One document matched: draft-ietf-tsvwg-byte-pkt-congest-02.xml
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<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="12" month="July" year="2010" />
<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 two strong recommendations: (1) packet size
should not be taken into account when transports read congestion
indications, not when network equipment writes them, and (2) byte-mode
packet drop variant of AQM algorithms, such as RED, should not be used
to drop fewer small packets.</t>
</abstract>
</front>
<middle>
<section anchor="pktb_Introduction" title="Introduction">
<t>When notifying congestion, the problem of how (and whether) to take
packet sizes into account has exercised the minds of researchers and
practitioners for as long as active queue management (AQM) has been
discussed. Indeed, one reason AQM was originally introduced was to
reduce the lock-out effects that small packets can have on large packets
in drop-tail queues. This memo aims to state the principles we should be
using and to come to conclusions on what these principles will mean for
future protocol design, taking into account the deployments we have
already.</t>
<t>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="Coding congestion notification into the wire protocol:">When
the congested resource decides whether to notify the level of
congestion on each particular packet. (When a queue considers
whether to notify 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 the transport 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 to do with
encoding congestion notification into packets: 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>. The primary purpose of this memo is to
build a definitive consensus against such deliberate preferential
treatment for small packets in AQM algorithms and to record this advice
within the RFC series. Fortunately all the implementers who responded to
our survey (<xref target="pktb_Coding_Status_Summary"></xref>) have not
followed the earlier advice, so the consensus this memo argues for seems
to already exist in implementations. </t>
<t>The primary conclusion of this memo is that packet size should be
taken into account when transports read congestion indications, not when
network equipment writes them. Reducing drop of small packets has some
tempting advantages: i) it drops less control packets, which tend to be
small and ii) it makes TCP's bit-rate less dependent on packet size.
However, there are ways of addressing these issues at the transport
layer, rather than reverse engineering network forwarding to fix
specific transport problems.</t>
<t>The second conclusion is that network layer algorithms like the
byte-mode packet drop variant of RED should not be used to drop fewer
small packets, because that creates a perverse incentive for transports
to use tiny segments, consequently also opening up a DoS
vulnerability.</t>
<t>This memo is initially concerned with how we should correctly scale
congestion control functions with packet size for the long term. But it
also recognises that expediency may be necessary to deal with existing
widely deployed protocols that don't live up to the long term goal. It
turns out that the 'correct' variant of RED to deploy seems to be the
one everyone has deployed, and no-one who responded to our survey has
implemented the other variant. However, at the transport layer, TCP
congestion control is a widely deployed protocol that we argue doesn't
scale correctly with packet size. To date this hasn't been a significant
problem because most TCPs have been used with similar packet sizes. But,
as we design new congestion controls, we should build in scaling with
packet size rather than assuming we should follow TCP's example.</t>
<!--
Then the body of the memo starts from first
principles, defining congestion notification in <xref
target="pktb_Congestion_Definition" /> then determining the correct way
to measure congestion (<xref target="pktb_Measure" />) and to design an
idealised congestion notification protocol (<xref
target="pktb_Ideal_Coding" />).
-->
<t>This memo continues as follows. Terminology and scoping are discussed
next, and the reasons to make the recommendations presented in this memo
now are given in <xref target="pktb_now"></xref>. Motivating arguments
for our advice are given in <xref target="pktb_Motivation"></xref>. We
then survey the advice given previously in the RFC series, the research
literature and the deployed legacy (<xref target="pktb_SotA"></xref>)
before listing outstanding issues (<xref target="pktb_Issues"></xref>)
that will need resolution both to inform future protocols designs and to
handle legacy. We then give concrete recommendations for the way forward
in (<xref target="pktb_Conclusions"></xref>). We finally give security
considerations in <xref target="pktb_Security_Considerations"></xref>.
The interested reader can also find further discussions about 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_Conclusions"></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" />) 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 fraction (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). <xref
target="pktb_CN_Definition"></xref> describes a further incidental
benefit that arises from using load as the denominator of
congestion notification.</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, 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 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 respectively called packet-mode or
byte-mode queue measurement. And if the probability of dropping a
packet depends on its byte-size it is called byte-mode drop,
whereas if the drop probability is independent of a packet's
byte-size it is called packet-mode drop.</t>
</list></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 the network layer, or at
the transport, for a number of reasons: <list style="numbers">
<t>The packet vs. byte issue requires speedy resolution because
the IETF pre-congestion notification (PCN) working group is
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;</t>
<t>Without the present memo, the only advice in the RFC series on
packet size bias in AQM algorithms would be a reference to an
archived email in <xref target="RFC2309"></xref> (including an
addendum at the end of the email to correct the original).</t>
<t>The IRTF Internet Congestion Control Research Group (ICCRG)
recently took on the challenge of building consensus on what
common congestion control support should be required from network
forwarding functions in future <xref
target="I-D.irtf-iccrg-welzl" />.
