One document matched: draft-briscoe-tsvwg-byte-pkt-mark-00.txt
Transport Area Working Group B. Briscoe
Internet-Draft BT & UCL
Intended status: Informational June 17, 2007
Expires: December 19, 2007
Byte and Packet Congestion Notification
draft-briscoe-tsvwg-byte-pkt-mark-00
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Copyright Notice
Copyright (C) The IETF Trust (2007).
Abstract
This memo was written to clarify how (and whether) to take packet
size into account when notifying congestion using active queue
management (AQM) such as random early detection (RED). The scope
includes resource congestion by bytes and by packet processing, even
though the latter is less common. It answers the question of whether
packet size should be taken into account when network equipment
writes congestion notification, or when transports read it. The
primary conclusion is that RED's byte-mode packet drop should not be
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used because it creates a perverse incentive for transports to use
tiny segments. TCP's lack of attention to packet size should be
fixed in TCP, not by reverse engineering network forwarding to fix
transport protocols.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Requirements notation . . . . . . . . . . . . . . . . . . . . 6
3. Working Definition of Congestion Notification . . . . . . . . 6
4. Congestion Measurement . . . . . . . . . . . . . . . . . . . . 7
5. Idealised Wire Protocol Coding . . . . . . . . . . . . . . . . 8
6. The State of the Art . . . . . . . . . . . . . . . . . . . . . 10
6.1. Congestion Measurement: Status . . . . . . . . . . . . . . 10
6.2. Congestion Coding: Status . . . . . . . . . . . . . . . . 11
6.2.1. Network Bias when Encoding . . . . . . . . . . . . . . 11
6.2.2. Transport Bias when Decoding . . . . . . . . . . . . . 12
6.2.3. Congestion Coding: Summary of Status . . . . . . . . . 14
7. Outstanding Issues and Next Steps . . . . . . . . . . . . . . 15
7.1. Bit-congestible World . . . . . . . . . . . . . . . . . . 15
7.2. Bit- & Packet-congestible World . . . . . . . . . . . . . 16
8. Security Considerations . . . . . . . . . . . . . . . . . . . 17
9. Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . 19
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
11. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 20
Appendix A. Example Scenarios . . . . . . . . . . . . . . . . . . 20
A.1. Notation . . . . . . . . . . . . . . . . . . . . . . . . . 20
A.2. Bit-congestible resource, equal bit rates (Ai) . . . . . . 21
A.3. Bit-congestible resource, equal packet rates (Bi) . . . . 22
A.4. Pkt-congestible resource, equal bit rates (Aii) . . . . . 22
A.5. Pkt-congestible resource, equal packet rates (Bii) . . . . 23
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23
12.1. Normative References . . . . . . . . . . . . . . . . . . . 23
12.2. Informative References . . . . . . . . . . . . . . . . . . 24
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 26
Intellectual Property and Copyright Statements . . . . . . . . . . 27
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1. Introduction
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. 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.
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.
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. 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.
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.
The byte vs. packet dilemma arises at three stages in the congestion
notification process:
Measuring congestion When the congested resource decides locally how
to measure how congested it is (should the queue be measured in
bytes or packets?);
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?);
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?).
In RED, whether to use packets or bytes when measuring queues is
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called packet-mode or byte-mode queue measurement. This choice is
now fairly well understood but is included in Section 4 to document
it in the RFC series.
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, this choice is termed
packet-mode or byte-mode drop as opposed to queue measurement, which
is an orthogonal choice. Note that this issue concerns how much each
congestion notification on a packet should be taken to mean,
irrespective of whether it is signalled implicitly by drop or
explicitly using ECN [RFC3168].
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 [Evol_cc][RFC3426]. 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. Encouraging a high volume of tiny packets
might in turn unnecessarily overload a completely unrelated part of
the system.
Currently, the paper trail of advice referenced from the RFC series
(sort of) recommends exactly such packet-size dependent drop,
although we believe implementers may have ignored the advice. The
primary purpose of this memo is to explain why that advice should be
reversed and eventually to record a definitive consensus within the
RFC series.
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. For bit-congestible resources, it is currently
recommended that RED should be 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 drop 1% of the smaller packets. The bit rate passed
to the line by the RED queue will therefore be 750k for the flow of
larger packets but 990k for flow of smaller packets, even though they
both arrived with the same bit rate.
