One document matched: draft-ietf-aqm-recommendation-07.xml
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<rfc category="bcp" docName="draft-ietf-aqm-recommendation-07"
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<front>
<title abbrev="Active Queue Management Recommendations">IETF
Recommendations Regarding Active Queue Management</title>
<author fullname="Fred Baker" initials="F." role="editor" surname="Baker">
<organization>Cisco Systems</organization>
<address>
<postal>
<street></street>
<city>Santa Barbara</city>
<code>93117</code>
<region>California</region>
<country>USA</country>
</postal>
<email>fred@cisco.com</email>
</address>
</author>
<author fullname="Godred Fairhurst" initials="G." role="editor"
surname="Fairhurst">
<organization>University of Aberdeen</organization>
<address>
<postal>
<street>School of Engineering</street>
<street>Fraser Noble Building</street>
<city>Aberdeen</city>
<region>Scotland</region>
<code>AB24 3UE</code>
<country>UK</country>
</postal>
<email>gorry@erg.abdn.ac.uk</email>
<uri>http://www.erg.abdn.ac.uk</uri>
</address>
</author>
<date day="05" month="August" year="2014" />
<area>Internet Engineering Task Force</area>
<workgroup></workgroup>
<abstract>
<t>This memo presents recommendations to the Internet community
concerning measures to improve and preserve Internet performance. It
presents a strong recommendation for testing, standardization, and
widespread deployment of active queue management (AQM) in network
devices, to improve the performance of today's Internet. It also urges a
concerted effort of research, measurement, and ultimate deployment of
AQM mechanisms to protect the Internet from flows that are not
sufficiently responsive to congestion notification.</t>
<t>The note largely repeats the recommendations of RFC 2309, and
replaces these after fifteen years of experience and new research.</t>
</abstract>
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<section title="Introduction">
<t>The Internet protocol architecture is based on a connectionless end-
to-end packet service using the Internet Protocol, whether <xref
target="RFC0791">IPv4</xref> or <xref target="RFC2460">IPv6</xref>. 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 in the mid 1980s <xref target="RFC0896"></xref><xref
target="RFC0970"></xref>, and is technically called "congestive
collapse" and was a key focus of RFC2309.</t>
<t>Since 1998, when RFC2309 was written, the Internet has become used
for a variety of traffic. In the current Internet low latency is
extremely important for many interactive and transaction-based
applications. The same type of technology that RFC2309 advocated for
combating congestion collapse is also effective at limiting delays to
reduce the interaction delay experienced by applications. While there is
still a need to avoid congestion collapse, there is now also a focus on
reducing network latency using the same technology.</t>
<section title="Congestion Collapse">
<t>The original fix for Internet meltdown was provided by Van
Jacobsen. Beginning in 1986, Jacobsen developed the congestion
avoidance mechanisms <xref target="Jacobson88"></xref> that are now
required for implementations of the Transport Control Protocol (TCP)
<xref target="RFC0768"></xref> <xref target="RFC1122"></xref>. These
mechanisms operate in Internet hosts to cause TCP connections to "back
off" during congestion. We say that TCP flows are "responsive" to
congestion signals (i.e., packets that are dropped or marked with
explicit congestion notification <xref target="RFC3168"></xref>). It
is primarily these TCP congestion avoidance algorithms that prevent
the congestive collapse of today's Internet. Similar algorithms are
specified for other non-TCP transports.</t>
<t>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 <xref target="RFC5681">congestion
avoidance mechanisms</xref>, while necessary and powerful, are not
sufficient to provide good service in all circumstances. Basically,
there is a limit to how much control can be accomplished from the
edges of the network. Some mechanisms are needed in the network
devices to complement the endpoint congestion avoidance mechanisms.
These mechanisms may be implemented in network devices that include
routers, switches, and other network middleboxes.</t>
</section>
<section title="Active Queue Management to Manage Latency">
<t>Internet latency has become a focus of attention to increase the
responsiveness of Internet applications and protocols. One major
source of delay is the build-up of queues in network devices. Queueing
occurs whenever the arrival rate of data at the ingress to a device
exceeds the current egress rate. Such queueing is normal in a
packet-switched network and often necessary to absorb bursts in
transmission and perform statistical multiplexing of traffic, but
excessive queueing can lead to unwanted delay, reducing the
performance of some Internet applications.</t>
<t>RFC 2309 introduced the concept of "Active Queue Management" (AQM),
a > class of technologies that, by signaling to common
congestion-controlled transports such as TCP, manages the size of
queues that build in network buffers. RFC 2309 also describes a
specific AQM algorithm, Random Early Detection (RED), and recommends
that this be widely implemented and used by default in routers. </t>
<t>With an appropriate set of parameters, RED is an effective
algorithm. However, dynamically predicting this set of parameters was
found to be difficult. As a result, RED has not been enabled by
default, and its present use in the Internet is limited. Other AQM
algorithms have been developed since RC2309 was published, some of
which are self-tuning within a range of applicability. Hence, while
this memo continues to recommend the deployment of AQM, it no longer
recommends that RED or any other specific algorithm is used as a
default; instead it provides recommendations on how to select
appropriate algorithms and recommends that algorithms should be used
that a recommened algorithm is able to automate any required tuning
for common deployment scenarios.</t>
<t>Deploying AQM in the network can significantly reduce the latency
across an Internet path and since writing RFC2309, this has become a
key motivation for using AQM in the Internet. In the context of AQM,
it is useful to distinguish between two related classes of algorithms:
"queue management" versus "scheduling" algorithms. To a rough
approximation, queue management algorithms manage the length of packet
queues by marking or dropping packets when necessary or appropriate,
while scheduling algorithms determine which packet to send next and
are used primarily to manage the allocation of bandwidth among flows.
