Internet Engineering Task Force M. Welzl
Internet-Draft University of Oslo
Intended status: Informational D. Ros
Expires: April 12, 2011 Telecom Bretagne
October 9, 2010
A Survey of Lower-than-Best-Effort Transport Protocols
draft-ietf-ledbat-survey-01.txt
Abstract
This document provides a survey of transport protocols which 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 low-priority "background" traffic,
as they provide what is sometimes called a "less than" (or "lower
than") best-effort service.
Status of this Memo
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This Internet-Draft will expire on April 12, 2011.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Delay-based transport protocols . . . . . . . . . . . . . . . 3
2.1. Accuracy of delay-based congestion predictors . . . . . . 6
2.2. Delay-based congestion control = LBE? . . . . . . . . . . 7
3. Non-delay-based transport protocols . . . . . . . . . . . . . 7
4. Application layer approaches . . . . . . . . . . . . . . . . . 8
5. Orthogonal work . . . . . . . . . . . . . . . . . . . . . . . 9
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 11
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
8. Security Considerations . . . . . . . . . . . . . . . . . . . 11
9. Informative References . . . . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
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1. Introduction
This document presents a brief survey of proposals to attain a Less
than Best Effort (LBE) service without help from routers. We loosely
define a 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 provide this
service as Less than Best Effort (LBE) systems. 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. Some
mechanisms achieve a LBE behavior at the application layer, e.g. by
changing the receiver window of standard TCP, and there is also a
substantial amount of work that is related to the LBE concept but not
presenting a solution that can be installed in end hosts or expected
to work over the Internet. According to this classification,
solutions have been categorized in this document as delay-based
transport protocols, non-delay-based transport protocols, application
layer approaches and orthogonal work.
2. Delay-based transport protocols
It is wrong to generally equate "little impact on standard TCP" with
"small sending rate". Unless the sender's maximum window is limited
for some reason, and in the absence of ECN support, standard TCP will
normally increase its rate until a queue overflows, causing one or
more packets to be dropped and the rate to be reduced. A protocol
which stops increasing the rate before this event happens can, in
principle, achieve a better performance than standard TCP. In the
absence of any other traffic, this is even true for TCP itself when
its maximum send window is limited to the bandwidth*round-trip time
(RTT) product.
TCP Vegas [Bra+94] is one of the first protocols that was known to
have a smaller sending rate than standard TCP when both protocols
share a bottleneck [Kur+00] -- yet it was designed to achieve more,
not less throughput than standard TCP. Indeed, when it is the only
protocol on the bottleneck, the throughput of TCP Vegas 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 RED Active Queue
Management mechanism [RFC2309].
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
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behavior [Hen+00]). 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 (measured) throughput. If the actual
throughput is smaller than the expected throughput minus a threshold
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 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
(which can be found in [Bra+94]) are not discussed here because of
the historical "landmark" role that TCP Vegas has taken in the
literature.
Transport protocols which were designed to be non-intrusive include
TCP-LP [Kuz+06] and TCP Nice [Ven+02]. Using a simple analytical
model, the authors of [Kuz+06] illustrate the feasibility of this
endeavor by showing that, due to the non-linear relationship between
throughput and RTT, it is possible to remain transparent to standard
TCP even when the flows under consideration have a larger RTT than
standard TCP flows.
TCP Nice [Ven+02] 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 [Ven+02] 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 uses only the one-way delay
(OWD) instead of the RTT as an indicator of incipient congestion.
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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 connections'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 halving the congestion window
like standard TCP. It does so by halving the window at first in
response to an early congestion indication, then initializing an
"interference time-out timer", and maintaining the window size until
this timer fires. If another early congestion indication appeared
during this "interference phase", the window is then set to 1;
otherwise, the window is maintained and TCP-LP continues to increase
it 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, like standard TCP, TCP-LP reacts to congestion at most
once per RTT.
With ns-2 simulations and real-life experiments using a Linux
implementation, the authors of [Kuz+06] 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 [Wei+05] 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 AIMD parameters depending
on the trend followed by the average delay over an observation
window. Even though the authors of [Wei+05] did not explicitely
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 at the basis of proposals
aiming at adapting TCP's congestion avoidance to very high-speed
networks. Some of these proposals [Tan+06][Sri+08][Liu+08] are
hybrid loss- and delay-based mechanisms, whereas others
[Dev+03][Wei+06][Cha+10] are variants of Vegas based solely 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. [Bia+03], [Mar+03],
[Pra+04], [Rew+06], [McC+08]. 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].
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 Fig. 2 of
[McC+08].
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 time scales [Gur+04].
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 therefore depend on the accuracy issues
listed above. 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
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the lifetime of the flow [Mo+99].
