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RFC 6297 - A Survey of Lower-than-Best-Effort Transport Protocols


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Internet Engineering Task Force (IETF)                          M. Welzl
Request for Comments: 6297                            University of Oslo
Category: Informational                                           D. Ros
ISSN: 2070-1721                                    IT / Telecom Bretagne
                                                               June 2011

         A Survey of Lower-than-Best-Effort Transport Protocols

Abstract

   This document provides a survey of transport protocols that are
   designed to have a smaller bandwidth and/or delay impact on standard
   TCP than standard TCP itself when they share a bottleneck with it.
   Such protocols could be used for delay-insensitive "background"
   traffic, as they provide what is sometimes called a "less than" (or
   "lower than") best-effort service.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc6297.

Copyright Notice

   Copyright (c) 2011 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
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
   2.  Delay-Based Transport Protocols  . . . . . . . . . . . . . . .  3
     2.1.  Accuracy of Delay-Based Congestion Predictors  . . . . . .  6
     2.2.  Potential Issues with Delay-Based Congestion Control
           for LBE Transport  . . . . . . . . . . . . . . . . . . . .  7
   3.  Non-Delay-Based Transport Protocols  . . . . . . . . . . . . .  8
   4.  Upper-Layer Approaches . . . . . . . . . . . . . . . . . . . .  8
     4.1.  Receiver-Oriented, Flow-Control-Based Approaches . . . . .  9
   5.  Network-Assisted Approaches  . . . . . . . . . . . . . . . . . 10
   6.  LEDBAT Considerations  . . . . . . . . . . . . . . . . . . . . 12
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 12
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 12
   9.  Informative References . . . . . . . . . . . . . . . . . . . . 12

1.  Introduction

   This document presents a brief survey of proposals to attain a Less-
   than-Best-Effort (LBE) service by means of end-host mechanisms.  We
   loosely define an LBE service as a service which results in smaller
   bandwidth and/or delay impact on standard TCP than standard TCP
   itself, when sharing a bottleneck with it.  We refer to systems that
   are designed to provide this service as LBE systems.  With the
   exception of TCP Vegas, which we present for historical reasons, we
   exclude systems that have been noted to exhibit LBE behavior under
   some circumstances but were not designed for this purpose (e.g.,
   RAPID [Kon09]).

   Generally, LBE behavior can be achieved by reacting to queue growth
   earlier than standard TCP would or by changing the congestion-
   avoidance behavior of TCP without utilizing any additional implicit
   feedback.  It is therefore assumed that readers are familiar with TCP
   congestion control [RFC5681].  Some mechanisms achieve an LBE
   behavior without modifying transport-protocol standards (e.g., by
   changing the receiver window of standard TCP), whereas others
   leverage network-level mechanisms at the transport layer for LBE
   purposes.  According to this classification, solutions have been
   categorized in this document as delay-based transport protocols, non-
   delay-based transport protocols, upper-layer approaches, and network-
   assisted approaches.  Some of the schemes in the first two categories
   could be implemented using TCP without changing its header format;
   this would facilitate their deployment in the Internet.  The schemes
   in the third category are, by design, supposed to be especially easy
   to deploy because they only describe a way in which existing
   transport protocols are used.  Finally, mechanisms in the last
   category require changes to equipment along the path, which can
   greatly complicate their deployment.

   This document is a product of the Low Extra Delay Background
   Transport (LEDBAT) working group.  It aims at putting the congestion
   control algorithm that the working group has specified [Sha11] in the
   context of the state of the art in LBE transport.  This survey is not
   exhaustive, as this would not be possible or useful; the authors have
   selected key, well-known, or otherwise interesting techniques for
   inclusion at their discretion.  There is also a substantial amount of
   work that is related to the LBE concept but does not present a
   solution that can be installed in end-hosts or expected to work over
   the Internet (e.g., there is a Diffserv-based, Lower-Effort service
   [RFC3662], and the IETF Congestion Exposure (CONEX) working group is
   developing a mechanism which can incentivize LEDBAT-like
   applications).  Such work is outside the scope of this document.