The wider Internet community needs to discuss whether the
complexity of adjusting for packet size should be in the network
or in transports;</t>
<t>Given there are many good reasons why larger path max
transmission units (PMTUs) would help solve a number of scaling
issues, we don't want to create any bias against large packets
that is greater than their true cost;</t>
<t>The IETF has started to consider the question of fairness
between flows that use different packet sizes (e.g. in the
small-packet variant of TCP-friendly rate control, TFRC-SP <xref
target="RFC4828"></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_Motivation" title="Motivating Arguments">
<section anchor="pktb_Scaling"
title="Scaling Congestion Control with Packet Size">
<t>There are two ways of interpreting a dropped or marked packet. It
can either be considered as a single loss event or as loss/marking of
the bytes in the packet. Here we try to design a test to see which
approach scales with packet size.</t>
<t>Given bit-congestible is the more common case (see <xref
target="pktb_term"></xref>), consider a bit-congestible link shared by
many flows, so that each busy period tends to cause packets to be lost
from different flows. The test compares two identical scenarios with
the same applications, the same numbers of sources and the same load.
But the sources break the load into large packets in one scenario and
small packets in the other. Of course, because the load is the same,
there will be proportionately more packets in the small packet
case.</t>
<t>The test of whether a congestion control scales with packet size is
that it should respond in the same way to the same congestion
excursion, irrespective of the size of the packets that the bytes
causing congestion happen to be broken down into.</t>
<t>A bit-congestible queue suffering a congestion excursion has to
drop or mark the same excess bytes whether they are in a few large
packets or many small packets. So for the same congestion excursion,
the same amount of bytes have to be shed to get the load back to its
operating point. But, of course, for smaller packets more packets will
have to be discarded to shed the same bytes.</t>
<t>If all the transports interpret each drop/mark as a single loss
event irrespective of the size of the packet dropped, those with
smaller packets will respond more to the same congestion excursion,
failing our test. On the other hand, if they respond proportionately
less when smaller packets are dropped/marked, overall they will be
able to respond the same to the same congestion excursion.</t>
<t>Therefore, for a congestion control to scale with packet size it
should respond to dropped or marked bytes (as TFRC-SP <xref
target="RFC4828"></xref> effectively does), not just to dropped or
marked packets irrespective of packet size (as TCP does).</t>
<t>The email <xref target="pktByteEmail"></xref> referred to by
RFC2309 says the question of whether a packet's own size should affect
its drop probability "depends on the dominant end-to-end congestion
control mechanisms". But we argue the network layer should not be
optimised for whatever transport is predominant.</t>
<t>TCP congestion control ensures that flows competing for the same
resource each maintain the same number of segments in flight,
irrespective of segment size. So under similar conditions, flows with
different segment sizes will get different bit rates. But even though
reducing the drop probability of small packets helps ensure TCPs with
different packet sizes will achieve similar bit rates, we argue this
correction should be made to TCP itself, not to the network in order
to fix one transport, no matter how prominent it is.</t>
<t>Effectively, favouring small packets is reverse engineering of the
network layer around 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>
</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="Normal 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>
<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. <vspace blankLines="1" />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>
</list> 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, if RED drops 25% of the larger packets, it will aim to
drop 1% of the smaller packets (but in practice it may drop more as
congestion increases <xref target="RFC4828"></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 750k for
the flow of larger packet but 990k for the smaller packets (but
because of rate variation it will be less than this target).</t>
<t>It can be seen that this behaviour reopens the same denial of
service vulnerability that drop tail queues offer to floods of small
packet, though not necessarily as strongly (see <xref
target="pktb_Security_Considerations"></xref>).</t>
</section>
<section anchor="pktb_Small.NE.Control" title="Small != Control">
<t>It is tempting to drop small packets with lower probability to
improve performance, because many control packets are small (TCP SYNs
& ACKs, DNS queries & responses, SIP messages, HTTP GETs, etc)
and dropping fewer control packets considerably improves performance.
