Internet DRAFT - draft-ietf-tsvwg-byte-pkt-congest
draft-ietf-tsvwg-byte-pkt-congest
Transport Area Working Group B. Briscoe
Internet-Draft BT
Updates: 2309 (if approved) J. Manner
Intended status: BCP Aalto University
Expires: May 11, 2014 November 07, 2013
Byte and Packet Congestion Notification
draft-ietf-tsvwg-byte-pkt-congest-12
Abstract
This document provides recommendations of best current practice for
dropping or marking packets using any active queue management (AQM)
algorithm, including random early detection (RED), BLUE, pre-
congestion notification (PCN) and newer schemes such as CoDel
(Controlled Delay) and PIE (Proportional Integral controller
Enhanced). We give three strong recommendations: (1) packet size
should be taken into account when transports detect and respond to
congestion indications, (2) packet size should not be taken into
account when network equipment creates congestion signals (marking,
dropping), and therefore (3) in the specific case of RED, the byte-
mode packet drop variant that drops fewer small packets should not be
used. This memo updates RFC 2309 to deprecate deliberate
preferential treatment of small packets in AQM algorithms.
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-
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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 May 11, 2014.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Terminology and Scoping . . . . . . . . . . . . . . . . . 6
1.2. Example Comparing Packet-Mode Drop and Byte-Mode Drop . . 7
2. Recommendations . . . . . . . . . . . . . . . . . . . . . . . 9
2.1. Recommendation on Queue Measurement . . . . . . . . . . . 9
2.2. Recommendation on Encoding Congestion Notification . . . . 10
2.3. Recommendation on Responding to Congestion . . . . . . . . 11
2.4. Recommendation on Handling Congestion Indications when
Splitting or Merging Packets . . . . . . . . . . . . . . . 12
3. Motivating Arguments . . . . . . . . . . . . . . . . . . . . . 12
3.1. Avoiding Perverse Incentives to (Ab)use Smaller Packets . 12
3.2. Small != Control . . . . . . . . . . . . . . . . . . . . . 14
3.3. Transport-Independent Network . . . . . . . . . . . . . . 14
3.4. Partial Deployment of AQM . . . . . . . . . . . . . . . . 15
3.5. Implementation Efficiency . . . . . . . . . . . . . . . . 17
4. A Survey and Critique of Past Advice . . . . . . . . . . . . . 17
4.1. Congestion Measurement Advice . . . . . . . . . . . . . . 18
4.1.1. Fixed Size Packet Buffers . . . . . . . . . . . . . . 18
4.1.2. Congestion Measurement without a Queue . . . . . . . . 19
4.2. Congestion Notification Advice . . . . . . . . . . . . . . 20
4.2.1. Network Bias when Encoding . . . . . . . . . . . . . . 20
4.2.2. Transport Bias when Decoding . . . . . . . . . . . . . 22
4.2.3. Making Transports Robust against Control Packet
Losses . . . . . . . . . . . . . . . . . . . . . . . . 23
4.2.4. Congestion Notification: Summary of Conflicting
Advice . . . . . . . . . . . . . . . . . . . . . . . . 24
5. Outstanding Issues and Next Steps . . . . . . . . . . . . . . 25
5.1. Bit-congestible Network . . . . . . . . . . . . . . . . . 25
5.2. Bit- & Packet-congestible Network . . . . . . . . . . . . 25
6. Security Considerations . . . . . . . . . . . . . . . . . . . 26
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 26
8. Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . 26
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 28
10. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 28
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 28
11.1. Normative References . . . . . . . . . . . . . . . . . . . 28
11.2. Informative References . . . . . . . . . . . . . . . . . . 28
Appendix A. Survey of RED Implementation Status . . . . . . . . . 32
Appendix B. Sufficiency of Packet-Mode Drop . . . . . . . . . . . 34
B.1. Packet-Size (In)Dependence in Transports . . . . . . . . . 35
B.2. Bit-Congestible and Packet-Congestible Indications . . . . 38
Appendix C. Byte-mode Drop Complicates Policing Congestion
Response . . . . . . . . . . . . . . . . . . . . . . 39
Appendix D. Changes from Previous Versions . . . . . . . . . . . 40
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1. Introduction
This document provides recommendations of best current practice for
how we should correctly scale congestion control functions with
respect to packet size for the long term. 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.
When signalling 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 outline how these principles will affect
future protocol design, taking into account the existing deployments
we have already.
The question of whether to take into account packet size arises at
three stages in the congestion notification process:
Measuring congestion: When a congested resource measures locally how
congested it is, should it measure its queue length in time, bytes
or packets?
Encoding congestion notification into the wire protocol: When a
congested network resource signals its level of congestion, should
it drop / mark each packet dependent on the size of the particular
packet in question?
Decoding congestion notification from the wire protocol: When a
transport interprets the notification in order to decide how much
to respond to congestion, should it take into account the size of
each missing or marked packet?
Consensus has emerged over the years concerning the first stage,
which Section 2.1 records in the RFC Series. In summary: If possible
it is best to measure congestion by time in the queue, but otherwise
the choice between bytes and packets solely depends on whether the
resource is congested by bytes or packets.
The controversy is mainly around the last two stages: 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
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recommends byte-mode (packet-size dependent) drop [pktByteEmail].
Reducing drop of small packets certainly 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 the
problems.
This memo updates [RFC2309] to deprecate deliberate preferential
treatment of packets in AQM algorithms solely because of their size.
It recommends that (1) packet size should be taken into account when
transports detect and respond to congestion indications, (2) not when
network equipment creates them. This memo also adds to the
congestion control principles enumerated in BCP 41 [RFC2914].
In the particular case of Random early Detection (RED), this means
that the byte-mode packet drop variant 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. Fortunately all the RED implementers who responded to
our admittedly limited survey (Section 4.2.4) have not followed the
earlier advice to use byte-mode drop, so the position this memo
argues for seems to already exist in implementations.
However, at the transport layer, TCP congestion control is a widely
deployed protocol that doesn't scale with packet size (i.e. its
reduction in rate does not take into account the size of a lost
packet). To date this hasn't been a significant problem because most
TCP implementations have been used with similar packet sizes. But,
as we design new congestion control mechanisms, this memo recommends
that we should build in scaling with packet size rather than assuming
we should follow TCP's example.
This memo continues as follows. First it discusses terminology and
scoping. Section 2 gives the concrete formal recommendations,
followed by motivating arguments in Section 3. We then critically
survey the advice given previously in the RFC series and the research
literature (Section 4), referring to an assessment of whether or not
this advice has been followed in production networks (Appendix A).
To wrap up, outstanding issues are discussed that will need
resolution both to inform future protocol designs and to handle
legacy (Section 5). Then security issues are collected together in
Section 6 before conclusions are drawn in Section 8. The interested
reader can find discussion of more detailed issues on the theme of
byte vs. packet in the appendices.
This memo intentionally includes a non-negligible amount of material
on the subject. For the busy reader Section 2 summarises the
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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].
This memo applies to the design of all AQM algorithms, for example,
Random Early Detection (RED) [RFC2309], BLUE [BLUE02], Pre-Congestion
Notification (PCN) [RFC5670], Controlled Delay (CoDel)
[I-D.nichols-tsvwg-codel] and the Proportional Integral controller
Enhanced (PIE) [I-D.pan-tsvwg-pie]. Throughout, RED is used as a
concrete example because it is a widely known and deployed AQM
algorithm. There is no intention to imply that the advice is any
less applicable to the other algorithms, nor that RED is preferred.