2.  Delay-Based Transport Protocols

   It is wrong to generally equate "little impact on standard TCP" with
   "small sending rate".  Without Explicit Congestion Notification (ECN)
   support, standard TCP will normally increase its congestion window
   (and effective sending rate) until a queue overflows, causing one or
   more packets to be dropped and the effective rate to be reduced.  A
   protocol that stops increasing the rate before this event happens
   can, in principle, achieve a better performance than standard TCP.

   TCP Vegas [Bra94] is one of the first protocols that was known to
   have a smaller sending rate than standard TCP when both protocols
   share a bottleneck [Kur00] -- yet, it was designed to achieve more,
   not less, throughput than standard TCP.  Indeed, when TCP Vegas is
   the only congestion control algorithm used by flows going through the
   bottleneck, its throughput is greater than the throughput of standard
   TCP.  Depending on the bottleneck queue length, TCP Vegas itself can
   be starved by standard TCP flows.  This can be remedied to some
   degree by the Random Early Detection (RED) Active Queue Management
   mechanism [RFC2309].  Vegas linearly increases or decreases the
   sending rate, based on the difference between the expected throughput
   and the actual throughput.  The estimation is based on RTT
   measurements.

   The congestion-avoidance behavior is the protocol's most important
   feature in terms of historical relevance as well as relevance in the
   context of this document (it has been shown that other elements of
   the protocol can sometimes play a greater role for its overall
   behavior [Hen00]).  In congestion avoidance, once per RTT, TCP Vegas
   calculates the expected throughput as WindowSize / BaseRTT, where
   WindowSize is the current congestion window and BaseRTT is the
   minimum of all measured RTTs.  The expected throughput is then
   compared with the actual throughput, measured based on recent
   acknowledgements.  If the actual throughput is smaller than the

   expected throughput minus a threshold called "beta", this is taken as
   a sign of congestion, causing the protocol to linearly decrease its
   rate.  If the actual throughput is greater than the expected
   throughput minus a threshold called "alpha" (with alpha < beta), this
   is taken as a sign that the network is underutilized, causing the
   protocol to linearly increase its rate.

   TCP Vegas has been analyzed extensively.  One of the most prominent
   properties of TCP Vegas is its fairness between multiple flows of the
   same kind, which does not penalize flows with large propagation
   delays in the same way as standard TCP.  While it was not the first
   protocol that uses delay as a congestion indication, its predecessors
   (like CARD [Jai89], Tri-S [Wan91], or DUAL [Wan92]) are not discussed
   here because of the historical "landmark" role that TCP Vegas has
   taken in the literature.

   Delay-based transport protocols that were designed to be non-
   intrusive include TCP Nice [Ven02] and TCP Low Priority (TCP-LP)
   [Kuz06].  TCP Nice [Ven02] follows the same basic approach as TCP
   Vegas but improves upon it in some aspects.  Because of its moderate
   linear-decrease congestion response, TCP Vegas can affect standard
   TCP despite its ability to detect congestion early.  TCP Nice removes
   this issue by halving the congestion window (at most once per RTT,
   like standard TCP) instead of linearly reducing it.  To avoid being
   too conservative, this is only done if a fixed predefined fraction of
   delay-based incipient congestion signals appears within one RTT.
   Otherwise, TCP Nice falls back to the congestion-avoidance rules of
   TCP Vegas if no packet was lost or standard TCP if a packet was lost.
   One more feature of TCP Nice is its ability to support a congestion
   window of less than one packet, by clocking out single packets over
   more than one RTT.  With ns-2 simulations and real-life experiments
   using a Linux implementation, the authors of [Ven02] show that TCP
   Nice achieves its goal of efficiently utilizing spare capacity while
   being non-intrusive to standard TCP.