However, we must not give control packets preference purely by virtue
of their smallness, otherwise it is too easy for any data source to
get the same preferential treatment simply by sending data in smaller
packets. Again we should not create perverse incentives to favour
small packets rather than to favour control packets, which is what we
intend.</t>
<t>Just because many control packets are small does not mean all small
packets are control packets.</t>
<t>So again, rather than fix these problems in the network layer, we
argue that the transport should be made more robust against losses of
control packets (see 'Making Transports Robust against Control Packet
Losses' in <xref
target="pktb_Transport_Robust_Ctrl_Loss"></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>
<!--
<section anchor="pktb_Measure" title="Congestion Measurement">
<section anchor="pktb_Measurement_Q"
title="Congestion Measurement by Queue Length">
<t>Queue length is usually the most correct and simplest way to
measure congestion of a resource. To avoid the pathological effects of
drop tail, an AQM function can then be used to transform queue length
into the probability of dropping or marking a packet (e.g. RED's
piecewise linear function between thresholds). If the resource is
bit-congestible, the length of the queue SHOULD be measured in bytes.
If the resource is packet-congestible, the length of the queue SHOULD
be measured in packets. No other choice makes sense, because the
number of packets waiting in the queue isn't relevant if the resource
gets congested by bytes and vice versa. We discuss the implications on
RED's byte mode and packet mode for measuring queue length in <xref
target="pktb_SotA" />.</t>
<section anchor="pktb_Fixed_Buffers" title="Fixed Size Packet Buffers">
<t>Some, mostly older, queuing hardware sets aside fixed sized
buffers in which to store each packet in the queue. Also, with some
hardware, any fixed sized buffers not completely filled by a packet
are padded when transmitted to the wire. If we imagine a theoretical
forwarding system with both queuing and transmission in fixed,
MTU-sized units, it should clearly be treated as packet-congestible,
because the queue length in packets would be a good model of
congestion of the lower layer link.</t>
<t>If we now imagine a hybrid forwarding system with transmission
delay largely dependent on the byte-size of packets but buffers of
one MTU per packet, it should strictly require a more complex
algorithm to determine the probability of congestion. It should be
treated as two resources in sequence, where the sum of the
byte-sizes of the packets within each packet buffer models
congestion of the line while the length of the queue in packets
models congestion of the queue. Then the probability of congesting
the forwarding buffer would be a conditional
probability—conditional on the previously calculated
probability of congesting the line.</t>
<t>However, in systems that use fixed size buffers, it is unusual
for all the buffers used by an interface to be the same size.
Typically pools of different sized buffers are provided (Cisco uses
the term 'buffer carving' for the process of dividing up memory into
these pools <xref target="IOSArch" />). Usually, if the pool of
small buffers is exhausted, arriving small packets can borrow space
in the pool of large buffers, but not vice versa. However, it is
easier to work out what should be done if we temporarily set aside
the possibility of such borrowing. Then, with fixed pools of buffers
for different sized packets and no borrowing, the size of each pool
and the current queue length in each pool would both be measured in
packets. So an AQM algorithm would have to maintain the queue length
for each pool, and judge whether to drop/mark a packet of a
particular size by looking at the pool for packets of that size and
using the length (in packets) of its queue.</t>
<t>We now return to the issue we temporarily set aside: small
packets borrowing space in larger buffers. In this case, the only
difference is that the pools for smaller packets have a maximum
queue size that includes all the pools for larger packets. And every
time a packet takes a larger buffer, the current queue size has to
be incremented for all queues in the pools of buffers less than or
equal to the buffer size used.</t>
<t>We will return to borrowing of fixed sized buffers when we
discuss biasing the drop/marking probability of a specific packet
because of its size in <xref target="pktb_Network_Bias" />. But here
we can give a simple summary of the present discussion on how to
measure the length of queues of fixed buffers: no matter how
complicated the scheme is, ultimately any fixed buffer system will
need to measure its queue length in packets not bytes.</t>
</section>
</section>
<section anchor="pktb_Measurement_NoQ"
title="Congestion Measurement without a Queue">
<t>AQM algorithms are nearly always described assuming there is a
queue for a congested resource and the algorithm can use the queue
length to determine the probability that it will drop or mark each
packet. But not all congested resources lead to queues. For instance,
wireless spectrum is bit-congestible (for a given coding scheme),
because interference increases with the rate at which bits are
transmitted. But wireless link protocols do not always maintain a
queue that depends on spectrum interference. Similarly, power limited
resources are also usually bit-congestible if energy is primarily
required for transmission rather than header processing, but it is
