One document matched: draft-ietf-tsvwg-byte-pkt-congest-12.xml
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<rfc category="bcp" docName="draft-ietf-tsvwg-byte-pkt-congest-12"
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@aalto.fi</email>
<uri>http://www.netlab.tkk.fi/~jmanner/</uri>
</address>
</author>
<date day="07" month="November" year="2013"/>
<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 document provides recommendations of best current practice for
dropping or marking packets using any active queue management (AQM)
algorithm, including random early detection (RED), BLUE, pre-congestion
notification (PCN) and newer schemes such as CoDel (Controlled Delay)
and PIE (Proportional Integral controller Enhanced). We give three
strong recommendations: (1) packet size should be taken into account
when transports detect and respond to congestion indications, (2) packet
size should not be taken into account when network equipment creates
congestion signals (marking, dropping), and therefore (3) in the
specific case of RED, the byte-mode packet drop variant that drops fewer
small packets should not be used. This memo updates RFC 2309 to
deprecate deliberate preferential treatment of small packets in AQM
algorithms.</t>
</abstract>
</front>
<middle>
<section anchor="pktb_Introduction" title="Introduction">
<t>This document provides recommendations of best current practice for
how we should correctly scale congestion control functions with respect
to packet size for the long term. 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 signalling 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 outline how these principles will affect future protocol
design, taking into account the existing deployments we have
already.</t>
<t>The question of whether to take into account packet size arises at
three stages in the congestion notification process: <list
style="hanging">
<t hangText="Measuring congestion:">When a congested resource
measures locally how congested it is, should it measure its queue
length in time, bytes or packets?</t>
<t
hangText="Encoding congestion notification into the wire protocol:">When
a congested network resource signals its level of congestion, should
it drop / mark each packet dependent on the size of the particular
packet in question?</t>
<t
hangText="Decoding congestion notification from the wire protocol:">When
a transport interprets the notification in order to decide how much
to respond to congestion, should it take into account the size of
each missing or marked packet?</t>
</list></t>
<t>Consensus has emerged over the years concerning the first stage,
which <xref target="pktb_Measure_Rec"/> records in the RFC Series. In
summary: If possible it is best to measure congestion by time in the
queue, but otherwise the choice between bytes and packets 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"/>, which
conditionally recommends byte-mode (packet-size dependent) drop <xref
target="pktByteEmail"/>. 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.</t>
<!-- 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.
-->
<t>This memo updates <xref target="RFC2309"/> to deprecate deliberate
preferential treatment of packets in AQM algorithms solely because of
their size. It recommends that (1) packet size should be taken into
account when transports detect and respond to congestion indications,
(2) not when network equipment creates them. This memo also adds to the
congestion control principles enumerated in BCP 41 <xref
target="RFC2914"/>.</t>
<t>In the particular case of Random early Detection (RED), this means
that the byte-mode packet drop variant 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 admittedly
limited survey (<xref target="pktb_Coding_Status_Summary"/>) have not
followed the earlier advice to use byte-mode drop, so the position 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 with packet size (i.e. its
reduction in rate does not take into account the size of a lost packet).
To date this hasn't been a significant problem because most TCP
implementations have been used with similar packet sizes. But, as we
design new congestion control mechanisms, this memo recommends that 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"/> gives the concrete formal
recommendations, followed by motivating arguments in <xref
target="pktb_Motivation"/>. We then critically survey the advice given
previously in the RFC series and the research literature (<xref
target="pktb_Critique_Advice"/>), referring to an assessment of whether
or not this advice has been followed in production networks (<xref
target="pktb_SotA"/>). To wrap up, outstanding issues are discussed that
will need resolution both to inform future protocol designs and to
handle legacy (<xref target="pktb_Issues"/>). Then security issues are
collected together in <xref target="pktb_Security_Considerations"/>
before conclusions are drawn in <xref target="pktb_Conclusions"/>. 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. For the busy reader <xref
target="pktb_Recommendations"/> summarises 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"/>.</t>
<t>This memo applies to the design of all AQM algorithms, for example,
Random Early Detection (RED) <xref target="RFC2309"/>, BLUE <xref
target="BLUE02"/>, Pre-Congestion Notification (PCN) <xref
target="RFC5670"/>, Controlled Delay (CoDel) <xref
target="I-D.nichols-tsvwg-codel"/> and the Proportional Integral
controller Enhanced (PIE) <xref target="I-D.pan-tsvwg-pie"/>.
Throughout, RED is used as a concrete example because it is a widely
known and deployed AQM algorithm. There is no intention to imply that
the advice is any less applicable to the other algorithms, nor that
RED is preferred.</t>
<!-- Old section 3 below ================================================================
<section anchor="pktb_Congestion_Definition"
title="Working Definition of Congestion Notification">
-->
<t><list style="hanging">
<t hangText="Congestion Notification:">Congestion notification is
a changing signal that aims to communicate the probability that
the network resource(s) will not be able to forward the level of
traffic load offered (or that there is an impending risk that they
will not be able to).<vspace blankLines="1"/> The `impending risk'
qualifier is added, because AQM systems set a virtual limit
smaller than the actual limit to the resource, then notify when
this virtual limit is exceeded in order to avoid uncontrolled
congestion of the actual capacity.<vspace
blankLines="1"/>Congestion notification communicates a real number
bounded by the range [ 0 , 1 ]. This ties in with the most
well-understood measure of congestion notification: drop
probability.</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"/> or PCN <xref
target="RFC5670"/>). 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"/>).<vspace blankLines="1"/>The path
through a machine will typically encounter both packet-congestible
and bit-congestible resources. However, currently, a design goal
of network processing equipment such as routers and firewalls is
to size the packet-processing engine(s) relative to the lines in
order to keep packet processing uncongested even under worst case
packet rates with runs of minimum size packets. Therefore,
packet-congestion is currently rare [<xref format="counter"
target="RFC6077"/>; §3.3], but there is no guarantee that it
will not become more common in future. <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 particular packet is independent or
dependent on its 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 anchor="pktb_Example"
title="Example Comparing Packet-Mode Drop and Byte-Mode Drop">
<t>Taking RED as a well-known example algorithm, a central question
addressed by this document is whether to recommend RED's packet-mode
drop variant and to deprecate byte-mode drop. <xref
target="pktb_Tab_Example"/> compares how packet-mode and byte-mode
drop affect two flows of different size packets. For each it gives the
expected number of packets and of bits dropped in one second. Each
example flow runs at the same bit-rate of 48Mb/s, but one is broken up
into small 60 byte packets and the other into large 1500 byte
packets.</t>
<t>To keep up the same bit-rate, in one second there are about 25
times more small packets because they are 25 times smaller. As can be
seen from the table, the packet rate is 100,000 small packets versus
4,000 large packets per second (pps).</t>
<?rfc needLines="18" ?>
<texttable anchor="pktb_Tab_Example" style="headers"
title="Example Comparing Packet-mode and Byte-mode Drop">
<ttcol>Parameter</ttcol>
<ttcol>Formula</ttcol>
<ttcol align="right">Small packets</ttcol>
<ttcol align="right">Large packets</ttcol>
<c>Packet size</c>
<c>s/8</c>
<c>60B</c>
<c>1,500B</c>
<c>Packet size</c>
<c>s</c>
<c>480b</c>
<c>12,000b</c>
<c>Bit-rate</c>
<c>x</c>
<c>48Mbps</c>
<c>48Mbps</c>
<c>Packet-rate</c>
<c>u = x/s</c>
<c>100kpps</c>
<c>4kpps</c>
<c> </c>
<c/>
<c/>
<c/>
<c>Packet-mode Drop</c>
<c/>
<c/>
<c/>
<c>Pkt loss probability</c>
<c>p</c>
<c>0.1%</c>
<c>0.1%</c>
<c>Pkt loss-rate</c>
<c>p*u</c>
<c>100pps</c>
<c>4pps</c>
<c>Bit loss-rate</c>
<c>p*u*s</c>
<c>48kbps</c>
<c>48kbps</c>
<c> </c>
<c/>
<c/>
<c/>
<c>Byte-mode Drop</c>
<c>MTU, M=12,000b</c>
<c/>
<c/>
<c>Pkt loss probability</c>
<c>b = p*s/M</c>
<c>0.004%</c>
<c>0.1%</c>
<c>Pkt loss-rate</c>
<c>b*u</c>
<c>4pps</c>
<c>4pps</c>
<c>Bit loss-rate</c>
<c>b*u*s</c>
<c>1.92kbps</c>
<c>48kbps</c>
</texttable>
<t>For packet-mode drop, we illustrate the effect of a drop
probability of 0.1%, which the algorithm applies to all packets
irrespective of size. Because there are 25 times more small packets in
one second, it naturally drops 25 times more small packets, that is
100 small packets but only 4 large packets. But if we count how many
bits it drops, there are 48,000 bits in 100 small packets and 48,000
bits in 4 large packets—the same number of bits of small packets
as large.<list style="empty">
