[Note that this file is a concatenation of more than one RFC.]
Network Working Group S. Floyd
Request for Comments: 2914 ACIRI
BCP: 41 September 2000
Category: Best Current Practice
Congestion Control Principles
Status of this Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2000). All Rights Reserved.
Abstract
The goal of this document is to explain the need for congestion
control in the Internet, and to discuss what constitutes correct
congestion control. One specific goal is to illustrate the dangers
of neglecting to apply proper congestion control. A second goal is
to discuss the role of the IETF in standardizing new congestion
control protocols.
1. Introduction
This document draws heavily from earlier RFCs, in some cases
reproducing entire sections of the text of earlier documents
[RFC2309, RFC2357]. We have also borrowed heavily from earlier
publications addressing the need for end-to-end congestion control
[FF99].
2. Current standards on congestion control
IETF standards concerning end-to-end congestion control focus either
on specific protocols (e.g., TCP [RFC2581], reliable multicast
protocols [RFC2357]) or on the syntax and semantics of communications
between the end nodes and routers about congestion information (e.g.,
Explicit Congestion Notification [RFC2481]) or desired quality-of-
service (diff-serv)). The role of end-to-end congestion control is
also discussed in an Informational RFC on "Recommendations on Queue
Management and Congestion Avoidance in the Internet" [RFC2309]. RFC
2309 recommends the deployment of active queue management mechanisms
in routers, and the continuation of design efforts towards mechanisms
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in routers to deal with flows that are unresponsive to congestion
notification. We freely borrow from RFC 2309 some of their general
discussion of end-to-end congestion control.
In contrast to the RFCs discussed above, this document is a more
general discussion of the principles of congestion control. One of
the keys to the success of the Internet has been the congestion
avoidance mechanisms of TCP. While TCP is still the dominant
transport protocol in the Internet, it is not ubiquitous, and there
are an increasing number of applications that, for one reason or
another, choose not to use TCP. Such traffic includes not only
multicast traffic, but unicast traffic such as streaming multimedia
that does not require reliability; and traffic such as DNS or routing
messages that consist of short transfers deemed critical to the
operation of the network. Much of this traffic does not use any form
of either bandwidth reservations or end-to-end congestion control.
The continued use of end-to-end congestion control by best-effort
traffic is critical for maintaining the stability of the Internet.
This document also discusses the general role of the IETF in the
standardization of new congestion control protocols.
The discussion of congestion control principles for differentiated
services or integrated services is not addressed in this document.
Some categories of integrated or differentiated services include a
guarantee by the network of end-to-end bandwidth, and as such do not
require end-to-end congestion control mechanisms.
3. The development of end-to-end congestion control.
3.1. Preventing congestion collapse.
The Internet protocol architecture is based on a connectionless end-
to-end packet service using the IP protocol. The advantages of its
connectionless design, flexibility and robustness, have been amply
demonstrated. However, these advantages are not without cost:
careful design is required to provide good service under heavy load.
In fact, lack of attention to the dynamics of packet forwarding can
result in severe service degradation or "Internet meltdown". This
phenomenon was first observed during the early growth phase of the
Internet of the mid 1980s [RFC896], and is technically called
"congestion collapse".
The original specification of TCP [RFC793] included window-based flow
control as a means for the receiver to govern the amount of data sent
by the sender. This flow control was used to prevent overflow of the
receiver's data buffer space available for that connection. [RFC793]
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reported that segments could be lost due either to errors or to
network congestion, but did not include dynamic adjustment of the
flow-control window in response to congestion.
The original fix for Internet meltdown was provided by Van Jacobson.
Beginning in 1986, Jacobson developed the congestion avoidance
mechanisms that are now required in TCP implementations [Jacobson88,
RFC 2581]. These mechanisms operate in the hosts to cause TCP
connections to "back off" during congestion. We say that TCP flows
are "responsive" to congestion signals (i.e., dropped packets) from
the network. It is these TCP congestion avoidance algorithms that
prevent the congestion collapse of today's Internet.
However, that is not the end of the story. Considerable research has
been done on Internet dynamics since 1988, and the Internet has
grown. It has become clear that the TCP congestion avoidance
mechanisms [RFC2581], while necessary and powerful, are not
sufficient to provide good service in all circumstances. In addition
to the development of new congestion control mechanisms [RFC2357],
router-based mechanisms are in development that complement the
endpoint congestion avoidance mechanisms.
A major issue that still needs to be addressed is the potential for
future congestion collapse of the Internet due to flows that do not
use responsible end-to-end congestion control. RFC 896 [RFC896]
suggested in 1984 that gateways should detect and `squelch'
misbehaving hosts: "Failure to respond to an ICMP Source Quench
message, though, should be regarded as grounds for action by a
gateway to disconnect a host. Detecting such failure is non-trivial
but is a worthwhile area for further research." Current papers
still propose that routers detect and penalize flows that are not
employing acceptable end-to-end congestion control [FF99].
3.2. Fairness
In addition to a concern about congestion collapse, there is a
concern about `fairness' for best-effort traffic. Because TCP "backs
off" during congestion, a large number of TCP connections can share a
single, congested link in such a way that bandwidth is shared
reasonably equitably among similarly situated flows. The equitable
sharing of bandwidth among flows depends on the fact that all flows
are running compatible congestion control algorithms. For TCP, this
means congestion control algorithms conformant with the current TCP
specification [RFC793, RFC1122, RFC2581].
The issue of fairness among competing flows has become increasingly
important for several reasons. First, using window scaling
[RFC1323], individual TCPs can use high bandwidth even over high-
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propagation-delay paths. Second, with the growth of the web,
Internet users increasingly want high-bandwidth and low-delay
communications, rather than the leisurely transfer of a long file in
the background. The growth of best-effort traffic that does not use
TCP underscores this concern about fairness between competing best-
effort traffic in times of congestion.
