Network Working Group                                      F. Baker, Ed.
Internet-Draft                                             Cisco Systems
Obsoletes: 2309 (if approved)                          G. Fairhurst, Ed.
Intended status: Best Current Practice            University of Aberdeen
Expires: August 3, 2014                                 January 30, 2014

         IETF Recommendations Regarding Active Queue Management


   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.

   The note largely repeats the recommendations of RFC 2309, updated
   after fifteen years of experience and new research.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 3, 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

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   ( 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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Requirements Language . . . . . . . . . . . . . . . . . .   4
   2.  The Need For Active Queue Management  . . . . . . . . . . . .   4
   3.  Managing Aggressive Flows . . . . . . . . . . . . . . . . . .   8
   4.  Conclusions and Recommendations . . . . . . . . . . . . . . .  10
     4.1.  Operational deployments SHOULD  use AQM procedures  . . .  11
     4.2.  Signaling to the transport endpoints  . . . . . . . . . .  11
       4.2.1.  AQM and ECN . . . . . . . . . . . . . . . . . . . . .  12
     4.3.  AQM algorithms deployed SHOULD NOT require operational
           tuning  . . . . . . . . . . . . . . . . . . . . . . . . .  13
     4.4.  AQM algorithms SHOULD respond to measured congestion, not
           application profiles. . . . . . . . . . . . . . . . . . .  14
     4.5.  AQM algorithms SHOULD NOT be dependent on specific
           transport protocol behaviours . . . . . . . . . . . . . .  15
     4.6.  Interactions with congestion control algorithms . . . . .  15
     4.7.  The need for further research . . . . . . . . . . . . . .  16
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  17
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  17
   7.  Privacy Considerations  . . . . . . . . . . . . . . . . . . .  18
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  18
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  18
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  18
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  19
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  21
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  22

1.  Introduction

   The Internet protocol architecture is based on a connectionless end-
   to-end packet service using the Internet Protocol, whether IPv4
   [RFC0791] or IPv6 [RFC2460].  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

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   mid 1980s [RFC0896][RFC0970], and is technically called "congestive

   The original fix for Internet meltdown was provided by Van Jacobsen.
   Beginning in 1986, Jacobsen developed the congestion avoidance
   mechanisms that are now required in TCP implementations [Jacobson88]
   [RFC1122].  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., marked or dropped
   packets) from the network.  It is primarily these TCP congestion
   avoidance algorithms that prevent the congestive collapse of today's

   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 [RFC5681], 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.

   It is useful to distinguish between two classes of algorithms related
   to congestion control: "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 AQM mechanisms
   are closely related, they address different performance issues.

   This memo highlights two performance issues:

   The first issue is the need for an advanced form of queue management
   that we call "active queue management."  Section 2 summarizes the
   benefits that active queue management can bring.  A number of Active
   Queue Management (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.

   The second issue, discussed in Section 3 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, there is no consensus solution to
   controlling congestion caused by such aggressive flows; significant

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   research and engineering will be required before any solution will be
   available.  It is imperative that this work be energetically pursued,
   to ensure the future stability of the Internet.

   Section 4 concludes the memo with a set of recommendations to the
   Internet community concerning these topics.

   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 RTT or
   a small number of RTTs, for elastic traffic [RFC1633].

   [RFC2309] resulted from past discussions of end-to-end performance,
   Internet congestion, and Random Early Discard (RED) in the End-to-End
   Research Group of the Internet Research Task Force (IRTF).  This
   update results from experience with this and other algorithms, and
   the AQM discussion within the IETF[AQM-WG].

1.1.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in [RFC2119].

2.  The Need For Active Queue Management

   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 two important drawbacks:

   1.  Lock-Out

       In some situations tail drop allows a single connection or a few
       flows to monopolize queue space, preventing other connections
       from getting room in the queue.  This "lock-out" phenomenon is
       often the result of synchronization or other timing effects.

   2.  Full Queues

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       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

       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.  Even though TCP constrains the congestion window of
       a flow, packets often arrive at network devices in bursts
       [Leland94].  If the queue is full or almost full, an arriving
       burst will cause multiple packets to be dropped.  This can result
       in a global synchronization of flows throttling back, followed by
       a sustained period of lowered link utilization, reducing overall

       The point 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 the transmission of bursty
       data.  Normally small queues are prefered 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.