TODO: incorporate [Bha+07] and any references therein that may be
missing.
2.2. Delay-based congestion control = LBE?
Regarding the issue of false positives/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 a 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.
Consider also 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
[Sha+05].
3. Non-delay-based transport protocols
4CP [Liu+07], which stands for "Competitive and Considerate
Congestion Control", is a protocol which provides a LBE service by
changing the window control rules of standard TCP. A "virtual
window" is maintained, which, 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. 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.
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Some work was done on applying weights to congestion control
mechanisms, allowing a flow to be as aggressive as a number of
parallel TCP flows at the same time. This is usually motivated by
the fact that users may want to assign different priorities to
different flows. The first, and best known, such protocol is MulTCP
[Cro+98], which emulates N TCPs in a rather simple fashion. Improved
versions of the parallel-TCP idea are presented in [Hac+04] and
[Hac+08], and there is also a variant where only one feedback loop is
applied to control a larger traffic aggregate by the name of Probe-
Aided (PA-)MulTCP [Kuo+08]. Another protocol, CP [Ott+04], applies
the same concept to the TFRC protocol [RFC5348] in order to provide
such fairness differentiation for multimedia flows.
The general assumption underlying all of the above work is that these
protocols are "N-TCP-friendly", i.e. they are as TCP-friendly as N
TCPs, where N is a positive (and possibly natural) number which is
greater than or equal to 1. The MulTFRC [Dam+09] protocol, another
extension of TFRC for multiple flows, is however able to support
values between 0 and 1, making it applicable as a mechanism for a LBE
service. Since it does not react to delay like the mechanisms above
but adjusts its rate like TFRC, it 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 aggression is tunable at a fine granularity even when N is
between 0 and 1.
4. Application layer approaches
A simplistic, application-level approach to a background transport
service may just consist in scheduling automated transfers at times
when the network is lightly loaded, as described in e.g. [Dyk+02].
An issue with such a technique is that it may not necessarily be
appropriate 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.
TCP's built-in flow control can be used as a means to achieve a low-
priority transport service. For instance, the mechanism described in
[Spr+00] controls the bandwidth by letting the receiver intelligently
manipulate the receiver window of standard TCP. This is done 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, it is shown that a significant improvement in
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packet latency can be attained over an unmodified system while
maintaining good link utilization.
A similar method is employed by Mehra et al. [Meh+03], where both
the advertised receiver window and the delay in sending ACK messages
are dynamically adapted to attain a given rate. As in [Spr+00],
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 to control the bandwidth allocated to different
flows, as well as to allot a minimum rate to some flows.
Receiver window tuning is also done in [Key+04], 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 [Key+04], 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 e.g. web
prefetching, is to rate-limit the transfer of such bursty traffic
[Cro+98b]. Note that one of the techniques used in [Cro+98b] is,
precisely, to have the downloading application adapt the TCP receiver
window, so as to reduce the data rate to the minimum needed.
The so-called Background Intelligent Transfer Service (BITS) [BITS],
implemented in several versions of Microsoft Windows, 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 give way to both high-priority
(foreground) transfers and traffic from interactive applications.
5. Orthogonal work
Various suggestions have been published for realizing a LBE service
by influencing the way packets are treated in routers. One example
is the Persistent Class Based Queuing (P-CBQ) scheme presented in
[Car+01], which is a variant of Class Based Queuing (CBQ) with per-
flow accounting. RFC 3662 [RFC3662] defines a DiffServ per-domain
behavior called "Lower Effort". Similar Lower-Effort PDBs have been
tested and deployed, at least in research networks [Cho+03], [QBSS].
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Harp [Kok+04] realizes a LBE service by dissipating background
traffic to less-utilized paths of the network. This is achieved
without changing 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
[Ven+02], and manages to improve utilization and fairness over pure
single-path solutions.
An entirely different approach is taken in [Egg+05]: 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 which
originates from any host where the presented scheduling scheme may
not be implemented.
In [Ven+08], Venkataraman et al. propose a transport-layer approach
to leverage an existing, network-layer LBE service based on priority
queueing. The 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. [Ott+03] proposes simple changes
to 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 [Ven+08] 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.
Another technique is that used by protocols like NF-TCP [Aru+10b],
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
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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.
6. Acknowledgements
The authors would like to thank Dragana Damjanovic, Melissa Chavez,
Yinxia Zhao and Mayutan Arumaithurai for reference pointers.
7. IANA Considerations
This memo includes no request to IANA.
8. Security Considerations
This document introduces no new security considerations.
9. Informative References
[Aru+10b] 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, <http://
www.net.informatik.uni-goettingen.de/publications/1718/
NF-TCP-techreport.pdf>.
[BITS] Microsoft, "Windows Background Intelligent Transfer
Service",
<http://msdn.microsoft.com/library/bb968799(VS.85).aspx>.
