Transport Area Working Group B. Briscoe
Internet-Draft BT & UCL
Intended status: Informational February 24, 2008
Expires: August 27, 2008
Byte and Packet Congestion Notification
draft-briscoe-tsvwg-byte-pkt-mark-02
Status of this Memo
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Copyright Notice
Copyright (C) The IETF Trust (2008).
Abstract
This memo concerns dropping or marking packets using active queue
management (AQM) such as random early detection (RED) or pre-
congestion notification (PCN). The primary conclusion is that packet
size should be taken into account when transports decode congestion
indications, not when network equipment writes them. Reducing drop
of small packets has some tempting advantages: i) it drops less
control packets, which tend to be small and ii) it makes TCP's bit-
rate less dependent on packet size. However, there are ways of
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addressing these issues at the transport layer, rather than reverse
engineering network forwarding to fix specific transport problems.
Network layer algorithms like the byte-mode packet drop variant of
RED should not be used to drop fewer small packets, because that
creates a perverse incentive for transports to use tiny segments,
consequently also opening up a DoS vulnerability.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Motivating Arguments . . . . . . . . . . . . . . . . . . . . . 9
2.1. Scaling Congestion Control with Packet Size . . . . . . . 9
2.2. Avoiding Perverse Incentives to (ab)use Smaller Packets . 10
2.3. Small != Control . . . . . . . . . . . . . . . . . . . . . 11
3. Working Definition of Congestion Notification . . . . . . . . 12
4. Congestion Measurement . . . . . . . . . . . . . . . . . . . . 12
4.1. Congestion Measurement by Queue Length . . . . . . . . . . 12
4.1.1. Fixed Size Packet Buffers . . . . . . . . . . . . . . 13
4.2. Congestion Measurement without a Queue . . . . . . . . . . 14
5. Idealised Wire Protocol Coding . . . . . . . . . . . . . . . . 14
6. The State of the Art . . . . . . . . . . . . . . . . . . . . . 16
6.1. Congestion Measurement: Status . . . . . . . . . . . . . . 17
6.2. Congestion Coding: Status . . . . . . . . . . . . . . . . 17
6.2.1. Network Bias when Encoding . . . . . . . . . . . . . . 17
6.2.2. Transport Bias when Decoding . . . . . . . . . . . . . 19
6.2.3. Making Transports Robust against Control Packet
Losses . . . . . . . . . . . . . . . . . . . . . . . . 20
6.2.4. Congestion Coding: Summary of Status . . . . . . . . . 21
7. Outstanding Issues and Next Steps . . . . . . . . . . . . . . 23
7.1. Bit-congestible World . . . . . . . . . . . . . . . . . . 23
7.2. Bit- & Packet-congestible World . . . . . . . . . . . . . 24
8. Security Considerations . . . . . . . . . . . . . . . . . . . 25
9. Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . 26
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 27
11. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 27
Editorial Comments . . . . . . . . . . . . . . . . . . . . . . . .
Appendix A. Example Scenarios . . . . . . . . . . . . . . . . . . 28
A.1. Notation . . . . . . . . . . . . . . . . . . . . . . . . . 28
A.2. Bit-congestible resource, equal bit rates (Ai) . . . . . . 28
A.3. Bit-congestible resource, equal packet rates (Bi) . . . . 29
A.4. Pkt-congestible resource, equal bit rates (Aii) . . . . . 30
A.5. Pkt-congestible resource, equal packet rates (Bii) . . . . 31
Appendix B. Congestion Notification Definition: Further
Justification . . . . . . . . . . . . . . . . . . . . 31
Appendix C. Byte-mode Drop Complicates Policing Congestion
Response . . . . . . . . . . . . . . . . . . . . . . 32
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 33
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12.1. Normative References . . . . . . . . . . . . . . . . . . . 33
12.2. Informative References . . . . . . . . . . . . . . . . . . 33
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 36
Intellectual Property and Copyright Statements . . . . . . . . . . 37
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Changes from Previous Versions
To be removed by the RFC Editor on publication.
Full incremental diffs between each version are available at
<http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#byte-pkt-mark>
(courtesy of the rfcdiff tool):
From -01 to -02 (this version):
Abstract reorganised to align with clearer separation of issue
in the memo.
Introduction reorganised with motivating arguments removed to
new Section 2.
Clarified avoiding lock-out of large packets is not the main or
only motivation for RED.
Mentioned choice of drop or marking explicitly throughout,
rather than trying to coin a word to mean either.
