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Versions: (draft-briscoe-tsvwg-byte-pkt-mark) 00 01 02 03 04 05 06 07 08 09 10 11 12 RFC 7141

Transport Area Working Group                                  B. Briscoe
Internet-Draft                                                        BT
Updates: 2309 (if approved)                             October 23, 2009
Intended status: Informational
Expires: April 26, 2010


                Byte and Packet Congestion Notification
                  draft-ietf-tsvwg-byte-pkt-congest-01

Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   Internet-Drafts are draft documents valid for a maximum of six months
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   The list of current Internet-Drafts can be accessed at
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   The list of Internet-Draft Shadow Directories can be accessed at
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   This Internet-Draft will expire on April 26, 2010.

Copyright Notice

   Copyright (c) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents in effect on the date of
   publication of this document (http://trustee.ietf.org/license-info).
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.

Abstract

   This memo concerns dropping or marking packets using active queue
   management (AQM) such as random early detection (RED) or pre-



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   congestion notification (PCN).  The primary conclusion is that packet
   size should be taken into account when transports read congestion
   indications, not when network equipment writes them.  Reducing drop
   of small packets 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 specific transport problems.
   Network layer algorithms like the byte-mode packet drop variant of
   RED 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.







































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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  6
     1.1.  Requirements Notation  . . . . . . . . . . . . . . . . . .  9
   2.  Motivating Arguments . . . . . . . . . . . . . . . . . . . . .  9
     2.1.  Scaling Congestion Control with Packet Size  . . . . . . .  9
     2.2.  Avoiding Perverse Incentives to (ab)use Smaller Packets  . 10
     2.3.  Small != Control . . . . . . . . . . . . . . . . . . . . . 12
     2.4.  Implementation Efficiency  . . . . . . . . . . . . . . . . 12
   3.  Working Definition of Congestion Notification  . . . . . . . . 12
   4.  Congestion Measurement . . . . . . . . . . . . . . . . . . . . 13
     4.1.  Congestion Measurement by Queue Length . . . . . . . . . . 13
       4.1.1.  Fixed Size Packet Buffers  . . . . . . . . . . . . . . 13
     4.2.  Congestion Measurement without a Queue . . . . . . . . . . 14
   5.  Idealised Wire Protocol Coding . . . . . . . . . . . . . . . . 15
   6.  The State of the Art . . . . . . . . . . . . . . . . . . . . . 17
     6.1.  Congestion Measurement: Status . . . . . . . . . . . . . . 17
     6.2.  Congestion Coding: Status  . . . . . . . . . . . . . . . . 18
       6.2.1.  Network Bias when Encoding . . . . . . . . . . . . . . 18
       6.2.2.  Transport Bias when Decoding . . . . . . . . . . . . . 20
       6.2.3.  Making Transports Robust against Control Packet
               Losses . . . . . . . . . . . . . . . . . . . . . . . . 21
       6.2.4.  Congestion Coding: Summary of Status . . . . . . . . . 22
   7.  Outstanding Issues and Next Steps  . . . . . . . . . . . . . . 24
     7.1.  Bit-congestible World  . . . . . . . . . . . . . . . . . . 24
     7.2.  Bit- & Packet-congestible World  . . . . . . . . . . . . . 24
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 25
   9.  Conclusions  . . . . . . . . . . . . . . . . . . . . . . . . . 26
   10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 28
   11. Comments Solicited . . . . . . . . . . . . . . . . . . . . . . 28
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 28
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 28
     12.2. Informative References . . . . . . . . . . . . . . . . . . 29
   Editorial Comments . . . . . . . . . . . . . . . . . . . . . . . .
   Appendix A.  Example Scenarios . . . . . . . . . . . . . . . . . . 32
     A.1.  Notation . . . . . . . . . . . . . . . . . . . . . . . . . 32
     A.2.  Bit-congestible resource, equal bit rates (Ai) . . . . . . 32
     A.3.  Bit-congestible resource, equal packet rates (Bi)  . . . . 33
     A.4.  Pkt-congestible resource, equal bit rates (Aii)  . . . . . 34
     A.5.  Pkt-congestible resource, equal packet rates (Bii) . . . . 35
   Appendix B.  Congestion Notification Definition: Further
                Justification . . . . . . . . . . . . . . . . . . . . 35
   Appendix C.  Byte-mode Drop Complicates Policing Congestion
                Response  . . . . . . . . . . . . . . . . . . . . . . 36
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 37






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Changes from Previous Versions

   To be removed by the RFC Editor on publication.

   Full incremental diffs between each version are available at
   <http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#byte-pkt-congest>
   or
   <http://tools.ietf.org/wg/tsvwg/draft-ietf-tsvwg-byte-pkt-congest/>
   (courtesy of the rfcdiff tool):

   From -00 to -01 (this version):

      *  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 2.

      *  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.

      *  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.




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      *  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 2.1).

      *  Separated out issue of normalising TCP's bit rate from issue of
         preference to control packets (Section 2.3).

      *  Divided up Congestion Measurement section for clarity,
         including new material on fixed size packet buffers and buffer
         carving (Section 4.1.1 & Section 6.2.1) and on congestion
         measurement in wireless link technologies without queues
         (Section 4.2).

      *  Added section on 'Making Transports Robust against Control
         Packet Losses' (Section 6.2.3) with existing & new material
         included.

      *  Added tabulated results of vendor survey on byte-mode drop
         variant of RED (Table 2).

      *

   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-
         ack).

      *  Updated survey results with newly arrived data.

      *  Pulled all recommendations together into the conclusions.

      *  Moved some detailed points into two additional appendices and a
         note.



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      *  Considerable clarifications throughout.

      *  Updated references


1.  Introduction

   When notifying 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 come to conclusions on what these
   principles will mean for future protocol design, taking into account
   the deployments we have already.

   Note that 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
   [I-D.ietf-pcn-marking-behaviour]).  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.

   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 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).

   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.

   Resources may be congestible at higher levels of granularity than
   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



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   be applicable for congestion notification protocols controlling per-
   flow and per-session processing or state.