While these two mechanisms are closely related, they address different
performance issues and operate on different timescales. Both may be
used in combination.</t>
</section>
<section title="Document Overview">
<t>The discussion in this memo applies to "best-effort" traffic, which
is to say, traffic generated by applications that accept the
occasional loss, duplication, or reordering of traffic in flight. It
also applies to other traffic, such as real-time traffic that can
adapt its sending rate to reduce loss and/or delay. It is most
effective when the adaption occurs on time scales of a single Round
Trip Time (RTT) or a small number of RTTs, for <xref
target="RFC1633">elastic traffic</xref>.</t>
<t>Two performance issues are highlighted:</t>
<t>The first issue is the need for an advanced form of queue
management that we call "Active Queue Management", AQM. <xref
target="Section2"></xref> summarizes the benefits that active queue
management can bring. A number of AQM procedures are described in the
literature, with different characteristics. This document does not
recommend any of them in particular, but does make recommendations
that ideally would affect the choice of procedure used in a given
implementation.</t>
<t>The second issue, discussed in <xref target="conclusion"></xref> of
this memo, is the potential for future congestive collapse of the
Internet due to flows that are unresponsive, or not sufficiently
responsive, to congestion indications. Unfortunately, while scheduling
can mitigate some of the side-effects of sharing a network queue with
an unresponsive flow, there is currently no consensus solution to
controlling the congestion caused by such aggressive flows. Methods
such as congestion exposure (ConEx) <xref target="RFC6789"></xref>
offer a framework <xref target="CONEX"></xref> that can update network
devices to alleviate these effects. Significant research and
engineering will be required before any solution will be available. It
is imperative that work to mitigate the impact of unresponsive flows
is energetically pursued, to ensure the future stability of the
Internet.</t>
<t><xref target="conclusion"></xref> concludes the memo with a set of
recommendations to the Internet community on the use of AQM and
recommendations for defining AQM algorithms.</t>
</section>
<section anchor="update-to-rfc2309"
title="Changes to the recommendations of RFC2309">
<t>This memo replaces the recommendations in <xref
target="RFC2309"></xref>, which resulted from past discussions of
end-to-end performance, Internet congestion, and RED in the End-to-End
Research Group of the Internet Research Task Force (IRTF). It follows
experience with this and other algorithms, and the AQM discussion
within the IETF <xref target="AQM-WG"></xref>.</t>
<t>While RFC2309 described AQM in terms of the length of a queue. This
memo changes this, to use AQM to refer to any method that allows
network devices to control either the queue length and/or the mean
time that a packet spends in a queue.</t>
<t>This memo also explicitly obsoletes the recommendation that Random
Early Detection (RED) was to be used as the default AQM mechanism for
the Internet. This is replaced by a detailed set of recommendations
for selecting an appropriate AQM algorithm. As in RFC2309, this memo
also motivates the need for continued research, but clarifies the
research with examples appropriate at the time that this memo is
published.</t>
</section>
<section title="Requirements Language">
<t>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 <xref
target="RFC2119"></xref>.</t>
</section>
</section>
<section anchor="Section2" title="The Need For Active Queue Management">
<t>Active Queue Management (AQM) is a method that allows network devices
to control the queue length or the mean time that a packet spends in a
queue. Although AQM can be applied across a range of deployment
environments, the recommendations in this document are directed to use
in the general Internet. It is expected that the principles and guidance
are also applicable to a wide range of environments, but may require
tuning for specific types of link/network (e.g. to accommodate the
traffic patterns found in data centres, the challenges of wireless
infrastructure, or the higher delay encountered on satellite Internet
links). The remainder of this section identifies the need for AQM and
the advantages of deploying AQM methods.</t>
<t>The traditional technique for managing the queue length in a network
device is to set a maximum length (in terms of packets) for each queue,
accept packets for the queue until the maximum length is reached, then
reject (drop) subsequent incoming packets until the queue decreases
because a packet from the queue has been transmitted. This technique is
known as "tail drop", since the packet that arrived most recently (i.e.,
the one on the tail of the queue) is dropped when the queue is full.
This method has served the Internet well for years, but it has four
important drawbacks:<list style="numbers">
<t>Full Queues <vspace blankLines="1" />The tail drop discipline
allows queues to maintain a full (or, almost full) status for long
periods of time, since tail drop signals congestion (via a packet
drop) only when the queue has become full. It is important to reduce
the steady-state queue size, and this is perhaps the most important
goal for queue management. <vspace blankLines="1" />The naive
assumption might be that there is a simple tradeoff between delay
and throughput, and that the recommendation that queues be
maintained in a "non-full" state essentially translates to a
recommendation that low end-to-end delay is more important than high
throughput. However, this does not take into account the critical
role that packet bursts play in Internet performance. For example,
even though TCP constrains the congestion window of a flow, packets
often arrive at network devices in bursts <xref
target="Leland94"></xref>. If the queue is full or almost full, an
arriving burst will cause multiple packets to be dropped from the
same flow. Bursts of loss can result in a global synchronization of
flows throttling back, followed by a sustained period of lowered
link utilization, reducing overall throughput <xref
target="Flo94"></xref>, <xref target="Zha90"></xref><vspace
blankLines="1" />The goal of buffering in the network is to absorb
data bursts and to transmit them during the (hopefully) ensuing
bursts of silence. This is essential to permit transmission of
bursts of data. Normally small queues are preferred in network
devices, with sufficient queue capacity to absorb the bursts. The
counter-intuitive result is that maintaining normally-small queues
can result in higher throughput as well as lower end-to-end delay.
In summary, queue limits should not reflect the steady state queues
we want to be maintained in the network; instead, they should
reflect the size of bursts that a network device needs to
absorb.</t>
<t>Lock-Out <vspace blankLines="1" />In some situations tail drop
allows a single connection or a few flows to monopolize the queue
space starving other connection preventing them from getting room in
the queue <xref target="Flo92"></xref>.</t>
<t>Mitigating the Impact of Packet Bursts <vspace
blankLines="1" />Large burst of packets can delay other packets,
disrupting the control loop (e.g. the pacing of flows by the TCP
ACK-Clock), and reducing the performance of flows that share a
common bottleneck.</t>
<t>Control loop synchronisation<vspace blankLines="1" />Congestion
control, like other end-to-end mechanisms, introduces a control loop
between hosts. Sessions that share a common network bottleneck can
therefore become synchronised, introducing periodic disruption (e.g.