Generalised the discussion throughout to any packet forwarding
function on any network equipment, not just routers.
Clarified the last point about why this is a good time to sort
out this issue: because it will be hard / impossible to design
new transports unless we decide whether the network or the
transport is allowing for packet size.
Added statement explaining the horizon of the memo is long
term, but with short term expediency in mind.
Added material on scaling congestion control with packet size
(Section 2.1).
Separated out issue of normalising TCP's bit rate from issue of
preference to control packets (Section 2.3).
Divided up Congestion Measurement section for clarity,
including new material on fixed size packet buffers and buffer
carving (Section 4.1.1 & Section 6.2.1) and on congestion
measurement in wireless link technologies without queues
(Section 4.2).
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Added section on 'Making Transports Robust against Control
Packet Losses' (Section 6.2.3) with existing & new material
included.
Added tabulated results of vendor survey on byte-mode drop
variant of RED (Table 2).
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.
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
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].
1. Introduction
When notifying congestion, the problem of how (and whether) to take
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packet sizes into account has exercised the minds of researchers and
practitioners for as long as active queue management (AQM) has been
discussed. Indeed, one reason AQM was originally introduced was to
reduce the lock-out effects that small packets can have on large
packets in drop-tail queues. This memo aims to state the principles
we should be using and to come to conclusions on what these
principles will mean for future protocol design, taking into account
the deployments we have already.
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] or PCN
[I-D.ietf-pcn-architecture]). Throughout this document, unless clear
from the context, the term marking will be used to mean notifying
congestion explicitly, while congestion notification will be used to
mean notifying congestion either implicitly by drop or explicitly by
marking.
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 most buffer memory, because the load depends
on how many bits they have to transmit or store. Some machine
architectures use fixed size packet buffers, so buffer memory in
these cases is packet-congestible (see Section 4.1.1).
Note that information is generally processed or transmitted with a
minimum granularity greater than a bit (e.g. octets). The
appropriate granularity for the resource in question SHOULD be used,
but for the sake of brevity we will talk in terms of bytes in this
memo.
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:
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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
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 RFC series is silent on this matter other than a paper
trail of advice referenced from [RFC2309], which conditionally
recommends byte-mode (packet-size dependent) drop [pktByteEmail].
However, all the implementers who responded to our survey have not
followed this advice. The primary purpose of this memo is to build a
definitive consensus against deliberate preferential treatment for
small packets in AQM algorithms and to record this advice within the
RFC series.
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 has been
chartered to produce a standards track specification of its
congestion notification (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
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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
drop in its last para, making the recommendation less clear-cut;
3. Without the present memo, the only advice in the RFC series on
packet size bias in AQM algorithms would be a reference to an
archived email in [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 network
forwarding functions in 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 in the network 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. The IETF has started to consider the question of fairness between
flows that use different packet sizes (e.g. in the small-packet
variant of TCP-friendly rate control, TFRC-SP [RFC4828]). Given
transports with different packet sizes, if we don't decide
whether the network or the transport should allow for packet
size, it will be hard if not impossible to design any transport
protocol so that its bit-rate relative to other transports meets
design guidelines [RFC5033] (Note however that, if the concern
were fairness between users, rather than between flows
[Rate_fair_Dis], relative rates between flows would have to come
under run-time control rather than being embedded in protocol
designs).
This memo is initially concerned with how we should correctly scale
congestion control functions with packet size for the long term. But
it also recognises that expediency may be necessary to deal with
existing widely deployed protocols that don't live up to the long
term goal. It turns out that the 'correct' variant of RED to deploy
seems to be the one everyone has deployed, and no-one who responded
to our survey has implemented the other variant. However, at the
transport layer, TCP congestion control is a widely deployed protocol
that we argue doesn't scale correctly with packet size. To date this
hasn't been a significant problem because most TCPs have been used
with similar packet sizes. But, as we design new congestion
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controls, we should build in scaling with packet size rather than
assuming we should follow TCP's example.
Motivating arguments for our advice are given next in Section 2.
Then the body of the 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. Motivating Arguments
2.1. Scaling Congestion Control with Packet Size
There are two ways of interpreting a dropped or marked packet. It
can either be considered as a single loss event or as loss/marking of
the bytes in the packet. Here we try to design a test to see which
approach scales with packet size.
Imagine a bit-congestible link shared by many flows, so that each
busy period tends to cause packets to be lost from different flows.
The test compares two identical scenarios with the same applications,
the same numbers of sources and the same load. But the sources break
the load into large packets in one scenario and small packets in the
other. Of course, because the load is the same, there will be
proportionately more packets in the small packet case.