   Other than TCP Vegas and TCP Nice, TCP-LP [Kuz06] uses only the one-
   way delay (OWD) instead of the RTT as an indicator of incipient
   congestion.  This is done to avoid reacting to delay fluctuations
   that are caused by reverse cross-traffic.  Using the TCP Timestamps
   option [RFC1323], the OWD is determined as the difference between the
   receiver's Timestamp value in the ACK and the original Timestamp
   value that the receiver copied into the ACK.  While the result of
   this subtraction can only precisely represent the OWD if clocks are
   synchronized, its absolute value is of no concern to TCP-LP, and
   hence clock synchronization is unnecessary.  Using a constant
   smoothing parameter, TCP-LP calculates an Exponentially Weighted
   Moving Average (EWMA) of the measured OWD and checks whether the
   result exceeds a threshold within the range of the minimum and

   maximum OWD that was seen during the connection's lifetime; if it
   does, this condition is interpreted as an "early congestion
   indication".  The minimum and maximum OWD values are initialized
   during the slow-start phase.

   Regarding its reaction to an early congestion indication, TCP-LP
   tries to strike a middle ground between the overly conservative
   choice of _immediately_ setting the congestion window to one packet,
   and the presumably too aggressive choice of simply halving the
   congestion window like standard TCP; TCP-LP tries to delay the former
   action by an additional RTT, to see if there is persistent congestion
   or not.  It does so by halving the window at first in response to an
   early congestion indication, then initializing an "inference time-out
   timer" and maintaining the current congestion window until this timer
   fires.  If another early congestion indication appeared during this
   "inference phase", the window is then set to 1; otherwise, the window
   is maintained and TCP-LP continues to increase it in the standard
   Additive-Increase fashion.  This method ensures that it takes at
   least two RTTs for a TCP-LP flow to decrease its window to 1, and
   that, like standard TCP, TCP-LP reacts to congestion at most once per
   RTT.

   Using a simple analytical model, the authors of TCP-LP [Kuz06]
   illustrate the feasibility of a delay-based LBE transport by showing
   that, due to the non-linear relationship between throughput and RTT,
   it is possible to avoid interfering with standard TCP traffic even
   when the flows under consideration have a larger RTT than standard
   TCP flows.  With ns-2 simulations and real-life experiments using a
   Linux implementation, the authors of [Kuz06] show that TCP-LP is
   largely non-intrusive to TCP traffic while at the same time enabling
   it to utilize a large portion of the excess network bandwidth, which
   is fairly shared among competing TCP-LP flows.  They also show that
   using their protocol for bulk data transfers greatly reduces file
   transfer times of competing best-effort web traffic.

   Sync-TCP [Wei05] follows a similar approach as TCP-LP, by adapting
   its reaction to congestion according to changes in the OWD.  By
   comparing the estimated (average) forward queuing delay to the
   maximum observed delay, Sync-TCP adapts the Additive-Increase
   Multiplicative-Decrease (AIMD) parameters depending on the trend
   followed by the average delay over an observation window.  Even
   though the authors of [Wei05] did not explicitly consider its use as
   an LBE protocol, Sync-TCP was designed to react early to incipient
   congestion, while grabbing available bandwidth more aggressively than
   a standard TCP in congestion-avoidance mode.

   Delay-based congestion control is also the basis of proposals that
   aim at adapting TCP's congestion avoidance to very high-speed
   networks.  Some of these proposals, like Compound TCP [Tan06] [Sri08]
   and TCP Illinois [Liu08], are hybrid loss- and delay-based
   mechanisms, whereas others (e.g., NewVegas [Dev03], FAST TCP [Wei06],
   or CODE TCP [Cha10]) are variants of Vegas based primarily on delays.