The reason it was recommended that RED should work like this is that
TCP has always been the predominant transport used in the Internet,
and TCP congestion control ensures that flows competing for the same
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resource each maintain the same number of segments in flight,
irrespective of segment size. Rather than discuss the possibility of
fixing the problem in TCP, it was recommended that routers should be
altered to reverse engineer the network layer around TCP, contrary to
the excellent advice in [RFC3426], which asks designers to question
"Why are you proposing a solution at this layer of the protocol
stack, rather than at another layer?" The implicit plan seems to
have been to use gradual RED deployment in the network as a way to
make the fairness that the TCP algorithm achieves gradually change
from equalising segment-rate to equalising bit-rate between flows.
This seems to be how we ended up recommending RED should use byte-
mode packet drop to discard equal numbers of packets, not bits, from
equal bit-rate flows.
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:
1. 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
[PCNcharter];
2. [RFC2309] says RED may either take account of packet size or not
when dropping, but gives no recommendation between the two,
referring instead to advice on the performance implications in an
email [pktByteEmail], which recommends byte-mode drop, but
without really discussing performance. 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;
3. Currently, no active queue management behaviour like RED has been
standardised, so implementers have no other standards guidance
than [RFC2309], which is informational;
4. 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;
5. The Internet community needs to discuss widely whether the
complexity of adjusting for packet size should be on routers or
in transports;
6. Given there are many good reasons why larger path max
transmission units (PMTUs) would help solve a number of scaling
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issues, we don't want to create any bias against large packets
that is greater than their true cost;
7. And finally, given it has recently been shown that TCP doesn't
achieve any meaningful fairness anyway
[I-D.briscoe-tsvarea-fair], because it doesn't consider fairness
over all the flows a user transmits nor over time, modifying the
network so as not to have to modify TCP still won't achieve
fairness. It seems more likely we have to face up to changing
TCP anyway.
This memo starts from first principles, defining congestion
notification in Section 3 then determining the correct way to measure
congestion (Section 4) and to design an idealised congestion
notification protocol (Section 5). It then surveys the advice given
previously in the RFC series, the research literature and the
deployed legacy (Section 6) before summarising the recommended way
forward and listing outstanding issues (Section 7) that will need
resolution both to achieve the ideal protocol and to handle legacy.
2. Requirements notation
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 [RFC2119].
3. Working Definition of Congestion Notification
Rather than aim to achieve what many have tried and failed, this memo
will not try to define congestion. It will give a working definition
of what congestion notification should be taken to mean for this
document. Congestion notification is a changing signal that aims to
communicate the ratio E/L, 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.
The phrase `would not serve' is added, because AQM systems (e.g.
RED, PCN [PCN]) use a virtual capacity smaller than actual capacity,
then notify congestion of this virtual capacity in order to avoid
congestion of the actual capacity.
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
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not being served (or being marked as at risk of not being served).
Incidentally, load being the denominator also has a subtle
significance in the related debate over whether desired flow rates
should be communicated between transport and network and whether
achievable flow rates should then be communicated back again (e.g. in
XCP [I-D.falk-xcp-spec] & Quickstart [RFC4782]). 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. Therefore the instantaneous excess flow rate an RTT ago is
implicitly communicated within this one scale-free dimensionless
fraction (and a lot more).
4. Congestion Measurement
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
Section 6.
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.
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.
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
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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.
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.
[ECNFixedWireless] 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.
5. Idealised Wire Protocol Coding
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 Appendix A.
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.
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.
However, although this idealised mechanism isn't intended for
implementation, we do want to emphasise that we must find a way to
implement it, because it could become necessary to somehow
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distinguish between bit and packet congestion [RFC3714]. Currently a
design goal of network processing equipment such as routers and
firewalls is to keep packet processing uncongested even under worst
case bit rates with minimum packet sizes. Therefore, packet-
congestion is currently rare, but there is no guarantee that it will
not become common with future technology trends.
The idealised wire protocol is given below. It allows for packet
size 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 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:
o A packet-congestible resource trying to code congestion level p_p
into a packet stream should mark the `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.
o 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.
The worked examples in Appendix A 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.
Sufficient and correct congestion information means that there is
sufficient information for the two different types of transport
requirements:
o Established transport congestion controls like TCP's [RFC2581] aim
to achieve equal segment rates per RTT through the same
bottleneck--TCP `fairness' [RFC3448]. 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.
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(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 [I-D.ietf-tcpm-rfc2581bis], mainly for orthogonal security
reasons.)
o Other congestion controls proposed in the research community aim
to limit the volume of congestion caused to a constant weight
parameter. [MulTCP][WindowPropFair] are examples of weighted
proportionally fair transports designed for cost-fair environments
[I-D.briscoe-tsvarea-fair]. In this case, the transport requires
a count (not a ratio) of marked bytes in the bit-congestible case
and of marked packets in the packet congestible case.