   The byte vs. packet dilemma arises at three stages in the congestion
   notification process:

   Measuring congestion  When the congested resource decides locally how
      to measure how congested it is.  (Should the queue be measured in
      bytes or packets?);

   Coding congestion notification into the wire protocol:  When the
      congested resource decides how to notify the level of congestion.
      (Should the level of notification depend on the byte-size of each
      particular packet carrying the notification?);

   Decoding congestion notification from the wire protocol:  When the
      transport interprets the notification.  (Should the byte-size of a
      missing or marked packet be taken into account?).

   In RED, whether to use packets or bytes when measuring queues is
   called packet-mode or byte-mode queue measurement.  This choice is
   now fairly well understood but is included in Section 4 to document
   it in the RFC series.

   The controversy is mainly around the other two stages: whether to
   allow for packet size when the network codes or when the transport
   decodes congestion notification.  In RED, the variant that reduces
   drop probability for packets based on their size in bytes is called
   byte-mode drop, while the variant that doesn't is called packet mode
   drop.  Whether queues are measured in bytes or packets is an
   orthogonal choice, termed byte-mode queue measurement or packet-mode
   queue measurement.

   Currently, the RFC series is silent on this matter other than a paper
   trail of advice referenced from [RFC2309], which conditionally
   recommends byte-mode (packet-size dependent) drop [pktByteEmail].
   However, all the implementers who responded to our survey
   (Section 6.2.4) have not followed this advice.  The primary purpose
   of this memo is to build a definitive consensus against deliberate
   preferential treatment for small packets in AQM algorithms and to
   record this advice within the RFC series.

   Now is a good time to discuss whether fairness between different
   sized packets would best be implemented in the network layer, or at
   the transport, for a number of reasons:

   1.  The packet vs. byte issue requires speedy resolution because the
       IETF pre-congestion notification (PCN) working group is about to



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       standardise the external behaviour of a PCN congestion
       notification (AQM) algorithm [I-D.ietf-pcn-marking-behaviour];

   2.  [RFC2309] says RED may either take account of packet size or not
       when dropping, but gives no recommendation between the two,
       referring instead to advice on the performance implications in an
       email [pktByteEmail], which recommends byte-mode drop.  Further,
       just before RFC2309 was issued, an addendum was added to the
       archived email that revisited the issue of packet vs. byte-mode
       drop in its last para, making the recommendation less clear-cut;

   3.  Without the present memo, the only advice in the RFC series on
       packet size bias in AQM algorithms would be a reference to an
       archived email in [RFC2309] (including an addendum at the end of
       the email to correct the original).

   4.  The IRTF Internet Congestion Control Research Group (ICCRG)
       recently took on the challenge of building consensus on what
       common congestion control support should be required from network
       forwarding functions in future
       [I-D.irtf-iccrg-welzl-congestion-control-open-research].  The
       wider Internet community needs to discuss whether the complexity
       of adjusting for packet size should be in the network or in
       transports;

   5.  Given there are many good reasons why larger path max
       transmission units (PMTUs) would help solve a number of scaling
       issues, we don't want to create any bias against large packets
       that is greater than their true cost;

   6.  The IETF has started to consider the question of fairness between
       flows that use different packet sizes (e.g. in the small-packet
       variant of TCP-friendly rate control, TFRC-SP [RFC4828]).  Given
       transports with different packet sizes, if we don't decide
       whether the network or the transport should allow for packet
       size, it will be hard if not impossible to design any transport
       protocol so that its bit-rate relative to other transports meets
       design guidelines [RFC5033] (Note however that, if the concern
       were fairness between users, rather than between flows
       [Rate_fair_Dis], relative rates between flows would have to come
       under run-time control rather than being embedded in protocol
       designs).

   This memo is initially concerned with how we should correctly scale
   congestion control functions with packet size for the long term.  But
   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.  It turns out that the 'correct' variant of RED to deploy



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   seems to be the one everyone has deployed, and no-one who responded
   to our survey has implemented the other variant.  However, at the
   transport layer, TCP congestion control is a widely deployed protocol
   that we argue doesn't scale correctly with packet size.  To date this
   hasn't been a significant problem because most TCPs have been used
   with similar packet sizes.  But, as we design new congestion
   controls, we should build in scaling with packet size rather than
   assuming we should follow TCP's example.

   Motivating arguments for our advice are given next in Section 2.
   Then the body of the memo starts from first principles, defining
   congestion notification in Section 3 then determining the correct way
   to measure congestion (Section 4) and to design an idealised
   congestion notification protocol (Section 5).  It then surveys the
   advice given previously in the RFC series, the research literature
   and the deployed legacy (Section 6) before listing outstanding issues
   (Section 7) that will need resolution both to achieve the ideal
   protocol and to handle legacy.  After discussing security
   considerations (Section 8) strong recommendations for the way forward
   are given in the conclusions (Section 9).

1.1.  Requirements Notation

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


2.  Motivating Arguments

2.1.  Scaling Congestion Control with Packet Size

   There are two ways of interpreting a dropped or marked packet.  It
   can either be considered as a single loss event or as loss/marking of
   the bytes in the packet.  Here we try to design a test to see which
   approach scales with packet size.

   Given bit-congestible is the more common case, consider a bit-
   congestible link shared by many flows, so that each busy period tends
   to cause packets to be lost from different flows.  The test compares
   two identical scenarios with the same applications, the same numbers
   of sources and the same load.  But the sources break the load into
   large packets in one scenario and small packets in the other.  Of
   course, because the load is the same, there will be proportionately
   more packets in the small packet case.

   The test of whether a congestion control scales with packet size is
   that it should respond in the same way to the same congestion



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   excursion, irrespective of the size of the packets that the bytes
   causing congestion happen to be broken down into.

   A bit-congestible queue suffering a congestion excursion has to drop
   or mark the same excess bytes whether they are in a few large packets
   or many small packets.  So for the same congestion excursion, the
   same amount of bytes have to be shed to get the load back to its
   operating point.  But, of course, for smaller packets more packets
   will have to be discarded to shed the same bytes.

   If all the transports interpret each drop/mark as a single loss event
   irrespective of the size of the packet dropped, those with smaller
   packets will respond more to the same congestion excursion, failing
   our test.  On the other hand, if they respond proportionately less
   when smaller packets are dropped/marked, overall they will be able to
   respond the same to the same congestion excursion.

