscoe-byte-pkt-mark-02 to ietf-byte-pkt-congest-00:"><list
style="symbols">
<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 style="symbols">
<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"></xref>.</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"></xref>).</t>
<t>Separated out issue of normalising TCP's bit rate from issue
of preference to control packets (<xref
target="pktb_Small.NE.Control"></xref>).</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> & <xref
target="pktb_Network_Bias"></xref>) and on congestion
measurement in wireless link technologies without queues (<xref
target="pktb_Measurement_NoQ"></xref>).</t>
<t>Added section on 'Making Transports Robust against Control
Packet Losses' (<xref
target="pktb_Transport_Robust_Ctrl_Loss"></xref>) 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"></xref>).</t>
</list></t>
<t hangText="From -00 to -01:"><list style="symbols">
<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>
</section>
</back>
</rfc>
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