The test of whether a congestion control scales with packet size is
that it should respond in the same way to the same congestion
excursion, irrespective of the size of the packets that the bytes
causing congestion happen to be broken down into.
A bit-congestible queue suffering a congestion excursion has to drop
or mark the same excess bytes whether they are in a few large packets
or many small packets. So for the same congestion excursion, the
same amount of bytes have to be shed to get the load back to its
operating point. But, of course, for smaller packets more packets
will have to be discarded to shed the same bytes.
If all the transports interpret each drop/mark as a single loss event
irrespective of the size of the packet dropped, they will respond
more to the same congestion excursion, failing our test. On the
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other hand, if they respond proportionately less when smaller packets
are dropped/marked, overall they will be able to respond the same to
the same congestion excursion.
Therefore, for a congestion control to scale with packet size it
should respond to dropped or marked bytes (as TFRC-SP [RFC4828]
effectively does), not just to dropped or marked packets irrespective
of packet size (as TCP does).
The email [pktByteEmail] 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.
TCP congestion control ensures that flows competing for the same
resource each maintain the same number of segments in flight,
irrespective of segment size. So under similar conditions, flows
with different segment sizes will get different bit rates. But even
though reducing the drop probability of small packets helps ensure
TCPs with different packet sizes will achieve similar bit rates, we
argue this should be achieved in TCP itself, not in the network.
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?"
2.2. Avoiding Perverse Incentives to (ab)use Smaller Packets
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]. Again,
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:
Malicious transports: A queue that gives an advantage to small
packets can be used to amplify the force of a flooding attack. By
sending a flood of small packets, the attacker can get the queue
to discard more traffic in large packets, allowing more attack
traffic to get through to cause further damage. Such a queue
allows attack traffic to have a disproportionately large effect on
regular traffic without the attacker having to do much work. The
byte-mode drop variant of RED amplifies small packet attacks.
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Drop-tail queues amplify small packet attacks even more than RED
byte-mode drop (see the Security Considerations section
Section 8). Wherever possible neither should be used.
Normal transports: Even if a transport is not malicious, if it finds
small packets go faster, it will tend to act in its own interest
and use them. Queues that give advantage to small packets create
an evolutionary pressure for transports to send at the same bit-
rate but break their data stream down into tiny segments to reduce
their drop rate. Encouraging a high volume of tiny packets might
in turn unnecessarily overload a completely unrelated part of the
system, perhaps more limited by header-processing than bandwidth.
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. Consider a scenario where
gentle RED [gentle_RED] is used, along with the variant of RED we
advise against, i.e. where the RED algorithm is configured to adjust
the drop probability of packets in proportion to each packet's size
(byte mode packet drop). In this case, 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).
2.3. Small != Control
It is tempting to drop small packets with lower probability to
improve performance, because many control packets are small (TCP SYNs
& ACKs, DNS queries & responses, SIP messages, HTTP GETs, etc) and
dropping fewer control packets considerably improves performance.
However, we must not give control packets preference purely by virtue
of their smallness, otherwise it is too easy for any data source to
get the same preferential treatment simply by sending data in smaller
packets. Again we should not create perverse incentives to favour
small packets rather than to favour control packets, which is what we
intend.
Just because many control packets are small does not mean all small
packets are control packets.
So again, rather than fix these problems in the network layer, we
argue that the transport should be made more robust against losses of
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control packets (see 'Making Transports Robust against Control Packet
Losses' in Section 6.2.3).
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
4.1. Congestion Measurement by Queue Length
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.
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4.1.1. Fixed Size Packet Buffers
Some, mostly older, queuing hardware sets aside fixed sized buffers
in which to store each packet in the queue. Also, with some
hardware, any fixed sized buffers not completely filled by a packet
are padded when transmitted to the wire. If we imagine a theoretical
forwarding system with both queuing and transmission in fixed, MTU-
sized units, it should clearly be treated as packet-congestible,
because the queue length in packets would be a good model of
congestion of the lower layer link.
If we now imagine a hybrid forwarding system with transmission delay
largely dependent on the byte-size of packets but buffers of one MTU
per packet, it should strictly require a more complex algorithm to
determine the probability of congestion. It should be treated as two
resources in sequence, where the sum of the byte-sizes of the packets
within each packet buffer models congestion of the line while the
length of the queue in packets models congestion of the queue. Then
the probability of congesting the forwarding buffer would be a
conditional probability--conditional on the previously calculated
probability of congesting the line.
