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Byte and Packet Congestion Notification

The information below is for an old version of the document.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 7141.
Authors Bob Briscoe , Jukka Manner
Last updated 2013-10-24 (Latest revision 2013-08-01)
Replaces draft-briscoe-tsvwg-byte-pkt-mark
RFC stream Internet Engineering Task Force (IETF)
Additional resources Mailing list discussion
Stream WG state WG Document
Document shepherd Gorry Fairhurst
Shepherd write-up Show Last changed 2013-08-28
IESG IESG state Became RFC 7141 (Best Current Practice)
Consensus boilerplate Yes
Telechat date (None)
Responsible AD Martin Stiemerling
Send notices to,
IANA IANA review state IANA OK - No Actions Needed
Transport Area Working Group                                  B. Briscoe
Internet-Draft                                                        BT
Updates: 2309 (if approved)                                    J. Manner
Intended status: BCP                                    Aalto University
Expires: February 2, 2014                                 August 1, 2013

                Byte and Packet Congestion Notification


   This document provides recommendations of best current practice for
   dropping or marking packets using any active queue management (AQM)
   algorithm, including random early detection (RED), BLUE, pre-
   congestion notification (PCN) and newer schemes such as CoDel and
   PIE.  We give three strong recommendations: (1) packet size should be
   taken into account when transports detect and respond to congestion
   indications, (2) packet size should not be taken into account when
   network equipment creates congestion signals (marking, dropping), and
   therefore (3) in the specific case of RED, the byte-mode packet drop
   variant that drops fewer small packets should not be used.  This memo
   updates RFC 2309 to deprecate deliberate preferential treatment of
   small packets in AQM algorithms.

Status of This Memo

   This 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

   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 February 2, 2014.

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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   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.1.  Terminology and Scoping  . . . . . . . . . . . . . . . . .  6
     1.2.  Example Comparing Packet-Mode Drop and Byte-Mode Drop  . .  7
   2.  Recommendations  . . . . . . . . . . . . . . . . . . . . . . .  9
     2.1.  Recommendation on Queue Measurement  . . . . . . . . . . .  9
     2.2.  Recommendation on Encoding Congestion Notification . . . . 10
     2.3.  Recommendation on Responding to Congestion . . . . . . . . 11
     2.4.  Recommendation on Handling Congestion Indications when
           Splitting or Merging Packets . . . . . . . . . . . . . . . 12
   3.  Motivating Arguments . . . . . . . . . . . . . . . . . . . . . 12
     3.1.  Avoiding Perverse Incentives to (Ab)use Smaller Packets  . 12
     3.2.  Small != Control . . . . . . . . . . . . . . . . . . . . . 14
     3.3.  Transport-Independent Network  . . . . . . . . . . . . . . 14
     3.4.  Partial Deployment of AQM  . . . . . . . . . . . . . . . . 15
     3.5.  Implementation Efficiency  . . . . . . . . . . . . . . . . 17
   4.  A Survey and Critique of Past Advice . . . . . . . . . . . . . 17
     4.1.  Congestion Measurement Advice  . . . . . . . . . . . . . . 17
       4.1.1.  Fixed Size Packet Buffers  . . . . . . . . . . . . . . 18
       4.1.2.  Congestion Measurement without a Queue . . . . . . . . 19
     4.2.  Congestion Notification Advice . . . . . . . . . . . . . . 20
       4.2.1.  Network Bias when Encoding . . . . . . . . . . . . . . 20
       4.2.2.  Transport Bias when Decoding . . . . . . . . . . . . . 21
       4.2.3.  Making Transports Robust against Control Packet
               Losses . . . . . . . . . . . . . . . . . . . . . . . . 23
       4.2.4.  Congestion Notification: Summary of Conflicting
               Advice . . . . . . . . . . . . . . . . . . . . . . . . 23
   5.  Outstanding Issues and Next Steps  . . . . . . . . . . . . . . 24
     5.1.  Bit-congestible Network  . . . . . . . . . . . . . . . . . 24
     5.2.  Bit- & Packet-congestible Network  . . . . . . . . . . . . 25
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 25
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 26
   8.  Conclusions  . . . . . . . . . . . . . . . . . . . . . . . . . 26
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 27
   10. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 28
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 28
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 28
     11.2. Informative References . . . . . . . . . . . . . . . . . . 28
   Appendix A.  Survey of RED Implementation Status . . . . . . . . . 32
   Appendix B.  Sufficiency of Packet-Mode Drop . . . . . . . . . . . 33
     B.1.  Packet-Size (In)Dependence in Transports . . . . . . . . . 34
     B.2.  Bit-Congestible and Packet-Congestible Indications . . . . 37
   Appendix C.  Byte-mode Drop Complicates Policing Congestion
                Response  . . . . . . . . . . . . . . . . . . . . . . 38
   Appendix D.  Changes from Previous Versions  . . . . . . . . . . . 39

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1.  Introduction

   This document provides recommendations of best current practice for
   how we should correctly scale congestion control functions with
   respect to packet size for the long term.  It also recognises that
   expediency may be necessary to deal with existing widely deployed
   protocols that don't live up to the long term goal.

   When signalling congestion, the problem of how (and whether) to take
   packet sizes into account has exercised the minds of researchers and
   practitioners for as long as active queue management (AQM) has been
   discussed.  Indeed, one reason AQM was originally introduced was to
   reduce the lock-out effects that small packets can have on large
   packets in drop-tail queues.  This memo aims to state the principles
   we should be using and to outline how these principles will affect
   future protocol design, taking into account the existing deployments
   we have already.

   The question of whether to take into account packet size arises at
   three stages in the congestion notification process:

   Measuring congestion:  When a congested resource measures locally how
      congested it is, should it measure its queue length in time, bytes
      or packets?

   Encoding congestion notification into the wire protocol:  When a
      congested network resource signals its level of congestion, should
      it drop / mark each packet dependent on the size of the particular
      packet in question?

   Decoding congestion notification from the wire protocol:  When a
      transport interprets the notification in order to decide how much
      to respond to congestion, should it take into account the size of
      each missing or marked packet?

   Consensus has emerged over the years concerning the first stage,
   which Section 2.1 records in the RFC Series.  In summary: If possible
   it is best to measure congestion by time in the queue, but otherwise
   the choice between bytes and packets solely depends on whether the
   resource is congested by bytes or packets.

   The controversy is mainly around the last two stages: whether to
   allow for the size of the specific packet notifying congestion i)
   when the network encodes or ii) when the transport decodes the
   congestion notification.

   Currently, the RFC series is silent on this matter other than a paper
   trail of advice referenced from [RFC2309], which conditionally

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   recommends byte-mode (packet-size dependent) drop [pktByteEmail].
   Reducing drop of small packets certainly has some tempting
   advantages: i) it drops less control packets, which tend to be small
   and ii) it makes TCP's bit-rate less dependent on packet size.
   However, there are ways of addressing these issues at the transport
   layer, rather than reverse engineering network forwarding to fix the

   This memo updates [RFC2309] to deprecate deliberate preferential
   treatment of packets in AQM algorithms solely because of their size.
   It recommends that (1) packet size should be taken into account when
   transports detect and respond to congestion indications, (2) not when
   network equipment creates them.  This memo also adds to the
   congestion control principles enumerated in BCP 41 [RFC2914].

   In the particular case of Random early Detection (RED), this means
   that the byte-mode packet drop variant should not be used to drop
   fewer small packets, because that creates a perverse incentive for
   transports to use tiny segments, consequently also opening up a DoS
   vulnerability.  Fortunately all the RED implementers who responded to
   our admittedly limited survey (Section 4.2.4) have not followed the
   earlier advice to use byte-mode drop, so the position this memo
   argues for seems to already exist in implementations.

   However, at the transport layer, TCP congestion control is a widely
   deployed protocol that doesn't scale with packet size.  To date this
   hasn't been a significant problem because most TCP implementations
   have been used with similar packet sizes.  But, as we design new
   congestion control mechanisms, this memo recommends that we should
   build in scaling with packet size rather than assuming we should
   follow TCP's example.

   This memo continues as follows.  First it discusses terminology and
   scoping.  Section 2 gives the concrete formal recommendations,
   followed by motivating arguments in Section 3.  We then critically
   survey the advice given previously in the RFC series and the research
   literature (Section 4), referring to an assessment of whether or not
   this advice has been followed in production networks (Appendix A).
   To wrap up, outstanding issues are discussed that will need
   resolution both to inform future protocol designs and to handle
   legacy (Section 5).  Then security issues are collected together in
   Section 6 before conclusions are drawn in Section 8.  The interested
   reader can find discussion of more detailed issues on the theme of
   byte vs. packet in the appendices.

   This memo intentionally includes a non-negligible amount of material
   on the subject.  For the busy reader Section 2 summarises the
   recommendations for the Internet community.

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1.1.  Terminology and Scoping

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

   This memo applies to the design of all AQM algorithms, for example,
   Random Early Detection (RED) [RFC2309], BLUE [BLUE02], Pre-Congestion
   Notification (PCN) [RFC5670], Controlled Delay (CoDel)
   [I-D.nichols-tsvwg-codel] and the Proportional Integral controller
   Enhanced (PIE) [I-D.pan-tsvwg-pie].  Throughout, RED is used as a
   concrete example because it is a widely known and deployed AQM
   algorithm.  There is no intention to imply that the advice is any
   less applicable to the other algorithms, nor that RED is preferred.

   Congestion Notification:  Congestion notification is a changing
      signal that aims to communicate the probability that the network
      resource(s) will not be able to forward the level of traffic load
      offered (or that there is an impending risk that they will not be
      able to).

      The `impending risk' qualifier is added, because AQM systems set a
      virtual limit smaller than the actual limit to the resource, then
      notify when this virtual limit is exceeded in order to avoid
      uncontrolled congestion of the actual capacity.

      Congestion notification communicates a real number bounded by the
      range [ 0 , 1 ].  This ties in with the most well-understood
      measure of congestion notification: drop probability.

   Explicit and Implicit Notification:  The byte vs. packet dilemma
      concerns congestion notification irrespective of whether it is
      signalled implicitly by drop or using explicit congestion
      notification (ECN [RFC3168] or PCN [RFC5670]).  Throughout this
      document, unless clear from the context, the term marking will be
      used to mean notifying congestion explicitly, while congestion
      notification will be used to mean notifying congestion either
      implicitly by drop or explicitly by marking.

   Bit-congestible vs. Packet-congestible:  If the load on a resource
      depends on the rate at which packets arrive, it is called packet-
      congestible.  If the load depends on the rate at which bits arrive
      it is called bit-congestible.