   Besides tail drop, two alternative queue disciplines that can be
   applied when a queue becomes full are "random drop on full" or "drop
   front on full".  Under the random drop on full discipline, a 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 the queue is full and a new packet
   arrives.  Under the "drop front on full" discipline [Lakshman96], the
   network device drops the packet at the front of the queue when the
   queue is full and a new packet arrives.  Both of these solve the
   lock-out problem, but neither solves the full-queues problem
   described above.

   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

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   provide a critical mechanism indicating congestion notification to
   hosts.  The solution to the full-queues problem is for network
   devices to drop 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 packets before buffers overflow,
   AQM allows network devices to control when and how many packets to

   In summary, an active queue management mechanism can provide the
   following advantages for responsive flows.

   1.  Reduce number of packets dropped in network devices

       Packet bursts are an unavoidable aspect of packet networks
       [Willinger95].  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.

       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
       possible waste of network capacity on the path before the drop

       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.

   2.  Provide a lower-delay interactive service

       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, Telnet
       traffic, or interactive audio-video sessions, whose subjective
       (and objective) performance is better when the end-to-end delay
       is low.

   3.  Avoid lock-out behavior

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       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.

       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 [Floyd91], and a flow that does not use congestion
       control may receive more capacity than a flow that does.  For
       example, a router may maintain per-flow state to achieve general
       fairness by a per-flow scheduling algorithm such as Fair Queueing
       (FQ) [Demers90], or a Class-Based Queue scheduling algorithm such
       as CBQ [Floyd95].

       In contrast, AQM is needed even for network devices that use per-
       flow scheduling algorithms such as FQ or class-based scheduling
       algorithms, such as CBQ.  This is because per-flow scheduling
       algorithms by themselves do not control the overall queue size or
       the size of individual queues.  AQM is needed to control the
       overall average queue sizes, so that arriving bursts can be
       accommodated without dropping packets.  In addition, AQM should
       be used to control the queue size for each individual flow or
       class, so that they do not experience unnecessarily high delay.
       Therefore, AQM should be applied across the classes or flows as
       well as within each class or flow.

   In short, scheduling algorithms and queue management should be seen
   as complementary, not as replacements for each other.

   It is also 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.  One the one hand, 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.
   Different types of traffic and deployment scenarios will lead to
   different requirements.  On the other hand, the choice of AQM
   algorithm and associated parameters is a function of the way in which
   congestion is experienced and the required reaction to achieve

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   acceptable performance.  This latter topic is the primary topic of
   the following sections.

3.  Managing Aggressive Flows

   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 [RFC5681].

   We call a flow "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 [RFC5348].  In this document, the term is used more
   generally to describe this and other algorithms that meet these

   It is convenient to divide flows into three classes: (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

   1.  TCP-Friendly flows

       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, MTU, etc.).  This is
       described in the remainder of the document.

   2.  Non-Responsive Flows

       The User Datagram Protocol (UDP) [RFC0768] 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 [RFC5405].

       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

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       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 [RFC2309].

       In general, UDP-based applications need to incorporate effective
       congestion avoidance mechanisms [RFC5405].  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, SCTP, DCCP) or by incorporating their own congestion
       control in the application.  [RFC5405].

   3.  Non-TCP-friendly Transport Protocols

       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

       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.

       Another example could be an RTP/UDP video flow that uses an
       adaptive codec, but responds incompletely to indications of
       congestion or over responds over an excessively long time period.
       Such flows are unlikely to be responsive to congestion signals in
       a time frame 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.

       Tunneled traffic aggregates carrying multiple (short) TCP flows
       can be more aggressive than standard bulk TCP.  Applications

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       (e.g. web browsers and peer-to-peer file-sharing) have exploited
       this by opening multiple connections to the same endpoint.

   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.

   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, DSCP); 2) a source/
   destination host pair (IP addresses, DSCP); 3) a given source host or
   a given destination host.  We suggest that the source/destination
   host pair gives the most appropriate granularity in many
   circumstances.  However, it is possible that different vendors/
   providers could set different granularities for defining a flow (as a
   way of "distinguishing" themselves from one another), or that
   different granularities could 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.

