One document matched: draft-ietf-tsvwg-byte-pkt-congest-02.txt
Differences from draft-ietf-tsvwg-byte-pkt-congest-01.txt
Transport Area Working Group B. Briscoe
Internet-Draft BT
Updates: 2309 (if approved) J. Manner
Intended status: Informational Aalto University
Expires: January 13, 2011 July 12, 2010
Byte and Packet Congestion Notification
draft-ietf-tsvwg-byte-pkt-congest-02
Abstract
This memo concerns dropping or marking packets using active queue
management (AQM) such as random early detection (RED) or pre-
congestion notification (PCN). We give two strong recommendations:
(1) packet size should not be taken into account when transports read
congestion indications, not when network equipment writes them, and
(2) byte-mode packet drop variant of AQM algorithms, such as RED,
should not be used to drop fewer small packets.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 13, 2011.
Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Terminology and Scoping . . . . . . . . . . . . . . . . . 6
1.2. Why now? . . . . . . . . . . . . . . . . . . . . . . . . . 7
2. Motivating Arguments . . . . . . . . . . . . . . . . . . . . . 8
2.1. Scaling Congestion Control with Packet Size . . . . . . . 8
2.2. Avoiding Perverse Incentives to (ab)use Smaller Packets . 10
2.3. Small != Control . . . . . . . . . . . . . . . . . . . . . 11
2.4. Implementation Efficiency . . . . . . . . . . . . . . . . 11
3. The State of the Art . . . . . . . . . . . . . . . . . . . . . 11
3.1. Congestion Measurement: Status . . . . . . . . . . . . . . 12
3.1.1. Fixed Size Packet Buffers . . . . . . . . . . . . . . 13
3.1.2. Congestion Measurement without a Queue . . . . . . . . 14
3.2. Congestion Coding: Status . . . . . . . . . . . . . . . . 14
3.2.1. Network Bias when Encoding . . . . . . . . . . . . . . 14
3.2.2. Transport Bias when Decoding . . . . . . . . . . . . . 16
3.2.3. Making Transports Robust against Control Packet
Losses . . . . . . . . . . . . . . . . . . . . . . . . 17
3.2.4. Congestion Coding: Summary of Status . . . . . . . . . 18
4. Outstanding Issues and Next Steps . . . . . . . . . . . . . . 20
4.1. Bit-congestible World . . . . . . . . . . . . . . . . . . 20
4.2. Bit- & Packet-congestible World . . . . . . . . . . . . . 21
5. Recommendation and Conclusions . . . . . . . . . . . . . . . . 22
5.1. Recommendation on Queue Measurement . . . . . . . . . . . 22
5.2. Recommendation on Notifying Congestion . . . . . . . . . . 23
5.3. Recommendation on Responding to Congestion . . . . . . . . 24
5.4. Recommended Future Research . . . . . . . . . . . . . . . 24
6. Security Considerations . . . . . . . . . . . . . . . . . . . 24
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
8. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 25
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 25
9.1. Normative References . . . . . . . . . . . . . . . . . . . 25
9.2. Informative References . . . . . . . . . . . . . . . . . . 26
Appendix A. Congestion Notification Definition: Further
Justification . . . . . . . . . . . . . . . . . . . . 30
Appendix B. Idealised Wire Protocol . . . . . . . . . . . . . . . 30
B.1. Protocol Coding . . . . . . . . . . . . . . . . . . . . . 30
B.2. Example Scenarios . . . . . . . . . . . . . . . . . . . . 32
B.2.1. Notation . . . . . . . . . . . . . . . . . . . . . . . 32
B.2.2. Bit-congestible resource, equal bit rates (Ai) . . . . 32
B.2.3. Bit-congestible resource, equal packet rates (Bi) . . 33
B.2.4. Pkt-congestible resource, equal bit rates (Aii) . . . 34
B.2.5. Pkt-congestible resource, equal packet rates (Bii) . . 35
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Appendix C. Byte-mode Drop Complicates Policing Congestion
Response . . . . . . . . . . . . . . . . . . . . . . 35
Appendix D. Changes from Previous Versions . . . . . . . . . . . 36
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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, 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.