      Examples of packet-congestible resources are route look-up engines
      and firewalls, because load depends on how many packet headers
      they have to process.  Examples of bit-congestible resources are
      transmission links, radio power and most buffer memory, because

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      the load depends on how many bits they have to transmit or store.
      Some machine architectures use fixed size packet buffers, so
      buffer memory in these cases is packet-congestible (see
      Section 4.1.1).

      The path through a machine will typically encounter both packet-
      congestible and bit-congestible resources.  However, currently, a
      design goal of network processing equipment such as routers and
      firewalls is to size the packet-processing engine(s) relative to
      the lines in order to keep packet processing uncongested even
      under worst case packet rates with runs of minimum size packets.
      Therefore, packet-congestion is currently rare [RFC6077; S.3.3],
      but there is no guarantee that it will not become more common in

      Note that information is generally processed or transmitted with a
      minimum granularity greater than a bit (e.g. octets).  The
      appropriate granularity for the resource in question should be
      used, but for the sake of brevity we will talk in terms of bytes
      in this memo.

   Coarser Granularity:  Resources may be congestible at higher levels
      of granularity than bits or packets, for instance stateful
      firewalls are flow-congestible and call-servers are session-
      congestible.  This memo focuses on congestion of connectionless
      resources, but the same principles may be applicable for
      congestion notification protocols controlling per-flow and per-
      session processing or state.

   RED Terminology:  In RED whether to use packets or bytes when
      measuring queues is called respectively "packet-mode queue
      measurement" or "byte-mode queue measurement".  And whether the
      probability of dropping a particular packet is independent or
      dependent on its size is called respectively "packet-mode drop" or
      "byte-mode drop".  The terms byte-mode and packet-mode should not
      be used without specifying whether they apply to queue measurement
      or to drop.

1.2.  Example Comparing Packet-Mode Drop and Byte-Mode Drop

   Taking RED as a well-known example algorithm, a central question
   addressed by this document is whether to recommend RED's packet-mode
   drop variant and to deprecate byte-mode drop.  Table 1 compares how
   packet-mode and byte-mode drop affect two flows of different size
   packets.  For each it gives the expected number of packets and of
   bits dropped in one second.  Each example flow runs at the same bit-
   rate of 48Mb/s, but one is broken up into small 60 byte packets and
   the other into large 1500 byte packets.

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   To keep up the same bit-rate, in one second there are about 25 times
   more small packets because they are 25 times smaller.  As can be seen
   from the table, the packet rate is 100,000 small packets versus 4,000
   large packets per second (pps).

      Parameter            Formula        Small packets Large packets
      -------------------- -------------- ------------- -------------
      Packet size          s/8                      60B        1,500B
      Packet size          s                       480b       12,000b
      Bit-rate             x                     48Mbps        48Mbps
      Packet-rate          u = x/s              100kpps         4kpps

      Packet-mode Drop
      Pkt loss probability p                       0.1%          0.1%
      Pkt loss-rate        p*u                   100pps          4pps
      Bit loss-rate        p*u*s                 48kbps        48kbps

      Byte-mode Drop       MTU, M=12,000b
      Pkt loss probability b = p*s/M             0.004%          0.1%
      Pkt loss-rate        b*u                     4pps          4pps
      Bit loss-rate        b*u*s               1.92kbps        48kbps

         Table 1: Example Comparing Packet-mode and Byte-mode Drop

   For packet-mode drop, we illustrate the effect of a drop probability
   of 0.1%, which the algorithm applies to all packets irrespective of
   size.  Because there are 25 times more small packets in one second,
   it naturally drops 25 times more small packets, that is 100 small
   packets but only 4 large packets.  But if we count how many bits it
   drops, there are 48,000 bits in 100 small packets and 48,000 bits in
   4 large packets--the same number of bits of small packets as large.

      The packet-mode drop algorithm drops any bit with the same
      probability whether the bit is in a small or a large packet.

   For byte-mode drop, again we use an example drop probability of 0.1%,
   but only for maximum size packets (assuming the link MTU is 1,500B or
   12,000b).  The byte-mode algorithm reduces the drop probability of
   smaller packets proportional to their size, making the probability
   that it drops a small packet 25 times smaller at 0.004%.  But there
   are 25 times more small packets, so dropping them with 25 times lower
   probability results in dropping the same number of packets: 4 drops
   in both cases.  The 4 small dropped packets contain 25 times less
   bits than the 4 large dropped packets: 1,920 compared to 48,000.

      The byte-mode drop algorithm drops any bit with a probability
      proportionate to the size of the packet it is in.

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2.  Recommendations

   This section gives recommendations related to network equipment in
   Sections 2.1 and 2.2, and in Sections 2.3 and 2.4 we discuss the
   implications on the transport protocols.

2.1.  Recommendation on Queue Measurement

   Ideally, an AQM would measure the service time of the queue to
   measure congestion of a resource.  However service time can only be
   measured as packets leave the queue, where it is not always feasible
   to implement a full AQM algorithm.  To predict the service time as
   packets join the queue, an AQM algorithm needs to measure the length
   of the queue.

   In this case, if the resource is bit-congestible, the AQM
   implementation SHOULD measure the length of the queue in bytes and,
   if the resource is packet-congestible, the implementation SHOULD
   measure the length of the queue in packets.  No other choice makes
   sense, because the number of packets waiting in the queue isn't
   relevant if the resource gets congested by bytes and vice versa.  For
   example, the length of the queue into a transmission line would be
   measured in bytes, while the length of the queue into a firewall
   would be measured in packets.

   To avoid the pathological effects of drop tail, the AQM can then
   transform this service time or queue length into the probability of
   dropping or marking a packet (e.g.  RED's piecewise linear function
   between thresholds).

   What this advice means for RED as a specific example:

   1.  A RED implementation SHOULD use byte mode queue measurement for
       measuring the congestion of bit-congestible resources and packet
       mode queue measurement for packet-congestible resources.

   2.  An implementation SHOULD NOT make it possible to configure the
       way a queue measures itself, because whether a queue is bit-
       congestible or packet-congestible is an inherent property of the

   Exceptions to these recommendations MAY be necessary, for instance
   where a packet-congestible resource has to be configured as a proxy
   bottleneck for a bit-congestible resource in an adjacent box that
   does not support AQM.

   The recommended approach in less straightforward scenarios, such as
   fixed size packet buffers, resources without a queue and buffers

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   comprising a mix of packet and bit-congestible resources, is
   discussed in Section 4.1.  For instance, Section 4.1.1 explains that
   the queue into a line should be measured in bytes even if the queue
   consists of fixed-size packet-buffers, because the root-cause of any
   congestion is bytes arriving too fast for the line--packets filling
   buffers are merely a symptom of the underlying congestion of the

2.2.  Recommendation on Encoding Congestion Notification

   When encoding congestion notification (e.g. by drop, ECN or PCN), the
   probability that network equipment drops or marks a particular packet
   to notify congestion SHOULD NOT depend on the size of the packet in
   question.  As the example in Section 1.2 illustrates, to drop any bit
   with probability 0.1% it is only necessary to drop every packet with
   probability 0.1% without regard to the size of each packet.

   This approach ensures the network layer offers sufficient congestion
   information for all known and future transport protocols and also
   ensures no perverse incentives are created that would encourage
   transports to use inappropriately small packet sizes.

   What this advice means for RED as a specific example:

   1.  The RED AQM algorithm SHOULD NOT use byte-mode drop, i.e. it
       ought to use packet-mode drop.  Byte-mode drop is more complex,
       it creates the perverse incentive to fragment segments into tiny
       pieces and it is vulnerable to floods of small packets.

   2.  If a vendor has implemented byte-mode drop, and an operator has
       turned it on, it is RECOMMENDED to switch it to packet-mode drop,
       after establishing if there are any implications on the relative
       performance of applications using different packet sizes.  The
       unlikely possibility of some application-specific legacy use of
       byte-mode drop is the only reason that all the above
       recommendations on encoding congestion notification are not
       phrased more strongly.

       RED as a whole SHOULD NOT be switched off.  Without RED, a drop
       tail queue biases against large packets and is vulnerable to
       floods of small packets.

   Note well that RED's byte-mode queue drop is completely orthogonal to
   byte-mode queue measurement and should not be confused with it.  If a
   RED implementation has a byte-mode but does not specify what sort of
   byte-mode, it is most probably byte-mode queue measurement, which is
   fine.  However, if in doubt, the vendor should be consulted.

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   A survey (Appendix A) showed that there appears to be little, if any,
   installed base of the byte-mode drop variant of RED.  This suggests
   that deprecating byte-mode drop will have little, if any, incremental
   deployment impact.

2.3.  Recommendation on Responding to Congestion

   When a transport detects that a packet has been lost or congestion
   marked, it SHOULD consider the strength of the congestion indication
   as proportionate to the size in octets (bytes) of the missing or
   marked packet.

   In other words, when a packet indicates congestion (by being lost or
   marked) it can be considered conceptually as if there is a congestion
   indication on every octet of the packet, not just one indication per

   To be clear, the above recommendation solely describes how a
   transport should interpret the meaning of a congestion indication.
   It makes no recommendation on whether a transport should act
   differently based on this interpretation.  It merely aids
   interoperablity between transports, if they choose to make their
   actions depend on the strength of congestion indications.

   This definition will be useful as the IETF transport area continues
   its programme of;

   o  updating host-based congestion control protocols to take account
      of packet size

   o  making transports less sensitive to losing control packets like
      SYNs and pure ACKs.

   What this advice means for the case of TCP:

   1.  If two TCP flows with different packet sizes are required to run
       at equal bit rates under the same path conditions, this SHOULD be
       done by altering TCP (Section 4.2.2), not network equipment (the
       latter affects other transports besides TCP).

   2.  If it is desired to improve TCP performance by reducing the
       chance that a SYN or a pure ACK will be dropped, this SHOULD be
       done by modifying TCP (Section 4.2.3), not network equipment.

   To be clear, we are not recommending at all that TCPs under
   equivalent conditions should aim for equal bit-rates.  We are merely
   saying that anyone trying to do such a thing should modify their TCP
   algorithm, not the network.

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   These recommendations are phrased as 'SHOULD' rather than 'MUST',
   because there may be cases where compatibility with pre-existing
   versions of a transport protocol make the recommendations

2.4.  Recommendation on Handling Congestion Indications when Splitting
      or Merging Packets

   Packets carrying congestion indications may be split or merged in
   some circumstances (e.g. at a RTP/RTCP transcoder or during IP
   fragment reassembly).  Splitting and merging only make sense in the
   context of ECN, not loss.

   The general rule to follow is that the number of octets in packets
   with congestion indications SHOULD be equivalent before and after
   merging or splitting.  This is based on the principle used above;
   that an indication of congestion on a packet can be considered as an
   indication of congestion on each octet of the packet.

