Transport Area Working Group B. Briscoe
Internet-Draft BT & UCL
Intended status: Informational November 19, 2007
Expires: May 22, 2008
Byte and Packet Congestion Notification
draft-briscoe-tsvwg-byte-pkt-mark-01
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Copyright Notice
Copyright (C) The IETF Trust (2007).
Abstract
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
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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.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Requirements notation . . . . . . . . . . . . . . . . . . . . 6
3. Working Definition of Congestion Notification . . . . . . . . 7
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 . . . . . . . . . . . . . 13
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 . . . . . . . . . . . . . . . . . . . . . . . . . 17
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19
11. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 19
Editorial Comments . . . . . . . . . . . . . . . . . . . . . . . .
Appendix A. Example Scenarios . . . . . . . . . . . . . . . . . . 19
A.1. Notation . . . . . . . . . . . . . . . . . . . . . . . . . 19
A.2. Bit-congestible resource, equal bit rates (Ai) . . . . . . 20
A.3. Bit-congestible resource, equal packet rates (Bi) . . . . 21
A.4. Pkt-congestible resource, equal bit rates (Aii) . . . . . 22
A.5. Pkt-congestible resource, equal packet rates (Bii) . . . . 22
Appendix B. Congestion Notification Definition: Further
Justification . . . . . . . . . . . . . . . . . . . . 23
Appendix C. Byte-mode Drop Complicates Policing Congestion
Response . . . . . . . . . . . . . . . . . . . . . . 23
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 25
12.1. Normative References . . . . . . . . . . . . . . . . . . . 25
12.2. Informative References . . . . . . . . . . . . . . . . . . 26
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 28
Intellectual Property and Copyright Statements . . . . . . . . . . 29
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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. 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.
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 [RFC3168]).
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.
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 (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.
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?);
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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
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, 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.
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.
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. 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.
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
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from RFC2309 is followed, gentle RED [gentle_RED] 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 [RFC4828](S.B.4)[Note_Variation]). 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
Section 8).
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 [RFC3426], which asks
designers to question "Why are you proposing a solution at this layer
of the protocol stack, rather than at another layer?"
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. Further,
just before RFC2309 was issued, an addendum was added to the
archived email that revisited the issue of packet vs. byte-mode
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drop in its last para, making the recommendation less clear-cut;
3. 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 [RFC2309] (including an addendum at the end of the email
to correct the original).
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
[I-D.irtf-iccrg-welzl-congestion-control-open-research]. The
wider Internet community needs to discuss whether the complexity
of adjusting for packet size should be on routers or in
transports;
5. 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;
6. And finally, given it has recently been pointed out that TCP
doesn't achieve any meaningful fairness anyway [Rate_fair_Dis],
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.
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 listing outstanding issues
(Section 7) that will need resolution both to achieve the ideal
protocol and to handle legacy. After discussing security
considerations (Section 8) strong recommendations for the way forward
are given in the conclusions (Section 9).
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].
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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 [I-D.ietf-pcn-architecture]) 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
not being served (or being marked as at risk of not being served).
Appendix B describes a further incidental benefit that arises from
using load as the denominator of congestion notification.
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
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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
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
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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 may need to find a
way to implement it, because it could become necessary to somehow
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 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:
o 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.
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
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requirements:
Ratio-based: Established transport congestion controls like TCP's
[RFC2581] aim to achieve equal segment rates per RTT through the
same bottleneck--TCP friendliness [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.
(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.)
Absolute-target-based: 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 [Rate_fair_Dis]. 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.
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 below) and whether the drop probability of an
individual packet should depend on its own size (Section 6.2 below).
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-
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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
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.
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.
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". [Section 1 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
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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.]
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.
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 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.
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]. On 10-Nov-2004, this variant of byte-mode packet drop
was made the default in the ns2 simulator.
More recently, two drafts have proposed changes to TCP that make it
more robust against losing small control packets
[I-D.ietf-tcpm-ecnsyn] [I-D.floyd-tcpm-ackcc]. 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.
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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!).
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 [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 [I-D.ietf-pcn-architecture] is a proposal
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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.
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 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 [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 a selection of 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 has been conducted of 84 vendors to assess how widely drop
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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.
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).
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.
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;
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 all but
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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
[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 a similar problem to that of
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,
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
[I-D.irtf-iccrg-welzl-congestion-control-open-research], we must not
give this problem no thought at all, just because it is hard and
currently hypothetical.