6. The State of the Art
The original 1993 paper on RED [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 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.
When RED was recommended for general deployment in 1998 [RFC2309],
the two modes were mentioned implying the choice between them was a
question of performance, referring to a 1997 email [pktByteEmail] for
advice on tuning. This email clarified that there were in fact two
orthogonal choices: whether to measure queue length in bytes or
packets (Section 6.1) and whether the drop probability of an
individual packet should depend on its own size (Section 6.2).
6.1. Congestion Measurement: Status
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
Section 4).
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
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algorithm on such a buffer will be oversensitive to high proportions
of small packets, 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.
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 it measures itself.
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
Section 4).
6.2. Congestion Coding: Status
6.2.1. Network Bias when Encoding
The previously mentioned email [pktByteEmail] referred to by
[RFC2309] said that the choice over whether a packet's own size
should affect its drop probability "depends on the dominant end-to-
end congestion control mechanisms". [This assumes the network should
be changed to accommodate the predominant transport, without
questioning whether the transport should be fixed instead.] The line
of reasoning went on to say that congestion control in protocols such
as TCP doesn't depend on the fraction of bytes or packets that are
dropped from a flow, but merely on whether or not one or more drops
were present in the most recent window [this is incorrect]. It
argued 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 [pBox]. [The problem could have
been solved by making the policing mechanism count the volume of
bytes randomly dropped, not the number of packets.]
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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. [One can only
assume that the reasoning for byte-mode drop in this case was still
to try to reverse engineer the network to allow for TCP not
accounting for packet size.]
In 2000, Cnodder et al [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 correct,... but only if one
agrees with the original principle behind RED's byte mode drop--that
we should reverse engineer the network in order to arrange for TCP
flows with different packet sizes to achieve equal rates through the
same bottleneck.
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 [REDbias]
that assessed how sensitive RED was to this parameter, amongst other
things. No-one seems to have pointed out that this changed algorithm
can often lead to drop probabilities of greater than 1 [which should
ring alarm bells hinting that there's a mistake in the theory
somewhere].
6.2.2. Transport Bias when Decoding
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!).
However, 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-
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friendly rate control [RFC3448] called TFRC-SP [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 [PktSizeEquCC]. There are also
other important differences of detail relative to TFRC, such as using
virtual packets [CCvarPktSize] to avoid responding to multiple losses
per round trip and using a minimum inter-packet interval.
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.
So the TFRC-SP spec doesn't address the issue of which of the network
or the transport _should_ 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.
The paper originally proposing TFRC with virtual packets (VP-TFRC)
[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 Section 6.2.3). 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).
Pre-congestion notification [PCN] 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.
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6.2.3. Congestion Coding: Summary of Status
+-----------+----------------+-----------------+--------------------+
| transport | RED_1 (packet | RED_4 (linear | RED_5 (square byte |
| cc | mode drop) | byte mode drop) | mode drop) |
+-----------+----------------+-----------------+--------------------+
| TCP or | s/sqrt(p) | sqrt(s/p) | 1/sqrt(p) |
| TFRC | | | |
| TFRC-SP | 1/sqrt(p) | 1/sqrt(sp) | 1/(s.sqrt(p)) |
+-----------+----------------+-----------------+--------------------+
Table 1: 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
Table 1 aims to summarise the positions we may now be in. Each
column shows a different possible AQM behaviour 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 [RFC2581] and TFRC [RFC3448] on the top row with
TFRC-SP [RFC4828] below. Suppressing all inessential details the
table shows that independence from packet size should either be
achievable by not altering the TCP transport in a RED_5 network, or
using the small packet TFRC-SP transport in a network without any
byte-mode dropping RED (top right and bottom left). Top left is the
`do nothing' scenario, while bottom right is the `do-both' scenario
in which bit-rate would become far too biased towards small packets.
Of course, if any form of byte-mode dropping RED has been deployed on
some congested routers, each path will present a different hybrid
scenario to its transport.
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 _all_
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).
A survey is being conducted of over a hundred 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 codebase 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
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implemented any packet-size bias.
Turning to our more formal survey, about 10% of those surveyed have
replied so far, giving a sample size of only about a dozen. 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
have confirmed that they have not implemented the variant of RED with
drop dependent on packet size. 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,
but not the same as 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.