Congestion Notification: Congestion notification is a changing
signal that aims to communicate the probability that the network
resource(s) will not be able to forward the level of traffic load
offered (or that there is an impending risk that they will not be
able to).
The `impending risk' qualifier is added, because AQM systems set a
virtual limit smaller than the actual limit to the resource, then
notify when this virtual limit is exceeded in order to avoid
uncontrolled congestion of the actual capacity.
Congestion notification communicates a real number bounded by the
range [ 0 , 1 ]. This ties in with the most well-understood
measure of congestion notification: drop probability.
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.
Examples of packet-congestible resources are route look-up engines
and firewalls, because load depends on how many packet headers
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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 4.1.1).
The path through a machine will typically encounter both packet-
congestible and bit-congestible resources. However, currently, a
design goal of network processing equipment such as routers and
firewalls is to size the packet-processing engine(s) relative to
the lines in order to keep packet processing uncongested even
under worst case packet rates with runs of minimum size packets.
Therefore, packet-congestion is currently rare [RFC6077; S.3.3],
but there is no guarantee that it will not become more common in
future.
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 bits or 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 called respectively "packet-mode queue
measurement" or "byte-mode queue measurement". And whether the
probability of dropping a particular packet is independent or
dependent on its size is called respectively "packet-mode drop" or
"byte-mode drop". The terms byte-mode and packet-mode should not
be used without specifying whether they apply to queue measurement
or to drop.
1.2. Example Comparing Packet-Mode Drop and Byte-Mode Drop
Taking RED as a well-known example algorithm, a central question
addressed by this document is whether to recommend RED's packet-mode
drop variant and to deprecate byte-mode drop. Table 1 compares how
packet-mode and byte-mode drop affect two flows of different size
packets. For each it gives the expected number of packets and of
bits dropped in one second. Each example flow runs at the same bit-
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rate of 48Mb/s, but one is broken up into small 60 byte packets and
the other into large 1500 byte packets.
To keep up the same bit-rate, in one second there are about 25 times
more small packets because they are 25 times smaller. As can be seen
from the table, the packet rate is 100,000 small packets versus 4,000
large packets per second (pps).
Parameter Formula Small packets Large packets
-------------------- -------------- ------------- -------------
Packet size s/8 60B 1,500B
Packet size s 480b 12,000b
Bit-rate x 48Mbps 48Mbps
Packet-rate u = x/s 100kpps 4kpps
Packet-mode Drop
Pkt loss probability p 0.1% 0.1%
Pkt loss-rate p*u 100pps 4pps
Bit loss-rate p*u*s 48kbps 48kbps
Byte-mode Drop MTU, M=12,000b
Pkt loss probability b = p*s/M 0.004% 0.1%
Pkt loss-rate b*u 4pps 4pps
Bit loss-rate b*u*s 1.92kbps 48kbps
Table 1: Example Comparing Packet-mode and Byte-mode Drop
For packet-mode drop, we illustrate the effect of a drop probability
of 0.1%, which the algorithm applies to all packets irrespective of
size. Because there are 25 times more small packets in one second,
it naturally drops 25 times more small packets, that is 100 small
packets but only 4 large packets. But if we count how many bits it
drops, there are 48,000 bits in 100 small packets and 48,000 bits in
4 large packets--the same number of bits of small packets as large.
The packet-mode drop algorithm drops any bit with the same
probability whether the bit is in a small or a large packet.
For byte-mode drop, again we use an example drop probability of 0.1%,
but only for maximum size packets (assuming the link maximum
transmission unit (MTU) is 1,500B or 12,000b). The byte-mode
algorithm reduces the drop probability of smaller packets
proportional to their size, making the probability that it drops a
small packet 25 times smaller at 0.004%. But there are 25 times more
small packets, so dropping them with 25 times lower probability
results in dropping the same number of packets: 4 drops in both
cases. The 4 small dropped packets contain 25 times less bits than
the 4 large dropped packets: 1,920 compared to 48,000.
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The byte-mode drop algorithm drops any bit with a probability
proportionate to the size of the packet it is in.
2. Recommendations
This section gives recommendations related to network equipment in
Sections 2.1 and 2.2, and in Sections 2.3 and 2.4 we discuss the
implications on the transport protocols.
2.1. Recommendation on Queue Measurement
Ideally, an AQM would measure the service time of the queue to
measure congestion of a resource. However service time can only be
measured as packets leave the queue, where it is not always expedient
to implement a full AQM algorithm. To predict the service time as
packets join the queue, an AQM algorithm needs to measure the length
of the queue.
In this case, if the resource is bit-congestible, the AQM
implementation SHOULD measure the length of the queue in bytes and,
if the resource is packet-congestible, the implementation SHOULD
measure the length of the queue in packets. Subject to the
exceptions below, 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. For example, the length of the
queue into a transmission line would be measured in bytes, while the
length of the queue into a firewall would be measured in packets.
To avoid the pathological effects of drop tail, the AQM can then
transform this service time or queue length into the probability of
dropping or marking a packet (e.g. RED's piecewise linear function
between thresholds).
What this advice means for RED as a specific example:
1. A RED implementation SHOULD use byte mode queue measurement for
measuring the congestion of bit-congestible resources and packet
mode queue measurement for packet-congestible resources.
2. An implementation SHOULD NOT make it possible to configure the
way a queue measures itself, because whether a queue is bit-
congestible or packet-congestible is an inherent property of the
queue.
Exceptions to these recommendations might be necessary, for instance
where a packet-congestible resource has to be configured as a proxy
bottleneck for a bit-congestible resource in an adjacent box that
does not support AQM.
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The recommended approach in less straightforward scenarios, such as
fixed size packet buffers, resources without a queue and buffers
comprising a mix of packet and bit-congestible resources, is
discussed in Section 4.1. For instance, Section 4.1.1 explains that
the queue into a line should be measured in bytes even if the queue
consists of fixed-size packet-buffers, because the root-cause of any
congestion is bytes arriving too fast for the line--packets filling
buffers are merely a symptom of the underlying congestion of the
line.
2.2. Recommendation on Encoding Congestion Notification
When encoding congestion notification (e.g. by drop, ECN or PCN), the
probability that network equipment drops or marks a particular packet
to notify congestion SHOULD NOT depend on the size of the packet in
question. As the example in Section 1.2 illustrates, to drop any bit
with probability 0.1% it is only necessary to drop every packet with
probability 0.1% without regard to the size of each packet.
This approach ensures the network layer offers sufficient 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.
What this advice means for RED as a specific example:
1. The RED AQM algorithm SHOULD NOT use byte-mode drop, i.e. it
ought to use packet-mode drop. Byte-mode drop is more complex,
it creates the perverse incentive to fragment segments into tiny
pieces and it is vulnerable to floods of small packets.
2. If a vendor has implemented byte-mode drop, and an operator has
turned it on, it is RECOMMENDED to switch it to packet-mode drop,
after establishing if there are any implications on the relative
performance of applications using different packet sizes. The
unlikely possibility of some application-specific legacy use of
byte-mode drop is the only reason that all the above
recommendations on encoding congestion notification are not
phrased more strongly.
RED as a whole SHOULD NOT be switched off. Without RED, a drop
tail queue biases against large packets and is vulnerable to
floods of small packets.
Note well that RED's byte-mode queue drop is completely orthogonal to
byte-mode queue measurement and should not be confused with it. 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
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fine. However, if in doubt, the vendor should be consulted.
A survey (Appendix A) showed that there appears to be little, if any,
installed base of the byte-mode drop variant of RED. This suggests
that deprecating byte-mode drop will have little, if any, incremental
deployment impact.