jitter/loss). "lock-out" is often also the result of synchronization
or other timing effects</t>
</list></t>
<t>Besides tail drop, two alternative queue management disciplines that
can be applied when a queue becomes full are "random drop on full" or
"head drop on full". When a new packet arrives at a full queue using the
random drop on full discipline, the network device drops a randomly
selected packet from the queue (which can be an expensive operation,
since it naively requires an O(N) walk through the packet queue). When a
new packet arrives at a full queue using the head drop on full
discipline, the network device drops the packet at the front of the
queue <xref target="Lakshman96"></xref>. Both of these solve the
lock-out problem, but neither solves the full-queues problem described
above.</t>
<t>We know in general how to solve the full-queues problem for
"responsive" flows, i.e., those flows that throttle back in response to
congestion notification. In the current Internet, dropped packets
provide a critical mechanism indicating congestion notification to
hosts. The solution to the full-queues problem is for network devices to
drop or ECN-mark packets before a queue becomes full, so that hosts can
respond to congestion before buffers overflow. We call such a proactive
approach AQM. By dropping or ECN-marking packets before buffers
overflow, AQM allows network devices to control when and how many
packets to drop.</t>
<t>In summary, an active queue management mechanism can provide the
following advantages for responsive flows. <list style="numbers">
<t>Reduce number of packets dropped in network devices <vspace
blankLines="1" />Packet bursts are an unavoidable aspect of packet
networks <xref target="Willinger95"></xref>. If all the queue space
in a network device is already committed to "steady state" traffic
or if the buffer space is inadequate, then the network device will
have no ability to buffer bursts. By keeping the average queue size
small, AQM will provide greater capacity to absorb
naturally-occurring bursts without dropping packets. <vspace
blankLines="1" />Furthermore, without AQM, more packets will be
dropped when a queue does overflow. This is undesirable for several
reasons. First, with a shared queue and the tail drop discipline,
this can result in unnecessary global synchronization of flows,
resulting in lowered average link utilization, and hence lowered
network throughput. Second, unnecessary packet drops represent a
waste of network capacity on the path before the drop point. <vspace
blankLines="1" />While AQM can manage queue lengths and reduce
end-to-end latency even in the absence of end-to-end congestion
control, it will be able to reduce packet drops only in an
environment that continues to be dominated by end-to-end congestion
control.</t>
<t>Provide a lower-delay interactive service <vspace
blankLines="1" />By keeping a small average queue size, AQM will
reduce the delays experienced by flows. This is particularly
important for interactive applications such as short web transfers,
POP/IMAP, DNS, terminal traffic (telnet, ssh, mosh, RDP, etc),
gaming or interactive audio-video sessions, whose subjective (and
objective) performance is better when the end-to-end delay is
low.</t>
<t>Avoid lock-out behavior <vspace blankLines="1" />AQM can prevent
lock-out behavior by ensuring that there will almost always be a
buffer available for an incoming packet. For the same reason, AQM
can prevent a bias against low capacity, but highly bursty, flows.
<vspace blankLines="1" />Lock-out is undesirable because it
constitutes a gross unfairness among groups of flows. However, we
stop short of calling this benefit "increased fairness", because
general fairness among flows requires per-flow state, which is not
provided by queue management. For example, in a network device using
AQM with only FIFO scheduling, two TCP flows may receive very
different share of the network capacity simply because they have
different round-trip times <xref target="Floyd91"></xref>, and a
flow that does not use congestion control may receive more capacity
than a flow that does. AQM can therefore be combined with a
scheduling mechanism that divides network traffic between multiple
queues (section 2.1).</t>
<t>Reduce the probability of control loop synchronisation<vspace
blankLines="1" />The probability of network control loop
synchronisation can be reduced if network devices introduce
randomness in the AQM functions that trigger congestion avoidance at
the sending host.</t>
</list></t>
<section title="AQM and Multiple Queues">
<t>A network device may use per-flow or per-class queuing with a
scheduling algorithm to either prioritise certain applications or
classes of traffic, limit the rate of transmission, or to provide
isolation between different traffic flows within a common class. For
example, a router may maintain per-flow state to achieve general
fairness by a per-flow scheduling algorithm such as various forms of
Fair Queueing (FQ) <xref target="Dem90"></xref> <xref
target="Sut99"></xref>, including Weighted Fair Queuing (WFQ),
Stochastic Fairness Queueing (SFQ) <xref target="McK90"></xref>
Deficit Round Robin (DRR) <xref target="Shr96"></xref>, <xref
target="Nic12"></xref>, and/or a Class-Based Queue scheduling
algorithm such as CBQ <xref target="Floyd95"></xref>. Hierarchical
queues may also be used e.g., as a part of a Hierarchical Token Bucket
(HTB), or Hierarchical Fair Service Curve (HFSC) <xref
target="Sto97"></xref>. These methods are also used to realise a range
of Quality of Service (QoS) behaviours designed to meet the need of
traffic classes (e.g. using the integrated or differentiated service
models).</t>
<t>AQM is needed even for network devices that use per-flow or
per-class queuing, because scheduling algorithms by themselves do not
control the overall queue size or the size of individual queues. AQM
mechanisms might need to control the overall queue sizes, to ensure
that arriving bursts can be accommodated without dropping packets. AQM
should also be used to control the queue size for each individual flow
or class, so that they do not experience unnecessarily high delay.
Using a combination of AQM and scheduling between multiple queues has
been shown to offer good results in experimental and some types of
operational use.</t>
<t>In short, scheduling algorithms and queue management should be seen
as complementary, not as replacements for each other.</t>
</section>
<section title="AQM and Explicit Congestion Marking (ECN)">
<t>An AQM method may use Explicit Congestion Notification (ECN) <xref
target="RFC3168"></xref> instead of dropping to mark packets under
mild or moderate congestion. ECN-marking can allow a network device to
signal congestion at a point before a transport experiences congestion
loss or additional queuing delay <xref target="ECN-Benefit"></xref>.