2.1.  Accuracy of Delay-Based Congestion Predictors

   The accuracy of delay-based congestion predictors has been the
   subject of a good deal of research, see, e.g., [Bia03], [Mar03],
   [Pra04], [Rew06], [McC08].  The main result of most of these studies
   is that delays (or, more precisely, round-trip times) are, in
   general, weakly correlated with congestion.  There are several
   factors that may induce such a poor correlation:

   o  Bottleneck buffer size: in principle, a delay-based mechanism
      could be made "more than TCP friendly" _if_ buffers are "large
      enough", so that RTT fluctuations and/or deviations from the
      minimum RTT can be detected by the end-host with reasonable
      accuracy.  Otherwise, it may be hard to distinguish real delay
      variations from measurement noise.

   o  RTT measurement issues: in principle, RTT samples may suffer from
      poor resolution, due to timers which are too coarse-grained with
      respect to the scale of delay fluctuations.  Also, a flow may
      obtain a very noisy estimate of RTTs due to undersampling, under
      some circumstances (e.g., the flow rate is much lower than the
      link bandwidth).  For TCP, other potential sources of measurement
      noise include TCP segmentation offloading (TSO) and the use of
      delayed ACKs [Hay10].  A congested reverse path may also result in
      an erroneous assessment of the congestion state of the forward
      path.  Finally, in the case of fast or short-distance links, the
      majority of the measured delay can in fact be due to processing in
      the involved hosts; typically, this processing delay is not of
      interest, and it can underlie fluctuations that are not related to
      the network at all.

   o  Level of statistical multiplexing and RTT sampling: it may be easy
      for an individual flow to "miss" loss/queue overflow events,
      especially if the number of flows sharing a bottleneck buffer is
      significant.  This is nicely illustrated, e.g., in Figure 1 of
      [McC08].

   o  Impact of wireless links: several mechanisms that are typical of
      wireless links, like link-layer scheduling and error recovery, may
      induce strong delay fluctuations over short timescales [Gur04].

   Interestingly, the results of Bhandarkar et al. [Bha07] seem to paint
   a slightly different picture, regarding the accuracy of delay-based
   congestion prediction.  Bhandarkar et al. claim that it is possible
   to significantly improve prediction accuracy by adopting some simple
   techniques (smoothing of RTT samples, increasing the RTT sampling
   frequency).  Nonetheless, they acknowledge that even with such
   techniques, it is not possible to eradicate detection errors.  Their
   proposed delay-based congestion-avoidance method, PERT (Probabilistic
   Early Response TCP), mitigates the impact of residual detection
   errors by means of a probabilistic response mechanism to congestion-
   detection events.

2.2.  Potential Issues with Delay-Based Congestion Control for LBE
      Transport

   Whether a delay-based protocol behaves in its intended manner (e.g.,
   it is "more than TCP friendly", or it grabs available bandwidth in a
   very aggressive manner) may depend on the accuracy issues listed in
   Section 2.1.  Moreover, protocols like Vegas need to keep an estimate
   of the minimum ("base") delay; this makes such protocols highly
   sensitive to eventual changes in the end-to-end route during the
   lifetime of the flow [Mo99].

   Regarding the issue of false positives or false negatives with a
   delay-based congestion detector, most studies focus on the loss of
   throughput coming from the erroneous detection of queue build-up and
   of alleviation of congestion.  Arguably, for an LBE transport
   protocol it's better to err on the "more-than-TCP-friendly side",
   that is, to always yield to _perceived_ congestion whether it is
   "real" or not; however, failure to detect congestion (due to one of
   the above accuracy problems) would result in behavior that is not
   LBE.  For instance, consider the case in which the bottleneck buffer
   is small, so that the contribution of queueing delay at the
   bottleneck to the global end-to-end delay is small.  In such a case,
   a flow using a delay-based mechanism might end up consuming a good
   deal of bandwidth with respect to a competing standard TCP flow,
   unless it also incorporates a suitable reaction to loss.