2.3. Recommendation on Responding to Congestion
When a transport detects that a packet has been lost or congestion
marked, it SHOULD consider the strength of the congestion indication
as proportionate to the size in octets (bytes) of the missing or
marked packet.
In other words, when a packet indicates congestion (by being lost or
marked) it can be considered conceptually as if there is a congestion
indication on every octet of the packet, not just one indication per
packet.
To be clear, the above recommendation solely describes how a
transport should interpret the meaning of a congestion indication, as
a long term goal. It makes no recommendation on whether a transport
should act differently based on this interpretation. It merely aids
interoperablity between transports, if they choose to make their
actions depend on the strength of congestion indications.
This definition will be useful as the IETF transport area continues
its programme of;
o updating host-based congestion control protocols to take account
of packet size
o making transports less sensitive to losing control packets like
SYNs and pure ACKs.
What this advice means for the case of TCP:
1. If two TCP flows with different packet sizes are required to run
at equal bit rates under the same path conditions, this SHOULD be
done by altering TCP (Section 4.2.2), not network equipment (the
latter affects other transports besides TCP).
2. If it is desired to improve TCP performance by reducing the
chance that a SYN or a pure ACK will be dropped, this SHOULD be
done by modifying TCP (Section 4.2.3), not network equipment.
To be clear, we are not recommending at all that TCPs under
equivalent conditions should aim for equal bit-rates. We are merely
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saying that anyone trying to do such a thing should modify their TCP
algorithm, not the network.
These recommendations are phrased as 'SHOULD' rather than 'MUST',
because there may be cases where expediency dictates that
compatibility with pre-existing versions of a transport protocol make
the recommendations impractical.
2.4. Recommendation on Handling Congestion Indications when Splitting
or Merging Packets
Packets carrying congestion indications may be split or merged in
some circumstances (e.g. at a RTP/RTCP transcoder or during IP
fragment reassembly). Splitting and merging only make sense in the
context of ECN, not loss.
The general rule to follow is that the number of octets in packets
with congestion indications SHOULD be equivalent before and after
merging or splitting. This is based on the principle used above;
that an indication of congestion on a packet can be considered as an
indication of congestion on each octet of the packet.
The above rule is not phrased with the word "MUST" to allow the
following exception. There are cases where pre-existing protocols
were not designed to conserve congestion marked octets (e.g. IP
fragment reassembly [RFC3168] or loss statistics in RTCP receiver
reports [RFC3550] before ECN was added [RFC6679]). When any such
protocol is updated, it SHOULD comply with the above rule to conserve
marked octets. However, the rule may be relaxed if it would
otherwise become too complex to interoperate with pre-existing
implementations of the protocol.
One can think of a splitting or merging process as if all the
incoming congestion-marked octets increment a counter and all the
outgoing marked octets decrement the same counter. In order to
ensure that congestion indications remain timely, even the smallest
positive remainder in the conceptual counter should trigger the next
outgoing packet to be marked (causing the counter to go negative).
3. Motivating Arguments
This section is informative. It justifies the recommendations given
in the previous section.
3.1. 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
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the protocol exchange perverse incentives [Evol_cc][RFC3426]. Given
there are many good reasons why larger path maximum transmission
units (PMTUs) would help solve a number of scaling issues, we do not
want to create any bias against large packets that is greater than
their true cost.
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 both the following cases:
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.
Non-malicious transports: Even if an application designer is not
actually malicious, if over time it is noticed that small packets
tend to go faster, designers will act in their own interest and
use smaller packets. Queues that give advantage to small packets
create an evolutionary pressure for applications or 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 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, RED aims
to drop 25x more of the larger packets than the smaller ones. Thus,
for example 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; Appx 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 750kbps for the flow of larger packets but
990kbps for the smaller packets (because of rate variations it will
actually be a little less than this target).
Note that, although the byte-mode drop variant of RED amplifies small
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packet attacks, drop-tail queues amplify small packet attacks even
more (see Security Considerations in Section 6). Wherever possible
neither should be used.
3.2. Small != Control
Dropping fewer control packets considerably improves performance. It
is tempting to drop small packets with lower probability in order to
improve performance, because many control packets tend to be smaller
(TCP SYNs & ACKs, DNS queries & responses, SIP messages, HTTP GETs,
etc). 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, rather than fix these problems in the network, 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 4.2.3).
3.3. Transport-Independent Network
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.
To counter this effect it seems tempting not to follow our
recommendation, and instead for the network to bias congestion
notification by packet size in order to equalise the bit-rates of
flows with different packet sizes. However, in order to do this, the
queuing algorithm has to make assumptions about the transport, which
become embedded in the network. Specifically:
o The queuing algorithm has to assume how aggressively the transport
will respond to congestion (see Section 4.2.4). If the network
assumes the transport responds as aggressively as TCP NewReno, it
will be wrong for Compound TCP and differently wrong for Cubic
TCP, etc. To achieve equal bit-rates, each transport then has to
guess what assumption the network made, and work out how to
replace this assumed aggressiveness with its own aggressiveness.
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o Also, if the network biases congestion notification by packet size
it has to assume a baseline packet size--all proposed algorithms
use the local MTU (for example see the byte-mode loss probability
formula in Table 1). Then if the non-Reno transports mentioned
above are trying to reverse engineer what the network assumed,
they also have to guess the MTU of the congested link.
Even though reducing the drop probability of small packets (e.g.
RED's byte-mode drop) helps ensure TCP flows with different packet
sizes will achieve similar bit rates, we argue this correction should
be made to any future transport protocols based on TCP, not to the
network in order to fix one transport, no matter how predominant it
is. Effectively, favouring small packets is reverse engineering of
network equipment around one particular transport protocol (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?"
In contrast, if the network never takes account of packet size, the
transport can be certain it will never need to guess any assumptions
the network has made. And the network passes two pieces of
information to the transport that are sufficient in all cases: i)
congestion notification on the packet and ii) the size of the packet.
Both are available for the transport to combine (by taking account of
packet size when responding to congestion) or not. Appendix B checks
that these two pieces of information are sufficient for all relevant
scenarios.
When the network does not take account of packet size, it allows
transport protocols to choose whether to take account of packet size
or not. However, if the network were to bias congestion notification
by packet size, transport protocols would have no choice; those that
did not take account of packet size themselves would unwittingly
become dependent on packet size, and those that already took account
of packet size would end up taking account of it twice.
3.4. Partial Deployment of AQM
In overview, the argument in this section runs as follows:
o Because the network does not and cannot always drop packets in
proportion to their size, it shouldn't be given the task of making
drop signals depend on packet size at all.
o Transports on the other hand don't always want to make their rate
response proportional to the size of dropped packets, but if they
want to, they always can.
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The argument is similar to the end-to-end argument that says "Don't
do X in the network if end-systems can do X by themselves, and they
want to be able to choose whether to do X anyway." Actually the
following argument is stronger; in addition it says "Don't give the
network task X that could be done by the end-systems, if X is not
deployed on all network nodes, and end-systems won't be able to tell
whether their network is doing X, or whether they need to do X
themselves." In this case, the X in question is "making the response
to congestion depend on packet size".
We will now re-run this argument taking each step in more depth. The
argument applies solely to drop, not to ECN marking.
A queue drops packets for either of two reasons: a) to signal to host
congestion controls that they should reduce the load and b) because
there is no buffer left to store the packets. Active queue
management tries to use drops as a signal for hosts to slow down
(case a) so that drop due to buffer exhaustion (case b) should not be
necessary.