The popularity of the Internet has caused a proliferation in the
number of TCP implementations. Some of these may fail to implement
the TCP congestion avoidance mechanisms correctly because of poor
implementation [RFC2525]. Others may deliberately be implemented
with congestion avoidance algorithms that are more aggressive in
their use of bandwidth than other TCP implementations; this would
allow a vendor to claim to have a "faster TCP". The logical
consequence of such implementations would be a spiral of increasingly
aggressive TCP implementations, or increasingly aggressive transport
protocols, leading back to the point where there is effectively no
congestion avoidance and the Internet is chronically congested.
There is a well-known way to achieve more aggressive performance
without even changing the transport protocol, by changing the level
of granularity: open multiple connections to the same place, as has
been done in the past by some Web browsers. Thus, instead of a
spiral of increasingly aggressive transport protocols, we would
instead have a spiral of increasingly aggressive web browsers, or
increasingly aggressive applications.
This raises the issue of the appropriate granularity of a "flow",
where we define a `flow' as the level of granularity appropriate for
the application of both fairness and congestion control. From RFC
2309: "There are a few `natural' answers: 1) a TCP or UDP connection
(source address/port, destination address/port); 2) a
source/destination host pair; 3) a given source host or a given
destination host. We would guess that the source/destination host
pair gives the most appropriate granularity in many circumstances.
The granularity of flows for congestion management is, at least in
part, a policy question that needs to be addressed in the wider IETF
community."
Again borrowing from RFC 2309, we use the term "TCP-compatible" for a
flow that behaves under congestion like a flow produced by a
conformant TCP. A TCP-compatible flow is responsive to congestion
notification, and in steady-state uses no more bandwidth than a
conformant TCP running under comparable conditions (drop rate, RTT,
MTU, etc.)
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It is convenient to divide flows into three classes: (1) TCP-
compatible flows, (2) unresponsive flows, i.e., flows that do not
slow down when congestion occurs, and (3) flows that are responsive
but are not TCP-compatible. The last two classes contain more
aggressive flows that pose significant threats to Internet
performance, as we discuss below.
In addition to steady-state fairness, the fairness of the initial
slow-start is also a concern. One concern is the transient effect on
other flows of a flow with an overly-aggressive slow-start procedure.
Slow-start performance is particularly important for the many flows
that are short-lived, and only have a small amount of data to
transfer.
3.3. Optimizing performance regarding throughput, delay, and loss.
In addition to the prevention of congestion collapse and concerns
about fairness, a third reason for a flow to use end-to-end
congestion control can be to optimize its own performance regarding
throughput, delay, and loss. In some circumstances, for example in
environments of high statistical multiplexing, the delay and loss
rate experienced by a flow are largely independent of its own sending
rate. However, in environments with lower levels of statistical
multiplexing or with per-flow scheduling, the delay and loss rate
experienced by a flow is in part a function of the flow's own sending
rate. Thus, a flow can use end-to-end congestion control to limit
the delay or loss experienced by its own packets. We would note,
however, that in an environment like the current best-effort
Internet, concerns regarding congestion collapse and fairness with
competing flows limit the range of congestion control behaviors
available to a flow.
4. The role of the standards process
The standardization of a transport protocol includes not only
standardization of aspects of the protocol that could affect
interoperability (e.g., information exchanged by the end-nodes), but
also standardization of mechanisms deemed critical to performance
(e.g., in TCP, reduction of the congestion window in response to a
packet drop). At the same time, implementation-specific details and
other aspects of the transport protocol that do not affect
interoperability and do not significantly interfere with performance
do not require standardization. Areas of TCP that do not require
standardization include the details of TCP's Fast Recovery procedure
after a Fast Retransmit [RFC2582]. The appendix uses examples from
TCP to discuss in more detail the role of the standards process in
the development of congestion control.
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4.1. The development of new transport protocols.
In addition to addressing the danger of congestion collapse, the
standardization process for new transport protocols takes care to
avoid a congestion control `arms race' among competing protocols. As
an example, in RFC 2357 [RFC2357] the TSV Area Directors and their
Directorate outline criteria for the publication as RFCs of
Internet-Drafts on reliable multicast transport protocols. From
[RFC2357]: "A particular concern for the IETF is the impact of
reliable multicast traffic on other traffic in the Internet in times
of congestion, in particular the effect of reliable multicast traffic
on competing TCP traffic.... The challenge to the IETF is to
encourage research and implementations of reliable multicast, and to
enable the needs of applications for reliable multicast to be met as
expeditiously as possible, while at the same time protecting the
Internet from the congestion disaster or collapse that could result
from the widespread use of applications with inappropriate reliable
multicast mechanisms."
The list of technical criteria that must be addressed by RFCs on new
reliable multicast transport protocols include the following: "Is
there a congestion control mechanism? How well does it perform? When
does it fail? Note that congestion control mechanisms that operate
on the network more aggressively than TCP will face a great burden of
proof that they don't threaten network stability."
It is reasonable to expect that these concerns about the effect of
new transport protocols on competing traffic will apply not only to
reliable multicast protocols, but to unreliable unicast, reliable
unicast, and unreliable multicast traffic as well.
4.2. Application-level issues that affect congestion control
The specific issue of a browser opening multiple connections to the
same destination has been addressed by RFC 2616 [RFC2616], which
states in Section 8.1.4 that "Clients that use persistent connections
SHOULD limit the number of simultaneous connections that they
maintain to a given server. A single-user client SHOULD NOT maintain
more than 2 connections with any server or proxy."
4.3. New developments in the standards process
The most obvious developments in the IETF that could affect the
evolution of congestion control are the development of integrated and
differentiated services [RFC2212, RFC2475] and of Explicit Congestion
Notification (ECN) [RFC2481]. However, other less dramatic
developments are likely to affect congestion control as well.
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One such effort is that to construct Endpoint Congestion Management
[BS00], to enable multiple concurrent flows from a sender to the same
receiver to share congestion control state. By allowing multiple
connections to the same destination to act as one flow in terms of
end-to-end congestion control, a Congestion Manager could allow
individual connections slow-starting to take advantage of previous
information about the congestion state of the end-to-end path.