[Bha+07] Bhandarkar, S., Reddy, A., Zhang, Y., and D. Loguinov,
"Emulating AQM from end hosts", Proceedings of ACM
SIGCOMM 2007, 2007.
[Bia+03] 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.
[Bra+94] Brakmo, L., O'Malley, S., and L. Peterson, "TCP Vegas: New
techniques for congestion detection and avoidance",
Proceedings of SIGCOMM '94, pages 24-35, August 1994.
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[Car+01] Carlberg, K., Gevros, P., and J. Crowcroft, "Lower than
best effort: a design and implementation", Workshop on
Data communication in Latin America and the
Caribbean 2001, San Jose, Costa Rica, Pages: 244 - 265,
2001.
[Cha+10] Chan, Y., Lin, C., Chan, C., and C. Ho, "CODE TCP: A
competitive delay-based TCP", Computer Communications ,
33(9):1013-1029, June 2010.
[Cho+03] Chown, T., Ferrari, T., Leinen, S., Sabatino, R., Simar,
N., and S. Venaas, "Less than Best Effort: Application
Scenarios and Experimental Results", Proceedings of
QoS-IP , pages 131-144, February 2003.
[Cro+98] Crowcroft, J. and P. Oechslin, "Differentiated end-to-end
Internet services using a weighted proportional fair
sharing TCP", ACM SIGCOMM Computer Communication
Review vol. 28, no. 3 (July 1998), pp. 53-69, 1998.
[Cro+98b] Crovella, M. and P. Barford, "The network effects of
prefetching", Proceedings of Infocom 1998, April 1998.
[Dam+09] Damjanovic, D. and M. Welzl, "MulTFRC: Providing Weighted
Fairness for Multimedia Applications (and others too!)",
ACM Computer Communication Review vol. 39, no. 3 (July
2009), 2009.
[Dev+03] De Vendictis, A., Baiocchi, A., and M. Bonacci, "Analysis
and enhancement of TCP Vegas congestion control in a mixed
TCP Vegas and TCP Reno network scenario", Performance
Evaluation , 53(3-4):225-253, 2003.
[Dyk+02] Dykes, S. and K. Robbins, "Limitations and benefits of
cooperative proxy caching", IEEE Journal on Selected Areas
in Communications 20(7):1290-1304, September 2002.
[Egg+05] Eggert, L. and J. Touch, "Idletime Scheduling with
Preemption Intervals", Proceedings of 20th ACM Symposium
on Operating Systems Principles SOSP 2005, Brighton,
United Kingdom, pp. 249/262, October 2005.
[Gur+04] Gurtov, A. and S. Floyd, "Modeling wireless links for
transport protocols", ACM SIGCOMM Computer Communications
Review 34(2):85-96, April 2004.
[Hac+04] Hacker, T., Noble, B., and B. Athey, "Improving Throughput
and Maintaining Fairness using Parallel TCP", Proceedings
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of Infocom 2004, March 2004.
[Hac+08] Hacker, T. and P. Smith, "Stochastic TCP: A Statistical
Approach to Congestion Avoidance", Proceedings of
PFLDnet 2008, March 2008.
[Hay10] Hayes, D., "Timing enhancements to the FreeBSD kernel to
support delay and rate based TCP mechanisms", Technical
Report 100219A , Centre for Advanced Internet
Architectures, Swinburne University of Technology,
February 2010.
[Hen+00] Hengartner, U., Bolliger, J., and T. Gross, "TCP Vegas
revisited", Proceedings of Infocom 2000, March 2000.
[Key+04] Key, P., Massoulie, L., and B. Wang, "Emulating Low-
Priority Transport at the Application Layer: a Background
Transfer Service", Proceedings of ACM SIGMETRICS 2004,
January 2004.
[Kok+04] Kokku, R., Bohra, A., Ganguly, S., and A. Venkataramani,
"A Multipath Background Network Architecture", Proceedings
of Infocom 2007, May 2007.
[Kuo+08] Kuo, F. and X. Fu, "Probe-Aided MulTCP: an aggregate
congestion control mechanism", ACM SIGCOMM Computer
Communication Review vol. 38, no. 1 (January 2008), pp.
17-28, 2008.
[Kur+00] Kurata, K., Hasegawa, G., and M. Murata, "Fairness
Comparisons Between TCP Reno and TCP Vegas for Future
Deployment of TCP Vegas", Proceedings of INET 2000,
July 2000.
[Kuz+06] Kuzmanovic, A. and E. Knightly, "TCP-LP: low-priority
service via end-point congestion control", IEEE/ACM
Transactions on Networking (ToN) Volume 14, Issue 4, pp.