However, in systems that use fixed size buffers, it is unusual for
all the buffers used by an interface to be the same size. Typically
pools of different sized buffers are provided (Cisco uses the term
'buffer carving' for the process of dividing up memory into these
pools [IOSArch]). Usually, if the pool of small buffers is
exhausted, arriving small packets can borrow space in the pool of
large buffers, but not vice versa. However, it is easier to work out
what should be done if we temporarily set aside the possibility of
such borrowing. Then, with fixed pools of buffers for different
sized packets and no borrowing, the size of each pool and the current
queue length in each pool would both be measured in packets. So an
AQM algorithm would have to maintain the queue length for each pool,
and judge whether to drop/mark a packet of a particular size by
looking at the pool for packets of that size and using the length (in
packets) of its queue.
We now return to the issue we temporarily set aside: small packets
borrowing space in larger buffers. In this case, the only difference
is that the pools for smaller packets have a maximum queue size that
includes all the pools for larger packets. And every time a packet
takes a larger buffer, the current queue size has to be incremented
for all queues in the pools of buffers less than or equal to the
buffer size used.
We will return to borrowing of fixed sized buffers when we discuss
biasing the drop/marking probability of a specific packet because of
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its size in Section 6.2.1. But here we can give a simple summary of
the present discussion on how to measure the length of queues of
fixed buffers: no matter how complicated the scheme is, ultimately
any fixed buffer system will need to measure its queue length in
packets not bytes.
4.2. Congestion Measurement without a Queue
AQM algorithms are nearly always described assuming there is a queue
for a congested resource and the algorithm can use the queue length
to determine the probability that it will drop or mark each packet.
But not all congested resources lead to queues. For instance,
wireless spectrum is bit-congestible (for a given coding scheme),
because interference increases with the rate at which bits are
transmitted. But wireless link protocols do not always maintain a
queue that depends on spectrum interference. Similarly, power
limited resources are also usually bit-congestible if energy is
primarily required for transmission rather than header processing,
but it is rare for a link protocol to build a queue as it approaches
maximum power.
However, AQM algorithms don't require a queue in order to work. For
instance spectrum congestion can be modelled by signal quality using
target bit-energy-to-noise-density ratio. And, to model radio power
exhaustion, transmission power levels can be measured and compared to
the maximum power available. [ECNFixedWireless] proposes a practical
and theoretically sound way to combine congestion notification for
different bit-congestible resources at different layers along an end
to end path, whether wireless or wired, and whether with or without
queues.
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
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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
the motivating argument about avoiding perverse incentives in
Section 2.2). Incidentally, it also ensures neither the network nor
the transport needs to do a multiply operation--multiplication by
packet size is effectively achieved as a repeated add when the
transport adds to its count of marked bytes as each congestion event
is fed to it:
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
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sufficient information for the two different types of transport
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 dropped to delivered segments (or marked to unmarked
segments in the case of ECN). The example scenarios show that
these ratio-based transports are effectively the same whether
counting in bytes or packets, because the units cancel out.
(Incidentally, this is why TCP's bit rate is still proportional to
packet size even when byte-counting is used, as recommended for
TCP in [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 dropped (or
marked) individual packets with a probability independent of their
size. Byte mode measured the queue length in bytes and marked an
individual packet with probability in proportion to its size
(relative to the maximum packet size). In the paper's outline of
further work, it was stated that no recommendation had been made on
whether the queue size should be measured in bytes or packets, but
noted that the difference could be significant.
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).
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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 don'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 queue implementations give a choice for setting RED's
thresholds in byte-mode or packet-mode. This may merely be an
administrator-interface preference, not altering how the queue itself
is measured but on some hardware it does actually change the way it
measures its queue. Whether a resource is bit-congestible or packet-
congestible is a property of the resource, so an admin SHOULD NOT
ever need to, or be able to, configure the way a queue measures
itself.
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). But there is no controversy over what should be done.
It's just you have to be an expert in probability to work out what
should be done and, even if you have, it's not always easy to find a
practical algorithm to implement it.
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 2 argues against this
approach, citing the excellent advice in RFC3246.] The referenced
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email went on to argue that drop probability should depend on the
size of the packet being considered for drop if the resource is bit-
congestible, but not if it is packet-congestible, but advised that
most scarce resources in the Internet were currently bit-congestible.