Further, the use of a Congestion Manager could remove the congestion
control dangers of multiple flows being opened between the same
source/destination pair, and could perhaps be used to allow a browser
to open many simultaneous connections to the same destination.
5. A description of congestion collapse
This section discusses congestion collapse from undelivered packets
in some detail, and shows how unresponsive flows could contribute to
congestion collapse in the Internet. This section draws heavily on
material from [FF99].
Informally, congestion collapse occurs when an increase in the
network load results in a decrease in the useful work done by the
network. As discussed in Section 3, congestion collapse was first
reported in the mid 1980s [RFC896], and was largely due to TCP
connections unnecessarily retransmitting packets that were either in
transit or had already been received at the receiver. We call the
congestion collapse that results from the unnecessary retransmission
of packets classical congestion collapse. Classical congestion
collapse is a stable condition that can result in throughput that is
a small fraction of normal [RFC896]. Problems with classical
congestion collapse have generally been corrected by the timer
improvements and congestion control mechanisms in modern
implementations of TCP [Jacobson88].
A second form of potential congestion collapse occurs due to
undelivered packets. Congestion collapse from undelivered packets
arises when bandwidth is wasted by delivering packets through the
network that are dropped before reaching their ultimate destination.
This is probably the largest unresolved danger with respect to
congestion collapse in the Internet today. Different scenarios can
result in different degrees of congestion collapse, in terms of the
fraction of the congested links' bandwidth used for productive work.
The danger of congestion collapse from undelivered packets is due
primarily to the increasing deployment of open-loop applications not
using end-to-end congestion control. Even more destructive would be
best-effort applications that *increase* their sending rate in
response to an increased packet drop rate (e.g., automatically using
an increased level of FEC).
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Table 1 gives the results from a scenario with congestion collapse
from undelivered packets, where scarce bandwidth is wasted by packets
that never reach their destination. The simulation uses a scenario
with three TCP flows and one UDP flow competing over a congested 1.5
Mbps link. The access links for all nodes are 10 Mbps, except that
the access link to the receiver of the UDP flow is 128 Kbps, only 9%
of the bandwidth of shared link. When the UDP source rate exceeds
128 Kbps, most of the UDP packets will be dropped at the output port
to that final link.
UDP
Arrival UDP TCP Total
Rate Goodput Goodput Goodput
--------------------------------------
0.7 0.7 98.5 99.2
1.8 1.7 97.3 99.1
2.6 2.6 96.0 98.6
5.3 5.2 92.7 97.9
8.8 8.4 87.1 95.5
10.5 8.4 84.8 93.2
13.1 8.4 81.4 89.8
17.5 8.4 77.3 85.7
26.3 8.4 64.5 72.8
52.6 8.4 38.1 46.4
58.4 8.4 32.8 41.2
65.7 8.4 28.5 36.8
75.1 8.4 19.7 28.1
87.6 8.4 11.3 19.7
105.2 8.4 3.4 11.8
131.5 8.4 2.4 10.7
Table 1. A simulation with three TCP flows and one UDP flow.
Table 1 shows the UDP arrival rate from the sender, the UDP goodput
(defined as the bandwidth delivered to the receiver), the TCP goodput
(as delivered to the TCP receivers), and the aggregate goodput on the
congested 1.5 Mbps link. Each rate is given as a fraction of the
bandwidth of the congested link. As the UDP source rate increases,
the TCP goodput decreases roughly linearly, and the UDP goodput is
nearly constant. Thus, as the UDP flow increases its offered load,
its only effect is to hurt the TCP and aggregate goodput. On the
congested link, the UDP flow ultimately `wastes' the bandwidth that
could have been used by the TCP flow, and reduces the goodput in the
network as a whole down to a small fraction of the bandwidth of the
congested link.
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The simulations in Table 1 illustrate both unfairness and congestion
collapse. As [FF99] discusses, compatible congestion control is not
the only way to provide fairness; per-flow scheduling at the
congested routers is an alternative mechanism at the routers that
guarantees fairness. However, as discussed in [FF99], per-flow
scheduling can not be relied upon to prevent congestion collapse.
There are only two alternatives for eliminating the danger of
congestion collapse from undelivered packets. The first alternative
for preventing congestion collapse from undelivered packets is the
use of effective end-to-end congestion control by the end nodes.
More specifically, the requirement would be that a flow avoid a
pattern of significant losses at links downstream from the first
congested link on the path. (Here, we would consider any link a
`congested link' if any flow is using bandwidth that would otherwise
be used by other traffic on the link.) Given that an end-node is
generally unable to distinguish between a path with one congested
link and a path with multiple congested links, the most reliable way
for a flow to avoid a pattern of significant losses at a downstream
congested link is for the flow to use end-to-end congestion control,
and reduce its sending rate in the presence of loss.
A second alternative for preventing congestion collapse from
undelivered packets would be a guarantee by the network that packets
accepted at a congested link in the network will be delivered all the
way to the receiver [RFC2212, RFC2475]. We note that the choice
between the first alternative of end-to-end congestion control and
the second alternative of end-to-end bandwidth guarantees does not
have to be an either/or decision; congestion collapse can be
prevented by the use of effective end-to-end congestion by some of
the traffic, and the use of end-to-end bandwidth guarantees from the
network for the rest of the traffic.
6. Forms of end-to-end congestion control
This document has discussed concerns about congestion collapse and
about fairness with TCP for new forms of congestion control. This
does not mean, however, that concerns about congestion collapse and
fairness with TCP necessitate that all best-effort traffic deploy
congestion control based on TCP's Additive-Increase Multiplicative-
Decrease (AIMD) algorithm of reducing the sending rate in half in
response to each packet drop. This section separately discusses the
implications of these two concerns of congestion collapse and
fairness with TCP.
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6.1. End-to-end congestion control for avoiding congestion collapse.