<t>The packet-mode drop algorithm drops any bit with the same
probability whether the bit is in a small or a large packet.</t>
</list></t>
<t>For byte-mode drop, again we use an example drop probability of
0.1%, but only for maximum size packets (assuming the link maximum
transmission unit (MTU) is 1,500B or 12,000b). The byte-mode algorithm
reduces the drop probability of smaller packets proportional to their
size, making the probability that it drops a small packet 25 times
smaller at 0.004%. But there are 25 times more small packets, so
dropping them with 25 times lower probability results in dropping the
same number of packets: 4 drops in both cases. The 4 small dropped
packets contain 25 times less bits than the 4 large dropped packets:
1,920 compared to 48,000.<list style="empty">
<t>The byte-mode drop algorithm drops any bit with a probability
proportionate to the size of the packet it is in.</t>
</list></t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Recommendations" title="Recommendations">
<t>This section gives recommendations related to network equipment in
Sections 2.1 and 2.2, and in Sections 2.3 and 2.4 we discuss the
implications on the transport protocols.</t>
<section anchor="pktb_Measure_Rec"
title="Recommendation on Queue Measurement">
<t>Ideally, an AQM would measure the service time of the queue to
measure congestion of a resource. However service time can only be
measured as packets leave the queue, where it is not always expedient
to implement a full AQM algorithm. To predict the service time as
packets join the queue, an AQM algorithm needs to measure the length
of the queue.</t>
<t>In this case, if the resource is bit-congestible, the AQM
implementation SHOULD measure the length of the queue in bytes and, if
the resource is packet-congestible, the implementation SHOULD measure
the length of the queue in packets. Subject to the exceptions below,
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. For example, the length of the queue into a transmission
line would be measured in bytes, while the length of the queue into a
firewall would be measured in packets.</t>
<t>To avoid the pathological effects of drop tail, the AQM can then
transform this service time or queue length into the probability of
dropping or marking a packet (e.g. RED's piecewise linear function
between thresholds).</t>
<t>What this advice means for RED as a specific example:<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 implementation SHOULD NOT make it possible to configure the
way a queue measures itself, because whether a queue is
bit-congestible or packet-congestible is an inherent property of
the queue.</t>
</list></t>
<t>Exceptions to these recommendations might be necessary, for
instance where a packet-congestible resource has to be configured as a
proxy bottleneck for a bit-congestible resource in an adjacent box
that does not support AQM.</t>
<t>The recommended approach in less straightforward scenarios, such as
fixed size packet buffers, resources without a queue and buffers
comprising a mix of packet and bit-congestible resources, is discussed
in <xref target="pktb_Measure_Status"/>. For instance, <xref
target="pktb_Fixed_Buffers"/> explains that the queue into a line
should be measured in bytes even if the queue consists of fixed-size
packet-buffers, because the root-cause of any congestion is bytes
arriving too fast for the line—packets filling buffers are
merely a symptom of the underlying congestion of the line.</t>
</section>
<section anchor="pktb_Notify_Rec"
title="Recommendation on Encoding Congestion Notification">
<t>When encoding congestion notification (e.g. by drop, ECN or PCN),
the probability that network equipment drops or marks a particular
packet to notify congestion SHOULD NOT depend on the size of the
packet in question. As the example in <xref target="pktb_Example"/>
illustrates, 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 approach ensures the network layer offers sufficient
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>What this advice means for RED as a specific example: <list
style="numbers">
<t>The RED AQM algorithm SHOULD NOT use byte-mode drop, i.e. it
ought to use packet-mode drop. Byte-mode drop is more complex, it
creates the perverse incentive to fragment segments into tiny
pieces and it is vulnerable to floods of small packets.</t>
<!-- OLD
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 RECOMMENDED to switch it to packet-mode drop,
after establishing if there are any implications on the relative
performance of applications using different packet sizes. The
unlikely possibility of some application-specific legacy use of
byte-mode drop is the only reason that all the above
recommendations on encoding congestion notification are not
phrased more strongly.<vspace blankLines="1"/> RED as a whole
SHOULD NOT be switched off. Without RED, a drop tail queue biases
against large packets and is vulnerable to floods of small
packets.</t>
<!-- OLD
If a vendor has implemented byte-mode drop, and an operator has
turned it on, it is RECOMMENDED to turn it 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 is expected that
byte-mode drop can be deprecated with little, if any, incremental deployment
impact.</t>
-->
<t>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>
<t>A survey (<xref target="pktb_SotA"/>) showed that there appears to
be little, if any, installed base of the byte-mode drop variant of
RED. This suggests that deprecating byte-mode drop will have 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 (bytes) 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>To be clear, the above recommendation solely describes how a
transport should interpret the meaning of a congestion indication, as
a long term goal. It makes no recommendation on whether a transport
should act differently based on this interpretation. It merely aids
interoperablity between transports, if they choose to make their
actions depend on the strength of congestion indications.</t>
<t>This definition will be useful as the IETF transport area continues
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>What this advice means for the case of TCP: <list style="numbers">
<t>If two TCP flows 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"/>),
not network equipment (the latter affects 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"/>), not network
equipment.</t>
</list></t>
<t>To be clear, we are not recommending at all that TCPs under
equivalent conditions should aim for equal bit-rates. We are merely
saying that anyone trying to do such a thing should modify their TCP
algorithm, not the network.</t>
<t>These recommendations are phrased as 'SHOULD' rather than 'MUST',
because there may be cases where expediency dictates that
compatibility with pre-existing versions of a transport protocol make
the recommendations impractical.</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 in
some circumstances (e.g. at a RTP/RTCP transcoder or during IP
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 equivalent 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>The above rule is not phrased with the word "MUST" to allow the
following exception. There are cases where pre-existing protocols were
not designed to conserve congestion marked octets (e.g. IP fragment
reassembly <xref target="RFC3168"/> or loss statistics in RTCP
receiver reports <xref target="RFC3550"/> before ECN was added <xref
target="RFC6679"/>). When any such protocol is updated, it SHOULD
comply with the above rule to conserve marked octets. However, the
rule may be relaxed if it would otherwise become too complex to
interoperate with pre-existing implementations of the protocol.</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>This section is informative. It justifies the recommendations given
in the previous section.</t>
<section anchor="pktb_Avoiding_Perverse_Incentives"
title="Avoiding Perverse Incentives to (Ab)use Smaller Packets">
<t>Increasingly, it is being recognised that a protocol design must
take care not to cause unintended consequences by giving the parties
in the protocol exchange perverse incentives <xref
target="Evol_cc"/><xref target="RFC3426"/>. Given there are many good
reasons why larger path maximum transmission units (PMTUs) would help
solve a number of scaling issues, we do not want to create any bias
against large packets that is greater than their true cost.</t>
<t>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 both the following
cases:</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 an application
designer is not actually malicious, if over time it is noticed
that small packets tend to go faster, designers will act in their
own interest and use smaller packets. Queues that give advantage
to small packets create an evolutionary pressure for applications
or 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"/> 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"/>; Appx 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 750kbps for the flow of larger packets but 990kbps
for the smaller packets (because of rate variations it will actually
be a little 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"/>). Wherever possible neither
should be used.</t>
</section>
<section anchor="pktb_Small.NE.Control" title="Small != Control">
<t>Dropping fewer control packets considerably improves performance.
It is tempting to drop small packets with lower probability in order
to improve performance, because many control packets tend to be
smaller (TCP SYNs & ACKs, DNS queries & responses, SIP
messages, HTTP GETs, etc). 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, 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"/>).</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>
<!-- OLD
<t>One motivation for the network biasing congestion notification by
packet size is to counter this effect and try to equalise the
bit-rates of flows with different packet sizes.