rare for a link protocol to build a queue as it approaches maximum
power.</t>
<t>However, AQM algorithms don't require a queue in order to work. For
instance spectrum congestion can be modelled by signal quality using
target bit-energy-to-noise-density ratio. And, to model radio power
exhaustion, transmission power levels can be measured and compared to
the maximum power available. <xref target="ECNFixedWireless" />
proposes a practical and theoretically sound way to combine congestion
notification for different bit-congestible resources at different
layers along an end to end path, whether wireless or wired, and
whether with or without queues.</t>
</section>
</section>
-->
<!-- ================================================================ -->
<section anchor="pktb_SotA" title="The State of the Art">
<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. This
email clarified that there were in fact two orthogonal choices: whether
to measure queue length in bytes or packets (<xref
target="pktb_Measure_Status"></xref> below) and whether the drop
probability of an individual packet should depend on its own size (<xref
target="pktb_Coding_Status"></xref> below).</t>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Measure_Status"
title="Congestion Measurement: Status">
<t>The choice of which metric to use to measure queue length was left
open in RFC2309. It is now well understood that queues for
bit-congestible resources should be measured in bytes, and queues for
packet-congestible resources should be measured in packets.</t>
<!-- (see <xref
target="pktb_Measure" />).</t>
-->
<t>Where buffers are not configured or legacy buffers cannot be
configured to the above guideline, we do not have to make allowances
for such legacy in future protocol design. If a bit-congestible buffer
is measured in packets, the operator will have set the thresholds
mindful of a typical mix of packets sizes. Any AQM algorithm on such a
buffer will be oversensitive to high proportions of small packets,
e.g. a DoS attack, and undersensitive to high proportions of large
packets. But an operator can safely keep such a legacy buffer because
any undersensitivity during unusual traffic mixes cannot lead to
congestion collapse given the buffer will eventually revert to tail
drop, discarding proportionately more large packets.</t>
<t>Some modern queue implementations give a choice for setting RED's
thresholds in byte-mode or packet-mode. This may merely be an
administrator-interface preference, not altering how the queue itself
is measured but on some hardware it does actually change the way it
measures its queue. Whether a resource is bit-congestible or
packet-congestible is a property of the resource, so an admin should
not ever need to, or be able to, configure the way a queue measures
itself.</t>
<t>We believe the question of whether to measure queues in bytes or
packets is fairly well understood these days. The only outstanding
issues concern how to measure congestion when the queue is bit
congestible but the resource is packet congestible or vice versa.
There is no controversy over what should be done. It's just you have
to be an expert in probability to work out what should be done
(summarised in the following section) and, even if you have, it's not
always easy to find a practical algorithm to implement it.</t>
<section anchor="pktb_Fixed_Buffers" title="Fixed Size Packet Buffers">
<t>Some, mostly older, queuing hardware sets aside fixed sized
buffers in which to store each packet in the queue. Also, with some
hardware, any fixed sized buffers not completely filled by a packet
are padded when transmitted to the wire. If we imagine a theoretical
forwarding system with both queuing and transmission in fixed,
MTU-sized units, it should clearly be treated as packet-congestible,
because the queue length in packets would be a good model of
congestion of the lower layer link.</t>
<t>If we now imagine a hybrid forwarding system with transmission
delay largely dependent on the byte-size of packets but buffers of
one MTU per packet, it should strictly require a more complex
algorithm to determine the probability of congestion. It should be
treated as two resources in sequence, where the sum of the
byte-sizes of the packets within each packet buffer models
congestion of the line while the length of the queue in packets
models congestion of the queue. Then the probability of congesting
the forwarding buffer would be a conditional
probability—conditional on the previously calculated
probability of congesting the line.</t>
<t>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 simple summary of the present discussion on how
to measure the length of queues of fixed buffers: no matter how
complicated the scheme is, ultimately any fixed buffer system will
need to measure its queue length in packets not bytes.</t>
</section>
<section 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 Coding: Status">
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<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> gave 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, but advised that most scarce resources in
the Internet were currently bit-congestible. The argument continued
that if packet drops were inflated by packet size (byte-mode
dropping), "a flow's fraction of the packet drops is then a good
indication of that flow's fraction of the link bandwidth in bits per
second". This was consistent with a referenced policing mechanism
being worked on at the time for detecting unusually high bandwidth
flows, eventually published in 1999 <xref target="pBox"></xref>.
