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Versions: (draft-baker-tsvwg-aqm-recommendation) 00 01 02 draft-ietf-aqm-recommendation

Network Working Group                                      F. Baker, Ed.
Internet-Draft                                             Cisco Systems
Obsoletes: 2309 (if approved)                             March 15, 2013
Intended status: Best Current Practice
Expires: September 16, 2013

         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 in routers, to
   improve the performance of today's Internet.  It also urges a
   concerted effort of research, measurement, and ultimate deployment of
   router mechanisms to protect the Internet from flows that are not
   sufficiently responsive to congestion notification.

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
   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 September 16, 2013.

Copyright Notice

   Copyright (c) 2013 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
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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect

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   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Requirements Language . . . . . . . . . . . . . . . . . .   4
   2.  The Need For Active Queue Management  . . . . . . . . . . . .   4
   3.  Managing Aggressive Flows . . . . . . . . . . . . . . . . . .   7
   4.  Conclusions and Recommendations . . . . . . . . . . . . . . .  10
     4.1.  Operational deployments SHOULD implement Active Queue
           Management procedures . . . . . . . . . . . . . . . . . .  10
     4.2.  Signaling to the endpoints of a session . . . . . . . . .  11
     4.3.  Active Queue Management algorithms deployed SHOULD NOT
           require operational tuning  . . . . . . . . . . . . . . .  12
     4.4.  Active Queue Management algorithms deployed SHOULD be
           effective on all common Internet traffic  . . . . . . . .  12
     4.5.  TCP and SCTP congestion control algorithms SHOULD
           maximize their use of available bandwidth without
           incurring loss or undue round trip delay  . . . . . . . .  12
     4.6.  The need for further research . . . . . . . . . . . . . .  12
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  13
   7.  Privacy Considerations  . . . . . . . . . . . . . . . . . . .  13
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13
   9.  Change Log  . . . . . . . . . . . . . . . . . . . . . . . . .  13
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  13
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  13
     10.2.  Informative References . . . . . . . . . . . . . . . . .  14
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  16

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

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   The original fix for Internet meltdown was provided by Van Jacobson.
   Beginning in 1986, Jacobson developed the congestion avoidance
   mechanisms that are now required in TCP implementations [Jacobson88]
   [RFC1122].  These mechanisms operate in the hosts to cause TCP
   connections to "back off" during congestion.  We say that TCP flows
   are "responsive" to congestion signals (i.e., 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 routers to
   complement the endpoint congestion avoidance mechanisms.

   It is useful to distinguish between two classes of router 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 router
   mechanisms are closely related, they address rather 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
   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
   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.

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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
   is most effective, on timescales of a single RTT or a small number of
   RTTs, for elastic traffic [RFC1633], but also impacts real time
   traffic generated by adaptive applications.

   [RFC2309] resulted from past discussions of end-to-end performance,
   Internet congestion, and RED in the End-to-End Research Group of the
   Internet Research Task Force (IRTF).  This update results from
   experience with that and other algorithms, and the Active Queue
   Management discussion within the IETF.

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 router queue lengths 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

       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 queue management's most important goal.

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       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 a flow's window size,
       packets often arrive at routers 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 throughput.

       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.  It should be clear why we would like to have normally-
       small queues in routers: we want to have 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 short, queue limits should not
       reflect the steady state queues we want maintained in the
       network; instead, they should reflect the size of bursts we need
       to absorb.

   Besides tail drop, two alternative queue disciplines that can be
   applied when the queue becomes full are "random drop on full" or
   "drop front on full".  Under the random drop on full discipline, a
   router 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
   router 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
   serve as a critical mechanism of congestion notification to end
   nodes.  The solution to the full-queues problem is for routers to
   drop packets before a queue becomes full, so that end nodes can
   respond to congestion before buffers overflow.  We call such a
   proactive approach "active queue management".  By dropping packets
   before buffers overflow, active queue management allows routers to
   control when and how many packets to drop.  The next section
   introduces RED, an active queue management mechanism that solves both
   problems listed above (given responsive flows).

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

   1.  Reduce number of packets dropped in routers

       Packet bursts are an unavoidable aspect of packet networks
       [Willinger95].  If all the queue space in a router is already
       committed to "steady state" traffic or if the buffer space is
       inadequate, then the router will have no ability to buffer
       bursts.  By keeping the average queue size small, active queue
       management will provide greater capacity to absorb naturally-
       occurring bursts without dropping packets.