-->
<t>To counter this effect it seems tempting not to follow our
recommendation, and instead for the network to bias congestion
notification by packet size in order to equalise the bit-rates of
flows with different packet sizes. However, in order to do this, the
queuing algorithm has to make assumptions about the transport, which
become embedded in the network. Specifically: <list style="symbols">
<t>The queuing algorithm has to assume how aggressively the
transport will respond to congestion (see <xref
target="pktb_Coding_Status_Summary"/>). If the network assumes the
transport responds as aggressively as TCP NewReno, it will be
wrong for Compound TCP and differently wrong for Cubic TCP, etc.
To achieve equal bit-rates, each transport then has to guess what
assumption the network made, and work out how to replace this
assumed aggressiveness with its own aggressiveness.</t>
<!--
<t>Also, if the network biases congestion notification by packet
size it has to assume a baseline packet size—all proposed
algorithms use the local MTU. Then transports have to guess which
link was congested and what its local MTU was, in order to know
how to tailor their congestion response to that link.</t>
-->
<t>Also, if the network biases congestion notification by packet
size it has to assume a baseline packet size—all proposed
algorithms use the local MTU (for example see the byte-mode loss
probability formula in Table 1). Then if the non-Reno transports
mentioned above are trying to reverse engineer what the network
assumed, they also have to guess the MTU of the congested
link.</t>
</list></t>
<t>Even though reducing the drop probability of small packets (e.g.
RED's byte-mode drop) helps ensure TCP flows 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 predominant 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"/>, 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>In contrast, if the network never takes account of packet size, the
transport can be certain it will never need to guess any assumptions
the network has made. And the network passes two pieces of information
to the transport that are sufficient in all cases: i) congestion
notification on the packet and ii) the size of the packet. Both are
available for the transport to combine (by taking account of packet
size when responding to congestion) or not. <xref
target="pktb_Ideal"/> checks that these two pieces of information are
sufficient for all relevant scenarios.</t>
<t>When the network does not take account of packet size, it allows
transport protocols to choose whether to take account of packet size
or not. However, if the network were to bias congestion notification
by packet size, transport protocols would have no choice; those that
did not take account of packet size themselves would unwittingly
become dependent on packet size, and those that already took account
of packet size would end up taking account of it twice.</t>
</section>
<section anchor="pktb_Scaling" title="Partial Deployment of AQM">
<t>In overview, the argument in this section runs as follows:</t>
<t><list style="symbols">
<t>Because the network does not and cannot always drop packets in
proportion to their size, it shouldn't be given the task of making
drop signals depend on packet size at all.</t>
<t>Transports on the other hand don't always want to make their
rate response proportional to the size of dropped packets, but if
they want to, they always can.</t>
</list></t>
<t>The argument is similar to the end-to-end argument that says "Don't
do X in the network if end-systems can do X by themselves, and they
want to be able to choose whether to do X anyway." Actually the
following argument is stronger; in addition it says "Don't give the
network task X that could be done by the end-systems, if X is not
deployed on all network nodes, and end-systems won't be able to tell
whether their network is doing X, or whether they need to do X
themselves." In this case, the X in question is "making the response
to congestion depend on packet size".</t>
<t>We will now re-run this argument taking each step in more depth.
The argument applies solely to drop, not to ECN marking.</t>
<t>A queue drops packets for either of two reasons: a) to signal to
host congestion controls that they should reduce the load and b)
because there is no buffer left to store the packets. Active queue
management tries to use drops as a signal for hosts to slow down (case
a) so that drop due to buffer exhaustion (case b) should not be
necessary.</t>
<t>AQM is not universally deployed in every queue in the Internet;
many cheap Ethernet bridges, software firewalls, NATs on consumer
devices, etc implement simple tail-drop buffers. Even if AQM were
universal, it has to be able to cope with buffer exhaustion (by
switching to a behaviour like tail-drop), in order to cope with
unresponsive or excessive transports. For these reasons networks will
sometimes be dropping packets as a last resort (case b) rather than
under AQM control (case a).</t>
<t>When buffers are exhausted (case b), they don't naturally drop
packets in proportion to their size. The network can only reduce the
probability of dropping smaller packets if it has enough space to
store them somewhere while it waits for a larger packet that it can
drop. If the buffer is exhausted, it does not have this choice.
Admittedly tail-drop does naturally drop somewhat fewer small packets,
but exactly how few depends more on the mix of sizes than the size of
the packet in question. Nonetheless, in general, if we wanted networks
to do size-dependent drop, we would need universal deployment of
(packet-size dependent) AQM code, which is currently unrealistic.</t>
<t>A host transport cannot know whether any particular drop was a
deliberate signal from an AQM or a sign of a queue shedding packets
due to buffer exhaustion. Therefore, because the network cannot
universally do size-dependent drop, it should not do it all.</t>
<t>Whereas universality is desirable in the network, diversity is
desirable between different transport layer protocols - some, like
NewReno TCP <xref target="RFC5681"/>, may not choose to make their
rate response proportionate to the size of each dropped packet, while
others will (e.g. TFRC-SP <xref target="RFC4828"/>).</t>
<!--
<t>Having so far justified only our recommendations for the network,
this section focuses on the host. We construct a scaling argument to
justify the recommendation that a host should respond to a dropped or
marked packet in proportion to its size, not just as a single
congestion event.</t>
<t>The argument assumes that we have already sufficiently justified
our recommendation that the network should not take account of packet
size. </t>
<t>Also, we assume bit-congestible links are the predominant source of
congestion. As the Internet stands, it is hard if not impossible to
know whether congestion notification is from a bit-congestible or a
packet-congestible resource (see <xref
target="pktb_bit_pkt-congestible"></xref>) so we have to assume the
most prevalent case (see <xref target="pktb_term"></xref>). If this
assumption is wrong, and particular congestion indications are
actually due to overload of packet-processing, there is no issue of
safety at stake. Any congestion control that triggers a multiplicative
decrease in response to a congestion indication will bring packet
processing back to its operating point just as quickly. The only issue
at stake is that the resource could be utilised more efficiently if
packet-congestion could be separately identified.</t>
<t>
-->
<!-- Here we try to design a test to see which
approach scales with packet size.</t>
-->
<!-- Imagine a bit-congestible link shared by many flows, so that each busy
period tends to cause packets to be lost from different flows.
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 notification, irrespective of
the size of the packets containing the bytes that contribute to
congestion.</t>
<t>A bit-congestible queue suffering congestion 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 amount of congestion overload,
the same amount of bytes has to be shed to get the load back to its
operating point. For smaller packets (B) more packets
will have to be discarded to shed the same bytes.</t>
<t>If both the transports interpret each drop/mark as a single loss
event irrespective of the size of the packet dropped, the flow of
smaller packets (B) will respond more times to the same congestion. On
the other hand, if a transport responds proportionately less when
smaller packets are dropped/marked, overall it will be able to respond
the same to the same amount of congestion.</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>
<t>For the avoidance of doubt, this is not a recommendation that TCP
should be changed so that it scales with packet size. It is a
recommendation that any future transport protocol proposal should
respond to dropped or marked bytes if it wishes to claim that it is
scalable.</t>
-->
</section>
<section anchor="pktb_Impl_Efficiency" title="Implementation Efficiency">
<t>Biasing against large packets typically requires an extra multiply
and divide in the network (see the example byte-mode drop formula in
Table 1). 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. Also the
work to do the biasing is spread over many hosts, rather than
concentrated in just the congested network element. These aren't
principled reasons in themselves, but they are a happy consequence of
the other principled reasons.</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Critique_Advice"
title="A Survey and Critique of Past Advice">
<t>This section is informative, not normative.</t>
<t>The original 1993 paper on RED <xref target="RED93"/> proposed two
options for the RED active queue management algorithm: packet mode and
byte mode. Packet mode measured the queue length in packets and dropped
(or marked) individual packets with a probability independent of their
size. Byte mode measured the queue length in bytes and marked an
individual packet with probability in proportion to its size (relative
to the maximum packet size). In the paper's outline of further work, it
was stated that no recommendation had been made on whether the queue
size should be measured in bytes or packets, but noted that the
difference could be significant.</t>
<t>When RED was recommended for general deployment in 1998 <xref
target="RFC2309"/>, the two modes were mentioned implying the choice
between them was a question of performance, referring to a 1997 email
<xref target="pktByteEmail"/> for advice on tuning. 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"/>)</t>
<t>whether the drop probability of an individual packet should
depend on its own size (<xref target="pktb_Coding_Status"/>).</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"/>.</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>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 under-sensitive 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 under-sensitivity 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>The question of whether to measure queues in bytes or packets
seems to be well understood. However, measuring congestion is
confusing when the resource is bit congestible but the queue into
the resource is packet congestible. This section outlines the
approach to take.</t>
<t>Some, mostly older, queuing hardware allocates fixed sized
buffers in which to store each packet in the queue. This hardware
forwards to the line in one of two ways:<list style="symbols">
<t>With some hardware, any fixed sized buffers not completely
filled by a packet are padded when transmitted to the wire. This
case, should clearly be treated as packet-congestible, because
both queuing and transmission are in fixed MTU-sized units.
