One document matched: draft-briscoe-tsvwg-byte-pkt-mark-01.xml
<?xml version="1.0" encoding="US-ASCII"?>
<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
<rfc category="info" docName="draft-briscoe-tsvwg-byte-pkt-mark-01"
ipr="full3978">
<?xml-stylesheet type='text/xsl' href='http://xml.resource.org/authoring/rfc2629.xslt' ?>
<!-- Alterations to I-D/RFC boilerplate -->
<?rfc private="" ?>
<!-- Default private="" Produce an internal memo 2.5pp shorter than an I-D or RFC -->
<?rfc topblock="yes" ?>
<!-- Default topblock="yes" put the famous header block on the first page -->
<?rfc footer="" ?>
<!-- Default footer="Expires <date>" override the center footer string -->
<?rfc header="" ?>
<!-- Default header="Internet-Draft" override the leftmost header string -->
<?rfc authorship="yes" ?>
<!-- Default authorship="yes" Render authors' addresses section -->
<?rfc rfcprocack="yes" ?>
<!-- Default rfcprocack="no" add a short sentence acknowledging xml2rfc -->
<?rfc strict="no" ?>
<!-- Default strict="no" Don't check I-D nits -->
<?rfc rfcedstyle="no" ?>
<!-- Default rfcedstyle="yes" attempt to closely follow finer details from the latest observable RFC-Editor style -->
<?rfc colonspace="no" ?>
<!-- Default colonspace="no" put two spaces instead of one after each colon (``:'') in txt or nroff files -->
<?rfc notedraftinprogress="yes" ?>
<!-- Default notedraftinprogress="yes" generates "(work in progress)", as appropriate -->
<?rfc refparent="References" ?>
<!-- Default refparent="References" title of the top-level section containing all references -->
<!-- IETF process -->
<?rfc iprnotified="no" ?>
<!-- Default iprnotified="no" I haven't disclosed existence of IPR to IETF -->
<!-- ToC format -->
<?rfc toc="yes" ?>
<!-- Default toc="no" No Table of Contents -->
<?rfc tocappendix="yes" ?>
<!-- Default tocappendix="yes" control whether the word `Appendix' appears in the table-of-content -->
<?rfc tocdepth="3" ?>
<!-- Default tocdepth="3" if toc is "yes", then this determines the depth of the table-of-contents -->
<?rfc tocindent="yes" ?>
<!-- Default tocindent="yes" if toc is "yes", indent subsections in the table-of-contents -->
<?rfc tocnarrow="yes" ?>
<!-- Default tocnarrow="yes" affects horizontal spacing in the table-of-content -->
<?rfc tocompact="yes" ?>
<!-- Default tocompact="yes" if toc is "yes", then setting this to "no" will make it a little less compact -->
<!-- Cross referencing, footnotes, comments -->
<?rfc symrefs="yes" ?>
<!-- Default symrefs="no" Don't use anchors, but use numbers for refs -->
<?rfc sortrefs="yes"?>
<!-- Default sortrefs="no" Don't sort references into order -->
<?rfc comments="yes" ?>
<!-- Default comments="yes" Don't render comments -->
<?rfc inline="no" ?>
<!-- Default inline="no" if comments is "yes", then render comments inline; otherwise render them in an `Editorial Comments' section -->
<?rfc editing="no" ?>
<!-- Default editing="no" Don't insert editing marks for ease of discussing draft versions -->
<!-- Pagination control -->
<?rfc compact="yes"?>
<!-- Default compact="no" Start sections on new pages -->
<?rfc subcompact="no"?>
<!-- Default subcompact="(as compact setting)" yes/no is not quite as compact as yes/yes -->
<?rfc autobreaks="yes" ?>
<!-- Default autobreaks="yes" avoid widows and orphans (not perfect) -->
<!-- HTML formatting control -->
<?rfc emoticonic="yes" ?>
<!-- Default emoticonic="no" Doesn't prettify HTML format -->
<?rfc background="" ?>
<!-- Default background="" when producing a html file, use this image -->
<?rfc useobject="no" ?>
<!-- Default useobject="no" use <object> not <src> when outputting HTML -->
<?rfc linkmailto="yes" ?>
<!-- Default linkmailto="yes" generate mailto: URL, as appropriate -->
<?rfc docmapping="no" ?>
<!-- Default docmapping="no" use hierarchical tags (e.g., <h1>, <h2>, etc.) for (sub)section titles -->
<?rfc slides="no" ?>
<!-- Default slides="no" when producing a html file, produce multiple files for a slide show -->
<front>
<title abbrev="Byte and Packet Congestion Notification">Byte and Packet
Congestion Notification</title>
<author fullname="Bob Briscoe" initials="B." surname="Briscoe">
<organization>BT & UCL</organization>
<address>
<postal>
<street>B54/77, Adastral Park</street>
<street>Martlesham Heath</street>
<city>Ipswich</city>
<code>IP5 3RE</code>
<country>UK</country>
</postal>
<phone>+44 1473 645196</phone>
<email>bob.briscoe@bt.com</email>
<uri>http://www.cs.ucl.ac.uk/staff/B.Briscoe/</uri>
</address>
</author>
<date day="19" month="November" year="2007" />
<area>Transport</area>
<workgroup>Transport Area Working Group</workgroup>
<keyword>Quality of Service</keyword>
<keyword>QoS</keyword>
<keyword>Congestion Control</keyword>
<keyword>Protocol</keyword>
<abstract>
<t>This memo concerns dropping or marking packets using active queue
management (AQM) such as random early detection (RED) or pre-congestion
notification (PCN). It answers the question of whether to take packet
size into account when network equipment writes congestion notification,
or when transports read it. The primary conclusion is that the variant
of RED that gives lower drop probability to smaller packets (byte-mode
packet drop) should not be used because it creates a perverse incentive
for transports to use tiny segments, consequently also opening up a DoS
vulnerability. TCP's lack of attention to packet size and its
sensitivity to loss of SYNs and ACKs should be fixed in TCP, not by
reverse engineering network forwarding to fix transport protocols.
Nonetheless raw drop-tail is just as vulnerable to gaming by small
packets, so AQM itself should not be turned off.</t>
</abstract>
</front>
<middle>
<!-- ================================================================ -->
<section anchor="pktb_Introduction" title="Introduction">
<t>When notifying congestion, the problem of how (and whether) to take
packet sizes into account has exercised the minds of researchers and
practitioners for as long as active queue management (AQM) has been
discussed. Indeed, AQM was originally introduced largely to remove the
advantage that small packets get from drop-tail queues. This memo aims
to state the principles we should be using and to come to conclusions on
what these principles will mean for future protocol design, taking into
account the deployments we have already.</t>
<t>Note that the byte vs. packet dilemma concerns congestion
notification irrespective of whether it is signalled implicitly by drop
or using explicit congestion notification (ECN <xref
target="RFC3168"></xref>). Throughout this document, unless clear from
the context, the term congestion marking, or just marking, will be used
to mean either drop or explicit congestion notification.</t>
<t>If the load on a resource depends on the rate at which packets
arrive, it is called packet-congestible. If the load depends on the rate
at which bits arrive it is called bit-congestible.</t>
<t>Examples of packet-congestible resources are route look-up engines
and firewalls, because load depends on how many packet headers they have
to process. Examples of bit-congestible resources are transmission
links, and buffer memory, because the load depends on how many bits they
have to transmit or store. Note that information is generally processed
or transmitted with a minimum granularity greater than a bit (e.g.