[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.</t>
<t>By 2003, a further change had been made to the adjustment for
packet size, this time in the RED algorithm of the ns2 simulator.
Instead of taking each packet's size relative to a `maximum packet
size' it was taken relative to a `mean packet size', intended to be
a static value representative of the `typical' packet size on the
link. We have not been able to find a justification for this change
in the literature, however Eddy and Allman conducted experiments
<xref target="REDbias"></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]. On
10-Nov-2004, this variant of byte-mode packet drop was made the
default in the ns2 simulator.</t>
<t>The byte-mode drop variant of RED is, of course, not the only
possible bias towards small packets in queueing algorithms. We have
already mentioned that tail-drop queues naturally tend to lock-out
large packets once they are full. But also queues with fixed sized
buffers reduce the probability that small packets will be dropped if
(and only if) they allow small packets to borrow buffers from the
pools for larger packets. As was explained in <xref
target="pktb_Fixed_Buffers"></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 RED_5, but apparently
unaware of it).</t>
<t>Pre-congestion notification <xref
target="I-D.ietf-pcn"></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.</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 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>
<t>The re-ECN protocol proposal <xref
target="I-D.briscoe-tsvwg-re-ecn-tcp"></xref> is designed so that
transports can be made more robust against losing control packets.
It gives queues an incentive to optionally give preference against
drop to packets with the 'feedback not established' codepoint in the
proposed 'extended ECN' field. Senders have incentives to use this
codepoint sparingly, but they can use it on control packets to
reduce their chance of being dropped. For instance, the proposed
modification to TCP for re-ECN uses this codepoint on the SYN and
SYN-ACK.</t>
<t>Although not brought to the IETF, a simple proposal from Wischik
<xref target="DupTCP"></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 Coding: Summary of Status">
<?rfc needLines="6" ?>
<texttable anchor="pktb_Tab_TFRC-SP"
title="Dependence of flow bit-rate per RTT on packet size s and drop rate p when network and/or transport bias towards small packets to varying degrees">
<ttcol align="right">transport cc</ttcol>
<ttcol align="center">RED_1 (packet mode drop)</ttcol>
<ttcol align="center">RED_4 (linear byte mode drop)</ttcol>
<ttcol align="center">RED_5 (square byte mode drop)</ttcol>
<c>TCP or TFRC</c>
<c>s/sqrt(p)</c>
<c>sqrt(s/p)</c>
<c>1/sqrt(p)</c>
<c>TFRC-SP</c>
<c>1/sqrt(p)</c>
<c>1/sqrt(sp)</c>
<c>1/(s.sqrt(p))</c>
</texttable>
<t><xref target="pktb_Tab_TFRC-SP"></xref> aims to summarise the
positions we may now be in. Each column shows a different possible
AQM behaviour in different queues in the network, using the
terminology of Cnodder et al outlined earlier (RED_1 is basic RED
with packet-mode drop). Each row shows a different transport
behaviour: TCP <xref target="RFC5681"></xref> and TFRC <xref
target="RFC3448"></xref> on the top row with TFRC-SP <xref
target="RFC4828"></xref> below. Suppressing all inessential details
the table shows that independence from packet size should either be
achievable by not altering the TCP transport in a RED_5 network, or
using the small packet TFRC-SP transport in a network without any
byte-mode dropping RED (top right and bottom left). Top left is the
`do nothing' scenario, while bottom right is the `do-both' scenario
in which bit-rate would become far too biased towards small packets.
Of course, if any form of byte-mode dropping RED has been deployed
on a selection of congested queues, each path will present a
different hybrid scenario to its transport.</t>
<t>Whatever, we can see that the linear byte-mode drop column in the
middle considerably complicates the Internet. It's a half-way house
that doesn't bias enough towards small packets even if one believes
the network should be doing the biasing. We argue below that <spanx
style="emph">all</spanx> network layer bias towards small packets
should be turned off—if indeed any equipment vendors have
implemented it—leaving packet size bias solely as the preserve
of the transport layer (solely the leftmost, packet-mode drop
column).</t>
<t>A survey has been conducted of 84 vendors to assess how widely
drop probability based on packet size has been implemented in RED.