AQM is not universally deployed in every queue in the Internet; many
cheap Ethernet bridges, software firewalls, NATs on consumer devices,
etc implement simple tail-drop buffers. Even if AQM were universal,
it has to be able to cope with buffer exhaustion (by switching to a
behaviour like tail-drop), in order to cope with unresponsive or
excessive transports. For these reasons networks will sometimes be
dropping packets as a last resort (case b) rather than under AQM
control (case a).
When buffers are exhausted (case b), they don't naturally drop
packets in proportion to their size. The network can only reduce the
probability of dropping smaller packets if it has enough space to
store them somewhere while it waits for a larger packet that it can
drop. If the buffer is exhausted, it does not have this choice.
Admittedly tail-drop does naturally drop somewhat fewer small
packets, but exactly how few depends more on the mix of sizes than
the size of the packet in question. Nonetheless, in general, if we
wanted networks to do size-dependent drop, we would need universal
deployment of (packet-size dependent) AQM code, which is currently
unrealistic.
A host transport cannot know whether any particular drop was a
deliberate signal from an AQM or a sign of a queue shedding packets
due to buffer exhaustion. Therefore, because the network cannot
universally do size-dependent drop, it should not do it all.
Whereas universality is desirable in the network, diversity is
desirable between different transport layer protocols - some, like
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NewReno TCP [RFC5681], may not choose to make their rate response
proportionate to the size of each dropped packet, while others will
(e.g. TFRC-SP [RFC4828]).
3.5. Implementation Efficiency
Biasing against large packets typically requires an extra multiply
and divide in the network (see the example byte-mode drop formula in
Table 1). 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. Also
the work to do the biasing is spread over many hosts, rather than
concentrated in just the congested network element. These aren't
principled reasons in themselves, but they are a happy consequence of
the other principled reasons.
4. A Survey and Critique of Past Advice
This section is informative, not normative.
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. A later addendum to this email introduced the
insight that there are in fact two orthogonal choices:
o whether to measure queue length in bytes or packets (Section 4.1)
o whether the drop probability of an individual packet should depend
on its own size (Section 4.2).
The rest of this section is structured accordingly.
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4.1. Congestion Measurement Advice
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
[pktByteEmail].
Congestion in some legacy bit-congestible buffers is only measured in
packets not bytes. In such cases, the operator has to 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 under-sensitive to high
proportions of large packets. However, there is no need to make
allowances for the possibility of such legacy in future protocol
design. This is safe because any under-sensitivity during unusual
traffic mixes cannot lead to congestion collapse given the buffer
will eventually revert to tail drop, discarding proportionately more
large packets.
4.1.1. Fixed Size Packet Buffers
The question of whether to measure queues in bytes or packets seems
to be well understood. However, measuring congestion is confusing
when the resource is bit congestible but the queue into the resource
is packet congestible. This section outlines the approach to take.
Some, mostly older, queuing hardware allocates fixed sized buffers in
which to store each packet in the queue. This hardware forwards to
the line in one of two ways:
o With some hardware, any fixed sized buffers not completely filled
by a packet are padded when transmitted to the wire. This case,
should clearly be treated as packet-congestible, because both
queuing and transmission are in fixed MTU-sized units. Therefore
the queue length in packets is a good model of congestion of the
link.
o More commonly, hardware with fixed size packet buffers transmits
packets to line without padding. This implies a hybrid forwarding
system with transmission congestion dependent on the size of
packets but queue congestion dependent on the number of packets,
irrespective of their size.
Nonetheless, there would be no queue at all unless the line had
become congested--the root-cause of any congestion is too many
bytes arriving for the line. Therefore, the AQM should measure
the queue length as the sum of all the packet sizes in bytes that
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are queued up waiting to be serviced by the line, irrespective of
whether each packet is held in a fixed size buffer.
In the (unlikely) first case where use of padding means the queue
should be measured in packets, further confusion is likely because
the fixed buffers are rarely all one 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, there is no need to consider all this
complexity, because the root-cause of any congestion is still line
overload--buffer consumption is only the symptom. Therefore, the
length of the queue should be measured as the sum of the bytes in the
queue that will be transmitted to line, including any padding. In
the (unusual) case of transmission with padding this means the sum of
the sizes of the small buffers queued plus the sum of the sizes of
the large buffers queued.
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 4.2.1. But here we can repeat the simple rule
for how to measure the length of queues of fixed buffers: no matter
how complicated the buffering scheme is, ultimately a transmission
line is nearly always bit-congestible so the number of bytes queued
up waiting for the line measures how congested the line is, and it is
rarely important to measure how congested the buffering system is.
4.1.2. 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, power
limited resources are 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.
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In wireless protocols that use request to send / clear to send (RTS /
CTS) control, such as some variants of IEEE802.11, it is reasonable
to base an AQM on the time spent waiting for transmission
opportunities (TXOPs) even though wireless spectrum is usually
regarded as congested by bits (for a given coding scheme). This is
because requests for TXOPs queue up as the spectrum gets congested by
all the bits being transferred. So the time that TXOPs are queued
directly reflects bit congestion of the spectrum.
4.2. Congestion Notification Advice
4.2.1. Network Bias when Encoding
4.2.1.1. Advice on Packet Size Bias in RED
The previously mentioned email [pktByteEmail] referred to by
[RFC2309] advised that most scarce resources in the Internet were
bit-congestible, which is still believed to be true (Section 1.1).
But it went on to offer advice that is updated by this memo. 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. 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]. However, 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 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
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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. This memo makes a different
recommendations in Section 2.
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 in the literature for this
change, however Eddy and Allman conducted experiments [REDbias] that
assessed how sensitive RED was to this parameter, amongst other
things. However, this changed algorithm can often lead to drop
probabilities of greater than 1 (which gives a hint that there is
probably 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. It seems unlikely that byte-mode drop
has ever been implemented in production networks (Appendix A),
therefore any conclusions based on ns2 simulations that use RED
without disabling byte-mode drop are likely to behave very
differently from RED in production networks.
4.2.1.2. Packet Size Bias Regardless of AQM
The byte-mode drop variant of RED (or a similar variant of other AQM
algorithms) is not the only possible bias towards small packets in
queueing systems. 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 (see
Section 4.1.1). 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. 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,
fixed size packet-buffers should be augmented with a good AQM
algorithm and packet-mode drop. If an AQM 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.
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4.2.2. 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.
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 [RFC5348] 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
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 Appendix A). Incidentally, VP-TFRC also proposed that byte-mode
RED dropping should really square the packet-size compensation-factor
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(like that of Cnodder's RED_5, but apparently unaware of it).
Pre-congestion notification [RFC5670] is an IETF technology 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 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, as
recommended here.
4.2.3. 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 two 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 there are no known proposals, it would also be possible and
perfectly valid to make control packets robust against drop by
requesting a scheduling class with lower drop probability, by re-
marking to a Diffserv code point [RFC2474] within the same behaviour
aggregate.
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. A similar but more extensive approach has
been evaluated on Google servers [GentleAggro].
The proposals discussed in this sub-section are experimental
approaches that are not yet in wide operational use, but they are
existence proofs that transports can make themselves robust against
loss of control packets. The examples are all TCP-based, but
applications over non-TCP transports could mitigate loss of control
packets by making similar use of Diffserv, data duplication, FEC etc.
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4.2.4. Congestion Notification: Summary of Conflicting Advice
+-----------+----------------+-----------------+--------------------+
| 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 2: Dependence of flow bit-rate per RTT on packet size, s, and
drop probability, p, when network and/or transport bias towards small
packets to varying degrees
Table 2 aims to summarise the potential effects of all the advice
from different sources. 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 [RFC5348] on the top row with TFRC-SP [RFC4828]
below. Each cell shows how the bits per round trip of a flow depends
on packet size, s, and drop probability, p. In order to declutter
the formulae to focus on packet-size dependence they are all given
per round trip, which removes any RTT term.