   Therefore, for a congestion control to scale with packet size it
   should respond to dropped or marked bytes (as TFRC-SP [RFC4828]
   effectively does), not just to dropped or marked packets irrespective
   of packet size (as TCP does).

   The email [pktByteEmail] referred to by RFC2309 says the question of
   whether a packet's own size should affect its drop probability
   "depends on the dominant end-to-end congestion control mechanisms".
   But we argue the network layer should not be optimised for whatever
   transport is predominant.

   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.  But even
   though reducing the drop probability of small packets helps ensure
   TCPs with different packet sizes will achieve similar bit rates, we
   argue this should be achieved in TCP itself, not in the network.

   Effectively, favouring small packets is reverse engineering of the
   network layer around 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?"

2.2.  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].  Again,
   imagine a scenario where the same bit rate of packets will contribute



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   the same to 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 this case:

   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.

      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 8).
      Wherever possible neither should be used.

   Normal transports:  Even if a transport is not malicious, if it finds
      small packets go faster, it will tend to act in its own interest
      and use them.  Queues that give advantage to small packets create
      an evolutionary pressure for 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, 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](S.B.4)[Note_Variation]).  Even though both flows
   arrive with the same bit rate, the bit rate the RED queue aims to
   pass to the line will be 750k for the flow of larger packet but 990k
   for the smaller packets (but because of rate variation it will be
   less than this target).

   It can be seen that this behaviour reopens the same denial of service
   vulnerability that drop tail queues offer to floods of small packet,
   though not necessarily as strongly (see Section 8).






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2.3.  Small != Control

   It is tempting to drop small packets with lower probability to
   improve performance, because many control packets are small (TCP SYNs
   & ACKs, DNS queries & responses, SIP messages, HTTP GETs, etc) and
   dropping fewer control packets considerably improves performance.
   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 again, rather than fix these problems in the network layer, 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 6.2.3).

2.4.  Implementation Efficiency

   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.  This isn't a
   principled reason in itself, but it is a happy consequence of the
   other principled reasons.


3.  Working Definition of Congestion Notification

   Rather than aim to achieve what many have tried and failed, this memo
   will not try to define congestion.  It will give a working definition
   of what congestion notification should be taken to mean for this
   document.  Congestion notification is a changing signal that aims to
   communicate the ratio E/L, where E is the instantaneous excess load
   offered to a resource that it cannot (or would not) serve and L is
   the instantaneous offered load.

   The phrase `would not serve' is added, because AQM systems (e.g.
   RED, PCN [I-D.ietf-pcn-marking-behaviour]) use a virtual capacity
   smaller than actual capacity, then notify congestion of this virtual
   capacity in order to avoid congestion of the actual capacity.

   Note that the denominator is offered load, not capacity.  Therefore



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   congestion notification is a real number bounded by the range [0,1].
   This ties in with the most well-understood measure of congestion
   notification: drop fraction (often loosely called loss rate).  It
   also means that congestion has a natural interpretation as a
   probability; the probability of offered traffic not being served (or
   being marked as at risk of not being served).  Appendix B describes a
   further incidental benefit that arises from using load as the
   denominator of congestion notification.


4.  Congestion Measurement

4.1.  Congestion Measurement by Queue Length

   Queue length is usually the most correct and simplest way to measure
   congestion of a resource.  To avoid the pathological effects of drop
   tail, an AQM function can then be used to transform queue length into
   the probability of dropping or marking a packet (e.g.  RED's
   piecewise linear function between thresholds).  If the resource is
   bit-congestible, the length of the queue SHOULD be measured in bytes.
   If the resource is packet-congestible, the length of the queue SHOULD
   be measured 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.  We discuss the implications
   on RED's byte mode and packet mode for measuring queue length in
   Section 6.

4.1.1.  Fixed Size Packet Buffers

   Some, mostly older, queuing hardware sets aside fixed sized buffers
   in which to store each packet in the queue.  Also, with some
   hardware, any fixed sized buffers not completely filled by a packet
   are padded when transmitted to the wire.  If we imagine a theoretical
   forwarding system with both queuing and transmission in fixed, MTU-
   sized units, it should clearly be treated as packet-congestible,
   because the queue length in packets would be a good model of
   congestion of the lower layer link.

   If we now imagine a hybrid forwarding system with transmission delay
   largely dependent on the byte-size of packets but buffers of one MTU
   per packet, it should strictly require a more complex algorithm to
   determine the probability of congestion.  It should be treated as two
   resources in sequence, where the sum of the byte-sizes of the packets
   within each packet buffer models congestion of the line while the
   length of the queue in packets models congestion of the queue.  Then
   the probability of congesting the forwarding buffer would be a
   conditional probability--conditional on the previously calculated
   probability of congesting the line.



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   However, in systems that use fixed size buffers, it is unusual for
   all the buffers used by an interface to be the same size.  Typically
   pools of 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, it is easier to work out
   what should be done if we temporarily set aside the possibility of
   such borrowing.  Then, with fixed pools of buffers for different
   sized packets and no borrowing, the size of each pool and the current
   queue length in each pool would both be measured in packets.  So an
   AQM algorithm would have to maintain the queue length for each pool,
   and judge whether to drop/mark a packet of a particular size by
   looking at the pool for packets of that size and using the length (in
   packets) of its queue.

   We now return to the issue we temporarily set aside: small packets
   borrowing space in larger buffers.  In this case, the only difference
   is that the pools for smaller packets have a maximum queue size that
   includes all the pools for larger packets.  And every time a packet
   takes a larger buffer, the current queue size has to be incremented
   for all queues in the pools of buffers less than or equal to the
   buffer size used.

   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 6.2.1.  But here we can give a simple summary of
   the present discussion on how to measure the length of queues of
   fixed buffers: no matter how complicated the scheme is, ultimately
   any fixed buffer system will need to measure its queue length in
   packets not bytes.

4.2.  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,
   wireless spectrum is bit-congestible (for a given coding scheme),
   because interference increases with the rate at which bits are
   transmitted.  But wireless link protocols do not always maintain a
   queue that depends on spectrum interference.  Similarly, power
   limited resources are also 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 power.

   However, AQM algorithms don't require a queue in order to work.  For



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   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.