Therefore the queue length in packets is a good model of
congestion of the link.</t>
<t>More commonly, hardware with fixed size packet buffers
transmits packets to line without padding. This implies a hybrid
forwarding system with transmission congestion dependent on the
size of packets but queue congestion dependent on the number of
packets, irrespective of their size. <vspace
blankLines="1"/>Nonetheless, there would be no queue at all
unless the line had become congested—the root-cause of any
congestion is too many bytes arriving for the line. Therefore,
the AQM should measure the queue length as the sum of all the
packet sizes in bytes that are queued up waiting to be serviced
by the line, irrespective of whether each packet is held in a
fixed size buffer.</t>
</list></t>
<t>In the (unlikely) first case where use of padding means the queue
should be measured in packets, further confusion is likely because
the fixed buffers are rarely all one 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, there is no need to
consider all this complexity, because the root-cause of any
congestion is still line overload—buffer consumption is only
the symptom. Therefore, the length of the queue should be measured
as the sum of the bytes in the queue that will be transmitted to
line, including any padding. In the (unusual) case of transmission
with padding this means the sum of the sizes of the small buffers
queued plus the sum of the sizes of the large buffers queued.</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 repeat the simple rule for how to measure the length of
queues of fixed buffers: no matter how complicated the buffering
scheme is, ultimately a transmission line is nearly always
bit-congestible so the number of bytes queued up waiting for the
line measures how congested the line is, and it is rarely important
to measure how congested the buffering system is.</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, power limited resources are 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"/> 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>
<t>In wireless protocols that use request to send / clear to send
(RTS / CTS) control, such as some variants of IEEE802.11, it is
reasonable to base an AQM on the time spent waiting for transmission
opportunities (TXOPs) even though wireless spectrum is usually
regarded as congested by bits (for a given coding scheme). <!--, because interference increases with the rate at which bits
are transmitted. -->This is because requests for TXOPs queue up as
the spectrum gets congested by all the bits being transferred. So
the time that TXOPs are queued directly reflects bit congestion of
the spectrum.</t>
</section>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Coding_Status"
title="Congestion Notification Advice">
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Network_Bias" title="Network Bias when Encoding">
<section anchor="" title="Advice on Packet Size Bias in RED">
<t>The previously mentioned email <xref target="pktByteEmail"/>
referred to by <xref target="RFC2309"/> advised that most scarce
resources in the Internet were bit-congestible, which is still
believed to be true (<xref target="pktb_term"/>). But it went on
to offer advice that is updated by this memo. 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"/>. 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 size of the packet in question. No example
was given the other way round.</t>
<t>In 2000, Cnodder et al <xref target="REDbyte"/> pointed out
that there was an error in the part of the original 1993 RED
algorithm that aimed to distribute drops uniformly, because it
didn't correctly take into account the adjustment for packet size.
They recommended an algorithm called RED_4 to fix this. But they
also recommended a further change, RED_5, to adjust drop rate
dependent on the square of relative packet size. This was indeed
consistent with one 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. This memo makes a different recommendations in <xref
target="pktb_Recommendations"/>.</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"/> that assessed how sensitive
RED was to this parameter, amongst other things. However, this
changed algorithm can often lead to drop probabilities of greater
than 1 (which gives a hint that there is probably 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. It seems unlikely that byte-mode
drop has ever been implemented in production networks (<xref
target="pktb_SotA"/>), therefore any conclusions based on ns2
simulations that use RED without disabling byte-mode drop are
likely to behave very differently from RED in production
networks.</t>
</section>
<section title="Packet Size Bias Regardless of AQM">
<t>The byte-mode drop variant of RED (or a similar variant of
other AQM algorithms) is 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.</t>
<t>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
(see <xref target="pktb_Fixed_Buffers"/>). 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. 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, fixed size packet-buffers should be augmented with a good AQM
algorithm and packet-mode drop. If an AQM 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>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<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. 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="RFC5348"/> called TFRC-SP
<xref target="RFC4828"/>. Essentially, it proposes a rate equation
that inflates the flow rate by the ratio of a typical TCP segment
size (1500B including TCP header) over the actual segment size <xref
target="PktSizeEquCC"/>. (There are also other important differences
of detail relative to TFRC, such as using virtual packets <xref
target="CCvarPktSize"/> to avoid responding to multiple losses per
round trip and using a minimum inter-packet interval.)</t>
<t>Section 4.5.1 of this TFRC-SP spec discusses the implications of
operating in an environment where queues have been configured to
drop smaller packets with proportionately lower probability than
larger ones. But it only discusses TCP operating in such an
environment, only mentioning TFRC-SP briefly when discussing how to
define fairness with TCP. And it only discusses the byte-mode
dropping version of RED as it was before Cnodder et al pointed out
it didn't sufficiently bias towards small packets to make TCP
independent of packet size.</t>
<t>So the TFRC-SP spec doesn't address the issue of which of the
network or the transport <spanx style="emph">should</spanx> handle
fairness between different packet sizes. In its Appendix B.4 it
discusses the possibility of both TFRC-SP and some network buffers
duplicating each other's attempts to deliberately bias towards small
packets. But the discussion is not conclusive, instead reporting
simulations of many of the possibilities in order to assess
performance but not recommending any particular course of
action.</t>
<t>The paper originally proposing TFRC with virtual packets
(VP-TFRC) <xref target="CCvarPktSize"/> proposed that there should
perhaps be two variants to cater for the different variants of RED.
However, as the TFRC-SP authors point out, there is no way for a
transport to know whether some queues on its path have deployed RED
with byte-mode packet drop (except if an exhaustive survey found
that no-one has deployed it!—see <xref target="pktb_SotA"/>).
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"/> is an IETF
technology 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 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 target="RFC5690"/>. 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 there are no known proposals, it would also be possible
and perfectly valid to make control packets robust against drop by
requesting a scheduling class with lower drop probability, by
re-marking to a Diffserv code point <xref target="RFC2474"/> within
the same behaviour aggregate.</t>
<t>Although not brought to the IETF, a simple proposal from Wischik
<xref target="DupTCP"/> suggests that the first three packets of
every TCP flow should be routinely duplicated after a short delay.
It shows that this would greatly improve the chances of short flows
completing quickly, but it would hardly increase traffic levels on
the Internet, because Internet bytes have always been concentrated
in the large flows. It further shows that the performance of many
typical applications depends on completion of long serial chains of
short messages. It argues that, given most of the value people get
from the Internet is concentrated within short flows, this simple
expedient would greatly increase the value of the best efforts
Internet at minimal cost. A similar but more extensive approach has
been evaluated on Google servers <xref target="GentleAggro"/>.</t>
<t>The proposals discussed in this sub-section are experimental
approaches that are not yet in wide operational use, but they are
existence proofs that transports can make themselves robust against
loss of control packets. The examples are all TCP-based, but
applications over non-TCP transports could mitigate loss of control
packets by making similar use of Diffserv, data duplication, FEC
etc.</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 probability, p, when network and/or transport bias towards small packets to varying degrees">
<ttcol align="right">transport cc</ttcol>
<ttcol align="center">RED_1 (packet mode drop)</ttcol>
<ttcol align="center">RED_4 (linear byte mode drop)</ttcol>
<ttcol align="center">RED_5 (square byte mode drop)</ttcol>
<c>TCP or TFRC</c>
<c>s/sqrt(p)</c>
<c>sqrt(s/p)</c>
<c>1/sqrt(p)</c>
<c>TFRC-SP</c>
<c>1/sqrt(p)</c>
<c>1/sqrt(sp)</c>
<c>1/(s.sqrt(p))</c>
</texttable>
<t><xref target="pktb_Tab_TFRC-SP"/> aims to summarise the 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"/> and TFRC
<xref target="RFC5348"/> on the top row with TFRC-SP <xref
target="RFC4828"/> below. Each cell shows how the bits per round
trip of a flow depends on packet size, s, and drop probability, p.