Let us assume that the goal is for the bit-rate of a flow to be
independent of packet size. Suppressing all inessential details, the
table shows that this should either be achievable by not altering the
TCP transport in a RED_5 network, or using the small packet TFRC-SP
transport (or similar) 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 subset
of queues that congest, each path through the network will present a
different hybrid scenario to its transport.
Whatever, we can see that the linear byte-mode drop column in the
middle would considerably complicate 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. Section 2
recommends that _all_ bias in network equipment 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).
In practice it seems that no deliberate bias towards small packets
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has been implemented for production networks. Of the 19% of vendors
who responded to a survey of 84 equipment vendors, none had
implemented byte-mode drop in RED (see Appendix A for details).
5. Outstanding Issues and Next Steps
5.1. Bit-congestible Network
For a connectionless network with nearly all resources being bit-
congestible the recommended position is clear--that the network
should not make allowance for packet sizes and the transport should.
This leaves two outstanding issues:
o How to handle any legacy of AQM with byte-mode drop already
deployed;
o The need to start a programme to update transport congestion
control protocol standards to take account of packet size.
A survey of equipment vendors (Section 4.2.4) found no evidence that
byte-mode packet drop had been implemented, so deployment will be
sparse at best. A migration strategy is not really needed to remove
an algorithm that may not even be deployed.
A programme of experimental updates to take account of packet size in
transport congestion control protocols has already started with
TFRC-SP [RFC4828].
5.2. Bit- & Packet-congestible Network
The position is much less clear-cut if the Internet becomes populated
by a more even mix of both packet-congestible and bit-congestible
resources (see Appendix B.2). This problem is not pressing, because
most Internet resources are designed to be bit-congestible before
packet processing starts to congest (see Section 1.1).
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 [RFC6077]). The research question of whether packet
congestion might become common and what to do if it does may in the
future be explored in the IRTF (the "Challenge 3: Packet Size" in
[RFC6077]).
Note that sometimes it seems that resources might be congested by
neither bits nor packets, e.g. where the queue for access to a
wireless medium is in units of transmission opportunities. However,
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the root cause of congestion of the underlying spectrum is overload
of bits (see Section 4.1.2).
6. Security Considerations
This memo recommends that queues do not bias drop probability due to
packets size. For instance dropping small packets less often than
large 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.
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.
7. IANA Considerations
This document has no actions for IANA.
8. Conclusions
This memo identifies the three distinct stages of the congestion
notification process where implementations need to decide whether to
take packet size into account. The recommendations provided in
Section 2 of this memo are different in each case:
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o When network equipment measures the length of a queue, if it is
not feasible to use time it is recommended to count in bytes if
the network resource is congested by bytes, or to count in packets
if is congested by packets.
o When network equipment decides whether to drop (or mark) a packet,
it is recommended that the size of the particular packet should
not be taken into account
o However, when a transport algorithm responds to a dropped or
marked packet, the size of the rate reduction should be
proportionate to the size of the packet.
In summary, the answers are 'it depends', 'no' and 'yes' respectively
For the specific case of RED, this means that byte-mode queue
measurement will often be appropriate but the use of byte-mode drop
is very strongly discouraged.
At the transport layer the IETF should continue updating congestion
control protocols to take account of the size of each packet that
indicates congestion. Also the IETF should continue to make
protocols less sensitive to losing control packets like SYNs, pure
ACKs and DNS exchanges. Although many control packets happen to be
small, the alternative of network equipment favouring all small
packets would be dangerous. That would create perverse incentives to
split data transfers into smaller packets.
The memo develops these recommendations from principled arguments
concerning scaling, layering, incentives, inherent efficiency,
security and policeability. But it also addresses practical issues
such as specific buffer architectures and incremental deployment.
Indeed a limited survey of RED implementations is discussed, which
shows there appears to be little, if any, installed base of RED's
byte-mode drop. Therefore it can be deprecated with little, if any,
incremental deployment complications.
The recommendations have been developed on the well-founded basis
that most Internet resources are bit-congestible not packet-
congestible. We need to know the likelihood that this assumption
will prevail longer term and, if it might not, what protocol changes
will be needed to cater for a mix of the two. The IRTF Internet
Congestion Control Research Group (ICCRG) is currently working on
these problems [RFC6077].
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9. Acknowledgements
Thank you to Sally Floyd, who gave extensive and useful review
comments. Also thanks for the reviews from Philip Eardley, David
Black, Fred Baker, David Taht, Toby Moncaster, Arnaud Jacquet and
Mirja Kuehlewind as well as helpful explanations of different
hardware approaches from Larry Dunn and Fred Baker. We are 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 were 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.
10. 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.
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to
Indicate Requirement Levels", BCP 14,
RFC 2119, March 1997.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black,
"The Addition of Explicit Congestion
Notification (ECN) to IP", RFC 3168,
September 2001.
11.2. Informative References
[BLUE02] Feng, W-c., Shin, K., Kandlur, D., and D.
Saha, "The BLUE active queue management
algorithms", IEEE/ACM Transactions on
Networking 10(4) 513--528, August 2002, <h
ttp://dx.doi.org/10.1109/
TNET.2002.801399>.
[CCvarPktSize] Widmer, J., Boutremans, C., and J-Y. Le
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Boudec, "Congestion Control for Flows with
Variable Packet Size", ACM CCR 34(2) 137--
151, 2004,
<http://doi.acm.org/10.1145/
997150.997162>.
[CHOKe_Var_Pkt] Psounis, K., Pan, R., and B. Prabhaker,
"Approximate Fair Dropping for Variable
Length Packets", IEEE Micro 21(1):48--56,
January-February 2001, <http://
www.stanford.edu/~balaji/papers/
01approximatefair.pdf}>.
[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",
Philosphical Transactions of the Royal
Society A 366(1872):1941-1953, June 2008,
<http://rsta.royalsocietypublishing.org/
content/366/1872/1941.full.pdf+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>.
[GentleAggro] Flach, T., Dukkipati, N., Terzis, A.,
Raghavan, B., Cardwell, N., Cheng, Y.,
Jain, A., Hao, S., Katz-Bassett, E., and
R. Govindan, "Reducing Web Latency: the
Virtue of Gentle Aggression", ACM SIGCOMM
CCR 43(4)159--170, August 2013, <http://
doi.acm.org/10.1145/2486001.2486014>.
[I-D.nichols-tsvwg-codel] Nichols, K. and V. Jacobson, "Controlled
Delay Active Queue Management",
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draft-nichols-tsvwg-codel-01 (work in
progress), February 2013.
[I-D.pan-tsvwg-pie] Pan, R., Natarajan, P., Piglione, C., and
M. Prabhu, "PIE: A Lightweight Control
Scheme To Address the Bufferbloat
Problem", draft-pan-tsvwg-pie-00 (work in
progress), December 2012.
[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.
[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, <ht
tp://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>.
[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",
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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.
[RFC2914] Floyd, S., "Congestion Control
Principles", BCP 41, RFC 2914,
September 2000.
[RFC3426] Floyd, S., "General Architectural and
Policy Considerations", RFC 3426,
November 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick,
R., and V. Jacobson, "RTP: A Transport
Protocol for Real-Time Applications",
STD 64, RFC 3550, July 2003.