It is RECOMMENDED that adjusting drop probability relative to packet
size (byte-mode dropping) SHOULD NOT be used in router AQM algorithms
and SHOULD be turned off wherever it has been deployed. Note that
RED as a whole SHOULD NOT be turned off, as without it, a drop tail
queue also biases against large packets. Also note 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.
Instead we argue that only transports, not AQM in the network, SHOULD
make allowance for the size of dropped or marked packets. If a
transport protocol doesn't take account of packet size when
controlling the rate of a flow, it SHOULD be corrected in that
transport protocol. No matter how predominant a transport protocol
is (even if it's TCP), trying to correct for its failings in the
network layer creates a perverse incentive to break down all flows
from all transports into tiny segments.
7. Outstanding Issues and Next Steps
7.1. Bit-congestible World
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:
o How to handle any legacy of AQM with byte-mode drop already
deployed;
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o The need to start a programme to update transport congestion
control protocol standards to take account of packet size.
The sample of returns from our vendor survey Section 6.2.3 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 nearly
nothing to nothing.
A programme of standards updates to take account of packet size in
transport congestion control protocols has started with TFRC-SP
[RFC4828], while weighted TCPs implemented in the research community
[MulTCP][WindowPropFair] could form the basis of a future change to
TCP congestion control [RFC2581] itself.
7.2. Bit- & Packet-congestible World
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.
The idealised wire protocol coding described in Section 5 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 the same problem we have
distinguishing wireless transmission losses from congestive losses.
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.
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,
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given the task is to reach consensus on generic router mechanisms
that are necessary and sufficient to support the Internet's future
congestion control requirements, we must not give this problem no
thought at all, just because it is hard and currently hypothetical.
8. Security Considerations
This draft recommends that routers do not bias drop probability
towards small packets as this creates a perverse incentive for
transports to break down their flows into tiny segments. Of course,
this still involves transports being trusted to adjust their rate to
take account of the size of dropped or marked packets. But, in the
current Internet architecture, transports are already trusted to act
against their own interests by reducing their rate in response to
congestion. Therefore at least this recommendation makes the problem
no worse.
Much more importantly though, the ability of networks to police the
response of _any_ transport to congestion depends on networks only
doing packet-mode not byte-mode drop, as we will now try to explain.
Byte-mode drop was originally proposed alongside a RED-based approach
to policing unusually high rate TCP flows [pBox] that has spawned
other similar approaches in the research community. The idea was to
place this policing function at any potential bottleneck. It was
crafted specifically around policing the bit-rate (not packet rate)
of TCP or TCP-friendly flows, by using its knowledge of its own local
MTU. If these bottleneck TCP policers were effective against
cheating (which [Re-TCP] has shown they are not), they would end up
embedding a TCP-fairness policy throughout the network layer.
[I-D.briscoe-tsvarea-fair] has recently shown that TCP fairness is an
insufficient basis for judging fairness because (amongst other
criticisms) it is instantaneous, myopically not taking account of
which individuals have congested resources more over time. If
fairness did take account of factors like duration, instantaneous
flow rates would necessarily have to be very _unequal_ to be fair.
So if TCP-fairness were to be embedded throughout the network layer,
it would prevent these highly unequal rate allocations that would be
essential for improving fairness.
So far, the argument goes that we will need transports that are not
TCP-`fair' in order to be more truly fair. So far this is only an
argument against bottleneck TCP-policers, not against byte-mode
packet drop.
The argument continues that, to be able to police a transport's
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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) is
causing due to all traffic entering the Internet from that
individual.
But with byte-mode drop, one marked packet is not necessarily
equivalent to another unless you know the MTU that caused it to be
marked. If congestion policing has to be located at an individual's
attachment point to the Internet, 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 any
of the flows.
That has been quite a specialised although strong argument against
byte-mode drop. The security/incentive argument _for_ packet-mode
drop is similar.
Firstly, confining RED to packet-mode drop would not preclude
bottleneck policing approaches such as [pBox] as it seems likely they
could work just as well by monitoring the volume of dropped bytes
rather than packets.
Secondly packet-mode drop naturally allows the congestion marking on
packets to be globally meaningful without relying on information held
elsewhere. Given this congestion marking has an economic
interpretation, it can be used as part of a globally distributed
incentive system to ensure the parties responsible for congestion can
be made accountable for it.