In order to declutter the formulae to focus on packet-size
dependence they are all given per round trip, which removes any RTT
term.</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 would considerably complicate 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"/> 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>
<t>In practice it seems that no deliberate bias towards small
packets has been implemented for production networks. Of the 19% of
vendors who responded to a survey of 84 equipment vendors, none had
implemented byte-mode drop in RED (see <xref target="pktb_SotA"/>
for details).</t>
</section>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Issues" title="Outstanding Issues and Next Steps">
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-World" title="Bit-congestible Network">
<t>For a connectionless network with nearly all resources being
bit-congestible the recommended position is 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>A survey of equipment vendors (<xref
target="pktb_Coding_Status_Summary"/>) found no evidence that
byte-mode packet drop had been implemented, so deployment will be
sparse at best. A migration strategy is not really needed to remove an
algorithm that may not even be deployed.</t>
<t>A programme of experimental updates to take account of packet size
in transport congestion control protocols has already started with
TFRC-SP <xref target="RFC4828"/>.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-Pkt-World"
title="Bit- & Packet-congestible Network">
<t>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 (see <xref
target="pktb_bit_pkt-congestible"/>). This problem is not pressing,
because most Internet resources are designed to be bit-congestible
before packet processing starts to congest (see <xref
target="pktb_term"/>).</t>
<t>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"/>). The research question of whether packet
congestion might become common and what to do if it does may in the
future be explored in the IRTF (the "Challenge 3: Packet Size" in
<xref target="RFC6077"/>).</t>
<t>Note that sometimes it seems that resources might be congested by
neither bits nor packets, e.g. where the queue for access to a
wireless medium is in units of transmission opportunities. However,
the root cause of congestion of the underlying spectrum is overload of
bits (see <xref target="pktb_Measurement_NoQ"/>).</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Security_Considerations"
title="Security Considerations">
<t>This memo recommends that queues do not bias drop probability due to
packets size. For instance dropping small packets less often than large
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.</t>
<!-- Of course, if
transports really want to make the greatest gains, they don't have to
respond to congestion anyway. But we don't want applications that are
trying to behave to discover that they can go faster by using smaller
packets.</t>
-->
<t>In practice, transports cannot all be trusted to respond to
congestion. So another reason for recommending that queues do not bias
drop probability towards small packets is to avoid the vulnerability to
small packet DDoS attacks that would otherwise result. One of the
benefits of implementing AQM was meant to be to remove drop-tail's DoS
vulnerability to small packets, so we shouldn't add it back again.</t>
<t>If most queues implemented AQM with byte-mode drop, the resulting
network would amplify the potency of a small packet DDoS attack. At the
first queue the stream of packets would push aside a greater proportion
of large packets, so more of the small packets would survive to attack
the next queue. Thus a flood of small packets would continue on towards
the destination, pushing regular traffic with large packets out of the
way in one queue after the next, but suffering much less drop
itself.</t>
<t><xref target="pktb_Policing_Congestion_Response"/> explains why the
ability of networks to police the response of <spanx style="emph">any</spanx>
transport to congestion depends on bit-congestible network resources
only doing packet-mode not byte-mode drop. In summary, it says that
making drop probability depend on the size of the packets that bits
happen to be divided into simply encourages the bits to be divided into
smaller packets. Byte-mode drop would therefore irreversibly complicate
any attempt to fix the Internet's incentive structures.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_IANA" title="IANA Considerations">
<t>This document has no actions for IANA.</t>
</section>
<section anchor="pktb_Conclusions" title="Conclusions">
<t>This memo identifies the three distinct stages of the congestion
notification process where implementations need to decide whether to
take packet size into account. The recommendations provided in Section 2
of this memo are different in each case:<list style="symbols">
<t>When network equipment measures the length of a queue, if it is
not feasible to use time it is recommended to count in bytes if the
network resource is congested by bytes, or to count in packets if is
congested by packets.</t>
<t>When network equipment decides whether to drop (or mark) a
packet, it is recommended that the size of the particular packet
should not be taken into account</t>
<t>However, when a transport algorithm responds to a dropped or
marked packet, the size of the rate reduction should be
proportionate to the size of the packet.</t>
</list>In summary, the answers are 'it depends', 'no' and 'yes'
respectively</t>
<t>For the specific case of RED, this means that byte-mode queue
measurement will often be appropriate but the use of byte-mode drop is
very strongly discouraged.</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 protocols
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 policeability. But it also addresses practical issues such as
specific buffer architectures and incremental deployment. Indeed a
limited survey of RED implementations is discussed, 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. The IRTF Internet Congestion Control Research
Group (ICCRG) is currently working on these problems <xref
target="RFC6077"/>.</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, David Black,
Fred Baker, David Taht, 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 were 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.3168" ?>
</references>
<references title="Informative References">
<?rfc include="reference.RFC.2309" ?>
<?rfc include="reference.RFC.2914" ?>
<?rfc include="reference.RFC.3426" ?>
<?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.5348" ?>
<?rfc include='reference.RFC.4828'?>
<?rfc include="localref.Eddy03.REDbias" ?>
<?rfc include="localref.Widmer04.CCvarPktSize" ?>
<?rfc include='localref.Feng02.BLUE'?>
<?rfc include='reference.I-D.nichols-tsvwg-codel'?>
<?rfc include='reference.I-D.pan-tsvwg-pie'?>
<!-- <?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.Wischik08.ShortMsgs'?>
<?rfc include='localref.Shin08.DRQ'?>
<?rfc include='localref.Bolla00.Cisco_IOS_Arch'?>
<?rfc include='localref.Psounis01.CHOKe_Var_Pkt'?>
<reference anchor="GentleAggro"
target="http://doi.acm.org/10.1145/2486001.2486014">
<front>
<title>Reducing Web Latency: the Virtue of Gentle Aggression</title>
<author fullname="Tobias Flach" initials="T" surname="Flach">
<organization>USC</organization>
</author>
<author fullname="Nandita Dukkipati" initials="N"
surname="Dukkipati">
<organization>Google</organization>
</author>
<author fullname="Andreas Terzis" initials="A" surname="Terzis">
<organization>Google</organization>
</author>
<author fullname="Barath Raghavan" initials="B" surname="Raghavan">
<organization>Google</organization>
</author>
<author fullname="Neal Cardwell" initials="N" surname="Cardwell">
<organization>Google</organization>
</author>
<author fullname="Yuchung Cheng" initials="Y" surname="Cheng">
<organization>Google</organization>
</author>
<author fullname="Ankur Jain" initials="A" surname="Jain">
<organization>USC</organization>
</author>
<author fullname="Shuai Hao" initials="S" surname="Hao">
<organization>USC</organization>
</author>
<author fullname="Ethan Katz-Bassett" initials="E"
surname="Katz-Bassett">
<organization>USC</organization>
</author>
<author fullname="Ramesh Govindan" initials="R" surname="Govindan">
<organization>USC</organization>
</author>
<date month="August" year="2013"/>
</front>
<seriesInfo name="ACM SIGCOMM CCR" value="43(4)159--170"/>
<format target="http://doi.acm.org/10.1145/2486001.2486014" type="PDF"/>
</reference>
<!--
<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.3550'?>
<?rfc include='reference.RFC.5670'?>
<?rfc include='reference.RFC.6077'?>
<?rfc include='reference.RFC.5562'?>
<?rfc include='reference.RFC.5690'?>
<?rfc include='reference.RFC.6679'?>
<?rfc include='reference.RFC.6789'?>
</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_SotA" title="Survey of RED Implementation Status">
<t>This Appendix is informative, not normative.</t>
<t>In May 2007 a survey was conducted of 84 vendors to assess how widely
drop probability based on packet size has been implemented in RED <xref
target="pktb_Tab_RED_Survey"/>. <!-- 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 the survey (<xref
target="pktb_Tab_RED_Survey"></xref>),
-->About 19% of those surveyed replied, giving a sample size of 16. Although
in most cases we do not have permission to identify the respondents, we
can say that those that have responded include most of the larger
equipment vendors, covering a large fraction of the market. The two who
gave permission to be identified were Cisco and Alcatel-Lucent. The
others 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. All those who
responded 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). At the time the survey was conducted,
Linux did not implement RED with packet-size bias of drop, 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/>
<ttcol align="right">Response</ttcol>
<ttcol align="right">No. of vendors</ttcol>
<ttcol align="right">%age of vendors</ttcol>
<c>Not implemented</c>
<c>14</c>
<c>17%</c>
<c>Not implemented (probably)</c>
<c>2</c>
<c>2%</c>
<c>Implemented</c>
<c>0</c>
<c>0%</c>
<c>No response</c>
<c>68</c>
<c>81%</c>
<c>Total companies/orgs surveyed</c>
<c>84</c>
<c>100%</c>
<postamble/>
</texttable>
<t>Where reasons have been given, the extra complexity of packet bias
code has been most prevalent, though one vendor had a more principled
reason for avoiding it—similar to the argument of this
document.</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 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"/>, 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 anchor="pktb_Ideal" title="Sufficiency of Packet-Mode Drop">
<t>This Appendix is informative, not normative.</t>
<t>Here we check that packet-mode drop (or marking) in the network gives
sufficiently generic information for the transport layer to use. We
check against a 2x2 matrix of four scenarios that may occur now or in
the future (<xref target="pktb_Tab_Main_Scenarios"/>). The horizontal
and vertical dimensions have been chosen because each tests extremes of
sensitivity to packet size in the transport and in the network
respectively.</t>
<t>Note that this section does not consider byte-mode drop at all.