4.  Conclusions and Recommendations

   The IRTF, in publishing [RFC2309], and the IETF in subsequent
   discussion, has developed a set of specific recommendations regarding
   the implementation and operational use of AQM procedures.  This
   document updates these to include:

   1.  Network devices SHOULD implement some AQM mechanism to manage
       queue lengths, reduce end-to-end latency, and avoid lock-out
       phenomena within the Internet.

   2.  Deployed AQM algorithms SHOULD support Explicit Congestion
       Notification (ECN) as well as loss to signal congestion to

   3.  The algorithms that the IETF recommends SHOULD NOT require
       operational (especially manual) configuration or tuning.

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   4.  AQM algorithms SHOULD respond to measured congestion, not
       application profiles.

   5.  AQM algorithms SHOULD NOT interpret specific transport protocol

   6.  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.

   7.  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.

   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.  However, 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 on.  To the
   contrary, available research [Papagiannaki] says that even high speed
   links in network cores that are normally very stable in depth and
   behavior experience occasional issues that need moderation.

4.1.  Operational deployments SHOULD use AQM procedures

   AQM procedures are designed to minimize delay 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.

4.2.  Signaling to the transport endpoints

   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:

   o  Delaying transport segments (packets) in flight, such as in a

   o  Dropping transport segments (packets) in transit.

   o  Marking transport segments (packets), such as using Explicit
      Congestion Control[RFC3168] [RFC4301] [RFC4774] [RFC6040]

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   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 [RFC2474] and [RFC2475].

   Increased network latency can be 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 RTP, network latency impacts the RTCP-reported RTT and increased
   latency can trigger a sender to adjust its rate.  Methods such as
   LEDBAT [RFC6817] assume increased latency as a primary signal of

   It is essential that all Internet hosts respond to loss [RFC5681],
   [RFC5405][RFC2960][RFC4340].  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 transport must the latter."
   Unreliable transports (e.g. using UDP) need to similarly react to
   loss [RFC5405]

   Network devices SHOULD use use an AQM algorithm to determine the
   packets that are marked or discarded due to congestion.

   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, loss is not
   entirely positive; it is a necessary evil.

4.2.1.  AQM and ECN

   Explicit Congestion Notification (ECN) [RFC4301] [RFC4774] [RFC6040]
   [RFC6679]. 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 whether to ECN-mark.

   Congestion for ECN-capable transports is signalled by a network
   device setting the "Congestion Experienced (CE)" codepoint in the IP

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   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.

   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.

   Safe deployment of ECN requires that network devices drop excessive
   traffic, even when marked as originating from an ECN-capable
   transport.  This is necessary because (1) A non-conformant, broken or
   malicious receiver could conceal an ECN mark, and not report this to
   the sender (2) A non-conformant, broken or malicious sender could
   ignore a reported ECN mark, as it could ignore a loss without using
   ECN (3) A malfunctioning or non-conforming network device may
   similarly "hide" an ECN mark.  In normal operation such cases should
   be very uncommon.

   Network devices SHOULD use an algorithm to drop excessive traffic,
   even when marked as originating from an ECN-capable transport.

4.3.  AQM algorithms deployed SHOULD NOT require operational tuning

   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.

   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 discards, etc.

   This document therefore recommends that AQM algorithm proposed for
   deployment in the Internet:

   o  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 conditions.  This
      is expected to ease deployment and operation.

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   o  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 autotuning is unlikely to achieve
      satisfactory performance and to identify the set of parameters
      that can be tuned.  This is expected to enable the algorithm to be
      deployed in networks that have specific characteristics (variable/
      larger delay; networks were capacity is impacted by interactions
      with lower layer mechanisms, etc)

   o  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.

   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.

4.4.  AQM algorithms SHOULD respond to measured congestion, not
      application profiles.

   Not all applications transmit packets of the same size.  Although
   applications may be characterised 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) [RFC5559], 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.

   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 actual or projected time 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 [Byte-pkt].

   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

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   marked packet, the size of the rate reduction should be proportionate
   to the size of the packet that was sent [Byte-pkt].