<xref target="ECN"> </xref> describes some of the benefits of using
ECN with AQM.</t>
</section>
<section title="AQM and Buffer Size">
<t>It is important to differentiate the choice of buffer size for a
queue in a switch/router or other network device, and the threshold(s)
and other parameters that determine how and when an AQM algorithm
operates. The optimum buffer size is a function of operational
requirements and should generally be sized to be sufficient to buffer
the largest normal traffic burst that is expected. This size depends
on the number and burstiness of traffic arriving at the queue and the
rate at which traffic leaves the queue.</t>
<t>One objective of AQM is to minimize the effect of lock-out, where
one flow prevents other flows from effectively gaining capacity. This
need can be illustrated by a simple example of drop-tail queuing when
a new TCP flow injects packets into a queue that happens to be almost
full. A TCP flow's congestion control algorithm <xref
target="RFC5681"></xref> increases the flow rate to maximize its
effective window. This builds a queue in the network, inducing latency
to the flow and other flows that share this queue. Once a drop-tail
queue fills, there will also be loss. A new flow, sending its initial
burst, has an enhanced probability of filling the remaining queue and
dropping packets. As a result, the new flow can be effectively
prevented from effectively sharing the queue for a period of many
RTTs. In contrast, AQM can minimize the mean queue depth and therefore
reducing the probability that competing sessions can materially
prevent each other from performing well.</t>
<t>AQM frees a designer from having to limit the buffer space assigned
to a queue to achieve acceptable performance, allowing allocation of
sufficient buffering to satisfy the needs of the particular traffic
pattern. Different types of traffic and deployment scenarios will lead
to different requirements. The choice of AQM algorithm and associated
parameters is therefore a function of the way in which congestion is
experienced and the required reaction to achieve acceptable
performance. This latter is the primary topic of the following
sections.</t>
</section>
</section>
<section anchor="Section4" title="Managing Aggressive Flows">
<t>One of the keys to the success of the Internet has been the
congestion avoidance mechanisms of TCP. Because TCP "backs off" during
congestion, a large number of TCP connections can share a single,
congested link in such a way that link bandwidth is shared reasonably
equitably among similarly situated flows. The equitable sharing of
bandwidth among flows depends on all flows running compatible congestion
avoidance algorithms, i.e., methods conformant with the current TCP
specification <xref target="RFC5681"></xref>.</t>
<t>In this document a flow is known as "TCP-friendly" when it has a
congestion response that approximates the average response expected of a
TCP flow. One example method of a TCP-friendly scheme is the
TCP-Friendly Rate Control algorithm <xref target="RFC5348"></xref>. In
this document, the term is used more generally to describe this and
other algorithms that meet these goals.</t>
<t>There are a variety of types of network flow. Some convenient classes
that describe flows are: (1) TCP Friendly 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-friendly. The last two classes
contain more aggressive flows that pose significant threats to Internet
performance, which we will now discuss. <list style="numbers">
<t>TCP-Friendly flows <vspace blankLines="1" />A TCP-friendly flow
responds to congestion notification within a small number of path
Round Trip Times (RTT), and in steady-state it uses no more capacity
than a conformant TCP running under comparable conditions (drop
rate, RTT, packet size, etc.). This is described in the remainder of
the document.</t>
<t>Non-Responsive Flows <vspace blankLines="1" />The User Datagram
Protocol (UDP) <xref target="RFC0768"></xref> provides a minimal,
best-effort transport to applications and upper-layer protocols
(both simply called "applications" in the remainder of this
document) and does not itself provide mechanisms to prevent
congestion collapse and establish a degree of fairness <xref
target="RFC5405"></xref>. <vspace blankLines="1" />There is a
growing set of UDP-based applications whose congestion avoidance
algorithms are inadequate or nonexistent (i.e, a flow that does not
throttle its sending rate when it experiences congestion). Examples
include some UDP streaming applications for packet voice and video,
and some multicast bulk data transport. If no action is taken, such
unresponsive flows could lead to a new congestive collapse.<vspace
blankLines="1" />In general, UDP-based applications need to
incorporate effective congestion avoidance mechanisms <xref
target="RFC5405"></xref>. Further research and development of ways
to accomplish congestion avoidance for presently unresponsive
applications continue to be important. Network devices need to be
able to protect themselves against unresponsive flows, and
mechanisms to accomplish this must be developed and deployed.
Deployment of such mechanisms would provide an incentive for all
applications to become responsive by either using a
congestion-controlled transport (e.g. TCP, <xref
target="RFC4960">SCTP</xref> and <xref
target="RFC4340">DCCP</xref>.) or by incorporating their own
congestion control in the application <xref
target="RFC5405"></xref>, <xref target="RFC6679"></xref>.<vspace
blankLines="1" />Lastly, some applications (e.g. current web
browsers) open a large numbers of short TCP flows for a single
session. This can lead to each individual flow spending the majority
of time in the exponential TCP slow start phase, rather than in TCP
congestion avoidance. The resulting traffic aggregate can therefore
be much less responsive than a single standard TCP flow.</t>
<t>Non-TCP-friendly Transport Protocols <vspace blankLines="1" />A
second threat is posed by transport protocol implementations that
are responsive to congestion, but, either deliberately or through
faulty implementation, are not TCP-friendly. Such applications may
gain an unfair share of the available network capacity. <vspace
blankLines="1" />For example, 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. Others may deliberately be
implemented with congestion avoidance algorithms that are more
aggressive in their use of capacity 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, leading back to the
point where there is effectively no congestion avoidance and the
Internet is chronically congested. <vspace blankLines="1" />Another
example could be an RTP/UDP video flow that uses an adaptive codec,
but responds incompletely to indications of congestion or responds
over an excessively long time period. Such flows are unlikely to be
responsive to congestion signals in a timeframe comparable to a
small number of end-to-end transmission delays. However, over a
longer timescale, perhaps seconds in duration, they could moderate
their speed, or increase their speed if they determine capacity to
be available. <vspace blankLines="1" />Tunneled traffic aggregates
carrying multiple (short) TCP flows can be more aggressive than
standard bulk TCP. Applications (e.g. web browsers and peer-to-peer
file-sharing) have exploited this by opening multiple connections to
the same endpoint.</t>
</list></t>
<t>The projected increase in the fraction of total Internet traffic for
more aggressive flows in classes 2 and 3 clearly poses a threat to
future Internet stability. There is an urgent need for measurements of
current conditions and for further research into the ways of managing
such flows. This raises many difficult issues in identifying and
isolating unresponsive or non-TCP-friendly flows at an acceptable
overhead cost. Finally, there is as yet little measurement or simulation
evidence available about the rate at which these threats are likely to
be realized, or about the expected benefit of algorithms for managing
such flows.</t>
<t>Another topic requiring consideration is the appropriate granularity
of a "flow" when considering a queue management method. There are a few
"natural" answers: 1) a transport (e.g. TCP or UDP) flow (source
address/port, destination address/port, protocol); 2) Differentiated
Services Code Point, DSCP; 3) a source/destination host pair (IP
address); 4) a given source host or a given destination host, or various
combinations of the above; 5) a subscriber or site receiving the
Internet service (enterprise or residential).</t>
<t>The source/destination host pair gives an appropriate granularity in
many circumstances, However, different vendors/providers use different
granularities for defining a flow (as a way of "distinguishing"
themselves from one another), and different granularities may be chosen
for different places in the network. It may be the case that the
granularity is less important than the fact that a network device needs
to be able to deal with more unresponsive flows at *some* granularity.