[RFC3714] Floyd, S. and J. Kempf, "IAB Concerns
Regarding Congestion Control for Voice
Traffic in the Internet", RFC 3714,
March 2004.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly
Rate Control (TFRC): The Small-Packet (SP)
Variant", RFC 4828, April 2007.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J.
Widmer, "TCP Friendly Rate Control (TFRC):
Protocol Specification", RFC 5348,
September 2008.
[RFC5562] Kuzmanovic, A., Mondal, A., Floyd, S., and
K. Ramakrishnan, "Adding Explicit
Congestion Notification (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.
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[RFC5690] Floyd, S., Arcia, A., Ros, D., and J.
Iyengar, "Adding Acknowledgement
Congestion Control to TCP", RFC 5690,
February 2010.
[RFC6077] Papadimitriou, D., Welzl, M., Scharf, M.,
and B. Briscoe, "Open Research Issues in
Internet Congestion Control", RFC 6077,
February 2011.
[RFC6679] Westerlund, M., Johansson, I., Perkins,
C., O'Hanlon, P., and K. Carlberg,
"Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012.
[RFC6789] Briscoe, B., Woundy, R., and A. Cooper,
"Congestion Exposure (ConEx) Concepts and
Use Cases", RFC 6789, December 2012.
[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>.
[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, <ht
tp://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>.
Appendix A. Survey of RED Implementation Status
This Appendix is informative, not normative.
In May 2007 a survey was conducted of 84 vendors to assess how widely
drop probability based on packet size has been implemented in RED
Table 3. About 19% of those surveyed replied, giving a sample size
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of 16. Although in most cases we do not have permission to identify
the respondents, we can say that those that have responded include
most of the larger equipment vendors, covering a large fraction of
the market. The two who gave permission to be identified were Cisco
and Alcatel-Lucent. The others 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. All those who responded 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). At the time the survey was conducted, Linux did not
implement RED with packet-size bias of drop, although we have not
investigated a wider range of open source code.
+-------------------------------+----------------+-----------------+
| 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 3: 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.
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 many tail-drop queues are bound to
still exist, 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 4.2.1, 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.
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Appendix B. Sufficiency of Packet-Mode Drop
This Appendix is informative, not normative.
Here we check that packet-mode drop (or marking) in the network gives
sufficiently generic information for the transport layer to use. We
check against a 2x2 matrix of four scenarios that may occur now or in
the future (Table 4). The horizontal and vertical dimensions have
been chosen because each tests extremes of sensitivity to packet size
in the transport and in the network respectively.
Note that this section does not consider byte-mode drop at all.
Having deprecated byte-mode drop, the goal here is to check that
packet-mode drop will be sufficient in all cases.
+-------------------------------+-----------------+-----------------+
| Transport | a) Independent | b) Dependent on |
| | of packet size | packet size of |
| Network | of congestion | congestion |
| | notifications | notifications |
+-------------------------------+-----------------+-----------------+
| 1) Predominantly | Scenario a1) | Scenario b1) |
| bit-congestible network | | |
| 2) Mix of bit-congestible and | Scenario a2) | Scenario b2) |
| pkt-congestible network | | |
+-------------------------------+-----------------+-----------------+
Table 4: Four Possible Congestion Scenarios
Appendix B.1 focuses on the horizontal dimension of Table 4 checking
that packet-mode drop (or marking) gives sufficient information,
whether or not the transport uses it--scenarios b) and a)
respectively.
Appendix B.2 focuses on the vertical dimension of Table 4, checking
that packet-mode drop gives sufficient information to the transport
whether resources in the network are bit-congestible or packet-
congestible (these terms are defined in Section 1.1).
Notation: To be concrete, we will compare two flows with different
packet sizes, s_1 and s_2. As an example, we will take s_1 = 60B
= 480b and s_2 = 1500B = 12,000b.
A flow's bit rate, x [bps], is related to its packet rate, u
[pps], by
x(t) = s.u(t).
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In the bit-congestible case, path congestion will be denoted by
p_b, and in the packet-congestible case by p_p. When either case
is implied, the letter p alone will denote path congestion.
B.1. Packet-Size (In)Dependence in Transports
In all cases we consider a packet-mode drop queue that indicates
congestion by dropping (or marking) packets with probability p
irrespective of packet size. We use an example value of loss
(marking) probability, p=0.1%.
A transport like RFC5681 TCP treats a congestion notification on any
packet whatever its size as one event. However, a network with just
the packet-mode drop algorithm does give more information if the
transport chooses to use it. We will use Table 5 to illustrate this.
We will set aside the last column until later. The columns labelled
"Flow 1" and "Flow 2" compare two flows consisting of 60B and 1500B
packets respectively. The body of the table considers two separate
cases, one where the flows have equal bit-rate and the other with
equal packet-rates. In both cases, the two flows fill a 96Mbps link.
Therefore, in the equal bit-rate case they each have half the bit-
rate (48Mbps). Whereas, with equal packet-rates, flow 1 uses 25
times smaller packets so it gets 25 times less bit-rate--it only gets
1/(1+25) of the link capacity (96Mbps/26 = 4Mbps after rounding). In
contrast flow 2 gets 25 times more bit-rate (92Mbps) in the equal
packet rate case because its packets are 25 times larger. The packet
rate shown for each flow could easily be derived once the bit-rate
was known by dividing bit-rate by packet size, as shown in the column
labelled "Formula".
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Parameter Formula Flow 1 Flow 2 Combined
----------------------- ----------- ------- ------- --------
Packet size s/8 60B 1,500B (Mix)
Packet size s 480b 12,000b (Mix)
Pkt loss probability p 0.1% 0.1% 0.1%
EQUAL BIT-RATE CASE
Bit-rate x 48Mbps 48Mbps 96Mbps
Packet-rate u = x/s 100kpps 4kpps 104kpps
Absolute pkt-loss-rate p*u 100pps 4pps 104pps
Absolute bit-loss-rate p*u*s 48kbps 48kbps 96kbps
Ratio of lost/sent pkts p*u/u 0.1% 0.1% 0.1%
Ratio of lost/sent bits p*u*s/(u*s) 0.1% 0.1% 0.1%
EQUAL PACKET-RATE CASE
Bit-rate x 4Mbps 92Mbps 96Mbps
Packet-rate u = x/s 8kpps 8kpps 15kpps
Absolute pkt-loss-rate p*u 8pps 8pps 15pps
Absolute bit-loss-rate p*u*s 4kbps 92kbps 96kbps
Ratio of lost/sent pkts p*u/u 0.1% 0.1% 0.1%
Ratio of lost/sent bits p*u*s/(u*s) 0.1% 0.1% 0.1%
Table 5: Absolute Loss Rates and Loss Ratios for Flows of Small and
Large Packets and Both Combined
So far we have merely set up the scenarios. We now consider
congestion notification in the scenario. Two TCP flows with the same
round trip time aim to equalise their packet-loss-rates over time.
That is the number of packets lost in a second, which is the packets
per second (u) multiplied by the probability that each one is dropped
(p). Thus TCP converges on the "Equal packet-rate" case, where both
flows aim for the same "Absolute packet-loss-rate" (both 8pps in the
table).
Packet-mode drop actually gives flows sufficient information to
measure their loss-rate in bits per second, if they choose, not just
packets per second. Each flow can count the size of a lost or marked
packet and scale its rate-response in proportion (as TFRC-SP does).
The result is shown in the row entitled "Absolute bit-loss-rate",
where the bits lost in a second is the packets per second (u)
multiplied by the probability of losing a packet (p) multiplied by
the packet size (s). Such an algorithm would try to remove any
imbalance in bit-loss-rate such as the wide disparity in the "Equal
packet-rate" case (4kbps vs. 92kbps). Instead, a packet-size-
dependent algorithm would aim for equal bit-loss-rates, which would
drive both flows towards the "Equal bit-rate" case, by driving them
to equal bit-loss-rates (both 48kbps in this example).