   The above rule is not phrased with the word "MUST" to allow the
   following exception.  There are cases where pre-existing protocols
   were not designed to conserve congestion marked octets (e.g.  IP
   fragment reassembly [RFC3168] or loss statistics in RTCP receiver
   reports [RFC3550] before ECN was added [RFC6679]).  When any such
   protocol is updated, it SHOULD comply with the above rule to conserve
   marked octets.  However, the rule may be relaxed if it would
   otherwise become too complex to interoperate with pre-existing
   implementations of the protocol.

   One can think of a splitting or merging process as if all the
   incoming congestion-marked octets increment a counter and all the
   outgoing marked octets decrement the same counter.  In order to
   ensure that congestion indications remain timely, even the smallest
   positive remainder in the conceptual counter should trigger the next
   outgoing packet to be marked (causing the counter to go negative).

3.  Motivating Arguments

   This section is informative.  It justifies the recommendations given
   in the previous section.

3.1.  Avoiding Perverse Incentives to (Ab)use Smaller Packets

   Increasingly, it is being recognised that a protocol design must take
   care not to cause unintended consequences by giving the parties in
   the protocol exchange perverse incentives [Evol_cc][RFC3426].  Given
   there are many good reasons why larger path maximum transmission
   units (PMTUs) would help solve a number of scaling issues, we do not

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   want to create any bias against large packets that is greater than
   their true cost.

   Imagine a scenario where the same bit rate of packets will contribute
   the same to bit-congestion of a link irrespective of whether it is
   sent as fewer larger packets or more smaller packets.  A protocol
   design that caused larger packets to be more likely to be dropped
   than smaller ones would be dangerous in both the following cases:

   Malicious transports:  A queue that gives an advantage to small
      packets can be used to amplify the force of a flooding attack.  By
      sending a flood of small packets, the attacker can get the queue
      to discard more traffic in large packets, allowing more attack
      traffic to get through to cause further damage.  Such a queue
      allows attack traffic to have a disproportionately large effect on
      regular traffic without the attacker having to do much work.

   Non-malicious transports:  Even if an application designer is not
      actually malicious, if over time it is noticed that small packets
      tend to go faster, designers will act in their own interest and
      use smaller packets.  Queues that give advantage to small packets
      create an evolutionary pressure for applications or transports to
      send at the same bit-rate but break their data stream down into
      tiny segments to reduce their drop rate.  Encouraging a high
      volume of tiny packets might in turn unnecessarily overload a
      completely unrelated part of the system, perhaps more limited by
      header-processing than bandwidth.

   Imagine two unresponsive flows arrive at a bit-congestible
   transmission link each with the same bit rate, say 1Mbps, but one
   consists of 1500B and the other 60B packets, which are 25x smaller.
   Consider a scenario where gentle RED [gentle_RED] is used, along with
   the variant of RED we advise against, i.e. where the RED algorithm is
   configured to adjust the drop probability of packets in proportion to
   each packet's size (byte mode packet drop).  In this case, RED aims
   to drop 25x more of the larger packets than the smaller ones.  Thus,
   for example if RED drops 25% of the larger packets, it will aim to
   drop 1% of the smaller packets (but in practice it may drop more as
   congestion increases [RFC4828; Appx B.4]).  Even though both flows
   arrive with the same bit rate, the bit rate the RED queue aims to
   pass to the line will be 750kbps for the flow of larger packets but
   990kbps for the smaller packets (because of rate variations it will
   actually be a little less than this target).

   Note that, although the byte-mode drop variant of RED amplifies small
   packet attacks, drop-tail queues amplify small packet attacks even
   more (see Security Considerations in Section 6).  Wherever possible
   neither should be used.

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3.2.  Small != Control

   Dropping fewer control packets considerably improves performance.  It
   is tempting to drop small packets with lower probability in order to
   improve performance, because many control packets tend to be smaller
   (TCP SYNs & ACKs, DNS queries & responses, SIP messages, HTTP GETs,
   etc).  However, we must not give control packets preference purely by
   virtue of their smallness, otherwise it is too easy for any data
   source to get the same preferential treatment simply by sending data
   in smaller packets.  Again we should not create perverse incentives
   to favour small packets rather than to favour control packets, which
   is what we intend.

   Just because many control packets are small does not mean all small
   packets are control packets.

   So, rather than fix these problems in the network, we argue that the
   transport should be made more robust against losses of control
   packets (see 'Making Transports Robust against Control Packet Losses'
   in Section 4.2.3).

3.3.  Transport-Independent Network

   TCP congestion control ensures that flows competing for the same
   resource each maintain the same number of segments in flight,
   irrespective of segment size.  So under similar conditions, flows
   with different segment sizes will get different bit-rates.

   To counter this effect it seems tempting not to follow our
   recommendation, and instead for the network to bias congestion
   notification by packet size in order to equalise the bit-rates of
   flows with different packet sizes.  However, in order to do this, the
   queuing algorithm has to make assumptions about the transport, which
   become embedded in the network.  Specifically:

   o  The queuing algorithm has to assume how aggressively the transport
      will respond to congestion (see Section 4.2.4).  If the network
      assumes the transport responds as aggressively as TCP NewReno, it
      will be wrong for Compound TCP and differently wrong for Cubic
      TCP, etc.  To achieve equal bit-rates, each transport then has to
      guess what assumption the network made, and work out how to
      replace this assumed aggressiveness with its own aggressiveness.

   o  Also, if the network biases congestion notification by packet size
      it has to assume a baseline packet size--all proposed algorithms
      use the local MTU (for example see the byte-mode loss probability
      formula in Table 1).  Then if the non-Reno transports mentioned
      above are trying to reverse engineer what the network assumed,

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      they also have to guess the MTU of the congested link.

   Even though reducing the drop probability of small packets (e.g.
   RED's byte-mode drop) helps ensure TCP flows with different packet
   sizes will achieve similar bit rates, we argue this correction should
   be made to any future transport protocols based on TCP, not to the
   network in order to fix one transport, no matter how predominant it
   is.  Effectively, favouring small packets is reverse engineering of
   network equipment around one particular transport protocol (TCP),
   contrary to the excellent advice in [RFC3426], which asks designers
   to question "Why are you proposing a solution at this layer of the
   protocol stack, rather than at another layer?"

   In contrast, if the network never takes account of packet size, the
   transport can be certain it will never need to guess any assumptions
   the network has made.  And the network passes two pieces of
   information to the transport that are sufficient in all cases: i)
   congestion notification on the packet and ii) the size of the packet.
   Both are available for the transport to combine (by taking account of
   packet size when responding to congestion) or not.  Appendix B checks
   that these two pieces of information are sufficient for all relevant

   When the network does not take account of packet size, it allows
   transport protocols to choose whether to take account of packet size
   or not.  However, if the network were to bias congestion notification
   by packet size, transport protocols would have no choice; those that
   did not take account of packet size themselves would unwittingly
   become dependent on packet size, and those that already took account
   of packet size would end up taking account of it twice.

3.4.  Partial Deployment of AQM

   In overview, the argument in this section runs as follows:

   o  Because the network does not and cannot always drop packets in
      proportion to their size, it shouldn't be given the task of making
      drop signals depend on packet size at all.

   o  Transports on the other hand don't always want to make their rate
      response proportional to the size of dropped packets, but if they
      want to, they always can.

   The argument is similar to the end-to-end argument that says "Don't
   do X in the network if end-systems can do X by themselves, and they
   want to be able to choose whether to do X anyway."  Actually the
   following argument is stronger; in addition it says "Don't give the
   network task X that could be done by the end-systems, if X is not

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   deployed on all network nodes, and end-systems won't be able to tell
   whether their network is doing X, or whether they need to do X
   themselves."  In this case, the X in question is "making the response
   to congestion depend on packet size".

   We will now re-run this argument taking each step in more depth.  The
   argument applies solely to drop, not to ECN marking.

   A queue drops packets for either of two reasons: a) to signal to host
   congestion controls that they should reduce the load and b) because
   there is no buffer left to store the packets.  Active queue
   management tries to use drops as a signal for hosts to slow down
   (case a) so that drop due to buffer exhaustion (case b) should not be

   AQM is not universally deployed in every queue in the Internet; many
   cheap Ethernet bridges, software firewalls, NATs on consumer devices,
   etc implement simple tail-drop buffers.  Even if AQM were universal,
   it has to be able to cope with buffer exhaustion (by switching to a
   behaviour like tail-drop), in order to cope with unresponsive or
   excessive transports.  For these reasons networks will sometimes be
   dropping packets as a last resort (case b) rather than under AQM
   control (case a).

   When buffers are exhausted (case b), they don't naturally drop
   packets in proportion to their size.  The network can only reduce the
   probability of dropping smaller packets if it has enough space to
   store them somewhere while it waits for a larger packet that it can
   drop.  If the buffer is exhausted, it does not have this choice.
   Admittedly tail-drop does naturally drop somewhat fewer small
   packets, but exactly how few depends more on the mix of sizes than
   the size of the packet in question.  Nonetheless, in general, if we
   wanted networks to do size-dependent drop, we would need universal
   deployment of (packet-size dependent) AQM code, which is currently

   A host transport cannot know whether any particular drop was a
   deliberate signal from an AQM or a sign of a queue shedding packets
   due to buffer exhaustion.  Therefore, because the network cannot
   universally do size-dependent drop, it should not do it all.

   Whereas universality is desirable in the network, diversity is
   desirable between different transport layer protocols - some, like
   NewReno TCP [RFC5681], may not choose to make their rate response
   proportionate to the size of each dropped packet, while others will
   (e.g.  TFRC-SP [RFC4828]).

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3.5.  Implementation Efficiency

   Biasing against large packets typically requires an extra multiply
   and divide in the network (see the example byte-mode drop formula in
   Table 1).  Allowing for packet size at the transport rather than in
   the network ensures that neither the network nor the transport needs
   to do a multiply operation--multiplication by packet size is
   effectively achieved as a repeated add when the transport adds to its
   count of marked bytes as each congestion event is fed to it.  Also
   the work to do the biasing is spread over many hosts, rather than
   concentrated in just the congested network element.  These aren't
   principled reasons in themselves, but they are a happy consequence of
   the other principled reasons.

4.  A Survey and Critique of Past Advice

   This section is informative, not normative.

   The original 1993 paper on RED [RED93] proposed two options for the
   RED active queue management algorithm: packet mode and byte mode.
   Packet mode measured the queue length in packets and dropped (or
   marked) individual packets with a probability independent of their
   size.  Byte mode measured the queue length in bytes and marked an
   individual packet with probability in proportion to its size
   (relative to the maximum packet size).  In the paper's outline of
   further work, it was stated that no recommendation had been made on
   whether the queue size should be measured in bytes or packets, but
   noted that the difference could be significant.

