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Versions: (draft-baker-aqm-recommendation) 00 01 02 03 04 05 06 07 08 09 10 11 RFC 7567

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: January 2, 2015                                    July 1, 2014


         IETF Recommendations Regarding Active Queue Management
                    draft-ietf-aqm-recommendation-06

Abstract

   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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on January 2, 2015.

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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   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Congestion Collapse . . . . . . . . . . . . . . . . . . .   3
     1.2.  Active Queue Management to Manage Latency . . . . . . . .   3
     1.3.  Document Overview . . . . . . . . . . . . . . . . . . . .   4
     1.4.  Requirements Language . . . . . . . . . . . . . . . . . .   5
   2.  The Need For Active Queue Management  . . . . . . . . . . . .   5
     2.1.  AQM and Multiple Queues . . . . . . . . . . . . . . . . .   9
     2.2.  AQM and Explicit Congestion Marking (ECN) . . . . . . . .   9
     2.3.  AQM and Buffer Size . . . . . . . . . . . . . . . . . . .   9
   3.  Managing Aggressive Flows . . . . . . . . . . . . . . . . . .  10
   4.  Conclusions and Recommendations . . . . . . . . . . . . . . .  13
     4.1.  Operational deployments SHOULD use AQM procedures . . . .  14
     4.2.  Signaling to the transport endpoints  . . . . . . . . . .  14
       4.2.1.  AQM and ECN . . . . . . . . . . . . . . . . . . . . .  15
     4.3.  AQM algorithms deployed SHOULD NOT require operational
           tuning  . . . . . . . . . . . . . . . . . . . . . . . . .  17
     4.4.  AQM algorithms SHOULD respond to measured congestion, not
           application profiles. . . . . . . . . . . . . . . . . . .  18
     4.5.  AQM algorithms SHOULD NOT be dependent on specific
           transport protocol behaviours . . . . . . . . . . . . . .  19
     4.6.  Interactions with congestion control algorithms . . . . .  19
     4.7.  The need for further research . . . . . . . . . . . . . .  20
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  21
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  21
   7.  Privacy Considerations  . . . . . . . . . . . . . . . . . . .  21
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  21
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  22
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  22
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  23
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  26
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  27

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.



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   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 [RFC0896][RFC0970], and is technically called "congestive
   collapse" and was a key focus of RFC2309.

   Since 1998, when RFC2309 was written, the Internet has become used
   for a variety traffic of traffic.  In the current Internet and 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.  This document replaces RFC2309, and while there is
   still a need to avoid congestion collapse, there is now also a focus
   on reducing network latency using the same technology.

1.1.  Congestion Collapse

   The original fix for Internet meltdown was provided by Van Jacobsen.
   Beginning in 1986, Jacobsen developed the congestion avoidance
   mechanisms [Jacobson88] that are now required for implementations of
   the Transport Control Protocol (TCP) [RFC0768] [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 Internet.  Similar
   algorithms are specified for other non-TCP transports.

   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 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.

1.2.  Active Queue Management to Manage Latency

   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



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   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.

   Active Queue Management (AQM) is a technology that manages the size
   of the queues that build in network buffers.  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.

1.3.  Document Overview

   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", AQM.  Section 2 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.

   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, 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) [RFC6789] offer a framework [CONEX] 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.




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   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 Round
   Trip Time (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].

1.4.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

2.  The Need For Active Queue Management

   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
   the method.

   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:



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   1.  Full Queues

       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.

       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 [Leland94].  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
       [Flo94], [Zha90]

       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.

   2.  Lock-Out

       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 [Flo92].

   3.  Mitigating the Impact of Packet Bursts

       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.




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   4.  Control loop synchronisation

       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

   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 [Lakshman96].  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
   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
   drop.

   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



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       throughput.  Second, unnecessary packet drops represent a waste
       of network capacity on the path before the drop point.

       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, 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.

   3.  Avoid lock-out behavior

       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.  AQM can
       therefore be combined with a scheduling mechanism that divides
       network traffic between multiple queues (section 2.1).

   4.  Reduce the probability of control loop synchronisation

       The probability of network control loop synchronisation can be
       reduced by introducing randomness in the AQM functions used by
       network devices that trigger congestion avoidance at the sending
       host.