octets). The appropriate granularity for the resource in question SHOULD
be used, but for the sake of brevity we will talk in terms of bytes in
this memo.</t>
<t>Resources may be congestible at higher levels of granularity than
packets, for instance stateful firewalls are flow-congestible and
call-servers are session-congestible. This memo focuses on congestion of
connectionless resources, but the same principles may be applied for
congestion notification protocols controlling per-flow and per-session
processing or state.</t>
<t>The byte vs. packet dilemma arises at three stages in the congestion
notification process: <list style="hanging">
<t hangText="Measuring congestion">When the congested resource
decides locally how to measure how congested it is. (Should the
queue be measured in bytes or packets?);</t>
<t
hangText="Coding congestion notification into the wire protocol:">When
the congested resource decides how to notify the level of
congestion. (Should the level of notification depend on the
byte-size of each particular packet carrying the notification?);</t>
<t
hangText="Decoding congestion notification from the wire protocol:">When
the transport interprets the notification. (Should the byte-size of
a missing or marked packet be taken into account?).</t>
</list>In RED, whether to use packets or bytes when measuring queues
is called packet-mode or byte-mode queue measurement. This choice is now
fairly well understood but is included in <xref
target="pktb_Measure"></xref> to document it in the RFC series.</t>
<t>The controversy is mainly around the other two stages: whether to
allow for packet size when the network codes or when the transport
decodes congestion notification. In RED, the variant that reduces drop
probability for packets based on their size in bytes is called byte-mode
drop, while the variant that doesn't is called packet mode drop. Whether
queues are measured in bytes or packets is an orthogonal choice, termed
byte-mode queue measurement or packet-mode queue measurement.</t>
<t>Currently, the paper trail of advice referenced from the RFC series
conditionally recommends byte-mode (packet-size dependent) drop,
although all the implementers who responded to our survey have ignored
this advice. The primary purpose of this memo is to build a definitive
consensus against allowing for packet size in AQM algorithms and record
this advice within the RFC series.</t>
<t>Increasingly, it is being recognised that a protocol design must take
care not to cause unintended consequences by giving the parties in the
protocol exchange perverse incentives <xref
target="Evol_cc"></xref><xref target="RFC3426"></xref>. For instance,
imagine a scenario where the same bit rate of packets will contribute
the same to congestion of a link irrespective of whether it is sent as
fewer larger packets or more smaller packets. A protocol design that
caused larger packets to be more likely to be dropped than smaller ones
would be dangerous in this case. Transports would tend to act in their
own interests by breaking their data stream down into tiny segments,
reducing their drop rate without reducing their bit rate. Further,
encouraging a high volume of tiny packets might in turn unnecessarily
overload a completely unrelated part of the system, perhaps more limited
by header-processing than bandwidth.</t>
<t>Imagine two flows arrive at a bit-congestible transmission link each
with the same bit rate, say 1Mbps, but one consists of 1500B and the
other 60B packets, which are 25x smaller. If the advice referred to from
RFC2309 is followed, gentle RED <xref target="gentle_RED"></xref> would
be used, configured to adjust the drop probability of packets in
proportion to each packet's size (byte mode packet drop). So in this
case, if RED drops 25% of the larger packets, it will aim to drop 1% of
the smaller packets (but in practice it may drop more as congestion
increases <xref target="RFC4828"></xref>(§B.4)<cref
anchor="Note_Variation">The algorithm of the byte-mode drop variant of
RED switches off any bias towards small packets whenever the smoothed
queue length dictates that the drop probability of large packets should
be 100%. In the example in the Introduction, as the large packet drop
probability varies around 25% the small packet drop probability will
vary around 1%, but with occasional jumps to 100% whenever the
instantaneous queue (after drop) manages to sustain a length above the
100% drop point for longer than the queue averaging period.</cref>).
Even though both flows arrive with the same bit rate, the bit rate the
RED queue aims to pass to the line will be 750k for the flow of larger
packet but 990k for the smaller packets (but because of rate variation
it will be less than this target). It can be seen that this behaviour
reopens the same denial of service vulnerability that drop tail queues
offer to floods of small packet, though not necessarily as strongly (see
<xref target="pktb_Security_Considerations"></xref>).</t>
<t>The above advice (that referred to by RFC2309) says the question of
whether a packet's own size should affect its drop probability "depends
on the dominant end-to-end congestion control mechanisms". But we argue
the network layer should not be optimised for whatever transport is
predominant. For instance, TCP congestion control ensures that flows
competing for the same resource each maintain the same number of
segments in flight, irrespective of segment size. Even though reducing
the drop probability of small packets helps correct this feature of TCP,
we argue it should be corrected in TCP itself, not in the network.
Favouring small packets also reduces the chance of dropping SYNs and
pure ACKs, which has a disproportionate effect on TCP performance. But
again, rather than fix these problems in the network, we argue that TCP
should be altered. Effectively, favouring small packets is reverse
engineering of the network layer around TCP, contrary to the excellent
advice in <xref target="RFC3426"></xref>, which asks designers to
question "Why are you proposing a solution at this layer of the protocol
stack, rather than at another layer?"</t>
<t>Now is a good time to discuss whether fairness between different
sized packets would best be implemented in the network layer, or at the
transport, for a number of reasons: <list style="numbers">
<t>The packet vs. byte issue requires speedy resolution because the
IETF pre-congestion notification (PCN) working group is in the
process of being chartered to produce a standards track
specification of its congestion marking (AQM) algorithm <xref
target="PCNcharter"></xref>;</t>
<t><xref target="RFC2309"></xref> says RED may either take account
of packet size or not when dropping, but gives no recommendation
between the two, referring instead to advice on the performance
implications in an email <xref target="pktByteEmail"></xref>, which
recommends byte-mode drop. Further, just before RFC2309 was issued,
an addendum was added to the archived email that revisited the issue
of packet vs. byte-mode drop in its last para, making the
recommendation less clear-cut;</t>
<t>Without this memo, the only advice in the RFC series on packet
size bias in AQM algorithms would be a reference to an archived
email in <xref target="RFC2309"></xref> (including an addendum at
the end of the email to correct the original).</t>
<t>The IRTF Internet Congestion Control Research Group (ICCRG)
recently took on the challenge of building consensus on what common
congestion control support should be required from forwarding
engines on routers in the future <xref
target="I-D.irtf-iccrg-welzl-congestion-control-open-research"></xref>.