Such a system has recently been proposed based on a protocol called
re-ECN [Re-TCP]. Re-ECN was designed to be robust to the self-
interest of the different parties providing and using the Internet,
based on this economic interpretation of congestion. Re-ECN policers
are specifically designed to allow evolution of new congestion
control protocols operating across multiple domains by confining
policing to the extreme edges of the Internet.
Because a marked packet is taken to mean all the bytes in the packet
are congestion marked the re-ECN system remains robust against bits
being re-divided into different size packets or across different size
flows [I-D.briscoe-tsvarea-fair]. Therefore it works naturally with
just simple packet-mode drop in RED.
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
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irreversibly complicate any attempt to fix the Internet's incentive
structures.
9. Conclusions
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.
The alternative of deflating RED's drop probability for smaller
packet sizes (byte-mode drop) has no enduring advantages. It is more
complex and creates the perverse incentive to fragment segments into
tiny pieces. It was proposed as a way for the network layer to make
allowance for an omission from the design of TCP, effectively reverse
engineering the network layer to contrive to make TCPs with different
packet sizes run at equal bit rates (rather than packet rates) under
the same path conditions. We SHOULD NOT hack the network layer to
fix a problem with certain transport protocols, even one as prevalent
as TCP.
So far, our survey of over 100 vendors across the industry has drawn
responses from about 10%, none of whom have implemented the byte mode
packet drop variant of RED.
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. 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.
Instead, the IETF transport area should continue its programme of
updating congestion control protocols to take account of packet size.
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.
The above conclusions cater for the Internet as it is today with
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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.
10. Acknowledgements
Sally Floyd and Arnaud Jacquet gave very useful review comments.
Bruce Davie and his colleagues provided a timely and efficient survey
of RED implementation in Cisco's product range. Toby Moncaster, Will
Dormann, John Regnault, Simon Carter and Stefaan De Cnodder further
helped survey the current status of RED implementation and
deployment.
11. Comments Solicited
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.
Appendix A. Example Scenarios
A.1. Notation
To prove the two sets of assertions in the idealised wire protocol
(Section 5) 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
x(t) = s.u(t).
We will consider a 2x2 matrix of four scenarios:
+-----------------------------+------------------+------------------+
| resource type and | A) Equal bit | B) Equal pkt |
| congestion level | rates | rates |
+-----------------------------+------------------+------------------+
| i) bit-congestible, p_b | (Ai) | (Bi) |
| ii) pkt-congestible, p_p | (Aii) | (Bii) |
+-----------------------------+------------------+------------------+
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Table 2
A.2. Bit-congestible resource, equal bit rates (Ai)
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.
In this scenario, the resource is bit-congestible, so it always uses
the bit-congestion field when it marks packets. Therefore the
transport should count marked bytes not packets. But it doesn't
actually matter. 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 (as used by TCP), the units cancel out.
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.
However, where the absolute rather than relative volume of congestion
caused is important, as it is for cost-fairness
[I-D.briscoe-tsvarea-fair], the transport must count marked bytes not
packets, in this bit-congestible case. Aside from the goal of cost-
fairness, 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 [WindowPropFair], rather than merely
responding to the ratio of marked and unmarked bytes.
Note the unit of byte-congestion volume is the byte.
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A.3. Bit-congestible resource, equal packet rates (Bi)
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.
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. 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.
But if the absolute volume of congestion is important, as it is to
achieve cost-fairness, 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.
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.
A.4. Pkt-congestible resource, equal bit rates (Aii)
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.
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.
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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 volume 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.
But if the transport is interested in the absolute volume 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.
Note the unit of packet congestion is packets.
A.5. Pkt-congestible resource, equal packet rates (Bii)
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.
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.
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).
12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
S., Wroclawski, J., and L. Zhang, "Recommendations on
Queue Management and Congestion Avoidance in the
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Internet", RFC 2309, April 1998.
[RFC2581] Allman, M., Paxson, V., and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, September 2001.
[RFC3426] Floyd, S., "General Architectural and Policy
Considerations", RFC 3426, November 2002.
[RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 3448, January 2003.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828,
April 2007.
12.2. Informative References
[CCvarPktSize]
Widmer, J., Boutremans, C., and J-Y. Le Boudec,
"Congestion Control for Flows with Variable Packet Size",
ACM CCR 34(2) 137--151, 2004,
<http://doi.acm.org/10.1145/997150.997162>.
[ECNFixedWireless]
Siris, V., "Resource Control for Elastic Traffic in CDMA
Networks", Proc. ACM MOBICOM'02 , September 2002, <http://
www.ics.forth.gr/netlab/publications/
resource_control_elastic_cdma.html>.