5.  Idealised Wire Protocol Coding

   We will start by inventing an idealised congestion notification
   protocol before discussing how to make it practical.  The idealised
   protocol is shown to be correct using examples in Appendix A.

   Congestion notification involves the congested resource coding a
   congestion notification signal into the packet stream and the
   transports decoding it.  The idealised protocol uses two different
   (imaginary) fields in each datagram to signal congestion: one for
   byte congestion and one for packet congestion.

   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 just a conceptual device.  They allow us to defer having to
   decide whether to distinguish between byte and packet congestion when
   the network resource codes the signal or when the transport decodes
   it.

   However, although this idealised mechanism isn't intended for
   implementation, we do want to emphasise that we may need to find a
   way to implement it, because it could become necessary to somehow
   distinguish between bit and packet congestion [RFC3714].  Currently a
   design goal of network processing equipment such as routers and
   firewalls is to keep packet processing uncongested even under worst
   case bit rates with minimum packet sizes.  Therefore, packet-
   congestion is currently rare, but there is no guarantee that it will
   not become common with future technology trends.

   The idealised wire protocol is given below.  It accounts for packet
   sizes at the transport layer, not in the network, and then only in
   the case of bit-congestible resources.  This avoids the perverse
   incentive to send smaller packets and the DoS vulnerability that
   would otherwise result if the network were to bias towards them (see
   the motivating argument about avoiding perverse incentives in
   Section 2.2):



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   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.

   The worked examples in Appendix A show that transports can extract
   sufficient and correct congestion notification from these protocols
   for cases when two flows with different packet sizes have matching
   bit rates or matching packet rates.  Examples are also given that mix
   these two flows into one to show that a flow with mixed packet sizes
   would still be able to extract sufficient and correct information.

   Sufficient and correct congestion information means that there is
   sufficient information for the two different types of transport
   requirements:

   Ratio-based:  Established transport congestion controls like TCP's
      [RFC5681] aim to achieve equal segment rates per RTT through the
      same bottleneck--TCP friendliness [RFC3448].  They work with the
      ratio of dropped to delivered segments (or marked to unmarked
      segments in the case of ECN).  The example scenarios show that
      these ratio-based transports are effectively the same whether
      counting in bytes or packets, because the units cancel out.
      (Incidentally, 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.)

   Absolute-target-based:  Other congestion controls proposed in the
      research community aim to limit the volume of congestion caused to
      a constant weight parameter.  [MulTCP][WindowPropFair] are
      examples of weighted proportionally fair transports designed for
      cost-fair environments [Rate_fair_Dis].  In this case, the
      transport requires a count (not a ratio) of dropped/marked bytes
      in the bit-congestible case and of dropped/marked packets in the
      packet congestible case.







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6.  The State of the Art

   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.  This email clarified that there were in fact two
   orthogonal choices: whether to measure queue length in bytes or
   packets (Section 6.1 below) and whether the drop probability of an
   individual packet should depend on its own size (Section 6.2 below).

6.1.  Congestion Measurement: Status

   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 (see
   Section 4).

   Where buffers are not configured or legacy buffers cannot be
   configured to the above guideline, we don't have to make allowances
   for such legacy in future protocol design.  If a bit-congestible
   buffer is measured in packets, the operator will have 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 undersensitive to high
   proportions of large packets.  But an operator can safely keep such a
   legacy buffer because any undersensitivity during unusual traffic
   mixes cannot lead to congestion collapse given the buffer will
   eventually revert to tail drop, discarding proportionately more large
   packets.

   Some modern queue implementations give a choice for setting RED's
   thresholds in byte-mode or packet-mode.  This may merely be an
   administrator-interface preference, not altering how the queue itself
   is measured but on some hardware it does actually change the way it
   measures its queue.  Whether a resource is bit-congestible or packet-
   congestible is a property of the resource, so an admin SHOULD NOT



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   ever need to, or be able to, configure the way a queue measures
   itself.

   We believe the question of whether to measure queues in bytes or
   packets is fairly well understood these days.  The only outstanding
   issues concern how to measure congestion when the queue is bit
   congestible but the resource is packet congestible or vice versa (see
   Section 4).  But there is no controversy over what should be done.
   It's just you have to be an expert in probability to work out what
   should be done and, even if you have, it's not always easy to find a
   practical algorithm to implement it.

6.2.  Congestion Coding: Status

6.2.1.  Network Bias when Encoding

   The previously mentioned email [pktByteEmail] referred to by
   [RFC2309] said that the choice over whether a packet's own size
   should affect its drop probability "depends on the dominant end-to-
   end congestion control mechanisms".  [Section 2 argues against this
   approach, citing the excellent advice in RFC3246.]  The referenced
   email went on to argue 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, but advised that
   most scarce resources in the Internet were currently bit-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].  [The problem
   could 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 byte-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



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   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 stated
   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.

   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 for this change in the
   literature, however Eddy and Allman conducted experiments [REDbias]
   that assessed how sensitive RED was to this parameter, amongst other
   things.  No-one seems to have pointed out that this changed algorithm
   can often lead to drop probabilities of greater than 1 [which should
   ring alarm bells hinting that there's 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.

   The byte-mode drop variant of RED is, of course, not the only
   possible bias towards small packets in queueing algorithms.  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.  As was explained in Section 4.1.1 on fixed
   size buffer carving, 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.

   However, in itself, the bias towards small packets caused by buffer
   borrowing is perfectly correct.  Lower drop probability for small
   packets is legitimate in buffer borrowing schemes, because small
   packets genuinely congest the machine's buffer memory less than large
   packets, given they can fit in more spaces.  The bias towards small
   packets is not artificially added (as it is in RED's byte-mode drop
   algorithm), it merely reflects the reality of the way fixed buffer
   memory gets congested.  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 a
   good AQM algorithm like RED with packet-mode drop should be used with
   fixed buffer memories where possible.  If RED is too complicated to



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   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.

6.2.2.  Transport Bias when Decoding

   The above proposals to alter the network layer to give a bias towards
   smaller packets have largely carried on outside the IETF process
   (unless one counts a reference in an informational RFC to an archived
   email!).  Whereas, within the IETF, there are many different
   proposals to alter 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 [RFC3448] 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



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   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 Section 6.2.4).  Incidentally, VP-TFRC also proposed that byte-
   mode RED dropping should really square the packet size compensation
   factor (like that of RED_5, but apparently unaware of it).