   When RED was recommended for general deployment in 1998 [RFC2309],
   the two modes were mentioned implying the choice between them was a
   question of performance, referring to a 1997 email [pktByteEmail] for
   advice on tuning.  A later addendum to this email introduced the
   insight that there are in fact two orthogonal choices:

   o  whether to measure queue length in bytes or packets (Section 4.1)

   o  whether the drop probability of an individual packet should depend
      on its own size (Section 4.2).

   The rest of this section is structured accordingly.

4.1.  Congestion Measurement Advice

   The choice of which metric to use to measure queue length was left
   open in RFC2309.  It is now well understood that queues for bit-
   congestible resources should be measured in bytes, and queues for
   packet-congestible resources should be measured in packets

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   Congestion in some legacy bit-congestible buffers is only measured in
   packets not bytes.  In such cases, the operator has to set the
   thresholds mindful of a typical mix of packets sizes.  Any AQM
   algorithm on such a buffer will be oversensitive to high proportions
   of small packets, e.g. a DoS attack, and under-sensitive to high
   proportions of large packets.  However, there is no need to make
   allowances for the possibility of such legacy in future protocol
   design.  This is safe because any under-sensitivity during unusual
   traffic mixes cannot lead to congestion collapse given the buffer
   will eventually revert to tail drop, discarding proportionately more
   large packets.

4.1.1.  Fixed Size Packet Buffers

   The question of whether to measure queues in bytes or packets seems
   to be well understood.  However, measuring congestion is confusing
   when the resource is bit congestible but the queue into the resource
   is packet congestible.  This section outlines the approach to take.

   Some, mostly older, queuing hardware allocates fixed sized buffers in
   which to store each packet in the queue.  This hardware forwards to
   the line in one of two ways:

   o  With some hardware, any fixed sized buffers not completely filled
      by a packet are padded when transmitted to the wire.  This case,
      should clearly be treated as packet-congestible, because both
      queuing and transmission are in fixed MTU-sized units.  Therefore
      the queue length in packets is a good model of congestion of the

   o  More commonly, hardware with fixed size packet buffers transmits
      packets to line without padding.  This implies a hybrid forwarding
      system with transmission congestion dependent on the size of
      packets but queue congestion dependent on the number of packets,
      irrespective of their size.

      Nonetheless, there would be no queue at all unless the line had
      become congested--the root-cause of any congestion is too many
      bytes arriving for the line.  Therefore, the AQM should measure
      the queue length as the sum of all the packet sizes in bytes that
      are queued up waiting to be serviced by the line, irrespective of
      whether each packet is held in a fixed size buffer.

   In the (unlikely) first case where use of padding means the queue
   should be measured in packets, further confusion is likely because
   the fixed buffers are rarely all one size.  Typically pools of

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   different sized buffers are provided (Cisco uses the term 'buffer
   carving' for the process of dividing up memory into these pools
   [IOSArch]).  Usually, if the pool of small buffers is exhausted,
   arriving small packets can borrow space in the pool of large buffers,
   but not vice versa.  However, there is no need to consider all this
   complexity, because the root-cause of any congestion is still line
   overload--buffer consumption is only the symptom.  Therefore, the
   length of the queue should be measured as the sum of the bytes in the
   queue that will be transmitted to line, including any padding.  In
   the (unusual) case of transmission with padding this means the sum of
   the sizes of the small buffers queued plus the sum of the sizes of
   the large buffers queued.

   We will return to borrowing of fixed sized buffers when we discuss
   biasing the drop/marking probability of a specific packet because of
   its size in Section 4.2.1.  But here we can repeat the simple rule
   for how to measure the length of queues of fixed buffers: no matter
   how complicated the buffering scheme is, ultimately a transmission
   line is nearly always bit-congestible so the number of bytes queued
   up waiting for the line measures how congested the line is, and it is
   rarely important to measure how congested the buffering system is.

4.1.2.  Congestion Measurement without a Queue

   AQM algorithms are nearly always described assuming there is a queue
   for a congested resource and the algorithm can use the queue length
   to determine the probability that it will drop or mark each packet.
   But not all congested resources lead to queues.  For instance, power
   limited resources are usually bit-congestible if energy is primarily
   required for transmission rather than header processing, but it is
   rare for a link protocol to build a queue as it approaches maximum

   Nonetheless, AQM algorithms do not require a queue in order to work.
   For instance spectrum congestion can be modelled by signal quality
   using target bit-energy-to-noise-density ratio.  And, to model radio
   power exhaustion, transmission power levels can be measured and
   compared to the maximum power available.  [ECNFixedWireless] proposes
   a practical and theoretically sound way to combine congestion
   notification for different bit-congestible resources at different
   layers along an end to end path, whether wireless or wired, and
   whether with or without queues.

   In wireless protocols that use request to send / clear to send (RTS /
   CTS) control, such as some variants of IEEE802.11, it is reasonable
   to base an AQM on the time spent waiting for transmission
   opportunities (TXOPs) even though wireless spectrum is usually
   regarded as congested by bits (for a given coding scheme).  This is

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   because requests for TXOPs queue up as the spectrum gets congested by
   all the bits being transferred.  So the time that TXOPs are queued
   directly reflects bit congestion of the spectrum.

4.2.  Congestion Notification Advice

4.2.1.  Network Bias when Encoding  Advice on Packet Size Bias in RED

   The previously mentioned email [pktByteEmail] referred to by
   [RFC2309] advised that most scarce resources in the Internet were
   bit-congestible, which is still believed to be true (Section 1.1).
   But it went on to offer advice that is updated by this memo.  It said
   that drop probability should depend on the size of the packet being
   considered for drop if the resource is bit-congestible, but not if it
   is packet-congestible.  The argument continued that if packet drops
   were inflated by packet size (byte-mode dropping), "a flow's fraction
   of the packet drops is then a good indication of that flow's fraction
   of the link bandwidth in bits per second".  This was consistent with
   a referenced policing mechanism being worked on at the time for
   detecting unusually high bandwidth flows, eventually published in
   1999 [pBox].  However, the problem could and should have been solved
   by making the policing mechanism count the volume of bytes randomly
   dropped, not the number of packets.

   A few months before RFC2309 was published, an addendum was added to
   the above archived email referenced from the RFC, in which the final
   paragraph seemed to partially retract what had previously been said.
   It clarified that the question of whether the probability of
   dropping/marking a packet should depend on its size was not related
   to whether the resource itself was bit congestible, but a completely
   orthogonal question.  However the only example given had the queue
   measured in packets but packet drop depended on the size of the
   packet in question.  No example was given the other way round.

   In 2000, Cnodder et al [REDbyte] pointed out that there was an error
   in the part of the original 1993 RED algorithm that aimed to
   distribute drops uniformly, because it didn't correctly take into
   account the adjustment for packet size.  They recommended an
   algorithm called RED_4 to fix this.  But they also recommended a
   further change, RED_5, to adjust drop rate dependent on the square of
   relative packet size.  This was indeed consistent with one implied
   motivation behind RED's byte mode drop--that we should reverse
   engineer the network to improve the performance of dominant end-to-
   end congestion control mechanisms.  This memo makes a different
   recommendations in Section 2.

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   By 2003, a further change had been made to the adjustment for packet
   size, this time in the RED algorithm of the ns2 simulator.  Instead
   of taking each packet's size relative to a `maximum packet size' it
   was taken relative to a `mean packet size', intended to be a static
   value representative of the `typical' packet size on the link.  We
   have not been able to find a justification in the literature for this
   change, however Eddy and Allman conducted experiments [REDbias] that
   assessed how sensitive RED was to this parameter, amongst other
   things.  However, this changed algorithm can often lead to drop
   probabilities of greater than 1 (which gives a hint that there is
   probably a mistake in the theory somewhere).

   On 10-Nov-2004, this variant of byte-mode packet drop was made the
   default in the ns2 simulator.  It seems unlikely that byte-mode drop
   has ever been implemented in production networks (Appendix A),
   therefore any conclusions based on ns2 simulations that use RED
   without disabling byte-mode drop are likely to behave very
   differently from RED in production networks.  Packet Size Bias Regardless of AQM

   The byte-mode drop variant of RED (or a similar variant of other AQM
   algorithms) is not the only possible bias towards small packets in
   queueing systems.  We have already mentioned that tail-drop queues
   naturally tend to lock-out large packets once they are full.

   But also queues with fixed sized buffers reduce the probability that
   small packets will be dropped if (and only if) they allow small
   packets to borrow buffers from the pools for larger packets (see
   Section 4.1.1).  Borrowing effectively makes the maximum queue size
   for small packets greater than that for large packets, because more
   buffers can be used by small packets while less will fit large
   packets.  Incidentally, the bias towards small packets from buffer
   borrowing is nothing like as large as that of RED's byte-mode drop.

   Nonetheless, fixed-buffer memory with tail drop is still prone to
   lock-out large packets, purely because of the tail-drop aspect.  So,
   fixed size packet-buffers should be augmented with a good AQM
   algorithm and packet-mode drop.  If an AQM is too complicated to
   implement with multiple fixed buffer pools, the minimum necessary to
   prevent large packet lock-out is to ensure smaller packets never use
   the last available buffer in any of the pools for larger packets.

4.2.2.  Transport Bias when Decoding

   The above proposals to alter the network equipment to bias towards
   smaller packets have largely carried on outside the IETF process.
   Whereas, within the IETF, there are many different proposals to alter

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   transport protocols to achieve the same goals, i.e. either to make
   the flow bit-rate take account of packet size, or to protect control
   packets from loss.  This memo argues that altering transport
   protocols is the more principled approach.

   A recently approved experimental RFC adapts its transport layer
   protocol to take account of packet sizes relative to typical TCP
   packet sizes.  This proposes a new small-packet variant of TCP-
   friendly rate control [RFC5348] called TFRC-SP [RFC4828].
   Essentially, it proposes a rate equation that inflates the flow rate
   by the ratio of a typical TCP segment size (1500B including TCP
   header) over the actual segment size [PktSizeEquCC].  (There are also
   other important differences of detail relative to TFRC, such as using
   virtual packets [CCvarPktSize] to avoid responding to multiple losses
   per round trip and using a minimum inter-packet interval.)

   Section 4.5.1 of this TFRC-SP spec discusses the implications of
   operating in an environment where queues have been configured to drop
   smaller packets with proportionately lower probability than larger
   ones.  But it only discusses TCP operating in such an environment,
   only mentioning TFRC-SP briefly when discussing how to define
   fairness with TCP.  And it only discusses the byte-mode dropping
   version of RED as it was before Cnodder et al pointed out it didn't
   sufficiently bias towards small packets to make TCP independent of
   packet size.