The avoidance of congestion collapse from undelivered packets
requires that flows avoid a scenario of a high sending rate, multiple
congested links, and a persistent high packet drop rate at the
downstream link. Because congestion collapse from undelivered
packets consists of packets that waste valuable bandwidth only to be
dropped downstream, this form of congestion collapse is not possible
in an environment where each flow traverses only one congested link,
or where only a small number of packets are dropped at links
downstream of the first congested link. Thus, any form of congestion
control that successfully avoids a high sending rate in the presence
of a high packet drop rate should be sufficient to avoid congestion
collapse from undelivered packets.
We would note that the addition of Explicit Congestion Notification
(ECN) to the IP architecture would not, in and of itself, remove the
danger of congestion collapse for best-effort traffic. ECN allows
routers to set a bit in packet headers as an indication of congestion
to the end-nodes, rather than being forced to rely on packet drops to
indicate congestion. However, with ECN, packet-marking would replace
packet-dropping only in times of moderate congestion. In particular,
when congestion is heavy, and a router's buffers overflow, the router
has no choice but to drop arriving packets.
6.2. End-to-end congestion control for fairness with TCP.
The concern expressed in [RFC2357] about fairness with TCP places a
significant though not crippling constraint on the range of viable
end-to-end congestion control mechanisms for best-effort traffic. An
environment with per-flow scheduling at all congested links would
isolate flows from each other, and eliminate the need for congestion
control mechanisms to be TCP-compatible. An environment with
differentiated services, where flows marked as belonging to a certain
diff-serv class would be scheduled in isolation from best-effort
traffic, could allow the emergence of an entire diff-serv class of
traffic where congestion control was not required to be TCP-
compatible. Similarly, a pricing-controlled environment, or a diff-
serv class with its own pricing paradigm, could supercede the concern
about fairness with TCP. However, for the current Internet
environment, where other best-effort traffic could compete in a FIFO
queue with TCP traffic, the absence of fairness with TCP could lead
to one flow `starving out' another flow in a time of high congestion,
as was illustrated in Table 1 above.
However, the list of TCP-compatible congestion control procedures is
not limited to AIMD with the same increase/ decrease parameters as
TCP. Other TCP-compatible congestion control procedures include
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rate-based variants of AIMD; AIMD with different sets of
increase/decrease parameters that give the same steady-state
behavior; equation-based congestion control where the sender adjusts
its sending rate in response to information about the long-term
packet drop rate; layered multicast where receivers subscribe and
unsubscribe from layered multicast groups; and possibly other forms
that we have not yet begun to consider.
7. Acknowledgements
Much of this document draws directly on previous RFCs addressing
end-to-end congestion control. This attempts to be a summary of
ideas that have been discussed for many years, and by many people.
In particular, acknowledgement is due to the members of the End-to-
End Research Group, the Reliable Multicast Research Group, and the
Transport Area Directorate. This document has also benefited from
discussion and feedback from the Transport Area Working Group.
Particular thanks are due to Mark Allman for feedback on an earlier
version of this document.
8. References
[BS00] Balakrishnan H. and S. Seshan, "The Congestion Manager",
Work in Progress.
[DMKM00] Dawkins, S., Montenegro, G., Kojo, M. and V. Magret,
"End-to-end Performance Implications of Slow Links",
Work in Progress.
[FF99] Floyd, S. and K. Fall, "Promoting the Use of End-to-End
Congestion Control in the Internet", IEEE/ACM
Transactions on Networking, August 1999. URL
http://www.aciri.org/floyd/end2end-paper.html
[HPF00] Handley, M., Padhye, J. and S. Floyd, "TCP Congestion
Window Validation", RFC 2861, June 2000.
[Jacobson88] V. Jacobson, Congestion Avoidance and Control, ACM
SIGCOMM '88, August 1988.
[RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.
[RFC896] Nagle, J., "Congestion Control in IP/TCP", RFC 896,
January 1984.
[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts --
Communication Layers", STD 3, RFC 1122, October 1989.
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RFC 2914 Congestion Control Principles September 2000
[RFC1323] Jacobson, V., Braden, R. and D. Borman, "TCP Extensions
for High Performance", RFC 1323, May 1992.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2212] Shenker, S., Partridge, C. and R. Guerin, "Specification
of Guaranteed Quality of Service", RFC 2212, September
1997.
[RFC2309] Braden, R., Clark, D., Crowcroft, J., Davie, B.,
Deering, S., Estrin, D., Floyd, S., Jacobson, V.,
Minshall, G., Partridge, C., Peterson, L., Ramakrishnan,
K.K., Shenker, S., Wroclawski, J., and L. Zhang,
"Recommendations on Queue Management and Congestion
Avoidance in the Internet", RFC 2309, April 1998.
[RFC2357] Mankin, A., Romanow, A., Bradner, S. and V. Paxson,
"IETF Criteria for Evaluating Reliable Multicast
Transport and Application Protocols", RFC 2357, June
1998.
[RFC2414] Allman, M., Floyd, S. and C. Partridge, "Increasing
TCP's Initial Window", RFC 2414, September 1998.
[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.
and W. Weiss, "An Architecture for Differentiated
Services", RFC 2475, December 1998.
[RFC2481] Ramakrishnan K. and S. Floyd, "A Proposal to add
Explicit Congestion Notification (ECN) to IP", RFC 2481,
January 1999.
[RFC2525] Paxson, V., Allman, M., Dawson, S., Fenner, W., Griner,
J., Heavens, I., Lahey, K., Semke, J. and B. Volz,
"Known TCP Implementation Problems", RFC 2525, March
1999.
[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[RFC2582] Floyd, S. and T. Henderson, "The NewReno Modification to
TCP's Fast Recovery Algorithm", RFC 2582, April 1999.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P. and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
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RFC 2914 Congestion Control Principles September 2000
[SCWA99] S. Savage, N. Cardwell, D. Wetherall, and T. Anderson,
TCP Congestion Control with a Misbehaving Receiver, ACM
Computer Communications Review, October 1999.