   Pre-congestion notification [I-D.ietf-pcn-marking-behaviour] is a
   proposal 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 proposed PCN 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.

6.2.3.  Making Transports Robust against Control Packet Losses

   Recently, two drafts have proposed changes to TCP that make it more
   robust against losing small control packets [I-D.ietf-tcpm-ecnsyn]
   [I-D.floyd-tcpm-ackcc].  In both cases they note that the case for
   these 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 no proposals exist as far as we know, 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.

   The re-ECN protocol proposal [I-D.briscoe-tsvwg-re-ecn-tcp] is
   designed so that transports can be made more robust against losing
   control packets.  It gives queues an incentive to optionally give
   preference against drop to packets with the 'feedback not
   established' codepoint in the proposed 'extended ECN' field.  Senders
   have incentives to use this codepoint sparingly, but they can use it
   on control packets to reduce their chance of being dropped.  For
   instance, the proposed modification to TCP for re-ECN uses this
   codepoint on the SYN and SYN-ACK.

   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



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   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.

6.2.4.  Congestion Coding: Summary of Status

   +-----------+----------------+-----------------+--------------------+
   | 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 1: Dependence of flow bit-rate per RTT on packet size s and
   drop rate p when network and/or transport bias towards small packets
                            to varying degrees

   Table 1 aims to summarise the positions we may now be in.  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 [RFC3448] on
   the top row with TFRC-SP [RFC4828] below.  Suppressing all
   inessential details the table shows that independence from packet
   size should either be achievable by not altering the TCP transport in
   a RED_5 network, or using the small packet TFRC-SP transport 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 selection of congested queues, each path
   will present a different hybrid scenario to its transport.

   Whatever, we can see that the linear byte-mode drop column in the
   middle considerably complicates 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.  We argue below that _all_
   network layer bias 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).

   A survey has been conducted of 84 vendors to assess how widely drop



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   probability based on packet size has been implemented in RED.  Prior
   to the survey, an individual approach to Cisco received confirmation
   that, having checked the code-base for each of the product ranges,
   Cisco has not implemented any discrimination based on packet size in
   any AQM algorithm in any of its products.  Also an individual
   approach to Alcatel-Lucent drew a confirmation that it was very
   likely that none of their products contained RED code that
   implemented any packet-size bias.

   Turning to our more formal survey (Table 2), about 19% of those
   surveyed have replied so far, giving a sample size of 16.  Although
   we do not have permission to identify the respondents, we can say
   that those that have responded include most of the larger vendors,
   covering a large fraction of the market.  They 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.  So far, all those who have
   responded have confirmed that they have not implemented the variant
   of RED with drop dependent on packet size (2 are fairly sure they
   haven't but need to check more thoroughly).

   +-------------------------------+----------------+-----------------+
   |                      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 2: 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.  We
   have established that Linux does not implement RED with packet size
   drop bias, although we have not investigated a wider range of open
   source code.

   Finally, we repeat that RED's byte mode drop is not the only way to
   bias towards small packets--tail-drop tends to lock-out large packets
   very effectively.  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.  My own company (BT) has
   widely deployed RED, but there are bound to be many tail-drop queues,
   particularly in access network equipment and on middleboxes like



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   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 6.2.1,
   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.


7.  Outstanding Issues and Next Steps

7.1.  Bit-congestible World

   For a connectionless network with nearly all resources being bit-
   congestible we believe the recommended position is now unarguably
   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
      deployed;

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

   The sample of returns from our vendor survey Section 6.2.4 suggest
   that byte-mode packet drop seems not to be implemented at all let
   alone deployed, or if it is, it is likely to be very sparse.
   Therefore, we do not really need a migration strategy from all but
   nothing to nothing.

   A programme of standards updates to take account of packet size in
   transport congestion control protocols has started with TFRC-SP
   [RFC4828], while weighted TCPs implemented in the research community
   [WindowPropFair] could form the basis of a future change to TCP
   congestion control [RFC5681] itself.

7.2.  Bit- & Packet-congestible World

   Nonetheless, a connectionless network with both bit-congestible and
   packet-congestible resources is a different matter.  If we believe we
   should allow for this possibility in the future, this space contains
   a truly open research issue.

   The idealised wire protocol coding described in Section 5 requires at
   least two flags for congestion of bit-congestible and packet-
   congestible resources.  This hides a fundamental problem--much more
   fundamental than whether we can magically create header space for yet
   another ECN flag in IPv4, or whether it would work while being



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   deployed incrementally.  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 dropped.  Also, it will not always be cost-effective to
   implement AQM at every low level resource, so drop will often have to
   suffice.  Distinguishing drop from delivery naturally provides just
   one congestion flag--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.

   We should also note that, strictly, packet-congestible resources are
   actually cycle-congestible because load also depends on the
   complexity of each look-up and whether the pattern of arrivals is
   amenable to caching or not.  Further, this 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.

   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 layer 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).

   The problem of signalling packet processing congestion is not
   pressing, as most Internet resources are designed to be bit-
   congestible before packet processing starts to congest.  However, 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
   [I-D.irtf-iccrg-welzl-congestion-control-open-research]).  Therefore,
   rather than not giving this problem any thought at all, just because
   it is hard and currently hypothetical, we defer the question of
   whether packet congestion might become common and what to do if it
   does to the IRTF (the 'Small Packets' challenge in
   [I-D.irtf-iccrg-welzl-congestion-control-open-research]).


8.  Security Considerations

   This draft recommends that queues do not bias drop probability
   towards small packets as this creates a perverse incentive for
   transports to break down their flows into tiny segments.  One of the



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   benefits of implementing AQM was meant to be to remove this perverse
   incentive that drop-tail queues gave to small packets.  Of course, if
   transports really want to make the greatest gains, they don't have to
   respond to congestion anyway.  But we don't want applications that
   are trying to behave to discover that they can go faster by using
   smaller 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
   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.


9.  Conclusions

   The strong conclusion is that AQM algorithms such as RED SHOULD NOT
   use byte-mode drop.  More generally, the Internet's congestion
   notification protocols (drop, ECN & PCN) SHOULD take account of
   packet size when the notification is read by the transport layer, NOT
   when it is written by the network layer.  This approach offers
   sufficient and correct 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.