   A delay-based mechanism may also suffer from the so-called "latecomer
   advantage" (or "latecomer unfairness") problem.  Consider the case in
   which the bottleneck link is already (very) congested.  In such a
   scenario, delay variations may be quite small; hence, it may be very
   difficult to tell an empty queue from a heavily-loaded queue, in
   terms of delay fluctuation.  Therefore, a newly-arriving delay-based
   flow may start sending faster when there is already heavy congestion,
   eventually driving away loss-based flows [Sha05] [Car10].

3.  Non-Delay-Based Transport Protocols

   There exist a few transport-layer proposals that achieve an LBE
   service without relying on delay as an indicator of congestion.  In
   the algorithms discussed below, the loss rate of the flow determines,
   either implicitly or explicitly, the sending rate (which is adapted
   so as to obtain a lower share of the available bandwidth than
   standard TCP); such mechanisms likely cause more queuing delay and
   react to congestion more slowly than delay-based ones.

   4CP [Liu07], which stands for "Competitive and Considerate Congestion
   Control", is a protocol that provides an LBE service by changing the
   window control rules of standard TCP.  A "virtual window" is
   maintained that, during a so-called "bad congestion phase", is
   reduced to less than a predefined minimum value of the actual
   congestion window.  The congestion window is only increased again
   once the virtual window exceeds this minimum, and in this way the
   virtual window controls the duration during which the sender
   transmits with a fixed minimum rate.  Whether the congestion state is
   "bad" or "good" depends on whether the loss event rate is above or
   below a threshold (or target) value.  The 4CP congestion-avoidance
   algorithm allows for setting a target average window and avoids
   starvation of "background" flows while bounding the impact on
   "foreground" flows.  Its performance was evaluated in ns-2
   simulations and in real-life experiments with a kernel-level
   implementation in Microsoft Windows Vista.

   The MulTFRC [Dam09] protocol is an extension of TCP-Friendly Rate
   Control (TFRC) [RFC5348] for multiple flows.  MulTFRC takes the main
   idea of MulTCP [Cro98] and similar proposals (e.g., [Hac04], [Hac08],
   [Kuo08]) a step further.  A single MulTCP flow tries to emulate (and
   be as friendly as) a number N > 1 of parallel TCP flows.  By
   supporting values of N between 0 and 1, MulTFRC can be used as a
   mechanism for an LBE service.  Since it does not react to delay like
   the protocols described in Section 2 but adjusts its rate like TFRC,
   MulTFRC can probably be expected to be more aggressive than
   mechanisms such as TCP Nice or TCP-LP.  This also means that MulTFRC
   is less likely to be prone to starvation, as its aggressiveness is
   tunable at a fine granularity, even when N is between 0 and 1.

4.  Upper-Layer Approaches

   The proposals described in this section do not require modifying
   transport-protocol standards.  Most of them can be regarded as
   running "on top" of an existing transport, even though they may be
   implemented either at the application layer (i.e., in user-level
   processes), or in the kernel of the end-hosts' operating systems.

   Such "upper-layer" mechanisms may arguably be easier to deploy than
   transport-layer approaches, since they do not require any changes to
   the transport itself.

   A simplistic, application-level approach to a background transport
   service may consist in scheduling automated transfers at times when
   the network is lightly loaded, e.g., as described in [Dyk02] for
   cooperative proxy caching.  An issue with such a technique is that it
   may not necessarily be applicable to applications like peer-to-peer
   file transfer, since the notion of an "off-peak hour" is not
   meaningful when end-hosts may be located anywhere in the world.

   The so-called Background Intelligent Transfer Service [BITS] is
   implemented in several versions of Microsoft Windows.  BITS uses a
   system of application-layer priority levels for file-transfer jobs,
   together with monitoring of bandwidth usage of the network interface
   (or, in more recent versions, of the network gateway connected to the
   end-host), so that low-priority transfers at a given end-host give
   way to both high-priority (foreground) transfers and traffic from
   interactive applications at the same host.