Prior to the survey, an individual approach to Cisco received
confirmation that, having checked the code-base for each of the
product ranges, Cisco has not implemented any discrimination based
on packet size in any AQM algorithm in any of its products. Also an
individual approach to Alcatel-Lucent drew a confirmation that it
was very likely that none of their products contained RED code that
implemented any packet-size bias.</t>
<t>Turning to our more formal survey (<xref
target="pktb_Tab_RED_Survey"></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).</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. We have established that Linux does not implement RED
with packet size drop bias, although we have not investigated a
wider range of open source code.</t>
<t>Finally, we repeat that RED's byte mode drop is not the only way
to bias towards small packets—tail-drop tends to lock-out
large packets very effectively. Our survey was of vendor
implementations, so we cannot be certain about operator deployment.
But we believe many queues in the Internet are still tail-drop. The
company of one of the co-authors (BT) has widely deployed RED, but
there are bound to be many tail-drop queues, particularly in access
network equipment and on middleboxes like firewalls, where RED is
not always available.</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, a connectionless network with both bit-congestible and
packet-congestible resources is a different matter. If we believe we
should allow for this possibility in the future, this space contains a
truly open research issue.</t>
<t>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>. The 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. A
congestion notification protocol must survive a transition from low
levels of congestion to high. Marking two states is feasible with
explicit marking, but much harder if packets are dropped. Also, it
will not always be cost-effective to implement AQM at every low level
resource, so drop will often have to suffice. Distinguishing drop from
delivery naturally provides just one congestion flag—it is hard
to drop a packet in two ways that are distinguishable remotely. This
is a similar problem to that of distinguishing wireless transmission
losses from congestive losses.</t>
<t>We should also note that, strictly, packet-congestible resources
are actually cycle-congestible because load also depends on the
complexity of each look-up and whether the pattern of arrivals is
amenable to caching or not. Further, this reminds us that any solution
must not require a forwarding engine to use excessive processor cycles
in order to decide how to say it has no spare processor cycles.</t>
<t>Recently, the dual resource queue (DRQ) proposal <xref
target="DRQ"></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 layer is desirable). Consequently the premise is that CPU
congestion will become more common. DRQ is a proposed modification to
the RED algorithm that folds both bit congestion and packet congestion
into one signal (either loss or ECN).</t>
<t>The problem of signalling packet processing congestion is not
pressing, as most Internet resources are designed to be
bit-congestible before packet processing starts to congest (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="I-D.irtf-iccrg-welzl"></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="I-D.irtf-iccrg-welzl"></xref>).</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Conclusions" title="Recommendation and Conclusions">
<t></t>
<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 length of the queue SHOULD
be measured in bytes. If the resource is packet-congestible, the
length of the queue SHOULD be measured in packets. No other choice
makes sense, because the number of packets waiting in the queue isn't
relevant if the resource gets congested by bytes and vice versa. We
discuss the implications on RED's byte mode and packet mode for
measuring queue length in <xref target="pktb_SotA"></xref>.</t>
<t>NOTE WELL that RED's byte-mode queue measurement is fine, being
completely orthogonal to byte-mode drop. If a RED implementation has a
byte-mode but does not specify what sort of byte-mode, it is most
probably byte-mode queue measurement, which is fine. However, if in
doubt, the vendor should be consulted.</t>
</section>
<section anchor="pktb_Notify_Rec"
title="Recommendation on Notifying Congestion">
<t>The strong recommendation is that AQM algorithms such as RED SHOULD
NOT use byte-mode drop. More generally, the Internet's congestion
notification protocols (drop, ECN & PCN) SHOULD take account of
packet size when the notification is read by the transport layer, NOT
when it is written by the network layer. This approach offers
sufficient and correct congestion information for all known and future
transport protocols and also ensures no perverse incentives are
created that would encourage transports to use inappropriately small
packet sizes.</t>
<t>The alternative of deflating RED's drop probability for smaller
packet sizes (byte-mode drop) has no enduring advantages. It is more
complex, it creates the perverse incentive to fragment segments into
tiny pieces and it reopens the vulnerability to floods of
small-packets that drop-tail queues suffered from and AQM was designed
to remove.</t>
<t>Byte-mode drop is a change to the network layer that makes
allowance for an omission from the design of TCP, effectively reverse
engineering the network layer to contrive to make two TCPs with
different packet sizes run at equal bit rates (rather than packet
rates) under the same path conditions.</t>
<t>It also improves TCP performance by reducing the chance that a SYN
or a pure ACK will be dropped, because they are small. But we SHOULD
NOT hack the network layer to improve or fix certain transport
protocols. No matter how predominant a transport protocol is (even if
it's TCP), trying to correct for its failings by biasing towards small
packets in the network layer creates a perverse incentive to break
down all flows from all transports into tiny segments.</t>
<t>So far, our survey of 84 vendors across the industry has drawn
responses from about 19%, none of whom have implemented the byte mode
packet drop variant of RED. Given there appears to be little, if any,
installed base it seems we can recommend removal of byte-mode drop
from RED with little, if any, incremental deployment impact.</t>
<t>If a vendor has implemented byte-mode drop, and an operator has
turned it on, it is strongly RECOMMENDED that it SHOULD be turned off.