The byte vs. packet dilemma arises at three stages in the congestion
notification process:
Measuring congestion: When the congested resource decides locally to
measure how congested it is. (Should the queue measure its length
in bytes or packets?);
Coding congestion notification into the wire protocol: When the
congested resource decides whether to notify the level of
congestion on each particular packet. (When a queue considers
whether to notify congestion by dropping or marking a particular
packet, should its decision depend on the byte-size of the
particular packet being dropped or marked?);
Decoding congestion notification from the wire protocol: When the
transport interprets the notification in order to decide how much
to respond to congestion. (Should the transport take into account
the byte-size of each missing or marked packet?).
Consensus has emerged over the years concerning the first stage:
whether queues are measured in bytes or packets, termed byte-mode
queue measurement or packet-mode queue measurement. This memo
records this consensus in the RFC Series. In summary the choice
solely depends on whether the resource is congested by bytes or
packets.
The controversy is mainly around the last two stages to do with
encoding congestion notification into packets: whether to allow for
the size of the specific packet notifying congestion i) when the
network encodes or ii) when the transport decodes the congestion
notification.
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].
The primary purpose of this memo is to build a definitive consensus
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against such deliberate preferential treatment for small packets in
AQM algorithms and to record this advice within the RFC series.
Fortunately all the implementers who responded to our survey
(Section 3.2.4) have not followed the earlier advice, so the
consensus this memo argues for seems to already exist in
implementations.
The primary conclusion of this memo is that packet size should be
taken into account when transports read 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 addressing these issues at
the transport layer, rather than reverse engineering network
forwarding to fix specific transport problems.
The second conclusion is that 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.
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
controls, we should build in scaling with packet size rather than
assuming we should follow TCP's example.
This memo continues as follows. Terminology and scoping are
discussed next, and the reasons to make the recommendations presented
in this memo now are given in Section 1.2. Motivating arguments for
our advice are given in Section 2. We then survey the advice given
previously in the RFC series, the research literature and the
deployed legacy (Section 3) before listing outstanding issues
(Section 4) that will need resolution both to inform future protocols
designs and to handle legacy. We then give concrete recommendations
for the way forward in (Section 5). We finally give security
considerations in Section 6. The interested reader can also find
further discussions about the theme of byte vs. packet in the
appendices.
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This memo intentionally includes a non-negligible amount of material
on the subject. A busy reader can jump right into Section 5 to read
a summary of the recommendations for the Internet community.
1.1. Terminology and Scoping
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].
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. E is the instantaneous excess load offered to a
resource that it is either incapable of serving or unwilling to
serve. L is the instantaneous offered load.
The phrase `unwilling to serve' is added, because AQM systems
(e.g. RED, PCN [RFC5670]) set a virtual limit smaller than the
actual limit to the resource, then notify when this virtual limit
is exceeded 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 measure
of congestion notification: drop fraction (often loosely called
loss 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 A describes a further incidental benefit that
arises from using load as the denominator of congestion
notification.
Explicit and Implicit Notification: The byte vs. packet dilemma
concerns congestion notification irrespective of whether it is
signalled implicitly by drop or using explicit congestion
notification (ECN [RFC3168] or PCN [RFC5670]). Throughout this
document, unless clear from the context, the term marking will be
used to mean notifying congestion explicitly, while congestion
notification will be used to mean notifying congestion either
implicitly by drop or explicitly by marking.
Bit-congestible vs. Packet-congestible: If the load on a resource
depends on the rate at which packets arrive, it is called packet-
congestible. If the load depends on the rate at which bits arrive
it is called bit-congestible.
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Examples of packet-congestible resources are route look-up engines
and firewalls, because load depends on how many packet headers
they have to process. Examples of bit-congestible resources are
transmission links, radio power and most buffer memory, because
the load depends on how many bits they have to transmit or store.
Some machine architectures use fixed size packet buffers, so
buffer memory in these cases is packet-congestible (see
Section 3.1.1).
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.
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.
Coarser granularity: 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 applicable for congestion notification
protocols controlling per-flow and per-session processing or
state.
RED Terminology: In RED, whether to use packets or bytes when
measuring queues is respectively called packet-mode or byte-mode
queue measurement. And if the probability of dropping a packet
depends on its byte-size it is called byte-mode drop, whereas if
the drop probability is independent of a packet's byte-size it is
called packet-mode drop.