Having deprecated byte-mode drop, the goal here is to check that
packet-mode drop will be sufficient in all cases.</t>
<?rfc needLines="6" ?>
<texttable anchor="pktb_Tab_Main_Scenarios"
title="Four Possible Congestion Scenarios">
<ttcol
align="left"> Transport
Network</ttcol>
<ttcol align="center">a) Independent of packet size of congestion
notifications</ttcol>
<ttcol align="center">b) Dependent on packet size of congestion
notifications</ttcol>
<c>1) Predominantly bit-congestible network</c>
<c>Scenario a1)</c>
<c>Scenario b1)</c>
<c>2) Mix of bit-congestible and pkt-congestible network</c>
<c>Scenario a2)</c>
<c>Scenario b2)</c>
</texttable>
<t><xref target="pktb_Size-Dependence_Transport"/> focuses on the
horizontal dimension of <xref target="pktb_Tab_Main_Scenarios"/>
checking that packet-mode drop (or marking) gives sufficient
information, whether or not the transport uses it—scenarios b) and
a) respectively.</t>
<t><xref target="pktb_bit_pkt-congestible"/> focuses on the vertical
dimension of <xref target="pktb_Tab_Main_Scenarios"/>, checking that
packet-mode drop gives sufficient information to the transport whether
resources in the network are bit-congestible or packet-congestible
(these terms are defined in <xref target="pktb_term"/>).</t>
<t><list style="hanging">
<t hangText="Notation:">To be concrete, we will compare two flows
with different packet sizes, s_1 and s_2. As an example, we will
take s_1 = 60B = 480b and s_2 = 1500B = 12,000b.</t>
<t hangText="">A flow's bit rate, x [bps], is related to its packet
rate, u [pps], by <list style="empty">
<t>x(t) = s.u(t).</t>
</list></t>
<t>In the bit-congestible case, path congestion will be denoted by
p_b, and in the packet-congestible case by p_p. When either case is
implied, the letter p alone will denote path congestion.</t>
</list></t>
<section anchor="pktb_Size-Dependence_Transport"
title="Packet-Size (In)Dependence in Transports">
<t>In all cases we consider a packet-mode drop queue that indicates
congestion by dropping (or marking) packets with probability p
irrespective of packet size. We use an example value of loss
(marking) probability, p=0.1%.</t>
<t>A transport like RFC5681 TCP treats a congestion notification on
any packet whatever its size as one event. However, a network with
just the packet-mode drop algorithm does give more information if the
transport chooses to use it. We will use <xref
target="pktb_Tab_Absolute_and_Ratio"/> to illustrate this.</t>
<t>We will set aside the last column until later. The columns labelled
"Flow 1" and "Flow 2" compare two flows consisting of 60B and 1500B
packets respectively. The body of the table considers two separate
cases, one where the flows have equal bit-rate and the other with
equal packet-rates. In both cases, the two flows fill a 96Mbps link.
Therefore, in the equal bit-rate case they each have half the bit-rate
(48Mbps). Whereas, with equal packet-rates, flow 1 uses 25 times
smaller packets so it gets 25 times less bit-rate—it only gets
1/(1+25) of the link capacity (96Mbps/26 = 4Mbps after rounding). In
contrast flow 2 gets 25 times more bit-rate (92Mbps) in the equal
packet rate case because its packets are 25 times larger. The packet
rate shown for each flow could easily be derived once the bit-rate was
known by dividing bit-rate by packet size, as shown in the column
labelled "Formula".</t>
<texttable anchor="pktb_Tab_Absolute_and_Ratio" style="headers"
title="Absolute Loss Rates and Loss Ratios for Flows of Small and Large Packets and Both Combined">
<ttcol>Parameter</ttcol>
<ttcol>Formula</ttcol>
<ttcol align="right">Flow 1</ttcol>
<ttcol align="right">Flow 2</ttcol>
<ttcol align="right">Combined</ttcol>
<c>Packet size</c>
<c>s/8</c>
<c>60B</c>
<c>1,500B</c>
<c>(Mix)</c>
<c>Packet size</c>
<c>s</c>
<c>480b</c>
<c>12,000b</c>
<c>(Mix)</c>
<c>Pkt loss probability</c>
<c>p</c>
<c>0.1%</c>
<c>0.1%</c>
<c>0.1%</c>
<c> </c>
<c/>
<c/>
<c/>
<c/>
<c>EQUAL BIT-RATE CASE</c>
<c/>
<c/>
<c/>
<c/>
<c>Bit-rate</c>
<c>x</c>
<c>48Mbps</c>
<c>48Mbps</c>
<c>96Mbps</c>
<c>Packet-rate</c>
<c>u = x/s</c>
<c>100kpps</c>
<c>4kpps</c>
<c>104kpps</c>
<c>Absolute pkt-loss-rate</c>
<c>p*u</c>
<c>100pps</c>
<c>4pps</c>
<c>104pps</c>
<c>Absolute bit-loss-rate</c>
<c>p*u*s</c>
<c>48kbps</c>
<c>48kbps</c>
<c>96kbps</c>
<c>Ratio of lost/sent pkts</c>
<c>p*u/u</c>
<c>0.1%</c>
<c>0.1%</c>
<c>0.1%</c>
<c>Ratio of lost/sent bits</c>
<c>p*u*s/(u*s)</c>
<c>0.1%</c>
<c>0.1%</c>
<c>0.1%</c>
<c> </c>
<c/>
<c/>
<c/>
<c/>
<c>EQUAL PACKET-RATE CASE</c>
<c/>
<c/>
<c/>
<c/>
<c>Bit-rate</c>
<c>x</c>
<c>4Mbps</c>
<c>92Mbps</c>
<c>96Mbps</c>
<c>Packet-rate</c>
<c>u = x/s</c>
<c>8kpps</c>
<c>8kpps</c>
<c>15kpps</c>
<c>Absolute pkt-loss-rate</c>
<c>p*u</c>
<c>8pps</c>
<c>8pps</c>
<c>15pps</c>
<c>Absolute bit-loss-rate</c>
<c>p*u*s</c>
<c>4kbps</c>
<c>92kbps</c>
<c>96kbps</c>
<c>Ratio of lost/sent pkts</c>
<c>p*u/u</c>
<c>0.1%</c>
<c>0.1%</c>
<c>0.1%</c>
<c>Ratio of lost/sent bits</c>
<c>p*u*s/(u*s)</c>
<c>0.1%</c>
<c>0.1%</c>
<c>0.1%</c>
</texttable>
<t>So far we have merely set up the scenarios. We now consider
congestion notification in the scenario. Two TCP flows with the same
round trip time aim to equalise their packet-loss-rates over time.
That is the number of packets lost in a second, which is the packets
per second (u) multiplied by the probability that each one is dropped
(p). Thus TCP converges on the "Equal packet-rate" case, where both
flows aim for the same "Absolute packet-loss-rate" (both 8pps in the
table).</t>
<t>Packet-mode drop actually gives flows sufficient information to
measure their loss-rate in bits per second, if they choose, not just
packets per second. Each flow can count the size of a lost or marked
packet and scale its rate-response in proportion (as TFRC-SP does).