[TCPB98] Hari Balakrishnan, Venkata N. Padmanabhan, Srinivasan
Seshan, Mark Stemm, and Randy H. Katz, TCP Behavior of a
Busy Internet Server: Analysis and Improvements, IEEE
Infocom, March 1998. Available from:
"http://www.cs.berkeley.edu/~hari/papers/infocom98.ps.gz".
[TCPF98] Dong Lin and H.T. Kung, TCP Fast Recovery Strategies:
Analysis and Improvements, IEEE Infocom, March 1998.
Available from:
"http://www.eecs.harvard.edu/networking/papers/infocom-
tcp-final-198.pdf".
9. TCP-Specific issues
In this section we discuss some of the particulars of TCP congestion
control, to illustrate a realization of the congestion control
principles, including some of the details that arise when
incorporating them into a production transport protocol.
9.1. Slow-start.
The TCP sender can not open a new connection by sending a large burst
of data (e.g., a receiver's advertised window) all at once. The TCP
sender is limited by a small initial value for the congestion window.
During slow-start, the TCP sender can increase its sending rate by at
most a factor of two in one roundtrip time. Slow-start ends when
congestion is detected, or when the sender's congestion window is
greater than the slow-start threshold ssthresh.
An issue that potentially affects global congestion control, and
therefore has been explicitly addressed in the standards process,
includes an increase in the value of the initial window
[RFC2414,RFC2581].
Issues that have not been addressed in the standards process, and are
generally considered not to require standardization, include such
issues as the use (or non-use) of rate-based pacing, and mechanisms
for ending slow-start early, before the congestion window reaches
ssthresh. Such mechanisms result in slow-start behavior that is as
conservative or more conservative than standard TCP.
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9.2. Additive Increase, Multiplicative Decrease.
In the absence of congestion, the TCP sender increases its congestion
window by at most one packet per roundtrip time. In response to a
congestion indication, the TCP sender decreases its congestion window
by half. (More precisely, the new congestion window is half of the
minimum of the congestion window and the receiver's advertised
window.)
An issue that potentially affects global congestion control, and
therefore would be likely to be explicitly addressed in the standards
process, would include a proposed addition of congestion control for
the return stream of `pure acks'.
An issue that has not been addressed in the standards process, and is
generally not considered to require standardization, would be a
change to the congestion window to apply as an upper bound on the
number of bytes presumed to be in the pipe, instead of applying as a
sliding window starting from the cumulative acknowledgement.
(Clearly, the receiver's advertised window applies as a sliding
window starting from the cumulative acknowledgement field, because
packets received above the cumulative acknowledgement field are held
in TCP's receive buffer, and have not been delivered to the
application. However, the congestion window applies to the number of
packets outstanding in the pipe, and does not necessarily have to
include packets that have been received out-of-order by the TCP
receiver.)
9.3. Retransmit timers.
The TCP sender sets a retransmit timer to infer that a packet has
been dropped in the network. When the retransmit timer expires, the
sender infers that a packet has been lost, sets ssthresh to half of
the current window, and goes into slow-start, retransmitting the lost
packet. If the retransmit timer expires because no acknowledgement
has been received for a retransmitted packet, the retransmit timer is
also "backed-off", doubling the value of the next retransmit timeout
interval.
An issue that potentially affects global congestion control, and
therefore would be likely to be explicitly addressed in the standards
process, might include a modified mechanism for setting the
retransmit timer that could significantly increase the number of
retransmit timers that expire prematurely, when the acknowledgement
has not yet arrived at the sender, but in fact no packets have been
dropped. This could be of concern to the Internet standards process
Floyd, ed. Best Current Practice [Page 14]
RFC 2914 Congestion Control Principles September 2000
because retransmit timers that expire prematurely could lead to an
increase in the number of packets unnecessarily transmitted on a
congested link.
9.4. Fast Retransmit and Fast Recovery.
After seeing three duplicate acknowledgements, the TCP sender infers
a packet loss. The TCP sender sets ssthresh to half of the current
window, reduces the congestion window to at most half of the previous
window, and retransmits the lost packet.
An issue that potentially affects global congestion control, and
therefore would be likely to be explicitly addressed in the standards
process, might include a proposal (if there was one) for inferring a
lost packet after only one or two duplicate acknowledgements. If
poorly designed, such a proposal could lead to an increase in the
number of packets unnecessarily transmitted on a congested path.
An issue that has not been addressed in the standards process, and
would not be expected to require standardization, would be a proposal
to send a "new" or presumed-lost packet in response to a duplicate or
partial acknowledgement, if allowed by the congestion window. An
example of this would be sending a new packet in response to a single
duplicate acknowledgement, to keep the `ack clock' going in case no
further acknowledgements would have arrived. Such a proposal is an
example of a beneficial change that does not involve interoperability
and does not affect global congestion control, and that therefore
could be implemented by vendors without requiring the intervention of
the IETF standards process. (This issue has in fact been addressed
in [DMKM00], which suggests that "researchers may wish to experiment
with injecting new traffic into the network when duplicate
acknowledgements are being received, as described in [TCPB98] and
[TCPF98]."
9.5. Other aspects of TCP congestion control.
Other aspects of TCP congestion control that have not been discussed
in any of the sections above include TCP's recovery from an idle or
application-limited period [HPF00].
10. Security Considerations
This document has been about the risks associated with congestion
control, or with the absence of congestion control. Section 3.2
discusses the potentials for unfairness if competing flows don't use
compatible congestion control mechanisms, and Section 5 considers the
dangers of congestion collapse if flows don't use end-to-end
congestion control.
Floyd, ed. Best Current Practice [Page 15]
RFC 2914 Congestion Control Principles September 2000
Because this document does not propose any specific congestion
control mechanisms, it is also not necessary to present specific
security measures associated with congestion control. However, we
would note that there are a range of security considerations
associated with congestion control that should be considered in IETF
documents.