1.2. Why now?
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
standardising the external behaviour of a PCN congestion
notification (AQM) algorithm [RFC5670];
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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
drop in its last paragraph, 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]. 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).
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.
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Given bit-congestible is the more common case (see Section 1.1),
consider 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, those with smaller
packets will respond more to the same congestion excursion, failing
our test. On the 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 correction should be made to TCP itself, not to the
network in order to fix one transport, no matter how prominent it is.
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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 bit-congestion of a link irrespective of whether it is
sent as fewer larger packets or more smaller packets. A protocol
design that caused larger packets to be more likely to be dropped
than smaller ones would be dangerous in this case:
Normal transports: Even if a transport is not actually malicious, if
it finds small packets go faster, over time 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.
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.
Note that, although the byte-mode drop variant of RED amplifies
small packet attacks, drop-tail queues amplify small packet
attacks even more (see Security Considerations in Section 6).
Wherever possible neither should be used.
Imagine two unresponsive flows arrive at a bit-congestible
transmission link each with the same bit rate, say 1Mbps, but one
consists of 1500B and the other 60B packets, which are 25x smaller.
Consider a scenario where gentle RED [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
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smaller packets (but in practice it may drop more as congestion
increases [RFC4828](S.B.4)). 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 6).
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
control packets (see 'Making Transports Robust against Control Packet
Losses' in Section 3.2.3).
2.4. Implementation Efficiency
Allowing for packet size at the transport rather than in the network
ensures that neither the network nor the transport needs to do a
multiply operation--multiplication by packet size is effectively
achieved as a repeated add when the transport adds to its count of
marked bytes as each congestion event is fed to it. This isn't a
principled reason in itself, but it is a happy consequence of the
other principled reasons.
3. 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
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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 3.1 below) and whether the drop probability of an
individual packet should depend on its own size (Section 3.2 below).
3.1. 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.
Where buffers are not configured or legacy buffers cannot be
configured to the above guideline, we do not 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.
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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
(summarised in the following section) and, even if you have, it's not
always easy to find a practical algorithm to implement it.
3.1.1. 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.
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
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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
its size in Section 3.2.1. 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.
3.1.2. 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.
Nonetheless, AQM algorithms do not require a queue in order to work.
For instance spectrum congestion can be modelled by signal quality
using target bit-energy-to-noise-density ratio. And, to model radio
power exhaustion, transmission power levels can be measured and
compared to the maximum power available. [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.
3.2. Congestion Coding: Status
3.2.1. Network Bias when Encoding
The previously mentioned email [pktByteEmail] referred to by
[RFC2309] gave advice we now disagree with. It said that drop
probability should depend on the size of the packet being considered
for drop if the resource is bit-congestible, but not if it is packet-
congestible, 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
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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 and should 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 implied
motivation behind RED's byte mode drop--that we should reverse
engineer the network to improve the performance of dominant end-to-
end congestion control mechanisms.
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.
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 3.1.1 on fixed
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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.
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.
3.2.2. Transport Bias when Decoding
The above proposals to alter the network equipment to 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
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
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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 3.2.4). 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] 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.
3.2.3. Making Transports Robust against Control Packet Losses
Recently, two RFCs have defined changes to TCP that make it more
robust against losing small control packets [RFC5562] [RFC5690]. 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.
Although no proposals exist as far as we know, it would also be
possible and perfectly valid to make control packets robust against
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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 [I-D.briscoe-tsvwg-re-ecn-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.
3.2.4. 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
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 [RFC5681] and TFRC [RFC3448] on
the top row with TFRC-SP [RFC4828] below. Suppressing all
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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 were fairly sure they
had not but needed 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. The company of one of
the co-authors (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 3.2.1, 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.
4. Outstanding Issues and Next Steps
4.1. Bit-congestible World
For a connectionless network with nearly all resources being bit-
congestible 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 3.2.4 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 [RFC5681] itself.
4.2. 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.
We develop the concept of an idealised congestion notification
protocol that supports both bit-congestible and packet-congestible
resources in Appendix B. The congestion notification 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.
Recently, the dual resource queue (DRQ) proposal [DRQ] has been made
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on the premise that, as network processors become more cost
effective, per packet operations will become more complex
(irrespective of whether more function in the network layer is
desirable). Consequently the premise is that CPU congestion will
become more common. DRQ is a proposed modification to the RED
algorithm that folds both bit congestion and packet congestion into
one signal (either loss or ECN).