The granularity of flows for congestion management is, at least in part,
a question of policy that needs to be addressed in the wider IETF
community.</t>
</section>
<section anchor="conclusion" title="Conclusions and Recommendations">
<t>The IRTF, in publishing <xref target="RFC2309"></xref>, and the IETF
in subsequent discussion, has developed a set of specific
recommendations regarding the implementation and operational use of AQM
procedures. The recommendations provided by this document are summarised
as: <list style="numbers">
<t>Network devices SHOULD implement some AQM mechanism to manage
queue lengths, reduce end-to-end latency, and avoid lock-out
phenomena within the Internet.</t>
<t>Deployed AQM algorithms SHOULD support Explicit Congestion
Notification (ECN) as well as loss to signal congestion to
endpoints.</t>
<t>The algorithms that the IETF recommends SHOULD NOT require
operational (especially manual) configuration or tuning.</t>
<t>AQM algorithms SHOULD respond to measured congestion, not
application profiles.</t>
<t>AQM algorithms SHOULD NOT interpret specific transport protocol
behaviours.</t>
<t>Transport protocol congestion control algorithms SHOULD maximize
their use of available capacity (when there is data to send) without
incurring undue loss or undue round trip delay.</t>
<t>Research, engineering, and measurement efforts are needed
regarding the design of mechanisms to deal with flows that are
unresponsive to congestion notification or are responsive, but are
more aggressive than present TCP.</t>
</list></t>
<t>These recommendations are expressed using the word "SHOULD". This is
in recognition that there may be use cases that have not been envisaged
in this document in which the recommendation does not apply. Therefore,
care should be taken in concluding that one's use case falls in that
category; during the life of the Internet, such use cases have been
rarely if ever observed and reported. To the contrary, available <xref
target="Choi04"> research </xref> says that even high speed links in
network cores that are normally very stable in depth and behavior
experience occasional issues that need moderation. The recommendations
are detailed in the following sections.</t>
<section anchor="useAQM"
title="Operational deployments SHOULD use AQM procedures">
<t>AQM procedures are designed to minimize the delay and buffer
exhaustion induced in the network by queues that have filled as a
result of host behavior. Marking and loss behaviors provide a signal
that buffers within network devices are becoming unnecessarily full,
and that the sender would do well to moderate its behavior.</t>
<t>The use of scheduling mechanisms, such as priority queuing,
classful queuing, and fair queuing, is often effective in networks to
help a network serve the needs of a range of applications. Network
operators can use these methods to manage traffic passing a choke
point. This is discussed in <xref target="RFC2474"></xref> and <xref
target="RFC2475"></xref>. When scheduling is used AQM should be
applied across the classes or flows as well as within each class or
flow:</t>
<t><list style="symbols">
<t>AQM mechanisms need to control the overall queue sizes, to
ensure that arriving bursts can be accommodated without dropping
packets.</t>
<t>AQM mechanisms need to allow combination with other mechanisms,
such as scheduling, to allow implementation of policies for
providing fairness between different flows.</t>
<t>AQM should be used to control the queue size for each
individual flow or class, so that they do not experience
unnecessarily high delay.</t>
</list></t>
</section>
<section anchor="signaling" title="Signaling to the transport endpoints">
<t>There are a number of ways a network device may signal to the end
point that the network is becoming congested and trigger a reduction
in rate. The signalling methods include:</t>
<t><list style="symbols">
<t>Delaying transport segments (packets) in flight, such as in a
queue.</t>
<t>Dropping transport segments (packets) in transit.</t>
<t>Marking transport segments (packets), such as using Explicit
Congestion Control<xref target="RFC3168"></xref> <xref
target="RFC4301"></xref> <xref target="RFC4774"></xref> <xref
target="RFC6040"></xref> <xref target="RFC6679"></xref>.</t>
</list>Increased network latency is used as an implicit signal of
congestion. E.g., in TCP additional delay can affect ACK Clocking and
has the result of reducing the rate of transmission of new data. In
the Real Time Protocol (RTP), network latency impacts the
RTCP-reported RTT and increased latency can trigger a sender to adjust
its rate. Methods such as Low Extra Delay Background Transport
(LEDBAT) <xref target="RFC6817"></xref> assume increased latency as a
primary signal of congestion. Appropriate use of delay-based methods
and the implications of AQM presently remains an area for further
research.</t>
<t>It is essential that all Internet hosts respond to loss <xref
target="RFC5681"> </xref>, <xref target="RFC5405"></xref><xref
target="RFC4960"></xref><xref target="RFC4340"></xref>. Packet
dropping by network devices that are under load has two effects: It
protects the network, which is the primary reason that network devices
drop packets. The detection of loss also provides a signal to a
reliable transport (e.g. TCP, SCTP) that there is potential congestion
using a pragmatic heuristic; "when the network discards a message in
flight, it may imply the presence of faulty equipment or media in a
path, and it may imply the presence of congestion. To be conservative,
a transport must assume it may be the latter." Unreliable transports
(e.g. using UDP) need to similarly react to loss <xref
target="RFC5405"></xref></t>
<t>Network devices SHOULD use an AQM algorithm to determine the
packets that are marked or discarded due to congestion. Procedures for
dropping or marking packets within the network need to avoid
increasing synchronisation events, and hence randomness SHOULD be
introduced in the algorithms that generate these congestion signals to
the endpoints.</t>
<t>Loss also has an effect on the efficiency of a flow and can
significantly impact some classes of application. In reliable
transports the dropped data must be subsequently retransmitted. While
other applications/transports may adapt to the absence of lost data,
this still implies inefficient use of available capacity and the
dropped traffic can affect other flows. Hence, congestion signalling
by loss is not entirely positive; it is a necessary evil.</t>
<section anchor="ECN" title="AQM and ECN">
<t>Explicit Congestion Notification (ECN) <xref
target="RFC4301"></xref> <xref target="RFC4774"></xref> <xref
target="RFC6040"></xref> <xref target="RFC6679"></xref> is a
network-layer function that allows a transport to receive network
congestion information from a network device without incurring the
unintended consequences of loss. ECN includes both transport
mechanisms and functions implemented in network devices, the latter
rely upon using AQM to decider when and whether to ECN-mark.</t>
<t>Congestion for ECN-capable transports is signalled by a network
device setting the "Congestion Experienced (CE)" codepoint in the IP
header. This codepoint is noted by the remote receiving end point
and signalled back to the sender using a transport protocol
mechanism, allowing the sender to trigger timely congestion control.