The argument continued that if packet drops were inflated by packet
size (byte-mode dropping), "a flow's fraction of the packet drops is
then a good indication of that flow's fraction of the link bandwidth
in bits per second". This was consistent with a referenced policing
mechanism being worked on at the time for detecting unusually high
bandwidth flows, eventually published in 1999 [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
dropping/marking a packet should depend on its size was not related
to whether the resource itself was bit congestible, but a completely
orthogonal question. However the only example given had the queue
measured in packets but packet drop depended on the byte-size of the
packet in question. No example was given the other way round.
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 one 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.
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The byte-mode drop variant of RED is, of course, not the only
possible bias towards small packets in queueing algorithms. We have
already mentioned that tail-drop queues naturally tend to lock-out
large packets once they are full. But also queues with fixed sized
buffers reduce the probability that small packets will be dropped if
(and only if) they allow small packets to borrow buffers from the
pools for larger packets. As was explained in Section 4.1.1 on fixed
size buffer carving, borrowing effectively makes the maximum queue
size for small packets greater than that for large packets, because
more buffers can be used by small packets while less will fit large
packets.
However, in itself, the bias towards small packets caused by buffer
borrowing is perfectly correct. Lower drop probability for small
packets is legitimate in buffer borrowing schemes, because small
packets genuinely congest the machine's buffer memory less than large
packets, given they can fit in more spaces. The bias towards small
packets is not artificially added (as it is in RED's byte-mode drop
algorithm), it merely reflects the reality of the way fixed buffer
memory gets congested. Incidentally, the bias towards small packets
from buffer borrowing is nothing like as large as that of RED's byte-
mode drop.
Nonetheless, fixed-buffer memory with tail drop is still prone to
lock-out large packets, purely because of the tail-drop aspect. So a
good AQM algorithm like RED with packet-mode drop should be used with
fixed buffer memories where possible. If RED is too complicated to
implement with multiple fixed buffer pools, the minimum necessary to
prevent large packet lock-out is to ensure smaller packets never use
the last available buffer in any of the pools for larger packets.
6.2.2. Transport Bias when Decoding
The above proposals to alter the network layer to give a bias towards
smaller packets have largely carried on outside the IETF process
(unless one counts a reference in an informational RFC to an archived
email!). Whereas, within the IETF, there are many different
proposals to alter transport protocols to achieve the same goals,
i.e. either to make the flow bit-rate take account of packet size, or
to protect control packets from loss. This memo argues that altering
transport protocols is the more principled approach.
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
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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 queues have been configured to drop
smaller packets with proportionately lower probability than larger
ones. But it only discusses TCP operating in such an environment,
only mentioning TFRC-SP briefly when discussing how to define
fairness with TCP. And it only discusses the byte-mode dropping
version of RED as it was before Cnodder et al pointed out it didn't
sufficiently bias towards small packets to make TCP independent of
packet size.
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 but not recommending any
particular course of 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 6.2.4). 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
to use a virtual queue for AQM marking for packets within one
Diffserv class in order to give early warning prior to any real
queuing. The proposed PCN marking algorithms have been designed not
to take account of packet size when forwarding through queues.
Instead the general principle has been to take account of the sizes
of marked packets when monitoring the fraction of marking at the edge
of the network.
6.2.3. Making Transports Robust against Control Packet Losses
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
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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.
Although no proposals exist as far as we know, it would also be
possible and perfectly valid to make control packets robust against
drop by explicitly requesting a lower drop probability using their
Diffserv code point [RFC2474] to request a scheduling class with
lower drop.
The re-ECN protocol proposal [Re-TCP] is designed so that transports
can be made more robust against losing control packets. It gives
queues an incentive to optionally give preference against drop to
packets with the 'feedback not established' codepoint in the proposed
'extended ECN' field. Senders have incentives to use this codepoint
sparingly, but they can use it on control packets to reduce their
chance of being dropped. For instance, the proposed modification to
TCP for re-ECN uses this codepoint on the SYN and SYN-ACK.
Although not brought to the IETF, a simple proposal from Wischik
[DupTCP] suggests that the first three packets of every TCP flow
should be routinely duplicated after a short delay. It shows that
this would greatly improve the chances of short flows completing
quickly, but it would hardly increase traffic levels on the Internet,
because Internet bytes have always been concentrated in the large
flows. It further shows that the performance of many typical
applications depends on completion of long serial chains of short
messages. It argues that, given most of the value people get from
the Internet is concentrated within short flows, this simple
expedient would greatly increase the value of the best efforts
Internet at minimal cost.