For example, individual congestion control mechanisms should be as
robust as possible to the attempts of individual end-nodes to subvert
end-to-end congestion control [SCWA99]. This is a particular concern
in multicast congestion control, because of the far-reaching
distribution of the traffic and the greater opportunities for
individual receivers to fail to report congestion.
RFC 2309 also discussed the potential dangers to the Internet of
unresponsive flows, that is, flows that don't reduce their sending
rate in the presence of congestion, and describes the need for
mechanisms in the network to deal with flows that are unresponsive to
congestion notification. We would note that there is still a need
for research, engineering, measurement, and deployment in these
areas.
Because the Internet aggregates very large numbers of flows, the risk
to the whole infrastructure of subverting the congestion control of a
few individual flows is limited. Rather, the risk to the
infrastructure would come from the widespread deployment of many
end-nodes subverting end-to-end congestion control.
AUTHOR'S ADDRESS
Sally Floyd
AT&T Center for Internet Research at ICSI (ACIRI)
Phone: +1 (510) 642-4274 x189
EMail: floyd@aciri.org
URL: http://www.aciri.org/floyd/
Floyd, ed. Best Current Practice [Page 16]
RFC 2914 Congestion Control Principles September 2000
Full Copyright Statement
Copyright (C) The Internet Society (2000). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
Floyd, ed. Best Current Practice [Page 17]
=========================================================================
Internet Engineering Task Force (IETF) B. Briscoe
Request for Comments: 7141 BT
BCP: 41 J. Manner
Updates: 2309, 2914 Aalto University
Category: Best Current Practice February 2014
ISSN: 2070-1721
Byte and Packet Congestion Notification
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 memo documents an Internet Best Current Practice.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
BCPs is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7141.
Briscoe & Manner Best Current Practice [Page 1]
RFC 7141 Byte and Packet Congestion Notification February 2014
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Briscoe & Manner Best Current Practice [Page 2]
RFC 7141 Byte and Packet Congestion Notification February 2014
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 . . . . . . . . . . . . . . . . . . . . 13
3.1. Avoiding Perverse Incentives to (Ab)use Smaller Packets . 13
3.2. Small != Control . . . . . . . . . . . . . . . . . . . . 14
3.3. Transport-Independent Network . . . . . . . . . . . . . . 14
3.4. Partial Deployment of AQM . . . . . . . . . . . . . . . . 16
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- and Packet-Congestible Network . . . . . . . . . . . 26
6. Security Considerations . . . . . . . . . . . . . . . . . . . 26
7. Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . 27
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 28
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 28
9.1. Normative References . . . . . . . . . . . . . . . . . . 28
9.2. Informative References . . . . . . . . . . . . . . . . . 29
Appendix A. Survey of RED Implementation Status . . . . . . . . 33
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
Briscoe & Manner Best Current Practice [Page 3]
RFC 7141 Byte and Packet Congestion Notification February 2014
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 tail-drop 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 pre-existing deployments.
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
the probability that it drops/marks each packet depend 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;
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
recommends byte-mode (packet-size dependent) drop [pktByteEmail].
Briscoe & Manner Best Current Practice [Page 4]
RFC 7141 Byte and Packet Congestion Notification February 2014
Reducing the number of small packets dropped certainly has some
tempting advantages: i) it drops fewer 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 build in scaling with packet size rather than
assuming that we should follow TCP's example.
This memo continues as follows. First, it discusses terminology and
scoping. Section 2 gives 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 AQM deployments
(Section 5). Then security issues are collected together in
Section 6 before conclusions are drawn in Section 7. 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
recommendations for the Internet community.
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RFC 7141 Byte and Packet Congestion Notification February 2014
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) [CoDel], and
the Proportional Integral controller Enhanced (PIE) [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 the transport 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 explicitly using 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'.
Briscoe & Manner Best Current Practice [Page 6]
RFC 7141 Byte and Packet Congestion Notification February 2014
Examples of packet-congestible resources are route look-up engines
and firewalls, because load depends on how many packet headers
they have to process. Examples of bit-congestible resources are
transmission links, radio power, and most buffer memory, because
the load depends on how many bits they have to transmit or store.
Some machine architectures use fixed-size packet buffers, so
buffer memory in these cases is packet-congestible (see
Section 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 (see Section 3.3 of
[RFC6077]), but there is no guarantee that it will not become more
common in the 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
Briscoe & Manner Best Current Practice [Page 7]
RFC 7141 Byte and Packet Congestion Notification February 2014
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
rate of 48 Mbps, but one is broken up into small 60 byte packets and
the other into large 1,500 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 60 B 1,500 B
Packet size s 480 b 12,000 b
Bit rate x 48 Mbps 48 Mbps
Packet rate u = x/s 100 kpps 4 kpps
Packet-mode Drop
Pkt-loss probability p 0.1% 0.1%
Pkt-loss rate p*u 100 pps 4 pps
Bit-loss rate p*u*s 48 kbps 48 kbps
Byte-mode Drop MTU, M=12,000 b
Pkt-loss probability b = p*s/M 0.004% 0.1%
Pkt-loss rate b*u 4 pps 4 pps
Bit-loss rate b*u*s 1.92 kbps 48 kbps
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 1,500 B or 12,000 b). 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
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RFC 7141 Byte and Packet Congestion Notification February 2014
cases. The 4 small dropped packets contain 25 times less bits than
the 4 large dropped packets: 1,920 compared to 48,000.
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 we discuss the implications on transport
protocols in Sections 2.3 and 2.4.
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 tail drop, 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.
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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.
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 that the operator use packet-mode
drop instead, 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.
Briscoe & Manner Best Current Practice [Page 10]
RFC 7141 Byte and Packet Congestion Notification February 2014
RED as a whole SHOULD NOT be switched off. Without RED, a tail-
drop 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
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
interoperability 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 packet
size into account, and
o making transports less sensitive to losing control packets like
SYNs and pure ACKs.