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The explanation so far has assumed that each flow consists of packets
of only one constant size. Nonetheless, it extends naturally to
flows with mixed packet sizes. In the right-most column of Table 5 a
flow of mixed size packets is created simply by considering flow 1
and flow 2 as a single aggregated flow. There is no need for a flow
to maintain an average packet size. It is only necessary for the
transport to scale its response to each congestion indication by the
size of each individual lost (or marked) packet. Taking for example
the "Equal packet-rate" case, in one second about 8 small packets and
8 large packets are lost (making closer to 15 than 16 losses per
second due to rounding). If the transport multiplies each loss by
its size, in one second it responds to 8*480b and 8*12,000b lost
bits, adding up to 96,000 lost bits in a second. This double checks
correctly, being the same as 0.1% of the total bit-rate of 96Mbps.
For completeness, the formula for absolute bit-loss-rate is p(u1*s1+
u2*s2).
Incidentally, a transport will always measure the loss probability
the same irrespective of whether it measures in packets or in bytes.
In other words, the ratio of lost to sent packets will be the same as
the ratio of lost to sent bytes. (This is why TCP's bit rate is
still proportional to packet size even when byte-counting is used, as
recommended for TCP in [RFC5681], mainly for orthogonal security
reasons.) This is intuitively obvious by comparing two example
flows; one with 60B packets, the other with 1500B packets. If both
flows pass through a queue with drop probability 0.1%, each flow will
lose 1 in 1,000 packets. In the stream of 60B packets the ratio of
bytes lost to sent will be 60B in every 60,000B; and in the stream of
1500B packets, the loss ratio will be 1,500B out of 1,500,000B. When
the transport responds to the ratio of lost to sent packets, it will
measure the same ratio whether it measures in packets or bytes: 0.1%
in both cases. The fact that this ratio is the same whether measured
in packets or bytes can be seen in Table 5, where the ratio of lost
to sent packets and the ratio of lost to sent bytes is always 0.1% in
all cases (recall that the scenario was set up with p=0.1%).
This discussion of how the ratio can be measured in packets or bytes
is only raised here to highlight that it is irrelevant to this memo!
Whether a transport depends on packet size or not depends on how this
ratio is used within the congestion control algorithm.
So far we have shown that packet-mode drop passes sufficient
information to the transport layer so that the transport can take
account of bit-congestion, by using the sizes of the packets that
indicate congestion. We have also shown that the transport can
choose not to take packet size into account if it wishes. We will
now consider whether the transport can know which to do.
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B.2. Bit-Congestible and Packet-Congestible Indications
As a thought-experiment, imagine an idealised congestion notification
protocol that supports both bit-congestible and packet-congestible
resources. It would require at least two ECN flags, one for each of
bit-congestible and packet-congestible resources.
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.
This hides a fundamental problem--much more fundamental than whether
we can magically create header space for yet another ECN flag, or
whether it would work while being deployed incrementally.
Distinguishing drop from delivery naturally provides just one
implicit bit of congestion indication information--the packet is
either dropped or not. 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.
This problem would not be solved even if ECN were universally
deployed. 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.
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 a conceptual device to illustrate a dilemma we could face in the
future. Section 3 gives four good reasons why it would be a bad idea
to allow for packet size by biasing drop probability in favour of
small packets within the network. The impracticality of our thought
experiment shows that it will be hard to give transports a practical
way to know whether to take account of the size of congestion
indication packets or not.
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Fortunately, this dilemma is not pressing because by design most
equipment becomes bit-congested before its packet-processing becomes
congested (as already outlined in Section 1.1). Therefore transports
can be designed on the relatively sound assumption that a congestion
indication will usually imply bit-congestion.
Nonetheless, although the above idealised protocol isn't intended for
implementation, we do want to emphasise that research is needed to
predict whether there are good reasons to believe that packet
congestion might become more common, and if so, to find a way to
somehow distinguish between bit and packet congestion [RFC3714].
Recently, the dual resource queue (DRQ) proposal [DRQ] has been made
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 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).
Finally, we note one further complication. Strictly, packet-
congestible resources are often cycle-congestible. For instance, for
routing look-ups load depends on the complexity of each look-up and
whether the pattern of arrivals is amenable to caching or not. This
also 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.
Appendix C. Byte-mode Drop Complicates Policing Congestion Response
This section is informative, not normative.
There are two main classes of approach to policing congestion
response: i) policing at each bottleneck link or ii) policing at the
edges of networks. Packet-mode drop in RED is compatible with
either, while byte-mode drop precludes edge policing.
The simplicity of an edge policer relies on one dropped or marked
packet being equivalent to another of the same size without having to
know which link the drop or mark occurred at. However, the byte-mode
drop algorithm has to depend on the local MTU of the line--it needs
to use some concept of a 'normal' packet size. Therefore, one
dropped or marked packet from a byte-mode drop algorithm is not
necessarily equivalent to another from a different link. A policing
function local to the link can know the local MTU where the
congestion occurred. However, a policer at the edge of the network
cannot, at least not without a lot of complexity.
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The early research proposals for type (i) policing at a bottleneck
link [pBox] used byte-mode drop, then detected flows that contributed
disproportionately to the number of packets dropped. However, with
no extra complexity, later proposals used packet mode drop and looked
for flows that contributed a disproportionate amount of dropped bytes
[CHOKe_Var_Pkt].
Work is progressing on the congestion exposure protocol (ConEx
[RFC6789]), which enables a type (ii) edge policer located at a
user's attachment point. The idea is to be able to take an
integrated view of the effect of all a user's traffic on any link in
the internetwork. However, byte-mode drop would effectively preclude
such edge policing because of the MTU issue above.
Indeed, making drop probability depend on the size of the packets
that bits happen to be divided into would simply encourage the bits
to be divided into smaller packets in order to confuse policing. In
contrast, as long as a dropped/marked packet is 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].
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://tools.ietf.org/wg/tsvwg/draft-ietf-tsvwg-byte-pkt-congest/>
(courtesy of the rfcdiff tool):
From -11 to -12: Following the second pass through the IESG:
* Section 2.1 [Barry Leiba]:
+ s/No other choice makes sense,/Subject to the exceptions
below, no other choice makes sense,/
+ s/Exceptions to these recommendations MAY be necessary
/Exceptions to these recommendations may be necessary /
* Sections 3.2 and 4.2.3 [Joel Jaeggli]:
+ Added comment to section 4.2.3 that the examples given are
not in widespread production use, but they give evidence
that it is possible to follow the advice given.
+ Section 4.2.3:
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- OLD: Although there are no known proposals, 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.
NEW: Although there are no known proposals, it would also
be possible and perfectly valid to make control packets
robust against drop by requesting a scheduling class with
lower drop probability, by re-marking to a Diffserv code
point [RFC2474] within the same behaviour aggregate.
- appended "Similarly applications, over non-TCP transports
could make any packets that are effectively control
packets more robust by using Diffserv, data duplication,
FEC etc."
+ Updated Wischik ref and added "Reducing Web Latency: the
Virtue of Gentle Aggression" ref.
* Expanded more abbreviations (CoDel, PIE, MTU).
* Section 1. Intro [Stephen Farrell]:
+ In the places where the doc desribes the dichotomy between
'long-term goal' and 'expediency' the words long term goal
and expedient have been introduced, to more explicitly refer
back to this introductory para (S.2.1 & S.2.3).