The result is shown in the row entitled "Absolute bit-loss-rate",
where the bits lost in a second is the packets per second (u)
multiplied by the probability of losing a packet (p) multiplied by the
packet size (s). Such an algorithm would try to remove any imbalance
in bit-loss-rate such as the wide disparity in the "Equal packet-rate"
case (4kbps vs. 92kbps). Instead, a packet-size-dependent algorithm
would aim for equal bit-loss-rates, which would drive both flows
towards the "Equal bit-rate" case, by driving them to equal
bit-loss-rates (both 48kbps in this example).</t>
<t>The explanation so far has assumed that each flow consists of
packets of only one constant size. Nonetheless, it extends naturally
to flows with mixed packet sizes. In the right-most column of <xref
target="pktb_Tab_Absolute_and_Ratio"/> a flow of mixed size packets is
created simply by considering flow 1 and flow 2 as a single aggregated
flow. There is no need for a flow to maintain an average packet size.
It is only necessary for the transport to scale its response to each
congestion indication by the size of each individual lost (or marked)
packet. Taking for example the "Equal packet-rate" case, in one second
about 8 small packets and 8 large packets are lost (making closer to
15 than 16 losses per second due to rounding). If the transport
multiplies each loss by its size, in one second it responds to 8*480b
and 8*12,000b lost bits, adding up to 96,000 lost bits in a second.
This double checks correctly, being the same as 0.1% of the total
bit-rate of 96Mbps. For completeness, the formula for absolute
bit-loss-rate is p(u1*s1+u2*s2).</t>
<t>Incidentally, a transport will always measure the loss probability
the same irrespective of whether it measures in packets or in bytes.
In other words, the ratio of lost to sent packets will be the same as
the ratio of lost to sent bytes. (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"/>, mainly for orthogonal
security reasons.) This is intuitively obvious by comparing two
example flows; one with 60B packets, the other with 1500B packets. If
both flows pass through a queue with drop probability 0.1%, each flow
will lose 1 in 1,000 packets. In the stream of 60B packets the ratio
of bytes lost to sent will be 60B in every 60,000B; and in the stream
of 1500B packets, the loss ratio will be 1,500B out of 1,500,000B.
When the transport responds to the ratio of lost to sent packets, it
will measure the same ratio whether it measures in packets or bytes:
0.1% in both cases. The fact that this ratio is the same whether
measured in packets or bytes can be seen in <xref
target="pktb_Tab_Absolute_and_Ratio"/>, where the ratio of lost to
sent packets and the ratio of lost to sent bytes is always 0.1% in all
cases (recall that the scenario was set up with p=0.1%).</t>
<t>This discussion of how the ratio can be measured in packets or
bytes is only raised here to highlight that it is irrelevant to this
memo! Whether a transport depends on packet size or not depends on how
this ratio is used within the congestion control algorithm.</t>
<t>So far we have shown that packet-mode drop passes sufficient
information to the transport layer so that the transport can take
account of bit-congestion, by using the sizes of the packets that
indicate congestion. We have also shown that the transport can choose
not to take packet size into account if it wishes. We will now
consider whether the transport can know which to do.</t>
</section>
<section anchor="pktb_bit_pkt-congestible"
title="Bit-Congestible and Packet-Congestible Indications">
<t>As a thought-experiment, imagine an idealised congestion
notification protocol that supports both bit-congestible and
packet-congestible resources. It would require at least two ECN flags,
one for each of bit-congestible and packet-congestible resources.
<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>This hides a fundamental problem—much more fundamental than
whether we can magically create header space for yet another ECN flag,
or whether it would work while being deployed incrementally.
Distinguishing drop from delivery naturally provides just one implicit
bit of congestion indication information—the packet is either
dropped or not. 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 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 a
conceptual device to illustrate a dilemma we could face in the future.
<xref target="pktb_Motivation"/> gives four good reasons why it would
be a bad idea to allow for packet size by biasing drop probability in
favour of small packets within the network. The impracticality of our
thought experiment shows that it will be hard to give transports a
practical way to know whether to take account of the size of
congestion indication packets or not.</t>
<t>Fortunately, this dilemma is not pressing because by design most
equipment becomes bit-congested before its packet-processing becomes
congested (as already outlined in <xref target="pktb_term"/>).
Therefore transports can be designed on the relatively sound
assumption that a congestion indication will usually imply
bit-congestion.</t>
<t>Nonetheless, although the above idealised protocol isn't intended
for implementation, we do want to emphasise that research is needed to
predict whether there are good reasons to believe that packet
congestion might become more common, and if so, to find a way to
somehow distinguish between bit and packet congestion <xref
target="RFC3714"/>.</t>
<t>Recently, the dual resource queue (DRQ) proposal <xref
target="DRQ"/> has been made on the premise that, as network
processors become more cost effective, per packet operations will
become more complex (irrespective of whether more function in the
network 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>Finally, we note one further complication. Strictly,
packet-congestible resources are often cycle-congestible. For
instance, for routing look-ups load depends on the complexity of each
look-up and whether the pattern of arrivals is amenable to caching or
not. This also 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>
</section>
</section>
<section anchor="pktb_Policing_Congestion_Response"
title="Byte-mode Drop Complicates Policing Congestion Response">
<t>This section is informative, not normative.</t>
<t>There are two main classes of approach to policing congestion
response: i) policing at each bottleneck link or ii) policing at the
edges of networks. Packet-mode drop in RED is compatible with either,
while byte-mode drop precludes edge policing.</t>
<t>The simplicity of an edge policer relies on one dropped or marked
packet being equivalent to another of the same size without having to
know which link the drop or mark occurred at. However, the byte-mode
drop algorithm has to depend on the local MTU of the line—it needs
to use some concept of a 'normal' packet size. Therefore, one dropped or
marked packet from a byte-mode drop algorithm is not necessarily
equivalent to another from a different link. A policing function local
to the link can know the local MTU where the congestion occurred.
However, a policer at the edge of the network cannot, at least not
without a lot of complexity.</t>
<t>The early research proposals for type (i) policing at a bottleneck
link <xref target="pBox"/> used byte-mode drop, then detected flows that
contributed disproportionately to the number of packets dropped.
However, with no extra complexity, later proposals used packet mode drop
and looked for flows that contributed a disproportionate amount of
dropped bytes <xref target="CHOKe_Var_Pkt"/>.</t>
<t>Work is progressing on the congestion exposure protocol (ConEx <xref
target="RFC6789"/>), which enables a type (ii) edge policer located at a
user's attachment point. The idea is to be able to take an integrated
view of the effect of all a user's traffic on any link in the
internetwork. However, byte-mode drop would effectively preclude such
edge policing because of the MTU issue above.</t>
<t>Indeed, making drop probability depend on the size of the packets
that bits happen to be divided into would simply encourage the bits to
be divided into smaller packets in order to confuse policing. In
contrast, as long as a dropped/marked packet is 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"/>.</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://tools.ietf.org/wg/tsvwg/draft-ietf-tsvwg-byte-pkt-congest/>
(courtesy of the rfcdiff tool): <list style="hanging">
<t hangText="From -11 to -12:">Following the second pass through the
IESG:<list style="symbols">
<t>Section 2.1 [Barry Leiba]:<list style="symbols">
<t>s/No other choice makes sense,/Subject to the exceptions
below, no other choice makes sense,/</t>
<t>s/Exceptions to these recommendations MAY be necessary
/Exceptions to these recommendations may be necessary /</t>
</list></t>
<t>Sections 3.2 and 4.2.3 [Joel Jaeggli]:<list style="symbols">
<t>Added comment to section 4.2.3 that the examples given
are not in widespread production use, but they give evidence
that it is possible to follow the advice given.</t>
<t>Section 4.2.3: <list style="symbols">
<t>OLD: Although there are no known proposals, 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
[RFC2474] to request a scheduling class with lower drop.