   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 DiffServ Code Point (DSCP) [RFC5559], setting of the ECN
   field[RFC3168] [RFC4774] or an equivalent codepoint at a lower layer.
   This recommendation goes beyond what is defined in RFC 3168, by
   allowing more than one instance of an AQM to handle both ECN-capable
   and non-ECN-capable packets.

4.5.  AQM algorithms SHOULD NOT be dependent on specific transport
      protocol behaviours

   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.

   AQM algorithms are often evaluated by considering TCP [RFC0793] 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 UDP [RFC0768] in voice and video services, and
   some applications find utility in SCTP [RFC4960] and DCCP [RFC4340].
   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.

   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).

4.6.  Interactions with congestion control algorithms

   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.

   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

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   difficult to detect tail losses at a higher layer and 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.

   A common objective is to deliver data from its source end point to
   its destination in the least possible time.  When speaking of TCP
   performance, the terms "knee" and "cliff" area defined by [Jain94].
   They respectively refer to the minimum congestion window that
   maximises throughput and the maximum congestion window that avoids
   loss.  An application that transmits at the rate determined by this
   window has the effect of maximizing the rate or throughput.  For the
   sender, exceeding the cliff is ineffective, as it (by definition)
   induces loss; operating at a point close to the cliff has a negative
   impact on other traffic and applications, triggering operator
   activities, such as those discussed in [RFC6057].  Operating below
   the knee reduces the throughput, since the sender fails to use
   available network capacity.  As a result, the behavior of any elastic
   transport congestion control algorithm designed to minimise delivery
   time should seek to use an effective window at or above the knee and
   well below the cliff.  Choice of an appropriate rate can
   significantly impact the loss and delay experienced not only by a
   flow, but by other flows that share the same queue.

   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 maximise throughput, but may desire
   a lower loss rate or bounded delay.

   The correct operation of an AQM-enabled network device MUST NOT rely
   upon specific transport responses to congestion signals.

4.7.  The need for further research

   The second recommendation of [RFC2309] 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.

   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.

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   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".  Where "Lemmings" are flash
   crowds of "mice" that the network inadvertently tries to signal to as
   if they were elephant flows, resulting in head of line blocking in
   data center applications.

   Examples of other required research include:

   o  Research into new AQM and scheduling algorithms.

   o  Research into the use of and deployment of ECN alongside AQM.

   o  Tools for enabling AQM (and ECN) deployment and measuring the

   o  Methods for mitigating the impact of non-conformant and malicious

   o  Research to understand the implications of using new network and
      transport methods on applications.

   Hence, this document therefore reiterates the call of RFC 2309: we
   need continuing research as applications develop.

5.  IANA Considerations

   This memo asks the IANA for no new parameters.

6.  Security Considerations

   While security is a very important issue, it is largely orthogonal to
   the performance issues discussed in this memo.

   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 may create unresponsive traffic flows that
   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.

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7.  Privacy Considerations

   This document, by itself, presents no new privacy issues.

8.  Acknowledgements

   The original recommendation in [RFC2309] 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.

   The need for an updated document was agreed to in the tsvarea meeting
   at IETF 86.  This document was reviewed on the list.
   Comments came from Colin Perkins, Richard Scheffenegger, and Dave

   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).

9.  References

9.1.  Normative References

              and Internet Engineering Task Force, Work in Progress,
              "Byte and Packet Congestion Notification (draft-ietf-
              tsvwg-byte-pkt-congest)", July 2013.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP", RFC
              3168, September 2001.

   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
              Internet Protocol", RFC 4301, December 2005.

   [RFC4774]  Floyd, S., "Specifying Alternate Semantics for the
              Explicit Congestion Notification (ECN) Field", BCP 124,
              RFC 4774, November 2006.

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   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

   [RFC6040]  Briscoe, B., "Tunnelling of Explicit Congestion
              Notification", RFC 6040, November 2010.

   [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.

9.2.  Informative References

   [AQM-WG]   "IETF AQM WG", .

              Demers, A., Keshav, S., and S. Shenker, "Analysis and
              Simulation of a Fair Queueing Algorithm, Internetworking:
              Research and Experience", SIGCOMM Symposium proceedings on
              Communications architectures and protocols , 1990.