       Furthermore, without active queue management, 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, an unnecessary global synchronization of flows
       cutting back can result in lowered average link utilization, and
       hence lowered network throughput.  Second, TCP recovers with more
       difficulty from a burst of packet drops than from a single packet
       drop.  Third, unnecessary packet drops represent a possible waste
       of bandwidth on the way to the drop point.

       We note that while Active Queue Management can manage queue
       lengths and reduce end- to-end latency even in the absence of
       end-to-end congestion control, Active Queue Management will be
       able to reduce packet dropping only in an environment that
       continues to be dominated by end-to-end congestion control.

   2.  Provide lower-delay interactive service

       By keeping the average queue size small, queue management will
       reduce the delays seen 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

       Active queue management can prevent lock-out behavior by ensuring
       that there will almost always be a buffer available for an
       incoming packet.  For the same reason, active queue management
       can prevent a router bias against low bandwidth but highly bursty

       It is clear that lock-out is undesirable because it constitutes a
       gross unfairness among groups of flows.  However, we stop short

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       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 router using
       queue management but only FIFO scheduling, two TCP flows may
       receive very different bandwidths simply because they have
       different round-trip times [Floyd91], and a flow that does not
       use congestion control may receive more bandwidth than a flow
       that does.  Per-flow state to achieve general fairness might be
       maintained by a per-flow scheduling algorithm such as Fair
       Queueing (FQ) [Demers90], or a class-based scheduling algorithm
       such as CBQ [Floyd95], for example.

       On the other hand, active queue management is needed even for
       routers 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 nothing to
       control the overall queue size or the size of individual queues.
       Active queue management is needed to control the overall average
       queue sizes, so that arriving bursts can be accommodated without
       dropping packets.  In addition, active queue management should be
       used to control the queue size for each individual flow or class,
       so that they do not experience unnecessarily high delays.
       Therefore, active queue management 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.

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 bandwidth is shared
   reasonably equitably among similarly situated flows.  The equitable
   sharing of bandwidth among flows depends on the fact that all flows
   are running basically the same congestion avoidance algorithms,
   conformant with the current TCP specification [RFC1122].

   Flows that behaves under congestion like a flow produced by a
   conformant TCP have come to be called "TCP Friendly" [RFC5348].  A
   TCP Friendly flow is responsive to congestion notification, and in
   steady-state it uses no more bandwidth than a conformant TCP running
   under comparable conditions (drop rate, RTT, MTU, etc.)

   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

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   Friendly.  The last two classes contain more aggressive flows that
   pose significant threats to Internet performance, as we will now

   o  Non-Responsive Flows

      There is a growing set of UDP-based applications whose congestion
      avoidance algorithms are inadequate or nonexistent (i.e, the flow
      does not throttle back upon receipt of congestion notification).
      Such UDP applications include streaming applications like packet
      voice and video, and also multicast bulk data transport [SRM96].
      If no action is taken, such unresponsive flows could lead to a new
      congestive collapse.

      In general, all UDP-based streaming applications should
      incorporate effective congestion avoidance mechanisms.  For
      example, recent research has shown the possibility of
      incorporating congestion avoidance mechanisms such as Receiver-
      driven Layered Multicast (RLM) within UDP-based streaming
      applications such as packet video [McCanne96] [Bolot94].  Further
      research and development on ways to accomplish congestion
      avoidance for streaming applications will be very important.

      However, it will also be important for the network to be able to
      protect itself against unresponsive flows, and mechanisms to
      accomplish this must be developed and deployed.  Deployment of
      such mechanisms would provide incentive for every streaming
      application to become responsive by incorporating its own
      congestion control.

   o  Non-TCP-Friendly Transport Protocols

      The second threat is posed by transport protocol implementations
      that are responsive to congestion notification but, either
      deliberately or through faulty implementations, are not TCP
      Friendly.  Such applications can grab an unfair share of the
      network bandwidth.

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      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 bandwidth than other TCP
      implementations; this would allow a vendor to claim to have a
      "faster TCP".  The logical consequence of such implementations
      would be a spiral of increasingly aggressive TCP implementations,
      leading back to the point where there is effectively no congestion
      avoidance and the Internet is chronically congested.