   The alternative of deflating RED's drop probability for smaller
   packet sizes (byte-mode drop) has no enduring advantages.  It is more



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   complex, it creates the perverse incentive to fragment segments into
   tiny pieces and it reopens the vulnerability to floods of small-
   packets that drop-tail queues suffered from and AQM was designed to
   remove.  Byte-mode drop is a change to the network layer that makes
   allowance for an omission from the design of TCP, effectively reverse
   engineering the network layer to contrive to make two TCPs with
   different packet sizes run at equal bit rates (rather than packet
   rates) under the same path conditions.  It also improves TCP
   performance by reducing the chance that a SYN or a pure ACK will be
   dropped, because they are small.  But we SHOULD NOT hack the network
   layer to improve or fix certain transport protocols.  No matter how
   predominant a transport protocol is (even if it's TCP), trying to
   correct for its failings by biasing towards small packets in the
   network layer creates a perverse incentive to break down all flows
   from all transports into tiny segments.

   So far, our survey of 84 vendors across the industry has drawn
   responses from about 19%, none of whom have implemented the byte mode
   packet drop variant of RED.  Given there appears to be little, if
   any, installed base it seems we can recommend removal of byte-mode
   drop from RED with little, if any, incremental deployment impact.

   If a vendor has implemented byte-mode drop, and an operator has
   turned it on, it is strongly RECOMMENDED that it SHOULD be turned
   off.  Note that RED as a whole SHOULD NOT be turned off, as without
   it, a drop tail queue also biases against large packets.  But note
   also that turning off byte-mode may alter the relative performance of
   applications using different packet sizes, so it would be advisable
   to establish the implications before turning it off.

   Instead, the IETF transport area should continue its programme of
   updating congestion control protocols to take account of packet size
   and to make transports less sensitive to losing control packets like
   SYNs and pure ACKS.

   NOTE WELL that RED's byte-mode queue measurement is fine, being
   completely orthogonal to byte-mode drop.  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.

   The above conclusions cater for the Internet as it is today with
   most, if not all, resources being primarily bit-congestible.  A
   secondary conclusion of this memo is that we may see more packet-
   congestible resources in the future, so research may be needed to
   extend the Internet's congestion notification (drop or ECN) so that
   it can handle a mix of bit-congestible and packet-congestible
   resources.



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10.  Acknowledgements

   Thank you to Sally Floyd, who gave extensive and useful review
   comments.  Also thanks for the reviews from Philip Eardley, Toby
   Moncaster and Arnaud Jacquet as well as helpful explanations of
   different hardware approaches from Larry Dunn and Fred Baker.  I am
   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 is 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
   author only.


11.  Comments Solicited

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


12.  References

12.1.  Normative References

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

   [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.

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

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

   [RFC5033]  Floyd, S. and M. Allman, "Specifying New Congestion



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              Control Algorithms", BCP 133, RFC 5033, August 2007.

12.2.  Informative References

   [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,
              <http://doi.acm.org/10.1145/997150.997162>.

   [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,
              <http://dx.doi.org/10.1109/TNET.2007.900415>.

   [DupTCP]   Wischik, D., "Short messages", Royal Society workshop on
              networks: modelling and control , September 2007, <http://
              www.cs.ucl.ac.uk/staff/ucacdjw/Research/shortmsg.html>.

   [ECNFixedWireless]
              Siris, V., "Resource Control for Elastic Traffic in CDMA
              Networks", Proc. ACM MOBICOM'02 , September 2002, <http://
              www.ics.forth.gr/netlab/publications/
              resource_control_elastic_cdma.html>.

   [Evol_cc]  Gibbens, R. and F. Kelly, "Resource pricing and the
              evolution of congestion control", Automatica 35(12)1969--
              1985, December 1999,
              <http://www.statslab.cam.ac.uk/~frank/evol.html>.

   [I-D.briscoe-tsvwg-re-ecn-tcp]
              Briscoe, B., Jacquet, A., Moncaster, T., and A. Smith,
              "Re-ECN: Adding Accountability for Causing Congestion to
              TCP/IP", draft-briscoe-tsvwg-re-ecn-tcp-07 (work in
              progress), March 2009.

   [I-D.floyd-tcpm-ackcc]
              Floyd, S., "Adding Acknowledgement Congestion Control to
              TCP", draft-floyd-tcpm-ackcc-06 (work in progress),
              July 2009.

   [I-D.ietf-pcn-marking-behaviour]
              Eardley, P., "Metering and marking behaviour of PCN-
              nodes", draft-ietf-pcn-marking-behaviour-05 (work in
              progress), August 2009.

   [I-D.ietf-tcpm-ecnsyn]
              Floyd, S., "Adding Explicit Congestion Notification (ECN)



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              Capability to TCP's SYN/ACK  Packets",
              draft-ietf-tcpm-ecnsyn-10 (work in progress), May 2009.

   [I-D.irtf-iccrg-welzl-congestion-control-open-research]
              Welzl, M., Scharf, M., Briscoe, B., and D. Papadimitriou,
              "Open Research Issues in Internet Congestion Control",
              draft-irtf-iccrg-welzl-congestion-control-open-research-05
              (work in progress), September 2009.

   [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.

   [MulTCP]   Crowcroft, J. and Ph. Oechslin, "Differentiated End to End
              Internet Services using a Weighted Proportional Fair
              Sharing TCP", CCR 28(3) 53--69, July 1998, <http://
              www.cs.ucl.ac.uk/staff/J.Crowcroft/hipparch/pricing.html>.

   [PktSizeEquCC]
              Vasallo, P., "Variable Packet Size Equation-Based
              Congestion Control", ICSI Technical Report tr-00-008,
              2000, <http://http.icsi.berkeley.edu/ftp/global/pub/
              techreports/2000/tr-00-008.pdf>.

   [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,
              <http://www.icir.org/floyd/papers/red/red.html>.

   [REDbias]  Eddy, W. and M. Allman, "A Comparison of RED's Byte and
              Packet Modes", Computer Networks 42(3) 261--280,
              June 2003,
              <http://www.ir.bbn.com/documents/articles/redbias.ps>.