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8. Security Considerations
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.
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.
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.
Appendix C explains why the ability of networks to police the
response of _any_ 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.
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
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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, 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.
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.
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.
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.
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
most, if not all, resources being primarily bit-congestible. A
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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
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.
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.
Editorial Comments
[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.
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
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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) |
+-----------------------------+------------------+------------------+
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
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 (Section 5). 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.
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
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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 an absolute target rather than relative volume of
congestion caused is important (Section 5), as it is for congestion
accountability [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 [WindowPropFair],
rather than merely responding to the ratio of marked and unmarked
bytes.
Note the unit of byte-congestion volume is the byte.
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 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.
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.
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
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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.
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.
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.
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.
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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).
Appendix B. Congestion Notification Definition: Further Justification
In Section 3 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
[I-D.falk-xcp-spec] & Quickstart [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.
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.
Appendix C. Byte-mode Drop Complicates Policing Congestion Response
This appendix explains why the ability of networks to police the
response of _any_ transport to congestion depends on bit-congestible
network resources only doing packet-mode not byte-mode drop.
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.
But with byte-mode drop, one dropped or marked packet is not
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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 [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.
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 marking naturally allows
the congestion marking on packets to be globally meaningful without
relying on MTU information held elsewhere.
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 [Rate_fair_Dis]. Therefore
policing would work 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
irreversibly complicate any attempt to fix the Internet's incentive
structures.
Changes from Previous Versions
To be removed by the RFC Editor on publication.
From -00 to -01:
Clarified applicability to drop as well as ECN.
Highlighted DoS vulnerability.
Emphasised that drop-tail suffers from similar problems to
byte-mode drop, so only byte-mode drop should be turned off,
not RED itself.
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.
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Added support for updates to TCP in progress (ackcc & ecn-syn-
ack).
Updated survey results with newly arrived data.
Pulled all recommendations together into the conclusions.
Moved some detailed points into two additional appendices and a
note.
Considerable clarifications throughout.
Updated references
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
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.
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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.falk-xcp-spec]
Falk, A., "Specification for the Explicit Control Protocol
(XCP)", draft-falk-xcp-spec-03 (work in progress),
July 2007.
[I-D.floyd-tcpm-ackcc]
Floyd, S. and I. Property, "Adding Acknowledgement
Congestion Control to TCP", draft-floyd-tcpm-ackcc-02
(work in progress), November 2007.
[I-D.ietf-pcn-architecture]
Eardley, P., "Pre-Congestion Notification Architecture",
draft-ietf-pcn-architecture-01 (work in progress),
October 2007.
[I-D.ietf-tcpm-ecnsyn]
Floyd, S. and I. Property, "Adding Explicit Congestion
Notification (ECN) Capability to TCP's SYN/ACK Packets",
draft-ietf-tcpm-ecnsyn-03 (work in progress),
November 2007.
[I-D.ietf-tcpm-rfc2581bis]
Allman, M., "TCP Congestion Control",
draft-ietf-tcpm-rfc2581bis-03 (work in progress),
September 2007.
[I-D.irtf-iccrg-welzl-congestion-control-open-research]
Papadimitriou, D., "Open Research Issues in Internet
Congestion Control",
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Internet-Draft Byte and Packet Congestion Notification November 2007
draft-irtf-iccrg-welzl-congestion-control-open-research-00
(work in progress), July 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>.
[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-
Start for TCP and IP", RFC 4782, January 2007.
[Rate_fair_Dis]
Briscoe, B., "Flow Rate Fairness: Dismantling a Religion",
ACM CCR 37(2)63--74, April 2007,
<http://portal.acm.org/citation.cfm?id=1232926>.
[Re-TCP] Briscoe, B., Jacquet, A., Salvatori, A., Koyabi, M., and
T. Moncaster, "Re-ECN: Adding Accountability for Causing
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Congestion to TCP/IP", draft-briscoe-tsvwg-re-ecn-tcp-04
(work in progress), July 2007.
[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>.
[gentle_RED]
Floyd, S., "Recommendation on using the "gentle_" variant
of RED", Web page , March 2000,
<http://www.icir.org/floyd/red/gentle.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
Phone: +44 1473 645196
Email: bob.briscoe@bt.com
URI: http://www.cs.ucl.ac.uk/staff/B.Briscoe/
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