   A different approach is taken in [Egg05] -- here, the priority of a
   flow is reduced via a generic idletime scheduling strategy in a
   host's operating system.  While results presented in this paper show
   that the new scheduler can effectively shield regular tasks from low-
   priority ones (e.g., TCP from greedy UDP) with only a minor
   performance impact, it is an underlying assumption that all involved
   end-hosts would use the idletime scheduler.  In other words, it is
   not the focus of this work to protect a standard TCP flow that
   originates from any host where the presented scheduling scheme may
   not be implemented.

4.1.  Receiver-Oriented, Flow-Control-Based Approaches

   Some proposals for achieving an LBE behavior work by exploiting
   existing transport-layer features -- typically, at the "receiving"
   side.  In particular, TCP's built-in flow control can be used as a
   means to achieve a low-priority transport service.

   The mechanism described in [Spr00] is an example of the above
   technique.  Such mechanism controls the bandwidth by letting the
   receiver intelligently manipulate the receiver window of standard
   TCP.  This is possible because the authors assume a client-server
   setting where the receiver's access link is typically the bottleneck.
   The scheme incorporates a delay-based calculation of the expected
   queue length at the bottleneck, which is quite similar to the
   calculation in the above delay-based protocols, e.g., TCP Vegas.
   Using a Linux implementation, where TCP flows are classified

   according to their application's needs, Spring et al. show in [Spr00]
   that a significant improvement in packet latency can be attained over
   an unmodified system, while maintaining good link utilization.

   A similar method is employed by Mehra et al. [Meh03], where both the
   advertised receiver window and the delay in sending ACK messages are
   dynamically adapted to attain a given rate.  As in [Spr00], Mehra et
   al. assume that the bottleneck is located at the receiver's access
   link.  However, the latter also propose a bandwidth-sharing system,
   allowing control of the bandwidth allocated to different flows, as
   well as allotment of a minimum rate to some flows.

   Receiver window tuning is also done in [Key04], where choosing the
   right value for the window is phrased as an optimization problem.  On
   this basis, two algorithms are presented, binary search (which is
   faster than the other one at achieving a good operation point but
   fluctuates) and stochastic optimization (which does not fluctuate but
   converges slower than binary search).  These algorithms merely use
   the previous receiver window and the amount of data received during
   the previous control interval as input.  According to [Key04], the
   encouraging simulation results suggest that such an application-level
   mechanism can work almost as well as a transport-layer scheme like
   TCP-LP.

   Another way of dealing with non-interactive flows, like web
   prefetching, is to rate-limit the transfer of such bursty traffic
   [Cro98b].  Note that one of the techniques used in [Cro98b] is,
   precisely, to have the downloading application adapt the TCP receiver
   window, so as to reduce the data rate to the minimum needed (thus
   disturbing other flows as little as possible while respecting a
   deadline for the transfer of the data).

5.  Network-Assisted Approaches

   Network-layer mechanisms, like active queue management (AQM) and
   packet scheduling in routers, can be exploited by a transport
   protocol for achieving an LBE service.  Such approaches may result in
   improved protection of non-LBE flows (e.g., when scheduling is used);
   besides, approaches using an explicit, AQM-based congestion signaling
   may arguably be more robust than, say, delay-based transports for
   detecting impending congestion.  However, an obvious drawback of any
   network-assisted approach is that, in principle, they need
   modifications in both end-hosts and intermediate network nodes.

   Harp [Kok04] realizes an LBE service by dissipating background
   traffic to less-utilized paths of the network, based on multipath
   routing and multipath congestion control.  This is achieved without
   changing all routers, by using edge nodes as relays.  According to

   the authors, these edge nodes should be gateways of organizations in
   order to align their scheme with usage incentives, but the technical
   solution would also work if Harp was only deployed in end-hosts.  It
   detects impending congestion by looking at delay, similar to TCP Nice
   [Ven02], and manages to improve the utilization and fairness of TCP
   over pure single-path solutions without requiring any changes to the
   TCP itself.