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RFC 7141 Byte and Packet Congestion Notification February 2014
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
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 an RTP / RTP Control Protocol (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 in which 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
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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 made
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
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 of 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 large-packet traffic, 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 that two unresponsive flows arrive at a bit-congestible
transmission link each with the same bit rate, say 1 Mbps, but one
consists of 1,500 B and the other 60 B packets, which are 25x
smaller. Consider a scenario where gentle RED [gentle_RED] is used,
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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; see Appendix B.4
of [RFC4828]). Even though both flows arrive with the same bit rate,
the bit rate the RED queue aims to pass to the line will be 750 kbps
for the flow of larger packets but 990 kbps 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-packet attacks, tail-drop 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 and ACKs, DNS queries and 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 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
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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.
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 that 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 packet size into account, the
transport can be certain it will never need to guess any assumptions
that 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 packet
size into account 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 packet size into account, it allows
transport protocols to choose whether or not to take packet size into
account. However, if the network were to bias congestion
notification by packet size, transport protocols would have no
choice; those that did not take into account packet size themselves
would unwittingly become dependent on packet size, and those that
already took packet size into account would end up taking it into
account twice.
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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.
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 reviewing 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 drops 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
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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
standards track TCP congestion control [RFC5681], may not choose to
make their rate response proportionate to the size of each dropped
packet, while others will (e.g., TCP-Friendly Rate Control for Small
Packets (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). Taking packet size into account 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.
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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),
and
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.
4.1. Congestion Measurement Advice
The choice of which metric to use to measure queue length was left
open in RFC 2309. 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 take into
account a typical mix of packet sizes when setting the thresholds.
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 a legacy in future
protocol design. This is safe because any under-sensitivity during
unusual traffic mixes cannot lead to congestion collapse given that
the buffer will eventually revert to tail drop, which discards
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-size buffers in
which to store each packet in the queue. This hardware forwards
packets to the line in one of two ways:
o With some hardware, any fixed-size 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
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queuing and transmission are in fixed MTU-size 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 the 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
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 the 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-size 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-
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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 the 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.
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 the 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.
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A few months before RFC 2309 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 the drop rate dependent on the
square of the relative packet size. This was indeed consistent with
one implied motivation behind RED's byte-mode drop -- that we should
reverse engineer the network to improve the performance of dominant
end-to-end congestion control mechanisms. 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. 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
queuing systems. We have already mentioned that tail-drop queues
naturally tend to lock out large packets once they are full.
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But also, queues with fixed-size 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 lockout is to ensure that smaller packets never
use the last available buffer in any of the pools for larger packets.
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 into account 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 into account packet sizes relative to typical TCP
packet sizes. This proposes a new small-packet variant of TCP-
friendly rate control (TFRC [RFC5348]), which is called TFRC-SP
[RFC4828]. Essentially, it proposes a rate equation that inflates
the flow rate by the ratio of a typical TCP segment size (1,500 B
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 the TFRC-SP specification 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
that it didn't sufficiently bias towards small packets to make TCP
independent of packet size.
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So the TFRC-SP specification doesn't address the issue of whether the
network or the transport _should_ handle fairness between different
packet sizes. In Appendix B.4 of RFC 4828, 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
(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 into account packet
size when forwarding through queues. Instead, the general principle
has been to take the sizes of marked packets into account 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, which
would be achieved 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
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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-effort
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.
4.2.4. Congestion Notification: Summary of Conflicting Advice
+-----------+-----------------+-----------------+-------------------+
| transport | RED_1 (packet- | RED_4 (linear | RED_5 (square |
| cc | mode drop) | byte-mode drop) | byte-mode drop) |
+-----------+-----------------+-----------------+-------------------+
| TCP or | s/sqrt(p) | sqrt(s/p) | 1/sqrt(p) |
| TFRC | | | |
| TFRC-SP | 1/sqrt(p) | 1/sqrt(s*p) | 1/(s*sqrt(p)) |
+-----------+-----------------+-----------------+-------------------+
Table 2: Dependence of flow bit rate per RTT on packet size, s, and
drop probability, p, when there is 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
Briscoe & Manner Best Current Practice [Page 24]
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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 the
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 the case, we can see that the linear byte-mode drop column
in the middle would considerably complicate the Internet. Even if
one believes the network should be doing the biasing, linear byte-
mode drop is a half-way house that doesn't bias enough towards small
packets. 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
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 -- the network should
not make allowance for packet sizes and the transport should. This
leaves two outstanding issues:
o The question of how to handle any legacy AQM deployments using
byte-mode drop;
o The need to start a programme to update transport congestion
control protocol standards to take packet size into account.
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 packet size into account
in transport congestion control protocols has already started with
TFRC-SP [RFC4828].
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5.2. Bit- and 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's 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,
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 ones 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 tail-drop
queues gave to small packets.
In practice, transports cannot all be trusted to respond to
congestion. So another reason for recommending that queues 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 tail drop'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.
Briscoe & Manner Best Current Practice [Page 26]
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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 drop, 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. 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:
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 into account 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'. It also addresses practical issues
Briscoe & Manner Best Current Practice [Page 27]
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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 in the 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].
8. 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 and
Trilogy 2, research projects (ICT-216372, ICT-317756) supported by
the European Community under its Seventh Framework Programme. The
views expressed here are those of the authors only.
9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
S., Wroclawski, J., and L. Zhang, "Recommendations on
Queue Management and Congestion Avoidance in the
Internet", RFC 2309, April 1998.
Briscoe & Manner Best Current Practice [Page 28]
RFC 7141 Byte and Packet Congestion Notification February 2014
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC
2914, September 2000.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
9.2. 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,
<http://dx.doi.org/10.1109/TNET.2002.801399>.
[CCvarPktSize]
Widmer, J., Boutremans, C., and J-Y. Le Boudec, "End-to-
end congestion control for TCP-friendly flows with
variable packet size", ACM CCR 34(2) 137-151, April 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://ieeexplore.ieee.org/xpl/
articleDetails.jsp?arnumber=903061>.