[Evol_cc] Gibbens, R. and F. Kelly, "Resource pricing and the
evolution of congestion control", Automatica 35(12)1969--
1985, December 1999,
<http://www.statslab.cam.ac.uk/~frank/evol.html>.
[I-D.briscoe-tsvarea-fair]
Briscoe, B., "Flow Rate Fairness: Dismantling a Religion",
draft-briscoe-tsvarea-fair-01 (work in progress),
March 2007.
[I-D.falk-xcp-spec]
Falk, A., "Specification for the Explicit Control Protocol
(XCP)", draft-falk-xcp-spec-02 (work in progress),
November 2006.
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[I-D.ietf-tcpm-rfc2581bis]
Allman, M., "TCP Congestion Control",
draft-ietf-tcpm-rfc2581bis-02 (work in progress),
February 2007.
[MulTCP] Crowcroft, J. and Ph. Oechslin, "Differentiated End to End
Internet Services using a Weighted Proportional Fair
Sharing TCP", CCR 28(3) 53--69, July 1998, <http://
www.cs.ucl.ac.uk/staff/J.Crowcroft/hipparch/pricing.html>.
[PCN] Briscoe, B., Eardley, P., Songhurst, D., Le Faucheur, F.,
Charny, A., Liatsos, V., Babiarz, J., Chan, K., Dudley,
S., Westberg, L., Bader, A., and G. Karagiannis, "Pre-
Congestion Notification Marking",
draft-briscoe-tsvwg-cl-phb-03 (work in progress),
October 2006.
[PCNcharter]
IETF, "Congestion and Pre-Congestion Notification (pcn)",
IETF w-g charter , Feb 2007,
<http://www.ietf.org/html.charters/pcn-charter.html>.
[PktSizeEquCC]
Vasallo, P., "Variable Packet Size Equation-Based
Congestion Control", ICSI Technical Report tr-00-008,
2000, <http://http.icsi.berkeley.edu/ftp/global/pub/
techreports/2000/tr-00-008.pdf>.
[RED93] Floyd, S. and V. Jacobson, "Random Early Detection (RED)
gateways for Congestion Avoidance", IEEE/ACM Transactions
on Networking 1(4) 397--413, August 1993,
<http://www.icir.org/floyd/papers/red/red.html>.
[REDbias] Eddy, W. and M. Allman, "A Comparison of RED's Byte and
Packet Modes", Computer Networks 42(3) 261--280,
June 2003,
<http://www.ir.bbn.com/documents/articles/redbias.ps>.
[REDbyte] De Cnodder, S., Elloumi, O., and K. Pauwels, "RED behavior
with different packet sizes", Proc. 5th IEEE Symposium on
Computers and Communications (ISCC) 793--799, July 2000,
<http://www.icir.org/floyd/red/Elloumi99.pdf>.
[RFC3714] Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
Control for Voice Traffic in the Internet", RFC 3714,
March 2004.
[RFC4782] Floyd, S., Allman, M., Jain, A., and P. Sarolahti, "Quick-
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Start for TCP and IP", RFC 4782, January 2007.
[Re-TCP] Briscoe, B., Jacquet, A., Salvatori, A., and M. Koyabi,
"Re-ECN: Adding Accountability for Causing Congestion to
TCP/IP", draft-briscoe-tsvwg-re-ecn-tcp-03 (work in
progress), October 2006.
[WindowPropFair]
Siris, V., "Service Differentiation and Performance of
Weighted Window-Based Congestion Control and Packet
Marking Algorithms in ECN Networks", Computer
Communications 26(4) 314--326, 2002, <http://
www.ics.forth.gr/netgroup/publications/
weighted_window_control.html>.
[pBox] Floyd, S. and K. Fall, "Promoting the Use of End-to-End
Congestion Control in the Internet", IEEE/ACM Transactions
on Networking 7(4) 458--472, August 1999,
<http://www.aciri.org/floyd/end2end-paper.html>.
[pktByteEmail]
Floyd, S., "RED: Discussions of Byte and Packet Modes",
email , March 1997,
<http://www-nrg.ee.lbl.gov/floyd/REDaveraging.txt>.
Author's Address
Bob Briscoe
BT & UCL
B54/77, Adastral Park
Martlesham Heath
Ipswich IP5 3RE
UK
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Email: bob.briscoe@bt.com
URI: http://www.cs.ucl.ac.uk/staff/B.Briscoe/
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Internet-Draft Byte and Packet Congestion Notification June 2007
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