6.2.4. 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
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Table 1 aims to summarise the positions we may now be in. Each
column shows a different possible AQM behaviour in different queues
in the network, using the terminology of Cnodder et al outlined
earlier (RED_1 is basic RED with packet-mode drop). Each row shows a
different transport behaviour: TCP [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 queues, 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 equipment 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
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 (Table 2), 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).
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+-------------------------------+----------------+-----------------+
| Response | No. of vendors | %age of vendors |
+-------------------------------+----------------+-----------------+
| Not implemented | 14 | 17% |
| Not implemented (probably) | 2 | 2% |
| Implemented | 0 | 0% |
| No response | 68 | 81% |
| Total companies/orgs surveyed | 84 | 100% |
+-------------------------------+----------------+-----------------+
Table 2: Vendor Survey on byte-mode drop variant of RED (lower drop
probability for small packets)
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.
Finally, we repeat that RED's byte mode drop is not the only way to
bias towards small packets--tail-drop tends to lock-out large packets
very effectively. Our survey was of vendor implementations, so we
cannot be certain about operator deployment. But we believe many
queues in the Internet are still tail-drop. My own company (BT) has
widely deployed RED, but there are bound to be many tail-drop queues,
particularly in access network equipment and on middleboxes like
firewalls, where RED is not always available. Routers using a memory
architecture based on fixed size buffers with borrowing may also
still be prevalent in the Internet. As explained in Section 6.2.1,
these also provide a marginal (but legitimate) bias towards small
packets. So even though RED byte-mode drop is not prevalent, it is
likely there is still some bias towards small packets in the Internet
due to tail drop and fixed buffer borrowing.
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 6.2.4 suggest
that byte-mode packet drop seems not to be implemented at all let
alone deployed, or if it is, it is likely to be very sparse.
Therefore, we do not really need a migration strategy from all but
nothing to nothing.
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,
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given the IRTF ICCRG has set itself the task of reaching consensus on
generic forwarding mechanisms that are necessary and sufficient to
support the Internet's future congestion control requirements
[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.
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.
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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, ECN & PCN) SHOULD take account of
packet size when the notification is read by the transport layer, NOT
when it is written by the network layer. This approach offers
sufficient and correct congestion information for all known and
future transport protocols and also ensures no perverse incentives
are created that would encourage transports to use inappropriately
small packet sizes.
The alternative of deflating RED's drop probability for smaller
packet sizes (byte-mode drop) has no enduring advantages. It is more
complex, it creates the perverse incentive to fragment segments into
tiny pieces and it reopens the vulnerability to floods of small-
packets that drop-tail queues suffered from and AQM was designed to
remove. 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 84 vendors across the industry has drawn
responses from about 19%, none of whom have implemented the byte mode
packet drop variant of RED. Given there appears to be little, if
any, installed base 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
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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
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 who
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
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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
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 3
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
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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
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.
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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
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.
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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).
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.
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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
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 queue that caused each drop/mark.
Therefore it cannot take an integrated approach to policing all the
responses to congestion of all the transports of one individual.
Therefore it cannot police anything.
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 dropping/marking naturally
allows the congestion notification of packets to be globally
meaningful without relying on MTU information held elsewhere.
Because we recommend that a dropped/marked packet should be taken to
mean that all the bytes in the packet are dropped/marked, a policer
can remain robust against bits being re-divided into different size
packets or across different size flows [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.
12. References
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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.
[RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474,
December 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.
[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion
Control Algorithms", BCP 133, RFC 5033, August 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>.
[DupTCP] Wischik, D., "Short messages", Royal Society workshop on
networks: modelling and control , September 2007, <http://
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www.cs.ucl.ac.uk/staff/ucacdjw/Research/shortmsg.html>.
[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-03 (work in progress),
February 2008.
[I-D.ietf-tcpm-ecnsyn]
Floyd, S., "Adding Explicit Congestion Notification (ECN)
Capability to TCP's SYN/ACK Packets",
draft-ietf-tcpm-ecnsyn-05 (work in progress),
February 2008.
[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",
draft-irtf-iccrg-welzl-congestion-control-open-research-00
(work in progress), July 2007.
[IOSArch] Bollapragada, V., White, R., and C. Murphy, "Inside Cisco
IOS Software Architecture", Cisco Press: CCIE Professional
Development ISBN13: 978-1-57870-181-0, July 2000.
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[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., Moncaster, T., and A. Smith,
"Re-ECN: Adding Accountability for Causing Congestion to
TCP/IP", draft-briscoe-tsvwg-re-ecn-tcp-05 (work in
progress), January 2008.
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[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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