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.
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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.
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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
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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
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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.
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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].
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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].
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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.
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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
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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
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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.
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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.
9. Informative References
[Aru10] Arumaithurai, M., Fu, X., and K. Ramakrishnan, "NF-TCP: A
Network Friendly TCP Variant for Background Delay-
Insensitive Applications", Technical Report No. IFI-TB-
2010-05, Institute of Computer Science, University of
Goettingen, Germany, September 2010, .
[BITS] Microsoft, "Windows Background Intelligent Transfer
Service",
.
[Bha07] Bhandarkar, S., Reddy, A., Zhang, Y., and D. Loguinov,
"Emulating AQM from end hosts", Proceedings of ACM
SIGCOMM 2007, 2007.
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[Bia03] Biaz, S. and N. Vaidya, "Is the round-trip time correlated
with the number of packets in flight?", Proceedings of the
3rd ACM SIGCOMM conference on Internet measurement (IMC
'03), pages 273-278, 2003.
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