      Another example of such flows is RTP/UDP video data flows in which
      the application uses an adaptive codec.  Such data flows are not
      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 will moderate
      their speed, or will increase their speed if they determine
      bandwidth to be available.

      Note that there is a well-known way to achieve more aggressive TCP
      performance without even changing TCP: open multiple connections
      to the same place, as has been done in multiple Web browsers and
      in peer-to-peer applications such as BitTorrent.

   The projected increase in more aggressive flows of both these
   classes, as a fraction of total Internet traffic, clearly poses a
   threat to the future Internet.  There is an urgent need for
   measurements of current conditions and for further research into the
   various ways of managing such flows.  There are many difficult issues
   in identifying and isolating unresponsive or non-TCP-Friendly flows
   at an acceptable router overhead cost.  Finally, there is little
   measurement or simulation evidence available about the rate at which
   these threats are likely to be realized, or about the expected
   benefit of router algorithms for managing such flows.

   There is an issue about the appropriate granularity of a "flow".
   There are a few "natural" answers: 1) a TCP or UDP connection (source
   address/port, destination address/port); 2) a source/destination host
   pair; 3) a given source host or a given destination host.  We would
   guess that the source/destination host pair gives the most
   appropriate granularity in many circumstances.  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 we are
   dealing with more unresponsive flows at *some* granularity.  The

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   granularity of flows for congestion management is, at least in part,
   a policy question that needs to be addressed in the wider IETF

4.  Conclusions and Recommendations

   The IRTF, in developing [RFC2309], and the IETF in subsequent
   discussion, has developed a set of specific recommendations regarding
   the implementation and operational use of Active Queue Management
   procedures.  These include:

   1.  Internet routers SHOULD implement some active queue management
       mechanism to manage queue lengths, reduce end-to-end latency,
       reduce packet dropping, and avoid lock-out phenomena within the

   2.  Deployed Active Queue Management SHOULD use ECN as well as loss
       in signaling congestion to endpoints.

   3.  Active Queue Management algorithms deployed SHOULD NOT require
       operational (especially manual) configuration or tuning.

   4.  Active Queue Management algorithms deployed SHOULD be effective
       on all common Internet traffic, including traffic that uses TCP,
       SCTP, UDP, and DCCP as transports.

   5.  TCP and SCTP congestion control algorithms SHOULD maximize their
       use of available bandwidth without incurring loss or undue round
       trip delay when possible.

   6.  It is urgent to continue research, engineering, and measurement
       efforts contributing to the design of mechanisms to deal with
       flows that are unresponsive to congestion notification or are
       responsive but more aggressive than TCP.

   These recommendations are expressed using the word "SHOULD".  This is
   in recognition that there may be use cases unenvisaged in this
   document in which the recommendation dos 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 implement Active Queue Management

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   In short, Active Queue Management procedures are designed to minimize
   delay induced in the network by queues which have filled as a result
   of host behavior.  Marking and loss behaviors signal to the senders
   of data that network buffers are becoming unnecessarily full, and
   they would do well to moderate their behavior.

4.2.  Signaling to the endpoints of a session

   Means of signaling to an endpoint regarding its effect on the network
   and how it might consider adapting include, at least:

   o  Delaying data segments in flight, such as in a queue, which
      affects Ack Clocking and as a result the transmission of new data.

   o  Marking traffic, such as using Explicit Congestion
      Control[RFC3168] [RFC4301] [RFC4774] [RFC6040]

   o  Dropping traffic in transit.

   The use of advanced scheduling mechanisms, such as priority queuing,
   classful queuing, and fair queuing, is often effective in networks to
   help a network to serve the needs of an application.  It can be used
   to manage traffic passing a choke point.  This is discussed in
   [RFC2474] and [RFC2475].  They are used operationally when an
   operator considers it important to do so.

   Loss has two effects.  It protects the network, which is the primary
   reason the network imposes it.  Its use as a signal to TCP or SCTP is
   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.  Presume the
   latter."  However, it also has an effect on the efficiency of the
   data flow.  The data in question must be retransmitted, or its
   absence must otherwise be adapted to by the application in question,
   which implies at least inefficient use of available bandwidth and may
   affect other data flows.  Hence, loss is not entirely positive; it is
   a necessary evil.

   Explicit Congestion Control, however, communicates information about
   network congestion that is assuredly about congestion, and avoids the
   unintended consequences of loss.