The wider Internet community needs to discuss whether the complexity
of adjusting for packet size should be on routers or in
transports;</t>
<t>Given there are many good reasons why larger path max
transmission units (PMTUs) would help solve a number of scaling
issues, we don't want to create any bias against large packets that
is greater than their true cost;</t>
<t>And finally, given it has recently been pointed out that TCP
doesn't achieve any meaningful fairness anyway <xref
target="Rate_fair_Dis"></xref>, because it doesn't consider fairness
over all the flows a user transmits nor over time, modifying the
network rather than modifying TCP still won't achieve fairness. It
seems more likely we have to face up to evolving beyond TCP
anyway.</t>
</list></t>
<t>This memo starts from first principles, defining congestion
notification in <xref target="pktb_Congestion_Definition"></xref> then
determining the correct way to measure congestion (<xref
target="pktb_Measure"></xref>) and to design an idealised congestion
notification protocol (<xref target="pktb_Ideal_Coding"></xref>). It
then surveys the advice given previously in the RFC series, the research
literature and the deployed legacy (<xref target="pktb_SotA"></xref>)
before listing outstanding issues (<xref target="pktb_Issues"></xref>)
that will need resolution both to achieve the ideal protocol and to
handle legacy. After discussing security considerations (<xref
target="pktb_Security_Considerations"></xref>) strong recommendations
for the way forward are given in the conclusions (<xref
target="pktb_Conclusions"></xref>).</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Reqs_notation" title="Requirements notation">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in <xref
target="RFC2119"></xref>.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Congestion_Definition"
title="Working Definition of Congestion Notification">
<t>Rather than aim to achieve what many have tried and failed, this memo
will not try to define congestion. It will give a working definition of
what congestion notification should be taken to mean for this document.
Congestion notification is a changing signal that aims to communicate
the ratio E/L, where E is the instantaneous excess load offered to a
resource that it cannot (or would not) serve and L is the instantaneous
offered load.</t>
<t>The phrase `would not serve' is added, because AQM systems (e.g. RED,
PCN <xref target="I-D.ietf-pcn-architecture"></xref>) use a virtual
capacity smaller than actual capacity, then notify congestion of this
virtual capacity in order to avoid congestion of the actual
capacity.</t>
<t>Note that the denominator is offered load, not capacity. Therefore
congestion notification is a real number bounded by the range [0,1].
This ties in with the most well-understood form of congestion
notification: drop rate. It also means that congestion has a natural
interpretation as a probability; the probability of offered traffic not
being served (or being marked as at risk of not being served). <xref
target="pktb_CN_Definition"></xref> describes a further incidental
benefit that arises from using load as the denominator of congestion
notification.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Measure" title="Congestion Measurement">
<t>Queue length is usually the most correct and simplest way to measure
congestion of a resource. To avoid the pathological effects of drop
tail, an AQM function can then be used to transform queue length into
the probability of dropping or marking a packet (e.g. RED's piecewise
linear function between thresholds). If the resource is bit-congestible,
the length of the queue SHOULD be measured in bytes. If the resource is
packet-congestible, the length of the queue SHOULD be measured in
packets. No other choice makes sense, because the number of packets
waiting in the queue isn't relevant if the resource gets congested by
bytes and vice versa. We discuss the implications on RED's byte mode and
packet mode for measuring queue length in <xref
target="pktb_SotA"></xref>.</t>
<t>There is a complication for some queuing hardware that consists of
fixed sized buffers. Each packet fills as many buffers as are necessary
leaving remaining space empty in the last buffer. Also, with some
hardware, any fixed sized buffers not completely filled by the end of a
packet are padded when transmitted to the wire.</t>
<t>Taking the extreme for the size of these buffers, a forwarding system
with both queuing and transmission in MTU-sized units should clearly be
treated as packet-congestible, because the queue length in packets would
be a good model of congestion of the lower layer link.</t>
<t>A hybrid forwarding system with transmission delay largely dependent
on the byte-size of packets but buffers of one MTU per packet would
strictly require a more complex algorithm to determine the probability
of congestion. It would have to be treated as two resources in sequence,
where the sum of the byte-sizes of the packets within each packet buffer
modelled congestion of the line while the length of the queue in packets
modelled congestion of the buffer. Then the probability of congesting
the forwarding buffer would have to be a conditional
probability—conditional on the previously calculated probability
of congesting the line. The sub-MTU-sized fixed buffers described above
would require a slightly more complex model to fully determine how best
to measure the queue. It would then be necessary to approximate this
back to some practical algorithm.</t>
<t>Not all congested resources lead to queues. For instance, wireless
spectrum is bit-congestible (for a given coding scheme), because
interference increases with the rate at which bits are transmitted. But
wireless link protocols do not always maintain a queue that depends on
spectrum interference. Similarly, power limited resources are also
usually bit-congestible if energy is primarily required for transmission
rather than header processing, but it is rare for a link protocol to
build a queue as it approaches maximum power.</t>
<t><xref target="ECNFixedWireless"></xref> proposes a practical and
theoretically sound way to combine congestion notification for different
bit-congestible resources along an end to end path, whether wireless or
wired, and whether with or without queues.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Ideal_Coding" title="Idealised Wire Protocol Coding">
<t>We will start by inventing an idealised congestion notification
protocol before discussing how to make it practical. The idealised
protocol is shown to be correct using examples in <xref
target="pktb_Scenarios"></xref>. Congestion notification involves the
congested resource coding a congestion notification signal into the
packet stream and the transports decoding it. The idealised protocol
uses two different fields in each datagram to signal congestion: one for
byte congestion and one for packet congestion.</t>
<t>We are not saying two ECN fields will be needed (and we are not
saying that somehow a resource should be able to drop a packet in one of
two different ways so that the transport can distinguish which sort of
drop it was!). These two congestion notification channels are just a
conceptual device. They allow us to defer having to decide whether to
distinguish between byte and packet congestion when the network resource
codes the signal or when the transport decodes it.</t>
<t>However, although this idealised mechanism isn't intended for
implementation, we do want to emphasise that we may need to find a way
to implement it, because it could become necessary to somehow
distinguish between bit and packet congestion <xref
target="RFC3714"></xref>. Currently a design goal of network processing
equipment such as routers and firewalls is to keep packet processing
uncongested even under worst case bit rates with minimum packet sizes.
Therefore, packet-congestion is currently rare, but there is no
guarantee that it will not become common with future technology
trends.</t>
<t>The idealised wire protocol is given below. It accounts for packet
sizes at the transport layer, not in the network, and then only in the
case of bit-congestible resources. This avoids the perverse incentive to
send smaller packets and the DoS vulnerability that would otherwise
result if the network were to bias towards them (see Introduction).
Incidentally, it also ensures neither the network nor the transport
needs to do a multiply—multiplication by packet size is
effectively achieved as a repeated add when the transport adds to its
count of marked bytes as each congestion event is fed to it: <list
style="symbols">
<t>A packet-congestible resource trying to code congestion level p_p
into a packet stream should mark the idealised `packet congestion'
field in each packet with probability p_p irrespective of the
packet's size. The transport should then take a packet with the
packet congestion field marked to mean just one mark, irrespective
of the packet size.</t>
<t>A bit-congestible resource trying to code time-varying
byte-congestion level p_b into a packet stream should mark the `byte
congestion' field in each packet with probability p_b, again
irrespective of the packet's size. Unlike before, the transport
should take a packet with the byte congestion field marked to count
as a mark on each byte in the packet.</t>
</list></t>
<t>The worked examples in <xref target="pktb_Scenarios"></xref> show
that transports can extract sufficient and correct congestion
notification from these protocols for cases when two flows with
different packet sizes have matching bit rates or matching packet rates.