739-752., August 2006,
<http://www.ece.rice.edu/networks/TCP-LP/>.
[Liu+07] Liu, S., Vojnovic, M., and D. Gunawardena, "Competitive
and Considerate Congestion Control for Bulk Data
Transfers", Proceedings of IWQoS 2007, June 2007.
[Liu+08] Liu, S., Basar, T., and R. Srikant, "TCP-Illinois: A loss-
and delay-based congestion control algorithm for high-
speed networks", Performance Evaluation , 65(6-7):417-440,
2008.
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[Mar+03] Martin, J., Nilsson, A., and I. Rhee, "Delay-based
congestion avoidance for TCP", IEEE/ACM Transactions on
Networking , 11(3):356-369, June 2003.
[McC+08] McCullagh, G. and D. Leith, "Delay-based congestion
control: Sampling and correlation issues revisited",
Technical report , Hamilton Institute, 2008.
[Meh+03] Mehra, P., Zakhor, A., and C. De Vleeschouwer, "Receiver-
Driven Bandwidth Sharing for TCP", Proceedings of
Infocom 2003, April 2003.
[Mo+99] Mo, J., La, R., Anantharam, V., and J. Walrand, "Analysis
and Comparison of TCP Reno and TCP Vegas", Proceedings of
Infocom 1999, March 1999.
[Ott+03] Ott, B., Warnky, T., and V. Liberatore, "Congestion
control for low-priority filler traffic", SPIE QoS 2003
(Quality of Service over Next-Generation Internet), In
Proc. SPIE, Vol. 5245, 154, Monterey (CA), USA, July 2003.
[Ott+04] Ott, D., Sparks, T., and K. Mayer-Patel, "Aggregate
congestion control for distributed multimedia
applications", Proceedings of Infocom 2004, March 2004.
[Pra+04] Prasad, R., Jain, M., and C. Dovrolis, "On the
effectiveness of delay-based congestion avoidance",
Proceedings of PFLDnet , 2004.
[QBSS] "QBone Scavenger Service (QBSS)", Internet2 QBone
Initiative .
[RFC1323] Jacobson, V., Braden, B., and D. Borman, "TCP Extensions
for High Performance", RFC 1323, May 1992.
[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.
[RFC3662] Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort
Per-Domain Behavior (PDB) for Differentiated Services",
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RFC 3662, December 2003.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, September 2008.
[Rew+06] Rewaskar, S., Kaur, J., and D. Smith, "Why don't delay-
based congestion estimators work in the real-world?",
Technical report TR06-001 , University of North Carolina
at Chapel Hill, Dept. of Computer Science, January 2006.
[Sha+05] Shalunov, S., Dunn, L., Gu, Y., Low, S., Rhee, I., Senger,
S., Wydrowski, B., and L. Xu, "Design Space for a Bulk
Transport Tool", Technical Report , Internet2 Transport
Group, May 2005.
[Spr+00] Spring, N., Chesire, M., Berryman, M., Sahasranaman, V.,
Anderson, T., and B. Bershad, "Receiver based management
of low bandwidth access links", Proceedings of
Infocom 2000, pp. 245-254, vol.1, 2000.
[Sri+08] Sridharan, M., Tan, K., Bansala, D., and D. Thaler,
"Compound TCP: A new TCP congestion control for high-speed
and long distance networks", Internet Draft
draft-sridharan-tcpm-ctcp , work in progress,
November 2008.
[Tan+06] Tan, K., Song, J., Zhang, Q., and M. Sridharan, "A
Compound TCP approach for high-speed and long distance
networks", Proceedings of IEEE INFOCOM 2006, Barcelona,
Spain, April 2008.
[Ven+02] Venkataramani, A., Kokku, R., and M. Dahlin, "TCP Nice: a
mechanism for background transfers", Proceedings of
OSDI '02, 2002.
[Ven+08] Venkataraman, V., Francis, P., Kodialam, M., and T.
Lakshman, "A priority-layered approach to transport for
high bandwidth-delay product networks", Proceedings of ACM
CoNEXT, Madrid, December 2008.
[Wei+05] Weigle, M., Jeffay, K., and F. Smith, "Delay-based early
congestion detection and adaptation in TCP: impact on web
performance", Computer Communications 28(8):837-850,
May 2005.
[Wei+06] Wei, D., Jin, C., Low, S., and S. Hegde, "FAST TCP:
Motivation, architecture, algorithms, performance", IEEE/
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ACM Transactions on Networking , 14(6):1246-1259,
December 2006.
Authors' Addresses
Michael Welzl
University of Oslo
Department of Informatics, PO Box 1080 Blindern
N-0316 Oslo,
Norway
Phone: +43 512 507 6110
Email: michawe@ifi.uio.no
David Ros
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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