   So the TFRC-SP spec doesn't address the issue of which of the network
   or the transport _should_ handle fairness between different packet
   sizes.  In its Appendix B.4 it discusses the possibility of both
   TFRC-SP and some network buffers duplicating each other's attempts to
   deliberately bias towards small packets.  But the discussion is not
   conclusive, instead reporting simulations of many of the
   possibilities in order to assess performance but not recommending any
   particular course of action.

   The paper originally proposing TFRC with virtual packets (VP-TFRC)
   [CCvarPktSize] proposed that there should perhaps be two variants to
   cater for the different variants of RED.  However, as the TFRC-SP
   authors point out, there is no way for a transport to know whether
   some queues on its path have deployed RED with byte-mode packet drop
   (except if an exhaustive survey found that no-one has deployed it!--
   see Appendix A).  Incidentally, VP-TFRC also proposed that byte-mode
   RED dropping should really square the packet-size compensation-factor
   (like that of Cnodder's RED_5, but apparently unaware of it).

   Pre-congestion notification [RFC5670] is an IETF technology to use a
   virtual queue for AQM marking for packets within one Diffserv class
   in order to give early warning prior to any real queuing.  The PCN

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   marking algorithms have been designed not to take account of packet
   size when forwarding through queues.  Instead the general principle
   has been to take account of the sizes of marked packets when
   monitoring the fraction of marking at the edge of the network, as
   recommended here.

4.2.3.  Making Transports Robust against Control Packet Losses

   Recently, two RFCs have defined changes to TCP that make it more
   robust against losing small control packets [RFC5562] [RFC5690].  In
   both cases they note that the case for these two TCP changes would be
   weaker if RED were biased against dropping small packets.  We argue
   here that these two proposals are a safer and more principled way to
   achieve TCP performance improvements than reverse engineering RED to
   benefit TCP.

   Although there are no known proposals, it would also be possible and
   perfectly valid to make control packets robust against drop by
   explicitly requesting a lower drop probability using their Diffserv
   code point [RFC2474] to request a scheduling class with lower drop.

   Although not brought to the IETF, a simple proposal from Wischik
   [DupTCP] suggests that the first three packets of every TCP flow
   should be routinely duplicated after a short delay.  It shows that
   this would greatly improve the chances of short flows completing
   quickly, but it would hardly increase traffic levels on the Internet,
   because Internet bytes have always been concentrated in the large
   flows.  It further shows that the performance of many typical
   applications depends on completion of long serial chains of short
   messages.  It argues that, given most of the value people get from
   the Internet is concentrated within short flows, this simple
   expedient would greatly increase the value of the best efforts
   Internet at minimal cost.

4.2.4.  Congestion Notification: Summary of Conflicting Advice

   | transport |  RED_1 (packet |  RED_4 (linear  | RED_5 (square byte |
   |        cc |   mode drop)   | byte mode drop) |     mode drop)     |
   |    TCP or |    s/sqrt(p)   |    sqrt(s/p)    |      1/sqrt(p)     |
   |      TFRC |                |                 |                    |
   |   TFRC-SP |    1/sqrt(p)   |    1/sqrt(sp)   |    1/(s.sqrt(p))   |

    Table 2: Dependence of flow bit-rate per RTT on packet size, s, and
   drop probability, p, when network and/or transport bias towards small
                        packets to varying degrees

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   Table 2 aims to summarise the potential effects of all the advice
   from different sources.  Each column shows a different possible AQM
   behaviour in different queues in the network, using the terminology
   of Cnodder et al outlined earlier (RED_1 is basic RED with packet-
   mode drop).  Each row shows a different transport behaviour: TCP
   [RFC5681] and TFRC [RFC5348] on the top row with TFRC-SP [RFC4828]
   below.  Each cell shows how the bits per round trip of a flow depends
   on packet size, s, and drop probability, p.  In order to declutter
   the formulae to focus on packet-size dependence they are all given
   per round trip, which removes any RTT term.

   Let us assume that the goal is for the bit-rate of a flow to be
   independent of packet size.  Suppressing all inessential details, the
   table shows that this should either be achievable by not altering the
   TCP transport in a RED_5 network, or using the small packet TFRC-SP
   transport (or similar) in a network without any byte-mode dropping
   RED (top right and bottom left).  Top left is the `do nothing'
   scenario, while bottom right is the `do-both' scenario in which bit-
   rate would become far too biased towards small packets.  Of course,
   if any form of byte-mode dropping RED has been deployed on a subset
   of queues that congest, each path through the network will present a
   different hybrid scenario to its transport.

   Whatever, we can see that the linear byte-mode drop column in the
   middle would considerably complicate the Internet.  It's a half-way
   house that doesn't bias enough towards small packets even if one
   believes the network should be doing the biasing.  Section 2
   recommends that _all_ bias in network equipment towards small packets
   should be turned off--if indeed any equipment vendors have
   implemented it--leaving packet-size bias solely as the preserve of
   the transport layer (solely the leftmost, packet-mode drop column).

   In practice it seems that no deliberate bias towards small packets
   has been implemented for production networks.  Of the 19% of vendors
   who responded to a survey of 84 equipment vendors, none had
   implemented byte-mode drop in RED (see Appendix A for details).

5.  Outstanding Issues and Next Steps

5.1.  Bit-congestible Network

   For a connectionless network with nearly all resources being bit-
   congestible the recommended position is clear--that the network
   should not make allowance for packet sizes and the transport should.
   This leaves two outstanding issues:

   o  How to handle any legacy of AQM with byte-mode drop already

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   o  The need to start a programme to update transport congestion
      control protocol standards to take account of packet size.

   A survey of equipment vendors (Section 4.2.4) found no evidence that
   byte-mode packet drop had been implemented, so deployment will be
   sparse at best.  A migration strategy is not really needed to remove
   an algorithm that may not even be deployed.

   A programme of experimental updates to take account of packet size in
   transport congestion control protocols has already started with
   TFRC-SP [RFC4828].

5.2.  Bit- & Packet-congestible Network

   The position is much less clear-cut if the Internet becomes populated
   by a more even mix of both packet-congestible and bit-congestible
   resources (see Appendix B.2).  This problem is not pressing, because
   most Internet resources are designed to be bit-congestible before
   packet processing starts to congest (see Section 1.1).

   The IRTF Internet congestion control research group (ICCRG) has set
   itself the task of reaching consensus on generic forwarding
   mechanisms that are necessary and sufficient to support the
   Internet's future congestion control requirements (the first
   challenge in [RFC6077]).  The research question of whether packet
   congestion might become common and what to do if it does may in the
   future be explored in the IRTF (the "Challenge 3: Packet Size" in

   Note that sometimes it seems that resources might be congested by
   neither bits nor packets, e.g. where the queue for access to a
   wireless medium is in units of transmission opportunities.  However,
   the root cause of congestion of the underlying spectrum is overload
   of bits (see Section 4.1.2).

6.  Security Considerations

   This memo recommends that queues do not bias drop probability due to
   packets size.  For instance dropping small packets less often than
   large creates a perverse incentive for transports to break down their
   flows into tiny segments.  One of the benefits of implementing AQM
   was meant to be to remove this perverse incentive that drop-tail
   queues gave to small packets.

   In practice, transports cannot all be trusted to respond to
   congestion.  So another reason for recommending that queues do not
   bias drop probability towards small packets is to avoid the
   vulnerability to small packet DDoS attacks that would otherwise

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   result.  One of the benefits of implementing AQM was meant to be to
   remove drop-tail's DoS vulnerability to small packets, so we
   shouldn't add it back again.

   If most queues implemented AQM with byte-mode drop, the resulting
   network would amplify the potency of a small packet DDoS attack.  At
   the first queue the stream of packets would push aside a greater
   proportion of large packets, so more of the small packets would
   survive to attack the next queue.  Thus a flood of small packets
   would continue on towards the destination, pushing regular traffic
   with large packets out of the way in one queue after the next, but
   suffering much less drop itself.

   Appendix C explains why the ability of networks to police the
   response of _any_ transport to congestion depends on bit-congestible
   network resources only doing packet-mode not byte-mode drop.  In
   summary, it says that making drop probability depend on the size of
   the packets that bits happen to be divided into simply encourages the
   bits to be divided into smaller packets.  Byte-mode drop would
   therefore irreversibly complicate any attempt to fix the Internet's
   incentive structures.

7.  IANA Considerations

   This document has no actions for IANA.

8.  Conclusions

   This memo identifies the three distinct stages of the congestion
   notification process where implementations need to decide whether to
   take packet size into account.  The recommendations provided in
   Section 2 of this memo are different in each case:

   o  When network equipment measures the length of a queue, if it is
      not feasible to use time it is recommended to count in bytes if
      the network resource is congested by bytes, or to count in packets
      if is congested by packets.

   o  When network equipment decides whether to drop (or mark) a packet,
      it is recommended that the size of the particular packet should
      not be taken into account

   o  However, when a transport algorithm responds to a dropped or
      marked packet, the size of the rate reduction should be
      proportionate to the size of the packet.

   In summary, the answers are 'it depends', 'no' and 'yes' respectively

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   For the specific case of RED, this means that byte-mode queue
   measurement will often be appropriate although byte-mode drop is
   strongly deprecated.

   At the transport layer the IETF should continue updating congestion
   control protocols to take account of the size of each packet that
   indicates congestion.  Also the IETF should continue to make
   protocols less sensitive to losing control packets like SYNs, pure
   ACKs and DNS exchanges.  Although many control packets happen to be
   small, the alternative of network equipment favouring all small
   packets would be dangerous.  That would create perverse incentives to
   split data transfers into smaller packets.

   The memo develops these recommendations from principled arguments
   concerning scaling, layering, incentives, inherent efficiency,
   security and policeability.  But it also addresses practical issues
   such as specific buffer architectures and incremental deployment.
   Indeed a limited survey of RED implementations is discussed, which
   shows there appears to be little, if any, installed base of RED's
   byte-mode drop.  Therefore it can be deprecated with little, if any,
   incremental deployment complications.

   The recommendations have been developed on the well-founded basis
   that most Internet resources are bit-congestible not packet-
   congestible.  We need to know the likelihood that this assumption
   will prevail longer term and, if it might not, what protocol changes
   will be needed to cater for a mix of the two.  The IRTF Internet
   Congestion Control Research Group (ICCRG) is currently working on
   these problems [RFC6077].

9.  Acknowledgements

   Thank you to Sally Floyd, who gave extensive and useful review
   comments.  Also thanks for the reviews from Philip Eardley, David
   Black, Fred Baker, David Taht, Toby Moncaster, Arnaud Jacquet and
   Mirja Kuehlewind as well as helpful explanations of different
   hardware approaches from Larry Dunn and Fred Baker.  We are grateful
   to Bruce Davie and his colleagues for providing a timely and
   efficient survey of RED implementation in Cisco's product range.
   Also grateful thanks to Toby Moncaster, Will Dormann, John Regnault,
   Simon Carter and Stefaan De Cnodder who further helped survey the
   current status of RED implementation and deployment and, finally,
   thanks to the anonymous individuals who responded.