Examples are also given that mix these two flows into one to show that a
flow with mixed packet sizes would still be able to extract sufficient
and correct information.</t>
<t>Sufficient and correct congestion information means that there is
sufficient information for the two different types of transport
requirements: <list style="hanging">
<t hangText="Ratio-based:">Established transport congestion controls
like TCP's <xref target="RFC2581"></xref> aim to achieve equal
segment rates per RTT through the same bottleneck—TCP
friendliness <xref target="RFC3448"></xref>. They work with the
ratio of marked to unmarked segments. The example scenarios show
that these ratio-based transports are effectively the same whether
counting in bytes or marks, because the units cancel out.
(Incidentally, this is why TCP's bit rate is still proportional to
packet size even when byte-counting is used, as recommended for TCP
in <xref target="I-D.ietf-tcpm-rfc2581bis"></xref>, mainly for
orthogonal security reasons.)</t>
<t hangText="Absolute-target-based:">Other congestion controls
proposed in the research community aim to limit the volume of
congestion caused to a constant weight parameter. <xref
target="MulTCP"></xref><xref target="WindowPropFair"></xref> are
examples of weighted proportionally fair transports designed for
cost-fair environments <xref target="Rate_fair_Dis"></xref>. In this
case, the transport requires a count (not a ratio) of dropped/marked
bytes in the bit-congestible case and of dropped/marked packets in
the packet congestible case.</t>
</list></t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_SotA" title="The State of the Art">
<t>The original 1993 paper on RED <xref target="RED93"></xref> proposed
two options for the RED active queue management algorithm: packet mode
and byte mode. Packet mode measured the queue length in packets and
marked (or dropped) individual packets with a probability independent of
their size. Byte mode measured the queue length in bytes and marked an
individual packet with probability in proportion to its size (relative
to the maximum packet size). In the paper's outline of further work, it
was stated that no recommendation had been made on whether the queue
size should be measured in bytes or packets, but noted that the
difference could be significant.</t>
<t>When RED was recommended for general deployment in 1998 <xref
target="RFC2309"></xref>, the two modes were mentioned implying the
choice between them was a question of performance, referring to a 1997
email <xref target="pktByteEmail"></xref> for advice on tuning. This
email clarified that there were in fact two orthogonal choices: whether
to measure queue length in bytes or packets (<xref
target="pktb_Measure_Status"></xref> below) and whether the drop
probability of an individual packet should depend on its own size (<xref
target="pktb_Coding_Status"></xref> below).</t>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Measure_Status"
title="Congestion Measurement: Status">
<t>The choice of which metric to use to measure queue length was left
open in RFC2309. It is now well understood that queues for
bit-congestible resources should be measured in bytes, and queues for
packet-congestible resources should be measured in packets (see <xref
target="pktb_Measure"></xref>).</t>
<t>Where buffers are not configured or legacy buffers cannot be
configured to the above guideline, we needn't have to make allowances
for such legacy in future protocol design. If a bit-congestible buffer
is measured in packets, the operator will have set the thresholds
mindful of a typical mix of packets sizes. Any AQM algorithm on such a
buffer will be oversensitive to high proportions of small packets,
e.g. a DoS attack, and undersensitive to high proportions of large
packets. But an operator can safely keep such a legacy buffer because
any undersensitivity during unusual traffic mixes cannot lead to
congestion collapse given the buffer will eventually revert to tail
drop, discarding proportionately more large packets.</t>
<t>Some modern router implementations give a choice for setting RED's
thresholds in byte-mode or packet-mode. This may merely be an
administrator-interface preference, not altering how the queue itself
is measured but on some hardware it does actually change the way it
measures its queue. Whether a resource is bit-congestible or
packet-congestible is a property of the resource, so an admin SHOULD
NOT ever need to, or be able to, configure the way a queue measures
itself.</t>
<t>We believe the question of whether to measure queues in bytes or
packets is fairly well understood these days. The only outstanding
issues concern how to measure congestion when the queue is bit
congestible but the resource is packet congestible or vice versa (see
<xref target="pktb_Measure"></xref>).</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Coding_Status" title="Congestion Coding: Status">
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Network_Bias" title="Network Bias when Encoding">
<t>The previously mentioned email <xref
target="pktByteEmail"></xref> referred to by <xref
target="RFC2309"></xref> said that the choice over whether a
packet's own size should affect its drop probability "depends on the
dominant end-to-end congestion control mechanisms". [<xref
target="pktb_Introduction"></xref> argues against this approach,
citing the excellent advice in RFC3246.] The referenced email went
on to argue that drop probability should depend on the size of the
packet being considered for drop if the resource is bit-congestible,
but not if it is packet-congestible, but advised that most scarce
resources in the Internet were currently bit-congestible. The
argument continued that if packet drops were inflated by packet size
(byte-mode dropping), "a flow's fraction of the packet drops is then
a good indication of that flow's fraction of the link bandwidth in
bits per second". This was consistent with a referenced policing
mechanism being worked on at the time for detecting unusually high
bandwidth flows, eventually published in 1999 <xref
target="pBox"></xref>. [The problem could have been solved by making
the policing mechanism count the volume of bytes randomly dropped,
not the number of packets.]</t>
<t>A few months before RFC2309 was published, an addendum was added
to the above archived email referenced from the RFC, in which the
final paragraph seemed to partially retract what had previously been
said. It clarified that the question of whether the probability of
marking a packet should depend on its size was not related to
whether the resource itself was bit congestible, but a completely
orthogonal question. However the only example given had the queue
measured in packets but packet drop depended on the byte-size of the
packet in question. No example was given the other way round.</t>
<t>In 2000, Cnodder et al <xref target="REDbyte"></xref> pointed out
that there was an error in the part of the original 1993 RED
algorithm that aimed to distribute drops uniformly, because it
didn't correctly take into account the adjustment for packet size.
They recommended an algorithm called RED_4 to fix this. But they
also recommended a further change, RED_5, to adjust drop rate
dependent on the square of relative packet size. This was indeed
consistent with the stated motivation behind RED's byte mode
drop—that we should reverse engineer the network to improve
the performance of dominant end-to-end congestion control
mechanisms.</t>
<t>By 2003, a further change had been made to the adjustment for
packet size, this time in the RED algorithm of the ns2 simulator.