Note that RED as a whole SHOULD NOT be turned off, as without it, a
drop tail queue also biases against large packets. But note also that
turning off byte-mode may alter the relative performance of
applications using different packet sizes, so it would be advisable to
establish the implications before turning it off.</t>
</section>
<section anchor="pktb_Respond_Rec"
title="Recommendation on Responding to Congestion">
<t>Instead of network equipment biasing its congestion notification
for small packets, the IETF transport area should continue its
programme of updating congestion control protocols to take account of
packet size and to make transports less sensitive to losing control
packets like SYNs and pure ACKS.</t>
</section>
<section anchor="pktb_Research_Rec" title="Recommended Future Research">
<t>The above conclusions cater for the Internet as it is today with
most, if not all, resources being primarily bit-congestible. A
secondary conclusion of this memo is that we may see more
packet-congestible resources in the future, so research may be needed
to extend the Internet's congestion notification (drop or ECN) so that
it can handle a mix of bit-congestible and packet-congestible
resources.</t>
</section>
</section>
<!-- ================================================================ -->
<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_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 and Arnaud Jacquet as well as helpful explanations of
different hardware approaches from Larry Dunn and Fred Baker. I am
grateful to Bruce Davie and his colleagues for providing a timely and
efficient survey of RED implementation in Cisco's product range. Also
grateful thanks to Toby Moncaster, Will Dormann, John Regnault, Simon
Carter and Stefaan De Cnodder who further helped survey the current
status of RED implementation and deployment and, finally, thanks to the
anonymous individuals who responded.</t>
<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="reference.I-D.ietf-pcn-marking-behaviour" ?>
<?rfc include="localref.Briscoe07.Rate_fair_Dis" ?>
<?rfc include="reference.RFC.5681" ?>
<?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'?>
<?rfc include='reference.I-D.irtf-iccrg-welzl-congestion-control-open-research'?>
<?rfc include='reference.RFC.5670'?>
<?rfc include='reference.RFC.5562'?>
<?rfc include='reference.RFC.5690'?>
</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.briscoe-tsvwg-re-ecn-tcp"></xref>. But from there it cannot
know the MTU of each remote queue that caused each drop/mark. Therefore
it cannot take an integrated approach to policing all the responses to
congestion of all the transports of one individual. Therefore it cannot
police anything.</t>
<t>The security/incentive argument <spanx style="emph">for</spanx>
packet-mode drop is similar. Firstly, confining RED to packet-mode drop
would not preclude bottleneck policing approaches such as <xref
target="pBox"></xref> as it seems likely they could work just as well by
monitoring the volume of dropped bytes rather than packets. Secondly
packet-mode dropping/marking naturally allows the congestion
notification of packets to be globally meaningful without relying on MTU
information held elsewhere.</t>
<t>Because we recommend that a dropped/marked packet should be taken to
mean that all the bytes in the packet are dropped/marked, a policer can
remain robust against bits being re-divided into different size packets
or across different size flows <xref target="Rate_fair_Dis"></xref>.
Therefore policing would work naturally with just simple packet-mode
drop in RED.</t>
<t>In summary, making drop probability depend on the size of the packets
that bits happen to be divided into simply encourages the bits to be
divided into smaller packets. Byte-mode drop would therefore
irreversibly complicate any attempt to fix the Internet's incentive
structures.</t>
</section>
<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 -01 to -02 (this version):"><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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