The decision to set the CE codepoint requires an AQM algorithm
configured with a threshold. Non-ECN capable flows (the default) are
dropped under congestion.</t>
<t>Network devices SHOULD use an AQM algorithm that marks
ECN-capable traffic when making decisions about the response to
congestion. Network devices need to implement this method by marking
ECN-capable traffic or by dropping non-ECN-capable traffic.</t>
<t>Safe deployment of ECN requires that network devices drop
excessive traffic, even when marked as originating from an
ECN-capable transport. This is a necessary safety precaution
because:</t>
<t><list style="numbers">
<t>A non-conformant, broken or malicious receiver could conceal
an ECN mark, and not report this to the sender;</t>
<t>A non-conformant, broken or malicious sender could ignore a
reported ECN mark, as it could ignore a loss without using
ECN;</t>
<t>A malfunctioning or non-conforming network device may "hide"
an ECN mark (or fail to correctly set the ECN codepoint at an
egress of a network tunnel).</t>
</list>In normal operation, such cases should be very uncommon,
however overload protection is desirable to protect traffic from
misconfigured or malicious use of ECN (e.g. a denial-of-service
attack that generates ECN-capable traffic that is unresponsive to
CE-marking).</t>
<t>An AQM algorithm that supports ECN needs to define the threshold
and algorithm for ECN-marking. This threshold MAY differ from that
used for dropping packets that are not marked as ECN-capable, and
SHOULD be configurable.</t>
<t>Network devices SHOULD use an algorithm to drop excessive traffic
(e.g. at some level above the threshold for CE-marking), even when
the packets are marked as originating from an ECN-capable
transport.</t>
</section>
</section>
<section anchor="autotuning"
title="AQM algorithms deployed SHOULD NOT require operational tuning">
<t>A number of AQM algorithms have been proposed. Many require some
form of tuning or setting of parameters for initial network
conditions. This can make these algorithms difficult to use in
operational networks.</t>
<t>AQM algorithms need to consider both "initial conditions" and
"operational conditions". The former includes values that exist before
any experience is gathered about the use of the algorithm, such as the
configured speed of interface, support for full duplex communication,
interface MTU and other properties of the link. The latter includes
information observed from monitoring the size of the queue,
experienced queueing delay, rate of packet discard, etc.</t>
<t>This document therefore specifies that AQM algorithms that are
proposed for deployment in the Internet have the following
properties:</t>
<t><list style="symbols">
<t>SHOULD NOT require tuning of initial or configuration
parameters. An algorithm needs to provide a default behaviour that
auto-tunes to a reasonable performance for typical network
operational conditions. This is expected to ease deployment and
operation. Initial conditions, such as the interface rate and MTU
size or other values derived from these, MAY be required by an AQM
algorithm.</t>
<t>MAY support further manual tuning that could improve
performance in a specific deployed network. Algorithms that lack
such variables are acceptable, but if such variables exist, they
SHOULD be externalized (made visible to the operator). Guidance
needs to be provided on the cases where auto-tuning is unlikely to
achieve satisfactory performance and to identify the set of
parameters that can be tuned. For example, the expected response
of an algorithm may need to be configured to accommodate the
largest expected Path RTT, since this value can not be known at
initialisation. This guidance is expected to enable the algorithm
to be deployed in networks that have specific characteristics
(paths with variable/larger delay; networks where capacity is
impacted by interactions with lower layer mechanisms, etc).</t>
<t>MAY provide logging and alarm signals to assist in identifying
if an algorithm using manual or auto-tuning is functioning as
expected. (e.g., this could be based on an internal consistency
check between input, output, and mark/drop rates over time). This
is expected to encourage deployment by default and allow operators
to identify potential interactions with other network
functions.</t>
</list>Hence, self-tuning algorithms are to be preferred. Algorithms
recommended for general Internet deployment by the IETF need to be
designed so that they do not require operational (especially manual)
configuration or tuning.</t>
</section>
<section title="AQM algorithms SHOULD respond to measured congestion, not application profiles.">
<t>Not all applications transmit packets of the same size. Although
applications may be characterized by particular profiles of packet
size this should not be used as the basis for AQM (see next section).
Other methods exist, e.g. Differentiated Services queueing,
Pre-Congestion Notification (PCN) <xref target="RFC5559"></xref>, that
can be used to differentiate and police classes of application.