   Another technique is that used by protocols like Network-Friendly TCP
   (NF-TCP) [Aru10], where a bandwidth-estimation module integrated into
   the transport protocol allows to rapidly take advantage of free
   capacity.  NF-TCP combines this with an early congestion detection
   based on Explicit Congestion Notification (ECN) [RFC3168] and RED
   [RFC2309]; when congestion starts building up, appropriate tuning of
   a RED queue allows to mark low-priority (i.e., NF-TCP) packets with a
   much higher probability than high-priority (i.e., standard TCP)
   packets, so low-priority flows yield up bandwidth before standard TCP
   flows.  NF-TCP could be implemented by adapting the congestion
   control behavior of TCP without requiring to change the protocol on
   the wire -- with the only exception that NF-TCP-capable routers must
   be able to somehow distinguish NF-TCP traffic from other TCP traffic.

   In [Ven08], Venkataraman et al. propose a transport-layer approach to
   leverage an existing, network-layer LBE service based on priority
   queueing.  Their transport protocol, which they call PLT (Priority-
   Layer Transport), splits a layer-4 connection into two flows, a high-
   priority one and a low-priority one.  The high-priority flow is sent
   over the higher-priority queueing class (in principle, offering a
   best-effort service) using an AIMD, TCP-like congestion control
   mechanism.  The low-priority flow, which is mapped to the LBE class,
   uses a non TCP-friendly congestion control algorithm.  The goal of
   PLT is thus to maximize its aggregate throughput by exploiting unused
   capacity in an aggressive way, while protecting standard TCP flows
   carried by the best-effort class.  Similar in spirit, [Ott03]
   proposes simple changes to only the AIMD parameters of TCP for use
   over a network-layer LBE service, so that such "filler" traffic may
   aggressively consume unused bandwidth.  Note that [Ven08] also
   considers a mechanism for detecting the lack of priority queueing in
   the network, so that the non-TCP friendly flow may be inhibited.  The
   PLT receiver monitors the loss rate of both flows; if the high-
   priority flow starts seeing losses while the low-priority one does
   not experience 100% loss, this is taken as an indication of the
   absence of strict priority queueing.

6.  LEDBAT Considerations

   The previous sections have shown that there is a large amount of work
   on attaining an LBE service, and that it is quite heterogeneous in
   nature.  The algorithm developed by the LEDBAT working group [Sha11]
   can be classified as a delay-based mechanism; as such, it is similar
   in spirit to the protocols presented in Section 2.  It is, however,
   not a protocol -- how it is actually applied to the Internet, i.e.,
   how to use existing or even new transport protocols together with the
   LEDBAT algorithm, is not defined by the LEDBAT working group.  As it
   heavily relies on delay, the discussion in Sections 2.1 and 2.2
   applies to it.  The performance of LEDBAT has been analyzed in
   comparison with some of the other work presented here in several
   articles, e.g.  [Aru10], [Car10], [Sch10], but these analyses have to
   be examined with care: at the time of writing, LEDBAT was still a
   moving target.

7.  Acknowledgements

   The authors would like to thank Melissa Chavez, Dragana Damjanovic,
   and Yinxia Zhao for reference pointers, as well as Jari Arkko,
   Mayutan Arumaithurai, Elwyn Davies, Wesley Eddy, Stephen Farrell,
   Mirja Kuehlewind, Tina Tsou, and Rolf Winter for their detailed
   reviews and suggestions.

8.  Security Considerations

   This document introduces no new security considerations.

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Authors' Addresses

   Michael Welzl
   University of Oslo
   Department of Informatics, PO Box 1080 Blindern
   N-0316 Oslo
   Norway

   Phone: +47 22 85 24 20
   EMail: michawe@ifi.uio.no

   David Ros
   Institut Telecom / Telecom Bretagne
   Rue de la Chataigneraie, CS 17607
   35576 Cesson Sevigne cedex
   France

   Phone: +33 2 99 12 70 46
   EMail: david.ros@telecom-bretagne.eu

 

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