   [Floyd91]  Floyd, S., "Connections with Multiple Congested Gateways
              in Packet-Switched Networks Part 1: One-way Traffic.",
              Computer Communications Review , October 1991.

   [Floyd95]  Floyd, S. and V. Jacobson, "Link-sharing and Resource
              Management Models for Packet Networks", IEEE/ACM
              Transactions on Networking , August 1995.

              Jacobson, V., "Congestion Avoidance and Control", SIGCOMM
              Symposium proceedings on Communications architectures and
              protocols , August 1988.

   [Jain94]   Jain, Raj., Ramakrishnan, KK., and Chiu. Dah-Ming,
              "Congestion avoidance scheme for computer networks", US
              Patent Office 5377327, December 1994.

              Lakshman, TV., Neidhardt, A., and T. Ott, "The Drop From
              Front Strategy in TCP Over ATM and Its Interworking with
              Other Control Features", IEEE Infocomm , 1996.

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              Leland, W., Taqqu, M., Willinger, W., and D. Wilson, "On
              the Self-Similar Nature of Ethernet Traffic (Extended
              Version)", IEEE/ACM Transactions on Networking , February

              Sprint ATL, KAIST, University of Minnesota, Sprint ATL,
              and Intel ResearchIETF, "Analysis of Point-To-Point Packet
              Delay In an Operational Network", IEEE Infocom 2004, March
              2004, <>.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              August 1980.

   [RFC0791]  Postel, J., "Internet Protocol", STD 5, RFC 791, September

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7, RFC
              793, September 1981.

   [RFC0896]  Nagle, J., "Congestion control in IP/TCP internetworks",
              RFC 896, January 1984.

   [RFC0970]  Nagle, J., "On packet switches with infinite storage", RFC
              970, December 1985.

   [RFC1122]  Braden, R., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122, October 1989.

   [RFC1633]  Braden, B., Clark, D., and S. Shenker, "Integrated
              Services in the Internet Architecture: an Overview", RFC
              1633, June 1994.

   [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.

   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", RFC 2460, December 1998.

   [RFC2474]  Nichols, K., Blake, S., Baker, F., and D. Black,
              "Definition of the Differentiated Services Field (DS
              Field) in the IPv4 and IPv6 Headers", RFC 2474, December

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   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, December 1998.

   [RFC2960]  Stewart, R., Xie, Q., Morneault, K., Sharp, C.,
              Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M.,
              Zhang, L., and V. Paxson, "Stream Control Transmission
              Protocol", RFC 2960, October 2000.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340, March 2006.

   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol", RFC
              4960, September 2007.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification", RFC
              5348, September 2008.

   [RFC5559]  Eardley, P., "Pre-Congestion Notification (PCN)
              Architecture", RFC 5559, June 2009.

   [RFC6057]  Bastian, C., Klieber, T., Livingood, J., Mills, J., and R.
              Woundy, "Comcast's Protocol-Agnostic Congestion Management
              System", RFC 6057, December 2010.

   [RFC6817]  Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
              "Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
              December 2012.

              Willinger, W., Taqqu, M., Sherman, R., Wilson, D., and V.
              Jacobson, "Self-Similarity Through High-Variability:
              Statistical Analysis of Ethernet LAN Traffic at the Source
              Level", SIGCOMM Symposium proceedings on Communications
              architectures and protocols , August 1995.

Appendix A.  Change Log

   Initial Version:  March 2013

   Minor update of the algorithms that the IETF recommends SHOULD NOT
   require operational (especially manual) configuration or tuningdate:

      April 2013

   Major surgery.  This draft is for discussion at IETF-87 and  expected
    to be further updated.

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      July 2013

   -00 WG Draft - Updated transport recommendations; revised deployment
   configuration section; numerous minor edits.
      Oct 2013

   -01 WG Draft - Updated transport recommendations; revised deployment
   configuration section; numerous minor edits.
      Jan 2014 - Feedback from WG.

Authors' Addresses

   Fred Baker (editor)
   Cisco Systems
   Santa Barbara, California  93117


   Godred Fairhurst (editor)
   University of Aberdeen
   School of Engineering
   Fraser Noble Building
   Aberdeen, Scotland  AB24 3UE


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