   Bob Briscoe and Jukka Manner were partly funded by Trilogy, a
   research project (ICT- 216372) supported by the European Community
   under its Seventh Framework Programme.  The views expressed here are
   those of the authors only.

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10.  Comments Solicited

   Comments and questions are encouraged and very welcome.  They can be
   addressed to the IETF Transport Area working group mailing list
   <>, and/or to the authors.

11.  References

11.1.  Normative References

   [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.

11.2.  Informative References

   [BLUE02]                   Feng, W-c., Shin, K., Kandlur, D., and D.
                              Saha, "The BLUE active queue management
                              algorithms", IEEE/ACM Transactions on
                              Networking 10(4) 513--528, August 2002, <h

   [CCvarPktSize]             Widmer, J., Boutremans, C., and J-Y. Le
                              Boudec, "Congestion Control for Flows with
                              Variable Packet Size", ACM CCR 34(2) 137--
                              151, 2004,

   [CHOKe_Var_Pkt]            Psounis, K., Pan, R., and B. Prabhaker,
                              "Approximate Fair Dropping for Variable
                              Length Packets", IEEE Micro 21(1):48--56,
                              January-February 2001, <http://

   [DRQ]                      Shin, M., Chong, S., and I. Rhee, "Dual-
                              Resource TCP/AQM for Processing-
                              Constrained Networks", IEEE/ACM
                              Transactions on Networking Vol 16, issue
                              2, April 2008, <

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   [DupTCP]                   Wischik, D., "Short messages", Royal
                              Society workshop on networks: modelling
                              and control , September 2007, <http://

   [ECNFixedWireless]         Siris, V., "Resource Control for Elastic
                              Traffic in CDMA Networks", Proc. ACM
                              MOBICOM'02 , September 2002, <http://

   [Evol_cc]                  Gibbens, R. and F. Kelly, "Resource
                              pricing and the evolution of congestion
                              control", Automatica 35(12)1969--1985,
                              December 1999, <http://

   [I-D.nichols-tsvwg-codel]  Nichols, K. and V. Jacobson, "Controlled
                              Delay Active Queue Management",
                              draft-nichols-tsvwg-codel-01 (work in
                              progress), February 2013.

   [I-D.pan-tsvwg-pie]        Pan, R., Natarajan, P., Piglione, C., and
                              M. Prabhu, "PIE: A Lightweight Control
                              Scheme To Address the Bufferbloat
                              Problem", draft-pan-tsvwg-pie-00 (work in
                              progress), December 2012.

   [IOSArch]                  Bollapragada, V., White, R., and C.
                              Murphy, "Inside Cisco IOS Software
                              Architecture", Cisco Press: CCIE
                              Professional Development ISBN13: 978-1-
                              57870-181-0, July 2000.

   [PktSizeEquCC]             Vasallo, P., "Variable Packet Size
                              Equation-Based Congestion Control", ICSI
                              Technical Report tr-00-008, 2000, <http://

   [RED93]                    Floyd, S. and V. Jacobson, "Random Early
                              Detection (RED) gateways for Congestion
                              Avoidance", IEEE/ACM Transactions on
                              Networking 1(4) 397--413, August 1993, <ht

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   [REDbias]                  Eddy, W. and M. Allman, "A Comparison of
                              RED's Byte and Packet Modes", Computer
                              Networks 42(3) 261--280, June 2003, <http:

   [REDbyte]                  De Cnodder, S., Elloumi, O., and K.
                              Pauwels, "RED behavior with different
                              packet sizes", Proc. 5th IEEE Symposium on
                              Computers and Communications (ISCC) 793--
                              799, July 2000, <

   [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.

   [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.

   [RFC2914]                  Floyd, S., "Congestion Control
                              Principles", BCP 41, RFC 2914,
                              September 2000.

   [RFC3426]                  Floyd, S., "General Architectural and
                              Policy Considerations", RFC 3426,
                              November 2002.

   [RFC3550]                  Schulzrinne, H., Casner, S., Frederick,
                              R., and V. Jacobson, "RTP: A Transport
                              Protocol for Real-Time Applications",
                              STD 64, RFC 3550, July 2003.

   [RFC3714]                  Floyd, S. and J. Kempf, "IAB Concerns
                              Regarding Congestion Control for Voice
                              Traffic in the Internet", RFC 3714,
                              March 2004.

   [RFC4828]                  Floyd, S. and E. Kohler, "TCP Friendly
                              Rate Control (TFRC): The Small-Packet (SP)
                              Variant", RFC 4828, April 2007.

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

   [RFC5562]                  Kuzmanovic, A., Mondal, A., Floyd, S., and
                              K. Ramakrishnan, "Adding Explicit
                              Congestion Notification (ECN) Capability
                              to TCP's SYN/ACK Packets", RFC 5562,
                              June 2009.

   [RFC5670]                  Eardley, P., "Metering and Marking
                              Behaviour of PCN-Nodes", RFC 5670,
                              November 2009.

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

   [RFC5690]                  Floyd, S., Arcia, A., Ros, D., and J.
                              Iyengar, "Adding Acknowledgement
                              Congestion Control to TCP", RFC 5690,
                              February 2010.

   [RFC6077]                  Papadimitriou, D., Welzl, M., Scharf, M.,
                              and B. Briscoe, "Open Research Issues in
                              Internet Congestion Control", RFC 6077,
                              February 2011.

   [RFC6679]                  Westerlund, M., Johansson, I., Perkins,
                              C., O'Hanlon, P., and K. Carlberg,
                              "Explicit Congestion Notification (ECN)
                              for RTP over UDP", RFC 6679, August 2012.

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

   [Rate_fair_Dis]            Briscoe, B., "Flow Rate Fairness:
                              Dismantling a Religion", ACM
                              CCR 37(2)63--74, April 2007, <http://

   [gentle_RED]               Floyd, S., "Recommendation on using the
                              "gentle_" variant of RED", Web page ,
                              March 2000, <

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   [pBox]                     Floyd, S. and K. Fall, "Promoting the Use
                              of End-to-End Congestion Control in the
                              Internet", IEEE/ACM Transactions on
                              Networking 7(4) 458--472, August 1999, <ht

   [pktByteEmail]             Floyd, S., "RED: Discussions of Byte and
                              Packet Modes", email , March 1997, <http:/

Appendix A.  Survey of RED Implementation Status

   This Appendix is informative, not normative.

   In May 2007 a survey was conducted of 84 vendors to assess how widely
   drop probability based on packet size has been implemented in RED
   Table 3.  About 19% of those surveyed replied, giving a sample size
   of 16.  Although in most cases we do not have permission to identify
   the respondents, we can say that those that have responded include
   most of the larger equipment vendors, covering a large fraction of
   the market.  The two who gave permission to be identified were Cisco
   and Alcatel-Lucent.  The others range across the large network
   equipment vendors at L3 & L2, firewall vendors, wireless equipment
   vendors, as well as large software businesses with a small selection
   of networking products.  All those who responded confirmed that they
   have not implemented the variant of RED with drop dependent on packet
   size (2 were fairly sure they had not but needed to check more
   thoroughly).  At the time the survey was conducted, Linux did not
   implement RED with packet-size bias of drop, although we have not
   investigated a wider range of open source code.

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   |                      Response | No. of vendors | %age of vendors |
   |               Not implemented |             14 |             17% |
   |    Not implemented (probably) |              2 |              2% |
   |                   Implemented |              0 |              0% |
   |                   No response |             68 |             81% |
   | Total companies/orgs surveyed |             84 |            100% |

    Table 3: Vendor Survey on byte-mode drop variant of RED (lower drop
                      probability for small packets)

   Where reasons have been given, the extra complexity of packet bias
   code has been most prevalent, though one vendor had a more principled
   reason for avoiding it--similar to the argument of this document.

   Our survey was of vendor implementations, so we cannot be certain
   about operator deployment.  But we believe many queues in the
   Internet are still tail-drop.  The company of one of the co-authors
   (BT) has widely deployed RED, but many tail-drop queues are bound to
   still exist, particularly in access network equipment and on
   middleboxes like firewalls, where RED is not always available.

   Routers using a memory architecture based on fixed size buffers with
   borrowing may also still be prevalent in the Internet.  As explained
   in Section 4.2.1, these also provide a marginal (but legitimate) bias
   towards small packets.  So even though RED byte-mode drop is not
   prevalent, it is likely there is still some bias towards small
   packets in the Internet due to tail drop and fixed buffer borrowing.

Appendix B.  Sufficiency of Packet-Mode Drop

   This Appendix is informative, not normative.

   Here we check that packet-mode drop (or marking) in the network gives
   sufficiently generic information for the transport layer to use.  We
   check against a 2x2 matrix of four scenarios that may occur now or in
   the future (Table 4).  The horizontal and vertical dimensions have
   been chosen because each tests extremes of sensitivity to packet size
   in the transport and in the network respectively.

   Note that this section does not consider byte-mode drop at all.
   Having deprecated byte-mode drop, the goal here is to check that
   packet-mode drop will be sufficient in all cases.

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   |                     Transport |  a) Independent | b) Dependent on |
   |                               |  of packet size |  packet size of |
   | Network                       |  of congestion  |    congestion   |
   |                               |  notifications  |  notifications  |
   | 1) Predominantly              |   Scenario a1)  |   Scenario b1)  |
   | bit-congestible network       |                 |                 |
   | 2) Mix of bit-congestible and |   Scenario a2)  |   Scenario b2)  |
   | pkt-congestible network       |                 |                 |

                Table 4: Four Possible Congestion Scenarios

   Appendix B.1 focuses on the horizontal dimension of Table 4 checking
   that packet-mode drop (or marking) gives sufficient information,
   whether or not the transport uses it--scenarios b) and a)

   Appendix B.2 focuses on the vertical dimension of Table 4, checking
   that packet-mode drop gives sufficient information to the transport
   whether resources in the network are bit-congestible or packet-
   congestible (these terms are defined in Section 1.1).

   Notation:  To be concrete, we will compare two flows with different
      packet sizes, s_1 and s_2.  As an example, we will take s_1 = 60B
      = 480b and s_2 = 1500B = 12,000b.

      A flow's bit rate, x [bps], is related to its packet rate, u
      [pps], by

         x(t) = s.u(t).

      In the bit-congestible case, path congestion will be denoted by
      p_b, and in the packet-congestible case by p_p.  When either case
      is implied, the letter p alone will denote path congestion.

B.1.  Packet-Size (In)Dependence in Transports

   In all cases we consider a packet-mode drop queue that indicates
   congestion by dropping (or marking) packets with probability p
   irrespective of packet size. We use an example value of loss
   (marking) probability, p=0.1%.