Instead of taking each packet's size relative to a `maximum packet
size' it was taken relative to a `mean packet size', intended to be
a static value representative of the `typical' packet size on the
link. We have not been able to find a justification for this change
in the literature, however Eddy and Allman conducted experiments
<xref target="REDbias"></xref> that assessed how sensitive RED was
to this parameter, amongst other things. No-one seems to have
pointed out that this changed algorithm can often lead to drop
probabilities of greater than 1 [which should ring alarm bells
hinting that there's a mistake in the theory somewhere]. On
10-Nov-2004, this variant of byte-mode packet drop was made the
default in the ns2 simulator.</t>
<t>More recently, two drafts have proposed changes to TCP that make
it more robust against losing small control packets <xref
target="I-D.ietf-tcpm-ecnsyn"></xref> <xref
target="I-D.floyd-tcpm-ackcc"></xref>. In both cases they note that
the case for these TCP changes would be weaker if RED were biased
against dropping small packets. We argue here that these two
proposals are a safer and more principled way to achieve TCP
performance improvements than reverse engineering RED to benefit
TCP.</t>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Transport_Bias"
title="Transport Bias when Decoding">
<t>The above proposals to alter the network layer to fix TCP's
insensitivity to segment size have largely carried on outside the
IETF process (unless one counts a reference in an informational RFC
to an archived email!).</t>
<t>Within the IETF, a recently approved experimental RFC adapts its
transport layer protocol to take account of packet sizes relative to
typical TCP packet sizes. This proposes a new small-packet variant
of TCP-friendly rate control <xref target="RFC3448"></xref> called
TFRC-SP <xref target="RFC4828"></xref>. Essentially, it proposes a
rate equation that inflates the flow rate by the ratio of a typical
TCP segment size (1500B including TCP header) over the actual
segment size <xref target="PktSizeEquCC"></xref>. (There are also
other important differences of detail relative to TFRC, such as
using virtual packets <xref target="CCvarPktSize"></xref> to avoid
responding to multiple losses per round trip and using a minimum
inter-packet interval.)</t>
<t>Section 4.5.1 of this TFRC-SP spec discusses the implications of
operating in an environment where routers have been configured to
drop smaller packets with proportionately lower probability than
larger ones. But surprisingly, 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 rather than recommending any action.</t>
<t>The paper originally proposing TFRC with virtual packets
(VP-TFRC) <xref target="CCvarPktSize"></xref> proposed that there
should perhaps be two variants to cater for the different variants
of RED. However, as the TFRC-SP authors point out, there is no way
for a transport to know whether some queues on its path have
deployed RED with byte-mode packet drop (except if an exhaustive
survey found that no-one has deployed it!—see <xref
target="pktb_Coding_Status_Summary"></xref>). Incidentally, VP-TFRC
also proposed that byte-mode RED dropping should really square the
packet size compensation factor (like that of RED_5, but apparently
unaware of it).</t>
<t>Pre-congestion notification <xref
target="I-D.ietf-pcn-architecture"></xref> is a proposal to use a
virtual queue for AQM marking for packets within one Diffserv class
in order to give early warning prior to any real queuing. The
proposed PCN marking algorithms have been designed not to take
account of packet size on routers. Instead the general principle has
been to take account of the sizes of marked packets when monitoring
the fraction of marking at the edge of the network.</t>
</section>
<!-- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -->
<section anchor="pktb_Coding_Status_Summary"
title="Congestion Coding: Summary of Status">
<t><?rfc needLines="6" ?><texttable anchor="pktb_Tab_TFRC-SP"
title="Dependence of flow bit-rate per RTT on packet size s and drop rate p when network and/or transport bias towards small packets to varying degrees">
<ttcol align="right">transport cc</ttcol>
<ttcol align="center">RED_1 (packet mode drop)</ttcol>
<ttcol align="center">RED_4 (linear byte mode drop)</ttcol>
<ttcol align="center">RED_5 (square byte mode drop)</ttcol>
<c>TCP or TFRC</c>
<c>s/sqrt(p)</c>
<c>sqrt(s/p)</c>
<c>1/sqrt(p)</c>
<c>TFRC-SP</c>
<c>1/sqrt(p)</c>
<c>1/sqrt(sp)</c>
<c>1/(s.sqrt(p))</c>
</texttable></t>
<t><xref target="pktb_Tab_TFRC-SP"></xref> aims to summarise the
positions we may now be in. Each column shows a different possible
AQM behaviour on different routers in the network, using the
terminology of Cnodder et al outlined earlier (RED_1 is basic RED
with packet-mode drop). Each row shows a different transport
behaviour: TCP <xref target="RFC2581"></xref> and TFRC <xref
target="RFC3448"></xref> on the top row with TFRC-SP <xref
target="RFC4828"></xref> below. Suppressing all inessential details
the table shows that independence from packet size should either be
achievable by not altering the TCP transport in a RED_5 network, or
using the small packet TFRC-SP transport in a network without any
byte-mode dropping RED (top right and bottom left). Top left is the
`do nothing' scenario, while bottom right is the `do-both' scenario
in which bit-rate would become far too biased towards small packets.
Of course, if any form of byte-mode dropping RED has been deployed
on a selection of congested routers, each path will present a
different hybrid scenario to its transport.</t>
<t>Whatever, we can see that the linear byte-mode drop column in the
middle considerably complicates the Internet. It's a half-way house
that doesn't bias enough towards small packets even if one believes
the network should be doing the biasing. We argue below that <spanx
style="emph">all</spanx> network layer bias towards small packets
should be turned off—if indeed any router vendors have
implemented it—leaving packet size bias solely as the preserve
of the transport layer (solely the leftmost, packet-mode drop
column).</t>
<t>A survey has been conducted of 84 vendors to assess how widely
drop probability based on packet size has been implemented in RED.
Prior to the survey, an individual approach to Cisco received
confirmation that, having checked the code-base for each of the
product ranges, Cisco has not implemented any discrimination based
on packet size in any AQM algorithm in any of its products. Also an
individual approach to Alcatel-Lucent drew a confirmation that it
was very likely that none of their products contained RED code that
implemented any packet-size bias.</t>
<t>Turning to our more formal survey, about 19% of those surveyed
have replied so far, giving a sample size of 16. Although we do not
have permission to identify the respondents, we can say that those
that have responded include most of the larger vendors, covering a
large fraction of the market. They range across the large network
equipment vendors at L3 & L2, firewall vendors, wireless
equipment vendors, as well as large software businesses with a small
selection of networking products. So far, all those who have
responded have confirmed that they have not implemented the variant
of RED with drop dependent on packet size (2 are fairly sure they
haven't but need to check more thoroughly).</t>
<!--{Todo: Tabulate survey results}-->
<t>Where reasons have been given, the extra complexity of packet
bias code has been most prevalent, though one vendor had a more
principled reason for avoiding it—similar to the argument of
this document. We have established that Linux does not implement RED
with packet size drop bias, although we have not investigated a
wider range of open source code.</t>
</section>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Issues" title="Outstanding Issues and Next Steps">
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-World" title="Bit-congestible World">
<t>For a connectionless network with only bit-congestible resources we
believe the recommended position is now unarguably clear—that
the network should not make allowance for packet sizes and the
transport should. This leaves two outstanding issues: <list
style="symbols">
<t>How to handle any legacy of AQM with byte-mode drop already
deployed;</t>
<t>The need to start a programme to update transport congestion
control protocol standards to take account of packet size.</t>
</list></t>
<t>The sample of returns from our vendor survey <xref
target="pktb_Coding_Status_Summary"></xref> suggest that byte-mode
packet drop seems not to be implemented at all let alone deployed, or
if it is, it is likely to be very sparse. Therefore, we do not really
need a migration strategy from all but nothing to nothing.</t>
<t>A programme of standards updates to take account of packet size in
transport congestion control protocols has started with TFRC-SP <xref
target="RFC4828"></xref>, while weighted TCPs implemented in the
research community <xref target="WindowPropFair"></xref> could form
the basis of a future change to TCP congestion control <xref
target="RFC2581"></xref> itself.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bit-Pkt-World"
title="Bit- & Packet-congestible World">