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2.1.  AQM and Multiple Queues

   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) [Dem90], including Weighted Fair Queuing (WFQ),
   Stochastic Fairness Queueing (SFQ) [McK90] Deficit Round Robin (DRR)
   [Shr96], and/or a Class-Based Queue scheduling algorithm such as CBQ
   [Floyd95].  Hierarchical queues may also be used e.g., as a part of a
   Hierarchical Token Bucket (HTB), or Hierarchical Fair Service Curve
   (HFSC) [Sto97] . 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).

   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 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.

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

2.2.  AQM and Explicit Congestion Marking (ECN)

   An AQM method may use Explicit Congestion Notification (ECN)
   [RFC3168] 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 [ECN-Benefit].  Section 4.2.1 describes
   some of the benefits of using ECN with AQM.

2.3.  AQM and Buffer Size

   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.  One the one hand, the optimum buffer size is a
   function of operational requirements and should generally be sized to



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   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.

   The simplest mechanism starts with a new or building session
   attacking a queue that is full.  One or more sessions, following
   algorithms similar to those of [RFC5681], maximizes its effective
   window, maximizing its impact on a queue somewhere in the network and
   the effect of that queue on both its own latency and that of
   competing sessions.  It also maximizes the probability of loss from
   that queue.  A new session, sending its initial burst, has an
   enhanced probability of filling the remaining queue and dropping
   packets.  As a result, the new session can be effectively prevented
   from sharing the queue effectively for a period of many RTTs.  One
   objective of AQM is to minimize the effect of lock-out by minimizing
   mean queue depth and therefore the probability that competing
   sessions can materially prevent each other from performing well.
   Different types of traffic and deployment scenarios will lead to
   different requirements.

   AQM frees a designer from having to the limit buffer space to achieve
   acceptable performance, allowing allocation of sufficient buffering
   to satisfy the needs of the particular traffic pattern.  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 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].

   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 [RFC5348].  In this document, the
   term is used more generally to describe this and other algorithms
   that meet these goals.

   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



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   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.

   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, packet size, 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
       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
       [RFC4960] and DCCP [RFC4340].) or by incorporating their own
       congestion control in the application [RFC5405].
       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.

   3.  Non-TCP-friendly Transport Protocols




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       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.

       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 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.

       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.

   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
   granugranularity 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/



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   destination host pair (IP address); 4) a given source host or a given
   destination host, or various combinations of the above.

   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.

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.  The
   updated recommendations provided by this document are summarised as:

   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
       endpoints.

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

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

   5.  AQM algorithms SHOULD NOT interpret specific transport protocol
       behaviours.

   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.





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   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 research [Choi04] 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.

4.1.  Operational deployments SHOULD use AQM procedures

   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.

   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].  When
   scheduling is used AQM should be applied across the classes or flows
   as well as within each class or flow:

   o  AQM mechanisms need to control the overall queue sizes, to ensure
      that arriving bursts can be accommodated without dropping packets.

   o  AQM mechanisms need to allow combination with other mechanisms,
      such as scheduling, to allow implementation of policies for
      providing fairness between different flows.

   o  AQM should be used to control the queue size for each individual
      flow or class, so that they do not experience unnecessarily high
      delay.

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
      queue.

   o  Dropping transport segments (packets) in transit.



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   o  Marking transport segments (packets), such as using Explicit
      Congestion Control[RFC3168] [RFC4301] [RFC4774] [RFC6040]
      [RFC6679].

   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)
   [RFC6817] 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.

   It is essential that all Internet hosts respond to loss [RFC5681],
   [RFC5405][RFC4960][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, a transport must assume it may be
   the latter."  Unreliable transports (e.g. using UDP) need to
   similarly react to loss [RFC5405]

   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.

   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.

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,



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   the latter rely upon using AQM to decider when and whether to ECN-
   mark.

   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.

   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 a necessary safety precaution 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 "hide" an
       ECN mark (or fail to correctly set the ECN codepoint at an egress
       of a network tunnel).

   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).

   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.

   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.






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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 discard, etc.

   This document therefore specifies that AQM algorithms that are
   proposed for deployment in the Internet have the following
   properties:

   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 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.

   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 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).

   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.




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   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 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) [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 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 [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
   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 Differentiated Services Code Point (DSCP) [RFC5559],
   enabling use of the ECN field (i.e. any of ECT(0), ECT(1) or
   CE)[RFC3168] [RFC4774], 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.