+ Added explanation of what scaling with packet size means.
* Conclusions [Benoit Claise]:
+ OLD: For the specific case of RED, this means that byte-mode
queue measurement will often be appropriate although byte-
mode drop is strongly deprecated.
NEW: For the specific case of RED, this means that byte-mode
queue measurement will often be appropriate but the use of
byte-mode drop is very strongly discouraged.
From -10 to -11: Following a further WGLC:
* Abstract: clarified that advice applies to all AQMs including
newer ones
* Abstract & Intro: changed 'read' to 'detect', because you don't
read losses, you detect them.
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* S.1. Introduction: Disambiguated summary of advice on queue
measurement.
* Clarified that the doc deprecates any preference based solely
on packet size, it's not only against preferring smaller
packets.
* S.4.1.2. Congestion Measurement without a Queue: Explained
that a queue of TXOPs represents a queue into spectrum
congested by too many bits.
* S.5.2: Bit- & Packet-congestible Network: Referred to
explanation in S.4.1.2 to make the point that TXOPs are not a
primary unit of workload like bits and packets are, even though
you get queues of TXOPs.
* 6. Security: Disambiguated 'bias towards'.
* 8. Conclusions: Made consistent with recommendation to use
time if possible for queue measurement.
From -09 to -10: Following IESG review:
* Updates 2309: Left header unchanged reflecting eventual IESG
consensus [Sean Turner, Pete Resnick].
* S.1 Intro: This memo adds to the congestion control principles
enumerated in BCP 41 [Pete Resnick]
* Abstract, S.1, S.1.1, s.1.2 Intro, Scoping and Example: Made
applicability to all AQMs clearer listing some more example
AQMs and explained that we always use RED for examples, but
this doesn't mean it's not applicable to other AQMs. [A number
of reviewers have described the draft as "about RED"]
* S.1 & S.2.1 Queue measurement: Explained that the choice
between measuring the queue in packets or bytes is only
relevant if measuring it in time units is infeasible [So as not
to imply that we haven't noticed the advances made by PDPC &
CoDel]
* S.1.1. Terminology: Better explained why hybrid systems
congested by both packets and bytes are often designed to be
treated as bit-congestible [Richard Barnes].
* S.2.1. Queue measurement advice: Added examples. Added a
counter-example to justify SHOULDs rather than MUSTs. Pointed
to S.4.1 for a list of more complicated scenarios. [Benson
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Schliesser, OpsDir]
* S2.2. Recommendation on Encoding Congestion Notification:
Removed SHOULD treat packets equally, leaving only SHOULD NOT
drop dependent on packet size, to avoid it sounding like we're
saying QoS is not allowed. Pointed to possible app-specific
legacy use of byte-mode as a counter-example that prevents us
saying MUST NOT. [Pete Resnick]
* S.2.3. Recommendation on Responding to Congestion: capitalised
the two SHOULDs in recommendations for TCP, and gave possible
counter-examples. [noticed while dealing with Pete Resnick's
point]
* S2.4. Splitting & Merging: RTCP -> RTP/RTCP [Pete McCann, Gen-
ART]
* S.3.2 Small != Control: many control packets are small ->
...tend to be small [Stephen Farrell]
* S.3.1 Perverse incentives: Changed transport designers to app
developers [Stephen Farrell]
* S.4.1.1. Fixed Size Packet Buffers: Nearly completely re-
written to simplify and to reverse the advice when the
underlying resource is bit-congestible, irrespective of whether
the buffer consists of fixed-size packet buffers. [Richard
Barnes & Benson Schliesser]
* S.4.2.1.2. Packet Size Bias Regardless of AQM: Largely re-
written to reflect the earlier change in advice about fixed-
size packet buffers, and to primarily focus on getting rid of
tail-drop, not various nuances of tail-drop. [Richard Barnes &
Benson Schliesser]
* Editorial corrections [Tim Bray, AppsDir, Pete McCann, Gen-ART
and others]
* Updated refs (two I-Ds have become RFCs). [Pete McCann]
From -08 to -09: Following WG last call:
* S.2.1: Made RED-related queue measurement recommendations
clearer
* S.2.3: Added to "Recommendation on Responding to Congestion" to
make it clear that we are definitely not saying transports have
to equalise bit-rates, just how to do it and not do it, if you
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want to.
* S.3: Clarified motivation sections S.3.3 "Transport-Independent
Network" and S.3.5 "Implementation Efficiency"
* S.3.4: Completely changed motivating argument from "Scaling
Congestion Control with Packet Size" to "Partial Deployment of
AQM".
From -07 to -08:
* Altered abstract to say it provides best current practice and
highlight that it updates RFC2309
* Added null IANA section
* Updated refs
From -06 to -07:
* A mix-up with the corollaries and their naming in 2.1 to 2.3
fixed.
From -05 to -06:
* Primarily editorial fixes.
From -04 to -05:
* Changed from Informational to BCP and highlighted non-normative
sections and appendices
* Removed language about consensus
* Added "Example Comparing Packet-Mode Drop and Byte-Mode Drop"
* Arranged "Motivating Arguments" into a more logical order and
completely rewrote "Transport-Independent Network" & "Scaling
Congestion Control with Packet Size" arguments. Removed "Why
Now?"
* Clarified applicability of certain recommendations
* Shifted vendor survey to an Appendix
* Cut down "Outstanding Issues and Next Steps"
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* Re-drafted the start of the conclusions to highlight the three
distinct areas of concern
* Completely re-wrote appendices
* Editorial corrections throughout.
From -03 to -04:
* Reordered Sections 2 and 3, and some clarifications here and
there based on feedback from Colin Perkins and Mirja
Kuehlewind.
From -02 to -03 (this version)
* Structural changes:
+ Split off text at end of "Scaling Congestion Control with
Packet Size" into new section "Transport-Independent
Network"
+ Shifted "Recommendations" straight after "Motivating
Arguments" and added "Conclusions" at end to reinforce
Recommendations
+ Added more internal structure to Recommendations, so that
recommendations specific to RED or to TCP are just
corollaries of a more general recommendation, rather than
being listed as a separate recommendation.
+ Renamed "State of the Art" as "Critical Survey of Existing
Advice" and retitled a number of subsections with more
descriptive titles.
+ Split end of "Congestion Coding: Summary of Status" into a
new subsection called "RED Implementation Status".
+ Removed text that had been in the Appendix "Congestion
Notification Definition: Further Justification".
* Reordered the intro text a little.
* Made it clearer when advice being reported is deprecated and
when it is not.
* Described AQM as in network equipment, rather than saying "at
the network layer" (to side-step controversy over whether
functions like AQM are in the transport layer but in network
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equipment).
* Minor improvements to clarity throughout
From -01 to -02:
* Restructured the whole document for (hopefully) easier reading
and clarity. The concrete recommendation, in RFC2119 language,
is now in Section 8.
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
* 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 3.
* 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.
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* 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 3.4).
* Separated out issue of normalising TCP's bit rate from issue of
preference to control packets (Section 3.2).
* Divided up Congestion Measurement section for clarity,
including new material on fixed size packet buffers and buffer
carving (Section 4.1.1 & Section 4.2.1) and on congestion
measurement in wireless link technologies without queues
(Section 4.1.2).
* Added section on 'Making Transports Robust against Control
Packet Losses' (Section 4.2.3) with existing & new material
included.
* Added tabulated results of vendor survey on byte-mode drop
variant of RED (Table 3).
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.
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* 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/
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@aalto.fi
URI: http://www.netlab.tkk.fi/~jmanner/
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