   A transport like RFC5681 TCP treats a congestion notification on any
   packet whatever its size as one event.  However, a network with just
   the packet-mode drop algorithm does give more information if the
   transport chooses to use it.  We will use Table 5 to illustrate this.

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   We will set aside the last column until later.  The columns labelled
   "Flow 1" and "Flow 2" compare two flows consisting of 60B and 1500B
   packets respectively.  The body of the table considers two separate
   cases, one where the flows have equal bit-rate and the other with
   equal packet-rates.  In both cases, the two flows fill a 96Mbps link.
   Therefore, in the equal bit-rate case they each have half the bit-
   rate (48Mbps).  Whereas, with equal packet-rates, flow 1 uses 25
   times smaller packets so it gets 25 times less bit-rate--it only gets
   1/(1+25) of the link capacity (96Mbps/26 = 4Mbps after rounding).  In
   contrast flow 2 gets 25 times more bit-rate (92Mbps) in the equal
   packet rate case because its packets are 25 times larger.  The packet
   rate shown for each flow could easily be derived once the bit-rate
   was known by dividing bit-rate by packet size, as shown in the column
   labelled "Formula".

       Parameter               Formula      Flow 1  Flow 2 Combined
       ----------------------- ----------- ------- ------- --------
       Packet size             s/8             60B  1,500B    (Mix)
       Packet size             s              480b 12,000b    (Mix)
       Pkt loss probability    p              0.1%    0.1%     0.1%

       Bit-rate                x            48Mbps  48Mbps   96Mbps
       Packet-rate             u = x/s     100kpps   4kpps  104kpps
       Absolute pkt-loss-rate  p*u          100pps    4pps   104pps
       Absolute bit-loss-rate  p*u*s        48kbps  48kbps   96kbps
       Ratio of lost/sent pkts p*u/u          0.1%    0.1%     0.1%
       Ratio of lost/sent bits p*u*s/(u*s)    0.1%    0.1%     0.1%

       Bit-rate                x             4Mbps  92Mbps   96Mbps
       Packet-rate             u = x/s       8kpps   8kpps   15kpps
       Absolute pkt-loss-rate  p*u            8pps    8pps    15pps
       Absolute bit-loss-rate  p*u*s         4kbps  92kbps   96kbps
       Ratio of lost/sent pkts p*u/u          0.1%    0.1%     0.1%
       Ratio of lost/sent bits p*u*s/(u*s)    0.1%    0.1%     0.1%

    Table 5: Absolute Loss Rates and Loss Ratios for Flows of Small and
                      Large Packets and Both Combined

   So far we have merely set up the scenarios.  We now consider
   congestion notification in the scenario.  Two TCP flows with the same
   round trip time aim to equalise their packet-loss-rates over time.
   That is the number of packets lost in a second, which is the packets
   per second (u) multiplied by the probability that each one is dropped
   (p).  Thus TCP converges on the "Equal packet-rate" case, where both
   flows aim for the same "Absolute packet-loss-rate" (both 8pps in the

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   Packet-mode drop actually gives flows sufficient information to
   measure their loss-rate in bits per second, if they choose, not just
   packets per second.  Each flow can count the size of a lost or marked
   packet and scale its rate-response in proportion (as TFRC-SP does).
   The result is shown in the row entitled "Absolute bit-loss-rate",
   where the bits lost in a second is the packets per second (u)
   multiplied by the probability of losing a packet (p) multiplied by
   the packet size (s).  Such an algorithm would try to remove any
   imbalance in bit-loss-rate such as the wide disparity in the "Equal
   packet-rate" case (4kbps vs. 92kbps).  Instead, a packet-size-
   dependent algorithm would aim for equal bit-loss-rates, which would
   drive both flows towards the "Equal bit-rate" case, by driving them
   to equal bit-loss-rates (both 48kbps in this example).

   The explanation so far has assumed that each flow consists of packets
   of only one constant size.  Nonetheless, it extends naturally to
   flows with mixed packet sizes.  In the right-most column of Table 5 a
   flow of mixed size packets is created simply by considering flow 1
   and flow 2 as a single aggregated flow.  There is no need for a flow
   to maintain an average packet size.  It is only necessary for the
   transport to scale its response to each congestion indication by the
   size of each individual lost (or marked) packet.  Taking for example
   the "Equal packet-rate" case, in one second about 8 small packets and
   8 large packets are lost (making closer to 15 than 16 losses per
   second due to rounding).  If the transport multiplies each loss by
   its size, in one second it responds to 8*480b and 8*12,000b lost
   bits, adding up to 96,000 lost bits in a second.  This double checks
   correctly, being the same as 0.1% of the total bit-rate of 96Mbps.
   For completeness, the formula for absolute bit-loss-rate is p(u1*s1+

   Incidentally, a transport will always measure the loss probability
   the same irrespective of whether it measures in packets or in bytes.
   In other words, the ratio of lost to sent packets will be the same as
   the ratio of lost to sent bytes.  (This is why TCP's bit rate is
   still proportional to packet size even when byte-counting is used, as
   recommended for TCP in [RFC5681], mainly for orthogonal security
   reasons.)  This is intuitively obvious by comparing two example
   flows; one with 60B packets, the other with 1500B packets.  If both
   flows pass through a queue with drop probability 0.1%, each flow will
   lose 1 in 1,000 packets.  In the stream of 60B packets the ratio of
   bytes lost to sent will be 60B in every 60,000B; and in the stream of
   1500B packets, the loss ratio will be 1,500B out of 1,500,000B. When
   the transport responds to the ratio of lost to sent packets, it will
   measure the same ratio whether it measures in packets or bytes: 0.1%
   in both cases.  The fact that this ratio is the same whether measured
   in packets or bytes can be seen in Table 5, where the ratio of lost
   to sent packets and the ratio of lost to sent bytes is always 0.1% in

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   all cases (recall that the scenario was set up with p=0.1%).

   This discussion of how the ratio can be measured in packets or bytes
   is only raised here to highlight that it is irrelevant to this memo!
   Whether a transport depends on packet size or not depends on how this
   ratio is used within the congestion control algorithm.

   So far we have shown that packet-mode drop passes sufficient
   information to the transport layer so that the transport can take
   account of bit-congestion, by using the sizes of the packets that
   indicate congestion.  We have also shown that the transport can
   choose not to take packet size into account if it wishes.  We will
   now consider whether the transport can know which to do.

B.2.  Bit-Congestible and Packet-Congestible Indications

   As a thought-experiment, imagine an idealised congestion notification
   protocol that supports both bit-congestible and packet-congestible
   resources.  It would require at least two ECN flags, one for each of
   bit-congestible and packet-congestible resources.

   1.  A packet-congestible resource trying to code congestion level p_p
       into a packet stream should mark the idealised `packet
       congestion' field in each packet with probability p_p
       irrespective of the packet's size.  The transport should then
       take a packet with the packet congestion field marked to mean
       just one mark, irrespective of the packet size.

   2.  A bit-congestible resource trying to code time-varying byte-
       congestion level p_b into a packet stream should mark the `byte
       congestion' field in each packet with probability p_b, again
       irrespective of the packet's size.  Unlike before, the transport
       should take a packet with the byte congestion field marked to
       count as a mark on each byte in the packet.

   This hides a fundamental problem--much more fundamental than whether
   we can magically create header space for yet another ECN flag, or
   whether it would work while being deployed incrementally.
   Distinguishing drop from delivery naturally provides just one
   implicit bit of congestion indication information--the packet is
   either dropped or not.  It is hard to drop a packet in two ways that
   are distinguishable remotely.  This is a similar problem to that of
   distinguishing wireless transmission losses from congestive losses.

   This problem would not be solved even if ECN were universally
   deployed.  A congestion notification protocol must survive a
   transition from low levels of congestion to high.  Marking two states
   is feasible with explicit marking, but much harder if packets are

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   dropped.  Also, it will not always be cost-effective to implement AQM
   at every low level resource, so drop will often have to suffice.

   We are not saying two ECN fields will be needed (and we are not
   saying that somehow a resource should be able to drop a packet in one
   of two different ways so that the transport can distinguish which
   sort of drop it was!).  These two congestion notification channels
   are a conceptual device to illustrate a dilemma we could face in the
   future.  Section 3 gives four good reasons why it would be a bad idea
   to allow for packet size by biasing drop probability in favour of
   small packets within the network.  The impracticality of our thought
   experiment shows that it will be hard to give transports a practical
   way to know whether to take account of the size of congestion
   indication packets or not.

   Fortunately, this dilemma is not pressing because by design most
   equipment becomes bit-congested before its packet-processing becomes
   congested (as already outlined in Section 1.1).  Therefore transports
   can be designed on the relatively sound assumption that a congestion
   indication will usually imply bit-congestion.

   Nonetheless, although the above idealised protocol isn't intended for
   implementation, we do want to emphasise that research is needed to
   predict whether there are good reasons to believe that packet
   congestion might become more common, and if so, to find a way to
   somehow distinguish between bit and packet congestion [RFC3714].

   Recently, the dual resource queue (DRQ) proposal [DRQ] has been made
   on the premise that, as network processors become more cost
   effective, per packet operations will become more complex
   (irrespective of whether more function in the network is desirable).
   Consequently the premise is that CPU congestion will become more
   common.  DRQ is a proposed modification to the RED algorithm that
   folds both bit congestion and packet congestion into one signal
   (either loss or ECN).

   Finally, we note one further complication.  Strictly, packet-
   congestible resources are often cycle-congestible.  For instance, for
   routing look-ups load depends on the complexity of each look-up and
   whether the pattern of arrivals is amenable to caching or not.  This
   also reminds us that any solution must not require a forwarding
   engine to use excessive processor cycles in order to decide how to
   say it has no spare processor cycles.

Appendix C.  Byte-mode Drop Complicates Policing Congestion Response

   This section is informative, not normative.

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   There are two main classes of approach to policing congestion
   response: i) policing at each bottleneck link or ii) policing at the
   edges of networks.  Packet-mode drop in RED is compatible with
   either, while byte-mode drop precludes edge policing.

   The simplicity of an edge policer relies on one dropped or marked
   packet being equivalent to another of the same size without having to
   know which link the drop or mark occurred at.  However, the byte-mode
   drop algorithm has to depend on the local MTU of the line--it needs
   to use some concept of a 'normal' packet size.  Therefore, one
   dropped or marked packet from a byte-mode drop algorithm is not
   necessarily equivalent to another from a different link.  A policing
   function local to the link can know the local MTU where the
   congestion occurred.  However, a policer at the edge of the network
   cannot, at least not without a lot of complexity.