<t>Nonetheless, a connectionless network with both bit-congestible and
packet-congestible resources is a different matter. If we believe we
should allow for this possibility in the future, this space contains a
truly open research issue.</t>
<t>The idealised wire protocol coding described in <xref
target="pktb_Ideal_Coding"></xref> requires at least two flags for
congestion of bit-congestible and packet-congestible resources. This
hides a fundamental problem—much more fundamental than whether
we can magically create header space for yet another ECN flag in IPv4,
or whether it would work while being deployed incrementally. A
congestion notification protocol must survive a transition from low
levels of congestion to high. Marking two states is feasible with
explicit marking, but much harder if packets are dropped. Also, it
will not always be cost-effective to implement AQM at every low level
resource, so drop will often have to suffice. Distinguishing drop from
delivery naturally provides just one congestion flag—it is hard
to drop a packet in two ways that are distinguishable remotely. This
is a similar problem to that of distinguishing wireless transmission
losses from congestive losses.</t>
<t>We should also note that, strictly, packet-congestible resources
are actually cycle-congestible because load also depends on the
complexity of each look-up and whether the pattern of arrivals is
amenable to caching or not. Further, this reminds us that any solution
must not require a forwarding engine to use excessive processor cycles
in order to decide how to say it has no spare processor cycles.</t>
<t>The problem of signalling packet processing congestion is not
pressing, as most if not all Internet resources are designed to be
bit-congestible before packet processing starts to congest. However,
given the IRTF ICCRG has set itself the task of reaching consensus on
generic router mechanisms that are necessary and sufficient to support
the Internet's future congestion control requirements <xref
target="I-D.irtf-iccrg-welzl-congestion-control-open-research"></xref>,
we must not give this problem no thought at all, just because it is
hard and currently hypothetical.</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Security_Considerations"
title="Security Considerations">
<t>This draft recommends that queues do not bias drop probability
towards small packets as this creates a perverse incentive for
transports to break down their flows into tiny segments. One of the
benefits of implementing AQM was meant to be to remove this perverse
incentive that drop-tail queues gave to small packets. Of course, if
transports really want to make the greatest gains, they don't have to
respond to congestion anyway. But we don't want applications that are
trying to behave to discover that they can go faster by using smaller
packets.</t>
<t>In practice, transports cannot all be trusted to respond to
congestion. So another reason for recommending that queues do not bias
drop probability towards small packets is to avoid the vulnerability to
small packet DDoS attacks that would otherwise result. One of the
benefits of implementing AQM was meant to be to remove drop-tail's DoS
vulnerability to small packets, so we shouldn't add it back again.</t>
<t>If most queues implemented AQM with byte-mode drop, the resulting
network would amplify the potency of a small packet DDoS attack. At the
first queue the stream of packets would push aside a greater proportion
of large packets, so more of the small packets would survive to attack
the next queue. Thus a flood of small packets would continue on towards
the destination, pushing regular traffic with large packets out of the
way in one queue after the next, but suffering much less drop
itself.</t>
<t><xref target="pktb_Policing_Congestion_Response"></xref> explains why
the ability of networks to police the response of <spanx
style="emph">any</spanx> transport to congestion depends on
bit-congestible network resources only doing packet-mode not byte-mode
drop. In summary, it says that making drop probability depend on the
size of the packets that bits happen to be divided into simply
encourages the bits to be divided into smaller packets. Byte-mode drop
would therefore irreversibly complicate any attempt to fix the
Internet's incentive structures.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Conclusions" title="Conclusions">
<t>The strong conclusion is that AQM algorithms such as RED SHOULD NOT
use byte-mode drop. More generally, the Internet's congestion
notification protocols (drop and ECN) SHOULD take account of packet size
when the notification is read by the transport layer, NOT when it is
written by the network layer. This approach offers sufficient and
correct congestion information for all known and future transport
protocols and also ensures no perverse incentives are created that would
encourage transports to use inappropriately small packet sizes.</t>
<t>The alternative of deflating RED's drop probability for smaller
packet sizes (byte-mode drop) has no enduring advantages. It is more
complex, it creates the perverse incentive to fragment segments into
tiny pieces and it reopens the vulnerability to foods of small-packets
that drop-tail queues suffered from and AQM was designed to remove.
Byte-mode drop is a change to the network layer that makes allowance for
an omission from the design of TCP, effectively reverse engineering the
network layer to contrive to make two TCPs with different packet sizes
run at equal bit rates (rather than packet rates) under the same path
conditions. It also improves TCP performance by reducing the chance that
a SYN or a pure ACK will be dropped, because they are small. But we
SHOULD NOT hack the network layer to improve or fix certain transport
protocols. No matter how predominant a transport protocol is (even if
it's TCP), trying to correct for its failings by biasing towards small
packets in the network layer creates a perverse incentive to break down
all flows from all transports into tiny segments.</t>
<t>So far, our survey of over 100 vendors across the industry has drawn
responses from about 19%, none of whom have implemented the byte mode
packet drop variant of RED. Given there appears to be little, if any,
installed base recommending removal of byte-mode drop from RED is
possibly only a paper exercise with few, if any, incremental deployment
issues.</t>
<t>If a vendor has implemented byte-mode drop, and an operator has
turned it on, it is strongly RECOMMENDED that it SHOULD be turned off.
Note that RED as a whole SHOULD NOT be turned off, as without it, a drop
tail queue also biases against large packets. But note also that turning
off byte-mode may alter the relative performance of applications using
different packet sizes, so it would be advisable to establish the
implications before turning it off.</t>
<t>Instead, the IETF transport area should continue its programme of
updating congestion control protocols to take account of packet size and
to make transports less sensitive to losing control packets like SYNs
and pure ACKS.</t>
<t>NOTE WELL that RED's byte-mode queue measurement is fine, being
completely orthogonal to byte-mode drop. If a RED implementation has a
byte-mode but does not specify what sort of byte-mode, it is most
probably byte-mode queue measurement, which is fine. However, if in
doubt, the vendor should be consulted.</t>
<t>The above conclusions cater for the Internet as it is today with
most, if not all, resources being primarily bit-congestible. A secondary
conclusion of this memo is that we may see more packet-congestible
resources in the future, so research may be needed to extend the
Internet's congestion notification (drop or ECN) so that it can handle a
mix of bit-congestible and packet-congestible resources.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Acknowledgements" title="Acknowledgements">
<t>Thank you to Sally Floyd, who gave extensive and useful review
comments. Also thanks for the reviews from Toby Moncaster and Arnaud
Jacquet. I am grateful to Bruce Davie and his colleagues for providing a
timely and efficient survey of RED implementation in Cisco's product
range. Also grateful thanks to Toby Moncaster, Will Dormann, John
Regnault, Simon Carter and Stefaan De Cnodder further helped survey the
current status of RED implementation and deployment and, finally, thanks
to the anonymous individuals who responded.</t>
</section>
<!-- ================================================================ -->
<section anchor="pktb_Comments_Solicited" title="Comments Solicited">
<t>Comments and questions are encouraged and very welcome. They can be
addressed to the IETF Transport Area working group mailing list
<tsvwg@ietf.org>, and/or to the authors.</t>
</section>
</middle>
<back>
<!-- ================================================================ -->
<section anchor="pktb_Scenarios" title="Example Scenarios">
<!--{ToDo: Tabulate these subsections}-->
<!-- ________________________________________________________________ -->
<section anchor="pktb_Notation" title="Notation">
<t>To prove the two sets of assertions in the idealised wire protocol
(<xref target="pktb_Ideal_Coding" />) are true, we will compare two
flows with different packet sizes, s_1 and s_2 [bit/pkt], to make sure
their transports each see the correct congestion notification.