Network devices may combine AQM with these traffic classification
mechanisms and perform AQM only on specific queues within a network
device.</t>
<t>An AQM algorithm should not deliberately try to prejudice the size
of packet that performs best (i.e. Preferentially drop/mark based only
on packet size). Procedures for selecting packets to mark/drop SHOULD
observe the actual or projected time that a packet is in a queue
(bytes at a rate being an analog to time). When an AQM algorithm
decides whether to drop (or mark) a packet, it is RECOMMENDED that the
size of the particular packet should not be taken into account <xref
target="RFC7141"></xref>.</t>
<t>Applications (or transports) generally know the packet size that
they are using and can hence make their judgments about whether to use
small or large packets based on the data they wish to send and the
expected impact on the delay or throughput, or other performance
parameter. When a transport or application responds to a dropped or
marked packet, the size of the rate reduction should be proportionate
to the size of the packet that was sent <xref
target="RFC7141"></xref>.</t>
<t>AQM-enabled system MAY instantiate different instances of an AQM
algorithm to be applied within the same traffic class. Traffic classes
may be differentiated based on an Access Control List (ACL), the
packet Differentiated Services Code Point (DSCP) <xref
target="RFC5559"></xref>, enabling use of the ECN field (i.e. any of
ECT(0), ECT(1) or CE)<xref target="RFC3168"></xref> <xref
target="RFC4774"> </xref>, a multi-field (MF) classifier that combines
the values of a set of protocol fields (e.g. IP address, transport,
ports) or an equivalent codepoint at a lower layer. This
recommendation goes beyond what is defined in RFC 3168, by allowing
that an implementation MAY use more than one instance of an AQM
algorithm to handle both ECN-capable and non-ECN-capable packets.</t>
</section>
<section anchor="alltraffic"
title="AQM algorithms SHOULD NOT be dependent on specific transport protocol behaviours">
<t>In deploying AQM, network devices need to support a range of
Internet traffic and SHOULD NOT make implicit assumptions about the
characteristics desired by the set transports/applications the network
supports. That is, AQM methods should be opaque to the choice of
transport and application.</t>
<t>AQM algorithms are often evaluated by considering <xref
target="RFC0793">TCP</xref> with a limited number of applications.
Although TCP is the predominant transport in the Internet today, this
no longer represents a sufficient selection of traffic for
verification. There is significant use of <xref
target="RFC0768">UDP</xref> in voice and video services, and some
applications find utility in <xref target="RFC4960">SCTP</xref> and
<xref target="RFC4340"> DCCP </xref>. Hence, AQM algorithms should
also demonstrate operation with transports other than TCP and need to
consider a variety of applications. Selection of AQM algorithms also
needs to consider use of tunnel encapsulations that may carry traffic
aggregates.</t>
<t>AQM algorithms SHOULD NOT target or derive implicit assumptions
about the characteristics desired by specific transports/applications.
Transports and applications need to respond to the congestion signals
provided by AQM (i.e. dropping or ECN-marking) in a timely manner
(within a few RTT at the latest).</t>
</section>
<section anchor="tcpcc"
title="Interactions with congestion control algorithms">
<t>Applications and transports need to react to received implicit or
explicit signals that indicate the presence of congestion. This
section identifies issues that can impact the design of transport
protocols when using paths that use AQM.</t>
<t>Transport protocols and applications need timely signals of
congestion. The time taken to detect and respond to congestion is
increased when network devices queue packets in buffers. It can be
difficult to detect tail losses at a higher layer and this may
sometimes require transport timers or probe packets to detect and
respond to such loss. Loss patterns may also impact timely detection,
e.g. the time may be reduced when network devices do not drop long
runs of packets from the same flow.</t>
<t>A common objective of an elastic transport congestion control
protocol is to allow an application to deliver the maximum rate of
data without inducing excessive delays when packets are queued in a
buffers within the network. To achieve this, a transport should try to
operate at rate below the inflexion point of the load/delay curve (the
bend of what is sometimes called a "hockey-stick" curve) <xref
target="Jain94"></xref>. When the congestion window allows the load to
approach this bend, the end-to-end delay starts to rise – a
result of congestion, as packets probabilistically arrive at
non-overlapping times. On the one hand, a transport that operates
above this point can experience congestion loss and could also trigger
operator activities, such as those discussed in <xref
target="RFC6057"></xref>. On the other hand, a flow may achieve both
near-maximum throughput and low latency when it operates close to this
knee point, with minimal contribution to router congestion. Choice of
an appropriate rate/congestion window can therefore significantly
impact the loss and delay experienced by a flow and will impact other
flows that share a common network queue.</t>
<t>Some applications may send less than permitted by the congestion
control window (or rate). Examples include multimedia codecs that
stream at some natural rate (or set of rates) or an application that
is naturally interactive (e.g., some web applications, gaming,
transaction-based protocols). Such applications may have different
objectives. They may not wish to maximize throughput, but may desire a
lower loss rate or bounded delay.</t>
<t>The correct operation of an AQM-enabled network device MUST NOT
rely upon specific transport responses to congestion signals.</t>
</section>
<section anchor="research" title="The need for further research">
<t><xref target="RFC2309">The second recommendation of </xref> called
for further research into the interaction between network queues and
host applications, and the means of signaling between them. This
research has occurred, and we as a community have learned a lot.
However, we are not done.</t>
<t>We have learned that the problems of congestion, latency and
buffer-sizing have not gone away, and are becoming more important to
many users. A number of self-tuning AQM algorithms have been found
that offer significant advantages for deployed networks. There is also
renewed interest in deploying AQM and the potential of ECN.</t>
<t>In 2013, an obvious example of further research is the need to
consider the use of Map/Reduce applications in data centers; do we
need to extend our taxonomy of TCP/SCTP sessions to include not only
"mice" and "elephants", but "lemmings"? "Lemmings" are flash crowds of
"mice" that the network inadvertently try to signal to as if they were
elephant flows, resulting in head of line blocking in data center
applications.</t>
<t>Examples of other required research include:</t>
<t><list style="symbols">
<t>Research into new AQM and scheduling algorithms.</t>
<t>Appropriate use of delay-based methods and the implications of
AQM.</t>
<t>Research into the use of and deployment of ECN alongside
AQM.</t>
<t>Tools for enabling AQM (and ECN) deployment and measuring the
performance.</t>
<t>Methods for mitigating the impact of non-conformant and
malicious flows.</t>
<t>Research to understand the implications of using new network
and transport methods on applications.</t>
</list>Hence, this document therefore reiterates the call of RFC
2309: we need continuing research as applications develop.</t>
</section>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This memo asks the IANA for no new parameters.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>While security is a very important issue, it is largely orthogonal to
the performance issues discussed in this memo.</t>
<t>Many deployed network devices use queueing methods that allow
unresponsive traffic to capture network capacity, denying access to
other traffic flows. This could potentially be used as a
denial-of-service attack. This threat could be reduced in network
devices deploy AQM or some form of scheduling. We note, however, that a
denial-of-service attack that results in unresponsive traffic flows may
be indistinguishable from other traffic flows (e.g. tunnels carrying
aggregates of short flows, high-rate isochronous applications). New
methods therefore may remain vulnerable, and this document recommends
that ongoing research should consider ways to mitigate such attacks.</t>
</section>
<section anchor="Privacy" title="Privacy Considerations">
<t>This document, by itself, presents no new privacy issues.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>The original recommendation in <xref target="RFC2309"></xref> was
written by the End-to-End Research Group, which is to say Bob Braden,
Dave Clark, Jon Crowcroft, Bruce Davie, Steve Deering, Deborah Estrin,
Sally Floyd, Van Jacobson, Greg Minshall, Craig Partridge, Larry
Peterson, KK Ramakrishnan, Scott Shenker, John Wroclawski, and Lixia
Zhang. This is an edited version of that document, with much of its text
and arguments unchanged.</t>
<t>The need for an updated document was agreed to in the tsvarea meeting
at IETF 86. This document was reviewed on the aqm@ietf.org list.