   [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,
              <http://www.icir.org/floyd/red/Elloumi99.pdf>.

   [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.

   [RFC3448]  Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 3448, January 2003.




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

   [RFC4782]  Floyd, S., Allman, M., Jain, A., and P. Sarolahti, "Quick-
              Start for TCP and IP", RFC 4782, January 2007.

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

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

   [Rate_fair_Dis]
              Briscoe, B., "Flow Rate Fairness: Dismantling a Religion",
              ACM CCR 37(2)63--74, April 2007,
              <http://portal.acm.org/citation.cfm?id=1232926>.

   [WindowPropFair]
              Siris, V., "Service Differentiation and Performance of
              Weighted Window-Based Congestion Control and Packet
              Marking Algorithms in ECN Networks", Computer
              Communications 26(4) 314--326, 2002, <http://
              www.ics.forth.gr/netgroup/publications/
              weighted_window_control.html>.

   [gentle_RED]
              Floyd, S., "Recommendation on using the "gentle_" variant
              of RED", Web page , March 2000,
              <http://www.icir.org/floyd/red/gentle.html>.

   [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,
              <http://www.aciri.org/floyd/end2end-paper.html>.

   [pktByteEmail]
              Floyd, S., "RED: Discussions of Byte and Packet Modes",
              email , March 1997,
              <http://www-nrg.ee.lbl.gov/floyd/REDaveraging.txt>.

   [xcp-spec]
              Falk, A., "Specification for the Explicit Control Protocol
              (XCP)", draft-falk-xcp-spec-03 (work in progress),
              July 2007.

              (Expired)



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Editorial Comments

   [Note_Variation]  The algorithm of the byte-mode drop variant of RED
                     switches off any bias towards small packets
                     whenever the smoothed queue length dictates that
                     the drop probability of large packets should be
                     100%. In the example in the Introduction, as the
                     large packet drop probability varies around 25% the
                     small packet drop probability will vary around 1%,
                     but with occasional jumps to 100% whenever the
                     instantaneous queue (after drop) manages to sustain
                     a length above the 100% drop point for longer than
                     the queue averaging period.


Appendix A.  Example Scenarios

A.1.  Notation

   To prove our idealised wire protocol (Section 5) is correct, we will
   compare two flows with different packet sizes, s_1 and s_2 [bit/pkt],
   to make sure their transports each see the correct congestion
   notification.  Initially, within each flow we will take all packets
   as having equal sizes, but later we will generalise to flows within
   which packet sizes vary.  A flow's bit rate, x [bit/s], is related to
   its packet rate, u [pkt/s], by

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

   We will consider a 2x2 matrix of four scenarios:

   +-----------------------------+------------------+------------------+
   |           resource type and |   A) Equal bit   |   B) Equal pkt   |
   |            congestion level |       rates      |       rates      |
   +-----------------------------+------------------+------------------+
   |     i) bit-congestible, p_b |       (Ai)       |       (Bi)       |
   |    ii) pkt-congestible, p_p |       (Aii)      |       (Bii)      |
   +-----------------------------+------------------+------------------+

                                  Table 3

A.2.  Bit-congestible resource, equal bit rates (Ai)

   Starting with the bit-congestible scenario, for two flows to maintain
   equal bit rates (Ai) the ratio of the packet rates must be the
   inverse of the ratio of packet sizes: u_2/u_1 = s_1/s_2.  So, for
   instance, a flow of 60B packets would have to send 25x more packets
   to achieve the same bit rate as a flow of 1500B packets.  If a



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   congested resource marks proportion p_b of packets irrespective of
   size, the ratio of marked packets received by each transport will
   still be the same as the ratio of their packet rates, p_b.u_2/p_b.u_1
   = s_1/s_2.  So of the 25x more 60B packets sent, 25x more will be
   marked than in the 1500B packet flow, but 25x more won't be marked
   too.

   In this scenario, the resource is bit-congestible, so it always uses
   our idealised bit-congestion field when it marks packets.  Therefore
   the transport should count marked bytes not packets.  But it doesn't
   actually matter for ratio-based transports like TCP (Section 5).  The
   ratio of marked to unmarked bytes seen by each flow will be p_b, as
   will the ratio of marked to unmarked packets.  Because they are
   ratios, the units cancel out.

   If a flow sent an inconsistent mixture of packet sizes, we have said
   it should count the ratio of marked and unmarked bytes not packets in
   order to correctly decode the level of congestion.  But actually, if
   all it is trying to do is decode p_b, it still doesn't matter.  For
   instance, imagine the two equal bit rate flows were actually one flow
   at twice the bit rate sending a mixture of one 1500B packet for every
   thirty 60B packets. 25x more small packets will be marked and 25x
   more will be unmarked.  The transport can still calculate p_b whether
   it uses bytes or packets for the ratio.  In general, for any
   algorithm which works on a ratio of marks to non-marks, either bytes
   or packets can be counted interchangeably, because the choice cancels
   out in the ratio calculation.

   However, where an absolute target rather than relative volume of
   congestion caused is important (Section 5), as it is for congestion
   accountability [Rate_fair_Dis], the transport must count marked bytes
   not packets, in this bit-congestible case.  Aside from the goal of
   congestion accountability, this is how the bit rate of a transport
   can be made independent of packet size; by ensuring the rate of
   congestion caused is kept to a constant weight [WindowPropFair],
   rather than merely responding to the ratio of marked and unmarked
   bytes.

   Note the unit of byte-congestion-volume is the byte.

A.3.  Bit-congestible resource, equal packet rates (Bi)

   If two flows send different packet sizes but at the same packet rate,
   their bit rates will be in the same ratio as their packet sizes, x_2/
   x_1 = s_2/s_1.  For instance, a flow sending 1500B packets at the
   same packet rate as another sending 60B packets will be sending at
   25x greater bit rate.  In this case, if a congested resource marks
   proportion p_b of packets irrespective of size, the ratio of packets



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   received with the byte-congestion field marked by each transport will
   be the same, p_b.u_2/p_b.u_1 = 1.