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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
   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.

   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).  When



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   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 [RFC6057].  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.

   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.

   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.

   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.

   Examples of other required research include:

   o  Research into new AQM and scheduling algorithms.




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   o  Appropriate use of delay-based methods and the implications of
      AQM.

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

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

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

   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 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.

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



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   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 aqm@ietf.org list.
   Comments were received from Colin Perkins, Richard Scheffenegger,
   Dave Taht, John Leslie, David Collier-Brown and many others.

   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

   [Byte-pkt]
              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.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November
              2008.

   [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.






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   [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", .

   [CONEX]    Mathis, M. and B. Briscoe, "The Benefits to Applications
              of using Explicit Congestion Notification (ECN)", IETF
              (Work-in-Progress) draft-ietf-conex-abstract-mech, March
              2014.

   [Choi04]   Choi, Baek-Young., Moon, Sue., Zhang, Zhi-Li.,
              Papagiannaki, K., and C. Diot, "Analysis of Point-To-Point
              Packet Delay In an Operational Network", March 2004.

   [Dem90]    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.

   [ECN-Benefit]
              Welzl, M. and G. Fairhurst, "The Benefits to Applications
              of using Explicit Congestion Notification (ECN)", IETF
              (Work-in-Progress) , February 2014.

   [Flo92]    Floyd, S. and V. Jacobsen, "On Traffic Phase Effects in
              Packet-Switched Gateways", 1992.

   [Flo94]    Floyd, S. and V. Jacobsen, "The Synchronization of
              Periodic Routing Messages,
              http://ee.lbl.gov/papers/sync_94.pdf", 1994.

   [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.

   [Jacobson88]
              Jacobson, V., "Congestion Avoidance and Control", SIGCOMM
              Symposium proceedings on Communications architectures and
              protocols , August 1988.





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

   [Lakshman96]
              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.

   [Leland94]
              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
              1994.

   [McK90]    McKenney, PE. and G. Varghese, "Stochastic Fairness
              Queuing",
              http://www2.rdrop.com/~paulmck/scalability/paper/
              sfq.2002.06.04.pdf , 1990.

   [Nic12]    Nichols, K., "Controlling Queue Delay", Communications of
              the ACM Vol. 55 No. 11, July, 2012, pp.42-50. , July 2002.

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

   [RFC0791]  Postel, J., "Internet Protocol", STD 5, RFC 791, September
              1981.

   [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.







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   [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
              1998.

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

   [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.

   [RFC6789]  Briscoe, B., Woundy, R., and A. Cooper, "Congestion
              Exposure (ConEx) Concepts and Use Cases", RFC 6789,
              December 2012.

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

   [Shr96]    Shreedhar, M. and G. Varghese, "Efficient Fair Queueing
              Using Deficit Round Robin", IEEE/ACM Transactions on
              Networking Vol 4, No. 3 , July 1996.




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   [Sto97]    Stoica, I. and H. Zhang, "A Hierarchical Fair Service
              Curve algorithm for Link sharing, real-time and priority
              services", ACM SIGCOMM , 1997.

   [Sut99]    Suter, B., "Buffer Management Schemes for Supporting TCP
              in Gigabit Routers with Per-flow Queueing", IEEE Journal
              on Selected Areas in Communications Vol. 17 Issue 6, June,
              1999, pp. 1159-1169. , 1999.

   [Willinger95]
              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.

   [Zha90]    Zhang, L. and D. Clark, "Oscillating Behavior of Network
              Traffic: A Case Study Simulation,
              http://groups.csail.mit.edu/ana/Publications/Zhang-DDC-
              Oscillating-Behavior-of-Network-Traffic-1990.pdf", 1990.

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.
      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.

   -02 WG Draft - Minor edits  Feb 2014 - Mainly language fixes.

   -03 WG Draft - Minor edits  Feb 2013 - Comments from David Collier-
      Brown and David Taht.





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   -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.

   -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.

   -06 WG Draft - Minor edits  July 2014 - Reorganised the introduction
      following WG feedback to better explain how this relates to the
      original goals of RGFC2309.  Added item on packet bursts.  Various
      minor corrections incorporatd - no change to main recommendations.

Authors' Addresses

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

   Email: fred@cisco.com


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

   Email: gorry@erg.abdn.ac.uk
   URI:   http://www.erg.abdn.ac.uk
















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