[CoDel] Nichols, K. and V. Jacobson, "Controlled Delay Active
Queue Management", Work in Progress, February 2013.
[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", Philosophical 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>.
Briscoe & Manner Best Current Practice [Page 29]
RFC 7141 Byte and Packet Congestion Notification February 2014
[Evol_cc] Gibbens, R. and F. Kelly, "Resource pricing and the
evolution of congestion control", Automatica
35(12)1969-1985, December 1999,
<http://www.sciencedirect.com/science/article/pii/
S0005109899001351>.
[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>.
[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.
[PIE] Pan, R., Natarajan, P., Piglione, C., Prabhu, M.,
Subramanian, V., Baker, F., and B. Steeg, "PIE: A
Lightweight Control Scheme To Address the Bufferbloat
Problem", Work in Progress, February 2014.
[PktSizeEquCC]
Vasallo, P., "Variable Packet Size Equation-Based
Congestion Control", ICSI Technical Report tr-00-008,
2000, <http://http.icsi.berkeley.edu/ftp/global/pub/
techreports/2000/tr-00-008.pdf>.
[RED93] Floyd, S. and V. Jacobson, "Random Early Detection (RED)
gateways for Congestion Avoidance", IEEE/ACM Transactions
on Networking 1(4) 397--413, August 1993,
<http://ieeexplore.ieee.org/xpls/
abs_all.jsp?arnumber=251892>.
[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, "Effect of
different packet sizes on RED performance", Proc. 5th IEEE
Symposium on Computers and Communications (ISCC) 793-799,
July 2000, <http://ieeexplore.ieee.org/xpls/
abs_all.jsp?arnumber=860741>.
Briscoe & Manner Best Current Practice [Page 30]
RFC 7141 Byte and Packet Congestion Notification February 2014
[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.
[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.
[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.
Briscoe & Manner Best Current Practice [Page 31]
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[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", IEEE/ACM Transactions on Networking
7(4) 458--472, August 1999, <http://ieeexplore.ieee.org/
xpls/abs_all.jsp?arnumber=793002>.
[pktByteEmail]
Floyd, S., "RED: Discussions of Byte and Packet Modes",
email, March 1997,
<http://ee.lbl.gov/floyd/REDaveraging.txt>.
Briscoe & Manner Best Current Practice [Page 32]
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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
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 | % 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 were given for why the byte-mode drop variant had not
been implemented, the extra complexity of packet-bias code was 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; however, 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.
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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.
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). Checking the two scenarios in each of the
horizontal and vertical dimensions tests the 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 bit- | Scenario a1) | Scenario b1) |
| 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).
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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 = 60 B = 480 b and s_2 = 1,500 B = 12,000 b.
A flow's bit rate, x [bps], is related to its packet rate, u
[pps], by
x(t) = s*u(t).
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 TCP as specified in RFC 5681 treats a congestion
notification on any packet whatever its size as one event. However,
a network with just the packet-mode drop algorithm gives 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 60 B and
1,500 B packets respectively. The body of the table considers two
separate cases, one where the flows have an equal bit rate and the
other with equal packet rates. In both cases, the two flows fill a
96 Mbps 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 (96 Mbps / 26 = 4 Mbps after
rounding). In contrast Flow 2 gets 25 times more bit rate (92 Mbps)
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 the bit rate by packet size,
as shown in the column labelled 'Formula'.
Briscoe & Manner Best Current Practice [Page 35]
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Parameter Formula Flow 1 Flow 2 Combined
----------------------- ----------- -------- -------- --------
Packet size s/8 60 B 1,500 B (Mix)
Packet size s 480 b 12,000 b (Mix)
Pkt loss probability p 0.1% 0.1% 0.1%
EQUAL BIT RATE CASE
Bit rate x 48 Mbps 48 Mbps 96 Mbps
Packet rate u = x/s 100 kpps 4 kpps 104 kpps
Absolute pkt-loss rate p*u 100 pps 4 pps 104 pps
Absolute bit-loss rate p*u*s 48 kbps 48 kbps 96 kbps
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 4 Mbps 92 Mbps 96 Mbps
Packet rate u = x/s 8 kpps 8 kpps 15 kpps
Absolute pkt-loss rate p*u 8 pps 8 pps 15 pps
Absolute bit-loss rate p*u*s 4 kbps 92 kbps 96 kbps
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 case labelled 'Equal packet rate' in
the table, where both flows aim for the same absolute packet-loss
rate (both 8 pps 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 the bit-loss rate such as the wide disparity in the case
labelled 'Equal packet rate' (4k bps vs. 92 kbps). Instead, a
packet-size-dependent algorithm would aim for equal bit-loss rates,
which would drive both flows towards the case labelled 'Equal bit
rate', by driving them to equal bit-loss rates (both 48 kbps in this
example).
Briscoe & Manner Best Current Practice [Page 36]
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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 case labelled 'Equal packet rate', 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*480 and
8*12,000 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 96 Mbps. 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 packets to sent packets will be the
same as the ratio of lost bytes 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 60 B packets, the other with
1,500 B 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 60 B packets, the ratio of lost bytes to sent bytes will be
60 B in every 60,000 B; and in the stream of 1,500 B packets, the
loss ratio will be 1,500 B out of 1,500,000 B. 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
packets to sent packets and the ratio of lost bytes 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 or not a transport depends on packet size 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 bit
congestion into account, by using the sizes of the packets that
indicate congestion. We have also shown that the transport can
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choose not to take packet size into account if it wishes. We will
now consider whether the transport can know which to do.
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
the 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 it is 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
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experiment shows that it will be hard to give transports a practical
way to know whether or not to take into account the size of
congestion indication packets.
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 lookups, load depends on the complexity of each lookup and
whether or not the pattern of arrivals is amenable to caching. 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
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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.
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 (ConEx) protocol
[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 sequences of bits being re-divided into different size
packets or across different size flows [Rate_fair_Dis].
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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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