The problem of signalling packet processing congestion is not
pressing, as most Internet resources are designed to be bit-
congestible before packet processing starts to congest (see
Section 1.1). However, the IRTF Internet congestion control research
group (ICCRG) has set itself the task of reaching consensus on
generic forwarding mechanisms that are necessary and sufficient to
support the Internet's future congestion control requirements (the
first challenge in [I-D.irtf-iccrg-welzl]). Therefore, rather than
not giving this problem any thought at all, just because it is hard
and currently hypothetical, we defer the question of whether packet
congestion might become common and what to do if it does to the IRTF
(the 'Small Packets' challenge in [I-D.irtf-iccrg-welzl]).
5. Recommendation and Conclusions
5.1. Recommendation on Queue 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 3.
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.
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5.2. Recommendation on Notifying Congestion
The strong recommendation 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 it seems we can recommend removal of byte-mode
drop from RED with little, if any, incremental deployment impact.
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.
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5.3. Recommendation on Responding to Congestion
Instead of network equipment biasing its congestion notification for
small packets, 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.
5.4. Recommended Future Research
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.
6. 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
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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.
7. Acknowledgements
Thank you to Sally Floyd, who gave extensive and useful review
comments. Also thanks for the reviews from Philip Eardley, Toby
Moncaster and Arnaud Jacquet as well as helpful explanations of
different hardware approaches from Larry Dunn and Fred Baker. 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.
Bob Briscoe and Jukka Manner are partly funded by Trilogy, a research
project (ICT- 216372) supported by the European Community under its
Seventh Framework Programme. The views expressed here are those of
the authors only.
8. 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.
9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in
RFCs to Indicate Requirement Levels",
BCP 14, RFC 2119, March 1997.
[RFC2309] Braden, B., Clark, D., Crowcroft, J.,
Davie, B., Deering, S., Estrin, D.,
Floyd, S., Jacobson, V., Minshall,
G., Partridge, C., Peterson, L.,
Ramakrishnan, K., Shenker, S.,
Wroclawski, J., and L. Zhang,
"Recommendations on Queue Management
and Congestion Avoidance in the
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Internet", RFC 2309, April 1998.
[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.
[RFC5033] Floyd, S. and M. Allman, "Specifying
New Congestion Control Algorithms",
BCP 133, RFC 5033, August 2007.
9.2. 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>.
[DRQ] Shin, M., Chong, S., and I. Rhee,
"Dual-Resource TCP/AQM for
Processing-Constrained Networks",
IEEE/ACM Transactions on
Networking Vol 16, issue 2,
April 2008, <http://dx.doi.org/
10.1109/TNET.2007.900415>.
[DupTCP] Wischik, D., "Short messages", Royal
Society workshop on networks:
modelling and control ,
September 2007, <http://
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",
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Automatica 35(12)1969--1985,
December 1999, <http://
www.statslab.cam.ac.uk/~frank/
evol.html>.
[I-D.briscoe-tsvwg-re-ecn-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-08
(work in progress), September 2009.
[I-D.ietf-pcn] Eardley, P., "Metering and marking
behaviour of PCN-nodes",
draft-ietf-pcn-marking-behaviour-05
(work in progress), August 2009.
[I-D.irtf-iccrg-welzl] Welzl, M., Scharf, M., Briscoe, B.,
and D. Papadimitriou, "Open Research
Issues in Internet Congestion
Control", draft-irtf-iccrg-welzl-
congestion-control-open-research-07
(work in progress), June 2010.
[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.
[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>.
[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
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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>.
[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.
[RFC3448] Handley, M., Floyd, S., Padhye, J.,
and J. Widmer, "TCP Friendly Rate
Control (TFRC): Protocol
Specification", RFC 3448,
January 2003.
[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.
[RFC4828] Floyd, S. and E. Kohler, "TCP
Friendly Rate Control (TFRC): The
Small-Packet (SP) Variant", RFC 4828,
April 2007.
[RFC5562] Kuzmanovic, A., Mondal, A., Floyd,
S., and K. Ramakrishnan, "Adding
Explicit Congestion Notification
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(ECN) Capability to TCP's SYN/ACK
Packets", RFC 5562, June 2009.