Initially, within each flow we will take all packets as having equal
sizes, but later we will generalise to flows within which packet sizes
vary. A flow's bit rate, x [bit/s], is related to its packet rate, u
[pkt/s], by <list style="empty">
<t>x(t) = s.u(t).</t>
</list></t>
<t>We will consider a 2x2 matrix of four scenarios:</t>
<?rfc needLines="6" ?>
<texttable anchor="pktb_Tab_Scenarios">
<ttcol align="right">resource type and congestion level</ttcol>
<ttcol align="center">A) Equal bit rates</ttcol>
<ttcol align="center">B) Equal pkt rates</ttcol>
<c>i) bit-congestible, p_b</c>
<c>(Ai)</c>
<c>(Bi)</c>
<c>ii) pkt-congestible, p_p</c>
<c>(Aii)</c>
<c>(Bii)</c>
</texttable>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Ai"
title="Bit-congestible resource, equal bit rates (Ai)">
<t>Starting with the bit-congestible scenario, for two flows to
maintain equal bit rates (Ai) the ratio of the packet rates must be
the inverse of the ratio of packet sizes: u_2/u_1 = s_1/s_2. So, for
instance, a flow of 60B packets would have to send 25x more packets to
achieve the same bit rate as a flow of 1500B packets. If a congested
resource marks proportion p_b of packets irrespective of size, the
ratio of marked packets received by each transport will still be the
same as the ratio of their packet rates, p_b.u_2/p_b.u_1 = s_1/s_2. So
of the 25x more 60B packets sent, 25x more will be marked than in the
1500B packet flow, but 25x more won't be marked too.</t>
<t>In this scenario, the resource is bit-congestible, so it always
uses our idealised bit-congestion field when it marks packets.
Therefore the transport should count marked bytes not packets. But it
doesn't actually matter for ratio-based transports like TCP (<xref
target="pktb_Ideal_Coding" />). The ratio of marked to unmarked bytes
seen by each flow will be p_b, as will the ratio of marked to unmarked
packets. Because they are ratios, the units cancel out.</t>
<t>If a flow sent an inconsistent mixture of packet sizes, we have
said it should count the ratio of marked and unmarked bytes not
packets in order to correctly decode the level of congestion. But
actually, if all it is trying to do is decode p_b, it still doesn't
matter. For instance, imagine the two equal bit rate flows were
actually one flow at twice the bit rate sending a mixture of one 1500B
packet for every thirty 60B packets. 25x more small packets will be
marked and 25x more will be unmarked. The transport can still
calculate p_b whether it uses bytes or packets for the ratio. In
general, for any algorithm which works on a ratio of marks to
non-marks, either bytes or packets can be counted interchangeably,
because the choice cancels out in the ratio calculation.</t>
<t>However, where an absolute target rather than relative volume of
congestion caused is important (<xref target="pktb_Ideal_Coding" />),
as it is for congestion accountability <xref
target="Rate_fair_Dis" />, the transport must count marked bytes not
packets, in this bit-congestible case. Aside from the goal of
congestion accountability, this is how the bit rate of a transport can
be made independent of packet size; by ensuring the rate of congestion
caused is kept to a constant weight <xref target="WindowPropFair" />,
rather than merely responding to the ratio of marked and unmarked
bytes.</t>
<t>Note the unit of byte-congestion volume is the byte.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bi"
title="Bit-congestible resource, equal packet rates (Bi)">
<t>If two flows send different packet sizes but at the same packet
rate, their bit rates will be in the same ratio as their packet sizes,
x_2/x_1 = s_2/s_1. For instance, a flow sending 1500B packets at the
same packet rate as another sending 60B packets will be sending at 25x
greater bit rate. In this case, if a congested resource marks
proportion p_b of packets irrespective of size, the ratio of packets
received with the byte-congestion field marked by each transport will
be the same, p_b.u_2/p_b.u_1 = 1.</t>
<t>Because the byte-congestion field is marked, the transport should
count marked bytes not packets. But because each flow sends
consistently sized packets it still doesn't matter for ratio-based
transports. The ratio of marked to unmarked bytes seen by each flow
will be p_b, as will the ratio of marked to unmarked packets.
Therefore, if the congestion control algorithm is only concerned with
the ratio of marked to unmarked packets (as is TCP), both flows will
be able to decode p_b correctly whether they count packets or
bytes.</t>
<t>But if the absolute volume of congestion is important, e.g. for
congestion accountability, the transport must count marked bytes not
packets. Then the lower bit rate flow using smaller packets will
rightly be perceived as causing less byte-congestion even though its
packet rate is the same.</t>
<t>If the two flows are mixed into one, of bit rate x1+x2, with equal
packet rates of each size packet, the ratio p_b will still be
measurable by counting the ratio of marked to unmarked bytes (or
packets because the ratio cancels out the units). However, if the
absolute volume of congestion is required, the transport must count
the sum of congestion marked bytes, which indeed gives a correct
measure of the rate of byte-congestion p_b(x_1 + x_2) caused by the
combined bit rate.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Aii"
title="Pkt-congestible resource, equal bit rates (Aii)">
<t>Moving to the case of packet-congestible resources, we now take two
flows that send different packet sizes at the same bit rate, but this
time the pkt-congestion field is marked by the resource with
probability p_p. As in scenario Ai with the same bit rates but a
bit-congestible resource, the flow with smaller packets will have a
higher packet rate, so more packets will be both marked and unmarked,
but in the same proportion.</t>
<t>This time, the transport should only count marks without taking
into account packet sizes. Transports will get the same result, p_p,
by decoding the ratio of marked to unmarked packets in either
flow.</t>
<t>If one flow imitates the two flows but merged together, the bit
rate will double with more small packets than large. The ratio of
marked to unmarked packets will still be p_p. But if the absolute
number of pkt-congestion marked packets is counted it will accumulate
at the combined packet rate times the marking probability,
p_p(u_1+u_2), 26x faster than packet congestion accumulates in the
single 1500B packet flow of our example, as required.</t>
<t>But if the transport is interested in the absolute number of packet
congestion, it should just count how many marked packets arrive. For
instance, a flow sending 60B packets will see 25x more marked packets
than one sending 1500B packets at the same bit rate, because it is
sending more packets through a packet-congestible resource.</t>
<t>Note the unit of packet congestion is packets.</t>
</section>
<!-- ________________________________________________________________ -->
<section anchor="pktb_Bii"
title="Pkt-congestible resource, equal packet rates (Bii)">
<t>Finally, if two flows with the same packet rate, pass through a
packet-congestible resource, they will both suffer the same proportion
of marking, p_p, irrespective of their packet sizes. On detecting that
the pkt-congestion field is marked, the transport should count
packets, and it will be able to extract the ratio p_p of marked to
unmarked packets from both flows, irrespective of packet sizes.</t>
<t>Even if the transport is monitoring the absolute amount of packets
congestion over a period, still it will see the same amount of packet
congestion from either flow.</t>
<t>And if the two equal packet rates of different size packets are
mixed together in one flow, the packet rate will double, so the
absolute volume of packet-congestion will accumulate at twice the rate
of either flow, 2p_p.u_1 = p_p(u_1+u_2).</t>
</section>
</section>
<!-- ================================================================ -->
<section anchor="pktb_CN_Definition"
title="Congestion Notification Definition: Further Justification">
<t>In <xref target="pktb_Congestion_Definition" /> on the definition of
congestion notification, load not capacity was used as the denominator.