   Because the byte-congestion field is marked, the transport should
   count marked bytes not packets.  But because each flow sends
   consistently sized packets it still doesn't matter for ratio-based
   transports.  The ratio of marked to unmarked bytes seen by each flow
   will be p_b, as will the ratio of marked to unmarked packets.
   Therefore, if the congestion control algorithm is only concerned with
   the ratio of marked to unmarked packets (as is TCP), both flows will
   be able to decode p_b correctly whether they count packets or bytes.

   But if the absolute volume of congestion is important, e.g. for
   congestion accountability, the transport must count marked bytes not
   packets.  Then the lower bit rate flow using smaller packets will
   rightly be perceived as causing less byte-congestion even though its
   packet rate is the same.

   If the two flows are mixed into one, of bit rate x1+x2, with equal
   packet rates of each size packet, the ratio p_b will still be
   measurable by counting the ratio of marked to unmarked bytes (or
   packets because the ratio cancels out the units).  However, if the
   absolute volume of congestion is required, the transport must count
   the sum of congestion marked bytes, which indeed gives a correct
   measure of the rate of byte-congestion p_b(x_1 + x_2) caused by the
   combined bit rate.

A.4.  Pkt-congestible resource, equal bit rates (Aii)

   Moving to the case of packet-congestible resources, we now take two
   flows that send different packet sizes at the same bit rate, but this
   time the pkt-congestion field is marked by the resource with
   probability p_p.  As in scenario Ai with the same bit rates but a
   bit-congestible resource, the flow with smaller packets will have a
   higher packet rate, so more packets will be both marked and unmarked,
   but in the same proportion.

   This time, the transport should only count marks without taking into
   account packet sizes.  Transports will get the same result, p_p, by
   decoding the ratio of marked to unmarked packets in either flow.

   If one flow imitates the two flows but merged together, the bit rate
   will double with more small packets than large.  The ratio of marked
   to unmarked packets will still be p_p.  But if the absolute number of
   pkt-congestion marked packets is counted it will accumulate at the
   combined packet rate times the marking probability, p_p(u_1+u_2), 26x
   faster than packet congestion accumulates in the single 1500B packet
   flow of our example, as required.



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   But if the transport is interested in the absolute number of packet
   congestion, it should just count how many marked packets arrive.  For
   instance, a flow sending 60B packets will see 25x more marked packets
   than one sending 1500B packets at the same bit rate, because it is
   sending more packets through a packet-congestible resource.

   Note the unit of packet congestion is a packet.

A.5.  Pkt-congestible resource, equal packet rates (Bii)

   Finally, if two flows with the same packet rate, pass through a
   packet-congestible resource, they will both suffer the same
   proportion of marking, p_p, irrespective of their packet sizes.  On
   detecting that the pkt-congestion field is marked, the transport
   should count packets, and it will be able to extract the ratio p_p of
   marked to unmarked packets from both flows, irrespective of packet
   sizes.

   Even if the transport is monitoring the absolute amount of packets
   congestion over a period, still it will see the same amount of packet
   congestion from either flow.

   And if the two equal packet rates of different size packets are mixed
   together in one flow, the packet rate will double, so the absolute
   volume of packet-congestion will accumulate at twice the rate of
   either flow, 2p_p.u_1 = p_p(u_1+u_2).


Appendix B.  Congestion Notification Definition: Further Justification

   In Section 3 on the definition of congestion notification, load not
   capacity was used as the denominator.  This also has a subtle
   significance in the related debate over the design of new transport
   protocols--typical new protocol designs (e.g. in XCP [xcp-spec] &
   Quickstart [RFC4782]) expect the sending transport to communicate its
   desired flow rate to the network and network elements to
   progressively subtract from this so that the achievable flow rate
   emerges at the receiving transport.

   Congestion notification with total load in the denominator can serve
   a similar purpose (though in retrospect not in advance like XCP &
   QuickStart).  Congestion notification is a dimensionless fraction but
   each source can extract necessary rate information from it because it
   already knows what its own rate is.  Even though congestion
   notification doesn't communicate a rate explicitly, from each
   source's point of view congestion notification represents the
   fraction of the rate it was sending a round trip ago that couldn't
   (or wouldn't) be served by available resources.  After they were



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   sent, all these fractions of each source's offered load added up to
   the aggregate fraction of offered load seen by the congested
   resource.  So, the source can also know the total excess rate by
   multiplying total load by congestion level.  Therefore congestion
   notification, as one scale-free dimensionless fraction, implicitly
   communicates the instantaneous excess flow rate, albeit a RTT ago.


Appendix C.  Byte-mode Drop Complicates Policing Congestion Response

   This appendix 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.

   To be able to police a transport's response to congestion when
   fairness can only be judged over time and over all an individual's
   flows, the policer has to have an integrated view of all the
   congestion an individual (not just one flow) has caused due to all
   traffic entering the Internet from that individual.  This is termed
   congestion accountability.

   But a byte-mode drop algorithm has to depend on the local MTU of the
   line - an algorithm needs to use some concept of a 'normal' packet
   size.  Therefore, one dropped or marked packet is not necessarily
   equivalent to another unless you know the MTU at the queue that where
   it was dropped/marked.  To have an integrated view of a user, we
   believe congestion policing has to be located at an individual's
   attachment point to the Internet [I-D.briscoe-tsvwg-re-ecn-tcp].  But
   from there it cannot know the MTU of each remote queue that caused
   each drop/mark.  Therefore it cannot take an integrated approach to
   policing all the responses to congestion of all the transports of one
   individual.  Therefore it cannot police anything.

   The security/incentive argument _for_ packet-mode drop is similar.
   Firstly, confining RED to packet-mode drop would not preclude
   bottleneck policing approaches such as [pBox] as it seems likely they
   could work just as well by monitoring the volume of dropped bytes
   rather than packets.  Secondly packet-mode dropping/marking naturally
   allows the congestion notification of packets to be globally
   meaningful without relying on MTU information held elsewhere.

   Because we recommend that a dropped/marked packet should be 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].  Therefore
   policing would work naturally with just simple packet-mode drop in
   RED.




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Internet-Draft   Byte and Packet Congestion Notification    October 2009


   In summary, 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.


Author's Address

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

   Phone: +44 1473 645196
   Email: bob.briscoe@bt.com
   URI:   http://bobbriscoe.net/
































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