   The early research proposals for type (i) policing at a bottleneck
   link [pBox] used byte-mode drop, then detected flows that contributed
   disproportionately to the number of packets dropped.  However, with
   no extra complexity, later proposals used packet mode drop and looked
   for flows that contributed a disproportionate amount of dropped bytes

   Work is progressing on the congestion exposure protocol (ConEx
   [RFC6789]), which enables a type (ii) edge policer located at a
   user's attachment point.  The idea is to be able to take an
   integrated view of the effect of all a user's traffic on any link in
   the internetwork.  However, byte-mode drop would effectively preclude
   such edge policing because of the MTU issue above.

   Indeed, making drop probability depend on the size of the packets
   that bits happen to be divided into would simply encourage the bits
   to be divided into smaller packets in order to confuse policing.  In
   contrast, as long as a dropped/marked packet is taken to mean that
   all the bytes in the packet are dropped/marked, a policer can remain
   robust against bits being re-divided into different size packets or
   across different size flows [Rate_fair_Dis].

Appendix D.  Changes from Previous Versions

   To be removed by the RFC Editor on publication.

   Full incremental diffs between each version are available at
   (courtesy of the rfcdiff tool):

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   From -10 to -11:  Following a further WGLC:

      *  Abstract: clarified that advice applies to all AQMs including
         newer ones

      *  Abstract & Intro: changed 'read' to 'detect', because you don't
         read losses, you detect them.

      *  S.1.  Introduction: Disambiguated summary of advice on queue

      *  Clarified that the doc deprecates any preference based solely
         on packet size, it's not only against preferring smaller

      *  S.4.1.2.  Congestion Measurement without a Queue: Explained
         that a queue of TXOPs represents a queue into spectrum
         congested by too many bits.

      *  S.5.2: Bit- & Packet-congestible Network: Referred to
         explanation in S.4.1.2 to make the point that TXOPs are not a
         primary unit of workload like bits and packets are, even though
         you get queues of TXOPs.

      *  6.  Security: Disambiguated 'bias towards'.

      *  8.  Conclusions: Made consistent with recommendation to use
         time if possible for queue measurement.

   From -09 to -10:  Following IESG review:

      *  Updates 2309: Left header unchanged reflecting eventual IESG
         consensus [Sean Turner, Pete Resnick].

      *  S.1 Intro: This memo adds to the congestion control principles
         enumerated in BCP 41 [Pete Resnick]

      *  Abstract, S.1, S.1.1, s.1.2 Intro, Scoping and Example: Made
         applicability to all AQMs clearer listing some more example
         AQMs and explained that we always use RED for examples, but
         this doesn't mean it's not applicable to other AQMs.  [A number
         of reviewers have described the draft as "about RED"]

      *  S.1 & S.2.1 Queue measurement: Explained that the choice
         between measuring the queue in packets or bytes is only
         relevant if measuring it in time units is infeasible [So as not
         to imply that we haven't noticed the advances made by PDPC &

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      *  S.1.1.  Terminology: Better explained why hybrid systems
         congested by both packets and bytes are often designed to be
         treated as bit-congestible [Richard Barnes].

      *  S.2.1.  Queue measurement advice: Added examples.  Added a
         counter-example to justify SHOULDs rather than MUSTs.  Pointed
         to S.4.1 for a list of more complicated scenarios.  [Benson
         Schliesser, OpsDir]

      *  S2.2.  Recommendation on Encoding Congestion Notification:
         Removed SHOULD treat packets equally, leaving only SHOULD NOT
         drop dependent on packet size, to avoid it sounding like we're
         saying QoS is not allowed.  Pointed to possible app-specific
         legacy use of byte-mode as a counter-example that prevents us
         saying MUST NOT.  [Pete Resnick]

      *  S.2.3.  Recommendation on Responding to Congestion: capitalised
         the two SHOULDs in recommendations for TCP, and gave possible
         counter-examples. [noticed while dealing with Pete Resnick's

      *  S2.4.  Splitting & Merging: RTCP -> RTP/RTCP [Pete McCann, Gen-

      *  S.3.2 Small != Control: many control packets are small ->
         ...tend to be small [Stephen Farrell]

      *  S.3.1 Perverse incentives: Changed transport designers to app
         developers [Stephen Farrell]

      *  S.4.1.1.  Fixed Size Packet Buffers: Nearly completely re-
         written to simplify and to reverse the advice when the
         underlying resource is bit-congestible, irrespective of whether
         the buffer consists of fixed-size packet buffers.  [Richard
         Barnes & Benson Schliesser]

      *  S.  Packet Size Bias Regardless of AQM: Largely re-
         written to reflect the earlier change in advice about fixed-
         size packet buffers, and to primarily focus on getting rid of
         tail-drop, not various nuances of tail-drop.  [Richard Barnes &
         Benson Schliesser]

      *  Editorial corrections [Tim Bray, AppsDir, Pete McCann, Gen-ART
         and others]

      *  Updated refs (two I-Ds have become RFCs).  [Pete McCann]

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   From -08 to -09:  Following WG last call:

      *  S.2.1: Made RED-related queue measurement recommendations

      *  S.2.3: Added to "Recommendation on Responding to Congestion" to
         make it clear that we are definitely not saying transports have
         to equalise bit-rates, just how to do it and not do it, if you
         want to.

      *  S.3: Clarified motivation sections S.3.3 "Transport-Independent
         Network" and S.3.5 "Implementation Efficiency"

      *  S.3.4: Completely changed motivating argument from "Scaling
         Congestion Control with Packet Size" to "Partial Deployment of

   From -07 to -08:

      *  Altered abstract to say it provides best current practice and
         highlight that it updates RFC2309

      *  Added null IANA section

      *  Updated refs

   From -06 to -07:

      *  A mix-up with the corollaries and their naming in 2.1 to 2.3

   From -05 to -06:

      *  Primarily editorial fixes.

   From -04 to -05:

      *  Changed from Informational to BCP and highlighted non-normative
         sections and appendices

      *  Removed language about consensus

      *  Added "Example Comparing Packet-Mode Drop and Byte-Mode Drop"

      *  Arranged "Motivating Arguments" into a more logical order and
         completely rewrote "Transport-Independent Network" & "Scaling
         Congestion Control with Packet Size" arguments.  Removed "Why

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      *  Clarified applicability of certain recommendations

      *  Shifted vendor survey to an Appendix

      *  Cut down "Outstanding Issues and Next Steps"

      *  Re-drafted the start of the conclusions to highlight the three
         distinct areas of concern

      *  Completely re-wrote appendices

      *  Editorial corrections throughout.

   From -03 to -04:

      *  Reordered Sections 2 and 3, and some clarifications here and
         there based on feedback from Colin Perkins and Mirja

   From -02 to -03  (this version)

      *  Structural changes:

         +  Split off text at end of "Scaling Congestion Control with
            Packet Size" into new section "Transport-Independent

         +  Shifted "Recommendations" straight after "Motivating
            Arguments" and added "Conclusions" at end to reinforce

         +  Added more internal structure to Recommendations, so that
            recommendations specific to RED or to TCP are just
            corollaries of a more general recommendation, rather than
            being listed as a separate recommendation.

         +  Renamed "State of the Art" as "Critical Survey of Existing
            Advice" and retitled a number of subsections with more
            descriptive titles.

         +  Split end of "Congestion Coding: Summary of Status" into a
            new subsection called "RED Implementation Status".

         +  Removed text that had been in the Appendix "Congestion
            Notification Definition: Further Justification".

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      *  Reordered the intro text a little.

      *  Made it clearer when advice being reported is deprecated and
         when it is not.

      *  Described AQM as in network equipment, rather than saying "at
         the network layer" (to side-step controversy over whether
         functions like AQM are in the transport layer but in network

      *  Minor improvements to clarity throughout

   From -01 to -02:

      *  Restructured the whole document for (hopefully) easier reading
         and clarity.  The concrete recommendation, in RFC2119 language,
         is now in Section 8.

   From -00 to -01:

      *  Minor clarifications throughout and updated references

   From briscoe-byte-pkt-mark-02 to ietf-byte-pkt-congest-00:

      *  Added note on relationship to existing RFCs

      *  Posed the question of whether packet-congestion could become
         common and deferred it to the IRTF ICCRG.  Added ref to the
         dual-resource queue (DRQ) proposal.

      *  Changed PCN references from the PCN charter & architecture to
         the PCN marking behaviour draft most likely to imminently
         become the standards track WG item.

   From -01 to -02:

      *  Abstract reorganised to align with clearer separation of issue
         in the memo.

      *  Introduction reorganised with motivating arguments removed to
         new Section 3.

      *  Clarified avoiding lock-out of large packets is not the main or
         only motivation for RED.

      *  Mentioned choice of drop or marking explicitly throughout,
         rather than trying to coin a word to mean either.

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      *  Generalised the discussion throughout to any packet forwarding
         function on any network equipment, not just routers.

      *  Clarified the last point about why this is a good time to sort
         out this issue: because it will be hard / impossible to design
         new transports unless we decide whether the network or the
         transport is allowing for packet size.

      *  Added statement explaining the horizon of the memo is long
         term, but with short term expediency in mind.

      *  Added material on scaling congestion control with packet size
         (Section 3.4).

      *  Separated out issue of normalising TCP's bit rate from issue of
         preference to control packets (Section 3.2).

      *  Divided up Congestion Measurement section for clarity,
         including new material on fixed size packet buffers and buffer
         carving (Section 4.1.1 & Section 4.2.1) and on congestion
         measurement in wireless link technologies without queues
         (Section 4.1.2).

      *  Added section on 'Making Transports Robust against Control
         Packet Losses' (Section 4.2.3) with existing & new material

      *  Added tabulated results of vendor survey on byte-mode drop
         variant of RED (Table 3).

   From -00 to -01:

      *  Clarified applicability to drop as well as ECN.

      *  Highlighted DoS vulnerability.

      *  Emphasised that drop-tail suffers from similar problems to
         byte-mode drop, so only byte-mode drop should be turned off,
         not RED itself.

      *  Clarified the original apparent motivations for recommending
         byte-mode drop included protecting SYNs and pure ACKs more than
         equalising the bit rates of TCPs with different segment sizes.
         Removed some conjectured motivations.

      *  Added support for updates to TCP in progress (ackcc & ecn-syn-

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      *  Updated survey results with newly arrived data.

      *  Pulled all recommendations together into the conclusions.

      *  Moved some detailed points into two additional appendices and a

      *  Considerable clarifications throughout.

      *  Updated references

Authors' Addresses

   Bob Briscoe
   B54/77, Adastral Park
   Martlesham Heath
   Ipswich  IP5 3RE

   Phone: +44 1473 645196

   Jukka Manner
   Aalto University
   Department of Communications and Networking (Comnet)
   P.O. Box 13000
   FIN-00076 Aalto

   Phone: +358 9 470 22481

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