This also has a subtle significance in the related debate over the
design of new transport protocols—typical new protocol designs
(e.g. in XCP <xref target="I-D.falk-xcp-spec" /> & Quickstart <xref
target="RFC4782" />) expect the sending transport to communicate its
desired flow rate to the network and network elements to progressively
subtract from this so that the achievable flow rate emerges at the
receiving transport.</t>
<t>Congestion notification with total load in the denominator can serve
a similar purpose (though in retrospect not in advance like XCP &
QuickStart). Congestion notification is a dimensionless fraction but
each source can extract necessary rate information from it because it
already knows what its own rate is. Even though congestion notification
doesn't communicate a rate explicitly, from each source's point of view
congestion notification represents the fraction of the rate it was
sending a round trip ago that couldn't (or wouldn't) be served by
available resources. After they were sent, all these fractions of each
source's offered load added up to the aggregate fraction of offered load
seen by the congested resource. So, the source can also know the total
excess rate by multiplying total load by congestion level. Therefore
congestion notification, as one scale-free dimensionless fraction,
implicitly communicates the instantaneous excess flow rate, albeit a RTT
ago.</t>
</section>
<section anchor="pktb_Policing_Congestion_Response"
title="Byte-mode Drop Complicates Policing Congestion Response">
<t>This appendix explains why the ability of networks to police the
response of <spanx style="emph">any</spanx> transport to congestion
depends on bit-congestible network resources only doing packet-mode not
byte-mode drop.</t>
<t>To be able to police a transport's response to congestion when
fairness can only be judged over time and over all an individual's
flows, the policer has to have an integrated view of all the congestion
an individual (not just one flow) has caused due to all traffic entering
the Internet from that individual. This is termed congestion
accountability.</t>
<t>But with byte-mode drop, one dropped or marked packet is not
necessarily equivalent to another unless you know the MTU that caused it
to be dropped/marked. To have an integrated view of a user, we believe
congestion policing has to be located at an individual's attachment
point to the Internet <xref target="Re-TCP" />. But from there it cannot
know the MTU of each remote router that caused each mark. Therefore it
cannot take an integrated approach to policing all the responses to
congestion of all the transports of one individual. Therefore it cannot
police anything.</t>
<t>The security/incentive argument <spanx style="emph">for</spanx>
packet-mode drop is similar. Firstly, confining RED to packet-mode drop
would not preclude bottleneck policing approaches such as <xref
target="pBox" /> as it seems likely they could work just as well by
monitoring the volume of dropped bytes rather than packets. Secondly
packet-mode marking naturally allows the congestion marking on packets
to be globally meaningful without relying on MTU information held
elsewhere.</t>
<t>Because we recommend that a marked packet should be taken to mean
that all the bytes in the packet are congestion 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" />.
Therefore policing would work naturally with just simple packet-mode
drop in RED.</t>
<t>In summary, making drop probability depend on the size of the packets
that bits happen to be divided into simply encourages the bits to be
divided into smaller packets. Byte-mode drop would therefore
irreversibly complicate any attempt to fix the Internet's incentive
structures.</t>
</section>
<note title="Changes from Previous Versions">
<t>To be removed by the RFC Editor on publication.<list style="hanging">
<t hangText="From -00 to -01:">
<list>
<t>Clarified applicability to drop as well as ECN.</t>
<t>Highlighted DoS vulnerability.</t>
<t>Emphasised that drop-tail suffers from similar problems to
byte-mode drop, so only byte-mode drop should be turned off, not
RED itself.</t>
<t>Clarified the original apparent motivations for recommending
byte-mode drop included protecting SYNs and pure ACKs more than
equalising the bit rates of TCPs with different segment sizes.
Removed some conjectured motivations.</t>
<t>Added support for updates to TCP in progress (ackcc &
ecn-syn-ack).</t>
<t>Updated survey results with newly arrived data.</t>
<t>Pulled all recommendations together into the conclusions.</t>
<t>Moved some detailed points into two additional appendices and
a note.</t>
<t>Considerable clarifications throughout.</t>
<t>Updated references</t>
</list>
</t>
</list></t>
</note>
<!-- ================================================================ -->
<references title="Normative References">
<?rfc include="reference.RFC.2119" ?>
<?rfc include="reference.RFC.2309" ?>
<?rfc include="reference.RFC.2581" ?>
<?rfc include="reference.RFC.3168" ?>
<?rfc include="reference.RFC.3426" ?>
<?rfc include="reference.RFC.3448" ?>
<?rfc include='reference.RFC.4828'?>
</references>
<references title="Informative References">
<?rfc include="localref.Floyd93.RED" ?>
<?rfc include="localref.Floyd97.REDPktByteEmail" ?>
<?rfc include="localref.Floyd99.Penalty_box" ?>
<?rfc include="localref.Crowcroft98.MulTCP" ?>
<?rfc include="localref.Gibbens99.Evol_cc" ?>
<?rfc include="localref.Elloumi00.REDbyte" ?>
<?rfc include="localref.Vasallo00.PktSizeEquCC" ?>
<?rfc include="localref.Siris02a.Window_ECN" ?>
<?rfc include="localref.Siris02.RscCtrlCDMA" ?>
<?rfc include="reference.RFC.3714" ?>
<?rfc include="localref.Eddy03.REDbias" ?>
<?rfc include="localref.Widmer04.CCvarPktSize" ?>
<?rfc include="localref.I-D.briscoe-tsvwg-re-ecn-tcp" ?>
<?rfc include="reference.I-D.ietf-pcn-architecture" ?>
<?rfc include="localref.Briscoe07.Rate_fair_Dis" ?>
<?rfc include="reference.I-D.ietf-tcpm-rfc2581bis" ?>
<?rfc include="reference.I-D.falk-xcp-spec" ?>
<?rfc include="reference.RFC.4782" ?>
<?rfc include="localref.IESG.PCN_charter" ?>
<?rfc include='localref.Floyd00.gentle_RED'?>
<?rfc include='reference.I-D.ietf-tcpm-ecnsyn'?>
<?rfc include='reference.I-D.floyd-tcpm-ackcc'?>
<?rfc include='reference.I-D.irtf-iccrg-welzl-congestion-control-open-research'?>
</references>
</back>
</rfc>| PAFTECH AB 2003-2026 | 2026-04-22 15:53:10 |