<vspace blankLines="0"/>NEW: Although there are no known
proposals, it would also be possible and perfectly valid
to make control packets robust against drop by
requesting a scheduling class with lower drop
probability, by re-marking to a Diffserv code point
[RFC2474] within the same behaviour aggregate.</t>
<t>appended "Similarly applications, over non-TCP
transports could make any packets that are effectively
control packets more robust by using Diffserv, data
duplication, FEC etc."</t>
</list></t>
<t>Updated Wischik ref and added "Reducing Web Latency: the
Virtue of Gentle Aggression" ref.</t>
</list></t>
<t>Expanded more abbreviations (CoDel, PIE, MTU).</t>
<t>Section 1. Intro [Stephen Farrell]: <list style="symbols">
<t>In the places where the doc desribes the dichotomy
between 'long-term goal' and 'expediency' the words long
term goal and expedient have been introduced, to more
explicitly refer back to this introductory para (S.2.1 &
S.2.3).</t>
<t>Added explanation of what scaling with packet size
means.</t>
</list></t>
<t>Conclusions [Benoit Claise]: <list style="symbols">
<t>OLD: For the specific case of RED, this means that
byte-mode queue measurement will often be appropriate
although byte-mode drop is strongly deprecated. <vspace
blankLines="0"/>NEW: For the specific case of RED, this
means that byte-mode queue measurement will often be
appropriate but the use of byte-mode drop is very strongly
discouraged.</t>
</list></t>
</list></t>
<t hangText="From -10 to -11:">Following a further WGLC:<list
style="symbols">
<t>Abstract: clarified that advice applies to all AQMs including
newer ones</t>
<t>Abstract & Intro: changed 'read' to 'detect', because you
don't read losses, you detect them.</t>
<t>S.1. Introduction: Disambiguated summary of advice on queue
measurement.</t>
<t>Clarified that the doc deprecates any preference based solely
on packet size, it's not only against preferring smaller
packets.</t>
<t>S.4.1.2. Congestion Measurement without a Queue: Explained
that a queue of TXOPs represents a queue into spectrum congested
by too many bits.</t>
<t>S.5.2: Bit- & Packet-congestible Network: Referred to
explanation in S.4.1.2 to make the point that TXOPs are not a
primary unit of workload like bits and packets are, even though
you get queues of TXOPs.</t>
<t>6. Security: Disambiguated 'bias towards'.</t>
<t>8. Conclusions: Made consistent with recommendation to use
time if possible for queue measurement.</t>
</list></t>
<t hangText="From -09 to -10:">Following IESG review:<list
style="symbols">
<t>Updates 2309: Left header unchanged reflecting eventual IESG
consensus [Sean Turner, Pete Resnick].</t>
<t>S.1 Intro: This memo adds to the congestion control
principles enumerated in BCP 41 [Pete Resnick]</t>
<t>Abstract, S.1, S.1.1, s.1.2 Intro, Scoping and Example: Made
applicability to all AQMs clearer listing some more example AQMs
and explained that we always use RED for examples, but this
doesn't mean it's not applicable to other AQMs. [A number of
reviewers have described the draft as "about RED"]</t>
<t>S.1 & S.2.1 Queue measurement: Explained that the choice
between measuring the queue in packets or bytes is only relevant
if measuring it in time units is infeasible [So as not to imply
that we haven't noticed the advances made by PDPC &
CoDel]</t>
<t>S.1.1. Terminology: Better explained why hybrid systems
congested by both packets and bytes are often designed to be
treated as bit-congestible [Richard Barnes].</t>
<t>S.2.1. Queue measurement advice: Added examples. Added a
counter-example to justify SHOULDs rather than MUSTs. Pointed to
S.4.1 for a list of more complicated scenarios. [Benson
Schliesser, OpsDir]</t>
<t>S2.2. Recommendation on Encoding Congestion Notification:
Removed SHOULD treat packets equally, leaving only SHOULD NOT
drop dependent on packet size, to avoid it sounding like we're
saying QoS is not allowed. Pointed to possible app-specific
legacy use of byte-mode as a counter-example that prevents us
saying MUST NOT. [Pete Resnick]</t>
<t>S.2.3. Recommendation on Responding to Congestion:
capitalised the two SHOULDs in recommendations for TCP, and gave
possible counter-examples. [noticed while dealing with Pete
Resnick's point]</t>
<t>S2.4. Splitting & Merging: RTCP -> RTP/RTCP [Pete
McCann, Gen-ART]</t>
<t>S.3.2 Small != Control: many control packets are small ->
...tend to be small [Stephen Farrell]</t>
<t>S.3.1 Perverse incentives: Changed transport designers to app
developers [Stephen Farrell]</t>
<t>S.4.1.1. Fixed Size Packet Buffers: Nearly completely
re-written to simplify and to reverse the advice when the
underlying resource is bit-congestible, irrespective of whether
the buffer consists of fixed-size packet buffers. [Richard
Barnes & Benson Schliesser]</t>
<t>S.4.2.1.2. Packet Size Bias Regardless of AQM: Largely
re-written to reflect the earlier change in advice about
fixed-size packet buffers, and to primarily focus on getting rid
of tail-drop, not various nuances of tail-drop. [Richard Barnes
& Benson Schliesser]</t>
<t>Editorial corrections [Tim Bray, AppsDir, Pete McCann,
Gen-ART and others]</t>
<t>Updated refs (two I-Ds have become RFCs). [Pete McCann]</t>
</list></t>
<t hangText="From -08 to -09:">Following WG last call:<list
style="symbols">
<t>S.2.1: Made RED-related queue measurement recommendations
clearer</t>
<t>S.2.3: Added to "Recommendation on Responding to Congestion"
to make it clear that we are definitely not saying transports
have to equalise bit-rates, just how to do it and not do it, if
you want to.</t>
<t>S.3: Clarified motivation sections S.3.3
"Transport-Independent Network" and S.3.5 "Implementation
Efficiency"</t>
<t>S.3.4: Completely changed motivating argument from "Scaling
Congestion Control with Packet Size" to "Partial Deployment of
AQM".</t>
</list></t>
<t hangText="From -07 to -08:"><list style="symbols">
<t>Altered abstract to say it provides best current practice and
highlight that it updates RFC2309</t>
<t>Added null IANA section</t>
<t>Updated refs</t>
</list></t>
<t hangText="From -06 to -07:"><list style="symbols">
<t>A mix-up with the corollaries and their naming in 2.1 to 2.3
fixed.</t>
</list></t>
<t hangText="From -05 to -06:"><list style="symbols">
<t>Primarily editorial fixes.</t>
</list></t>
<t hangText="From -04 to -05:"><list style="symbols">
<t>Changed from Informational to BCP and highlighted
non-normative sections and appendices</t>
<t>Removed language about consensus</t>
<t>Added "Example Comparing Packet-Mode Drop and Byte-Mode
Drop"</t>
<t>Arranged "Motivating Arguments" into a more logical order and
completely rewrote "Transport-Independent Network" &
"Scaling Congestion Control with Packet Size" arguments. Removed
"Why Now?"</t>
<t>Clarified applicability of certain recommendations</t>
<t>Shifted vendor survey to an Appendix</t>
<t>Cut down "Outstanding Issues and Next Steps"</t>
<t>Re-drafted the start of the conclusions to highlight the
three distinct areas of concern</t>
<t>Completely re-wrote appendices</t>
<t>Editorial corrections throughout.</t>
</list></t>
<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"/>.</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"/>.</t>
<t>Clarified avoiding lock-out of large packets is not the main
or only motivation for RED.</t>
<t>Mentioned choice of drop or marking explicitly throughout,
rather than trying to coin a word to mean either.</t>
<t>Generalised the discussion throughout to any packet
forwarding function on any network equipment, not just
routers.</t>
<t>Clarified the last point about why this is a good time to
sort out this issue: because it will be hard / impossible to
design new transports unless we decide whether the network or
the transport is allowing for packet size.</t>
<t>Added statement explaining the horizon of the memo is long
term, but with short term expediency in mind.</t>
<t>Added material on scaling congestion control with packet size
(<xref target="pktb_Scaling"/>).</t>
<t>Separated out issue of normalising TCP's bit rate from issue
of preference to control packets (<xref
target="pktb_Small.NE.Control"/>).</t>
<t>Divided up Congestion Measurement section for clarity,
including new material on fixed size packet buffers and buffer
carving (<xref target="pktb_Fixed_Buffers"/> & <xref
target="pktb_Network_Bias"/>) and on congestion measurement in
wireless link technologies without queues (<xref
target="pktb_Measurement_NoQ"/>).</t>
<t>Added section on 'Making Transports Robust against Control
Packet Losses' (<xref
target="pktb_Transport_Robust_Ctrl_Loss"/>) with existing &
new material included.</t>
<t>Added tabulated results of vendor survey on byte-mode drop
variant of RED (<xref target="pktb_Tab_RED_Survey"/>).</t>
</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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