Comments were received from Colin Perkins, Richard Scheffenegger, Dave
Taht, John Leslie, David Collier-Brown and many others.</t>
<t>Gorry Fairhurst was in part supported by the European Community under
its Seventh Framework Programme through the Reducing Internet Transport
Latency (RITE) project (ICT-317700).</t>
</section>
</middle>
<back>
<!-- references split to informative and normative -->
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include="reference.RFC.3168" ?>
<?rfc include="reference.RFC.6679" ?>
<?rfc include="reference.RFC.4301" ?>
<?rfc include="reference.RFC.4774" ?>
<?rfc include="reference.RFC.5405"
?>
<?rfc include="reference.RFC.5681"
?>
<?rfc include="reference.RFC.6040"
?>
<?rfc include="reference.RFC.7141"
?>
</references>
<references title="Informative References">
<?rfc include="reference.RFC.0768" ?>
<?rfc include="reference.RFC.0791" ?>
<?rfc include="reference.RFC.0793" ?>
<?rfc include="reference.RFC.0896" ?>
<?rfc include="reference.RFC.0970" ?>
<?rfc include="reference.RFC.1122" ?>
<?rfc include="reference.RFC.1633"?>
<?rfc include="reference.RFC.2309"?>
<?rfc include="reference.RFC.2460" ?>
<?rfc include="reference.RFC.2474" ?>
<?rfc include="reference.RFC.2475"?>
<?rfc include="reference.RFC.4340" ?>
<?rfc include="reference.RFC.4960" ?>
<?rfc include="reference.RFC.5348"?>
<?rfc include="reference.RFC.5559"?>
<?rfc include="reference.RFC.6057" ?>
<?rfc include="reference.RFC.6817" ?>
<?rfc include="reference.RFC.6789" ?>
<reference anchor="Floyd91">
<front>
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<date month="October" year="1991" />
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<organization></organization>
</author>
<date month="August" year="1995" />
</front>
<seriesInfo name="IEEE/ACM Transactions on Networking" value="" />
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<front>
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<author fullname="W. Willinger" initials="W" surname="Willinger">
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<date month="August" year="1995" />
</front>
<seriesInfo name="SIGCOMM Symposium proceedings on Communications architectures and protocols"
value="" />
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<reference anchor="Jacobson88">
<front>
<title>Congestion Avoidance and Control</title>
<author fullname="Van Jacobson" initials="V" surname="Jacobson">
<organization>Lawrence Berkeley Network Labs</organization>
</author>
<date month="August" year="1988" />
</front>
<seriesInfo name="SIGCOMM Symposium proceedings on Communications architectures and protocols"
value="" />
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<reference anchor="Lakshman96">
<front>
<title>The Drop From Front Strategy in TCP Over ATM and Its
Interworking with Other Control Features</title>
<author fullname="T. V. Lakshman" initials="TV" surname="Lakshman">
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<organization></organization>
</author>
<date year="1996" />
</front>
<seriesInfo name="IEEE Infocomm" value="" />
</reference>
<reference anchor="Leland94">
<front>
<title>On the Self-Similar Nature of Ethernet Traffic (Extended
Version)</title>
<author fullname="W. Leland" initials="W" surname="Leland">
<organization></organization>
</author>
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<organization></organization>
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<organization></organization>
</author>
<author fullname="D. Wilson" initials="D" surname="Wilson">
<organization></organization>
</author>
<date month="February" year="1994" />
</front>
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<?rfc ?>
<section anchor="log" title="Change Log">
<t><list style="hanging">
<t hangText="Initial Version:">March 2013</t>
<t
hangText="Minor update of the algorithms that the IETF recommends SHOULD NOT require operational (especially manual) configuration or tuningdate:">April
2013</t>
<t
hangText="Major surgery. This draft is for discussion at IETF-87 and expected to be further updated.">July
2013</t>
<t
hangText="-00 WG Draft - Updated transport recommendations; revised deployment configuration section; numerous minor edits.">Oct
2013</t>
<t
hangText="-01 WG Draft - Updated transport recommendations; revised deployment configuration section; numerous minor edits.">Jan
2014 - Feedback from WG.</t>
<t hangText="-02 WG Draft - Minor edits">Feb 2014 - Mainly language
fixes.</t>
<t hangText="-03 WG Draft - Minor edits">Feb 2013 - Comments from
David Collier-Brown and David Taht.</t>
<t hangText="-04 WG Draft - Minor edits">May 2014 - Comments during
WGLC: Provided some introductory subsections to help people (with
subsections and better text). - Written more on the role scheduling.
- Clarified that ECN mark threshold needs to be configurable. -
Reworked your "knee" para. Various updates in response to
feedback.</t>
<t hangText="-05 WG Draft - Minor edits">June 2014 - New text added
to address further comments, and improve introduction - adding
context, reference to Conex, linking between sections, added text on
synchronisation.</t>
<t hangText="-06 WG Draft - Minor edits">July 2014 - Reorganised the
introduction following WG feedback to better explain how this
relates to the original goals of RFC2309. Added item on packet
bursts. Various minor corrections incorporated - no change to main
recommendations.</t>
<t hangText="-07 WG Draft - Minor edits">July 2014 - Replaced ID REF
by RFC 7141. Changes made to introduction following inputs from Wes
Eddy and John Leslie. Corrections and additions proposed by Bob
Briscoe. </t>
</list></t>
</section>
</back>
</rfc>
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