[RFC5670] Eardley, P., "Metering and Marking
Behaviour of PCN-Nodes", RFC 5670,
November 2009.
[RFC5681] Allman, M., Paxson, V., and E.
Blanton, "TCP Congestion Control",
RFC 5681, September 2009.
[RFC5690] Floyd, S., Arcia, A., Ros, D., and J.
Iyengar, "Adding Acknowledgement
Congestion Control to TCP", RFC 5690,
February 2010.
[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>.
[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] Yes and J. Doe, "Missing for now",
RFC 0000, May 2006.
[xcp-spec] Falk, A., "Specification for the
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Explicit Control Protocol (XCP)",
draft-falk-xcp-spec-03 (work in
progress), July 2007.
Appendix A. Congestion Notification Definition: Further Justification
In Section 1.1 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 [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.
Appendix B. Idealised Wire Protocol
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 later in this
appendix.
B.1. Protocol Coding
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
(imaginary) fields in each datagram to signal congestion: one for
byte congestion and one for packet congestion.
We are not saying two ECN fields will be needed (and we are not
saying that somehow a resource should be able to drop a packet in one
of two different ways so that the transport can distinguish which
sort of drop it was!). These two congestion notification channels
are just a conceptual device. They allow us to defer having to
decide whether to distinguish between byte and packet congestion when
the network resource codes the signal or when the transport decodes
it.
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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,
packet-congestion is not the common case, 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):
1. 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.
2. 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 B.2 show that transports can extract
sufficient and correct congestion notification from these protocols
for cases when two flows with different packet sizes have matching
bit rates or matching packet rates. Examples are also given that mix
these two flows into one to show that a flow with mixed packet sizes
would still be able to extract sufficient and correct information.
Sufficient and correct congestion information means that there is
sufficient information for the two different types of transport
requirements:
Ratio-based: Established transport congestion controls like TCP's
[RFC5681] 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
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packet size even when byte-counting is used, as recommended for
TCP in [RFC5681], 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.
B.2. Example Scenarios
B.2.1. Notation
To prove our idealised wire protocol (Appendix B.1) is correct, 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
B.2.2. 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
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= 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 (Appendix B.1).
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
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 (Appendix B.1), 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.
B.2.3. 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.
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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
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.
B.2.4. 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
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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 a packet.
B.2.5. 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 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 a byte-mode drop algorithm has to depend on the local MTU of the
line - an algorithm needs to use some concept of a 'normal' packet
size. Therefore, one dropped or marked packet is not necessarily
equivalent to another unless you know the MTU at the queue where it
was 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 [I-D.briscoe-tsvwg-re-ecn-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
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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.
Appendix D. 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-congest>
or
<http://tools.ietf.org/wg/tsvwg/draft-ietf-tsvwg-byte-pkt-congest/>
(courtesy of the rfcdiff tool):
From -01 to -02 (this version):
* Restructured the whole document for (hopefully) easier reading
and clarity. The concrete recommendation, in RFC2119 language,
is now in Section 5.
From -00 to -01:
* Minor clarifications throughout and updated references
From briscoe-byte-pkt-mark-02 to ietf-byte-pkt-congest-00:
* Added note on relationship to existing RFCs
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* Posed the question of whether packet-congestion could become
common and deferred it to the IRTF ICCRG. Added ref to the
dual-resource queue (DRQ) proposal.
* Changed PCN references from the PCN charter & architecture to
the PCN marking behaviour draft most likely to imminently
become the standards track WG item.
From -01 to -02:
* 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 3.1.1 & Section 3.2.1) and on congestion
measurement in wireless link technologies without queues
(Section 3.1.2).
* Added section on 'Making Transports Robust against Control
Packet Losses' (Section 3.2.3) with existing & new material
included.
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* 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
Authors' Addresses
Bob Briscoe
BT
B54/77, Adastral Park
Martlesham Heath
Ipswich IP5 3RE
UK
Phone: +44 1473 645196
EMail: bob.briscoe@bt.com
URI: http://bobbriscoe.net/
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Jukka Manner
Aalto University
Department of Communications and Networking (Comnet)
P.O. Box 13000
FIN-00076 Aalto
Finland
Phone: +358 9 470 22481
EMail: jukka.manner@tkk.fi
URI: http://www.netlab.tkk.fi/~jmanner/
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