   Hence, network communication to the host regarding the moderation of
   its traffic flow SHOULD use an AQM algorithm to determine which
   packets it should affect, and then implement that effect by marking
   ECN-capable traffic "Congestion Experienced (CE)" or dropping non-
   ECN-capable traffic.

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   Due to the possibility of abuse, the queue must also impose an upper
   bound, so that even ECN-capable traffic experiences tail-drop if
   necessary; this possibility, while equipment must design for the end
   case, should in theory be very uncommon.

4.3.  Active Queue Management algorithms deployed SHOULD NOT require
      operational tuning

   A number of algorithms have been proposed.  Many require some form of
   tuning or initial condition, which makes them difficult to use
   operationally.  Hence, self-tuning algorithms are to be preferred.

4.4.  Active Queue Management algorithms deployed SHOULD be effective on
      all common Internet traffic

   Active Queue Management algorithms often target TCP [RFC0793], as it
   is by far the predominant transport in the Internet today.  However,
   we have significant use of UDP [RFC0768] in voice and video services,
   and find utility in SCTP [RFC4960] and DCCP [RFC4340].  Hence, Active
   Queue Management algorithms that are effective with all of those
   transports and the applications that use them are to be preferred.

4.5.  TCP and SCTP congestion control algorithms SHOULD maximize their
      use of available bandwidth without incurring loss or undue round
      trip delay

   The terms "knee" and "cliff" area defined by [Jain94].  They
   respectively refer to the minimum and maximum values of the effective
   window that have the effect of maximizing transmission rate in a
   congestion control algorithm such as is used by TCP or SCTP.  For the
   sender of data, 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 discussed in [RFC6057].

   Operating below the knee is also ineffective, as it fails to use
   available network capacity.  If the objective is to deliver data from
   its source to its recipient in the least possible time, as a result,
   the behavior of any TCP/SCTP congestion control algorithm SHOULD be
   to seek and use effective window values at or above the knee and well
   below the cliff.

4.6.  The need for further research

   [RFC2309] called for, as its second recommendation, further research
   in the interaction between network queues and host applications, and
   the means of signaling between them.  This research occurred, and we
   as a community have learned a lot.  However, we are not done.  An

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   obvious example in 2013 is in 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 - 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?

   Hence, this document reiterates the call: 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.  We note, however,
   that denial-of-service attacks may create unresponsive traffic flows
   that are indistinguishable from flows from normal high-bandwidth
   isochronous applications, and the mechanism suggested in The
   recommendation in support of ongoing research will be equally
   applicable to 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
   Ramakrishnan, Scott Shenker, John Wroclawski, and Lixia Zhang.  This
   is an edited version of that document, with much of its text and
   arguments unchanged.

9.  Change Log

   Initial Version:  March 2013

10.  References

10.1.  Normative References

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

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   [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.

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

10.2.  Informative References

   [Bolot94]  Bolot, JC., Turletti, T., and T. Wakeman, "Scalable
              Feedback Control for Multicast Video Distribution in the
              Internet", SIGCOMM Symposium proceedings on Communications
              architectures and protocols , August 1994.

              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

              McCanne, S., Jacobson, V., and M. Vetterli, "Receiver-
              driven Layered Multicast", SIGCOMM Symposium proceedings
              on Communications architectures and protocols , August

              Sprint ATL, KAIST, University of Minnesota, Sprint ATL,
              Intel Research, "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.D., Crowcroft, J., Davie, B.,
              Deering, S., Estrin, D., Floyd, S., Jacobson, V.,
              Minshall, G., Partridge, C., Peterson, L., Ramakrishnan,
              K.K., Shenker, S., Wroclawski, J., and L. Zhang,
              "Recommendations on Queue Management and Congestion
              Avoidance in the Internet", RFC 2309, April 1998.

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

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

   [RFC2475]  Blake, S., Black, D.L., Carlson, M.A., 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.

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

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

   [SRM96]    Floyd, S., Jacobson, V., McCanne, S., Liu, C., and L.
              Zhang, "A Reliable Multicast Framework for Light-weight
              Sessions and Application Level Framing", SIGCOMM Symposium
              proceedings on Communications architectures and protocols
              , 1996.

              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.

Author's Address

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

   Email: fred@cisco.com

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