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
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




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.

本文件是互联网工程任务组(IETF)的产品。它代表了IETF社区的共识。它已经接受了公众审查,并已被互联网工程指导小组(IESG)批准出版。并非IESG批准的所有文件都适用于任何级别的互联网标准;见RFC 5741第2节。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


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 . . . . . . . . . . . . . . . . . . . . . . . . .  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
1. 介绍

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]).

本文件简要概述了通过终端主机机制实现不尽力而为(LBE)服务的建议。我们松散地将LBE服务定义为在与标准TCP共享瓶颈时,对标准TCP的带宽和/或延迟影响小于标准TCP本身的服务。我们将设计用于提供此服务的系统称为LBE系统。除了TCP Vegas(我们出于历史原因提出)之外,我们排除了在某些情况下表现出LBE行为但并非为此目的设计的系统(例如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
2. 基于延迟的传输协议

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.

TCP Vegas[Bra94]是已知在两个协议共享瓶颈[Kur00]时具有比标准TCP更小的发送速率的首批协议之一——然而,它的设计目的是实现比标准TCP更多而不是更少的吞吐量。事实上,当TCP Vegas是通过瓶颈的流使用的唯一拥塞控制算法时,其吞吐量大于标准TCP的吞吐量。根据瓶颈队列长度,TCP Vegas本身可能会被标准TCP流耗尽。这可以通过随机早期检测(RED)主动队列管理机制[RFC2309]在一定程度上解决。Vegas根据预期吞吐量和实际吞吐量之间的差异线性增加或减少发送速率。该估算基于RTT测量。

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

拥塞避免行为是协议在历史相关性以及本文档上下文中的相关性方面最重要的特征(已经证明,协议的其他元素有时可以对其整体行为发挥更大的作用[Hen00])。在拥塞避免中,每RTT一次,TCP Vegas将预期吞吐量计算为WindowsSize/BaseRTT,其中WindowsSize是当前拥塞窗口,BaseRTT是所有测量RTT的最小值。然后将预期吞吐量与根据最近的确认测量的实际吞吐量进行比较。如果实际吞吐量小于

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.

TCP Vegas已被广泛分析。TCP Vegas最突出的特性之一是它在相同类型的多个流之间的公平性,这不会像标准TCP那样惩罚具有大传播延迟的流。虽然这不是第一个使用延迟作为拥塞指示的协议,但由于TCP Vegas在文献中扮演的历史性“里程碑”角色,其前身(如CARD[Jai89]、Tri-S[Wan91]或DUAL[Wan92])在此不作讨论。

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.

设计为非侵入性的基于延迟的传输协议包括TCP Nice[Ven02]和TCP低优先级(TCP-LP)[Kuz06]。TCP Nice[Ven02]遵循与TCP Vegas相同的基本方法,但在某些方面有所改进。由于其适度的线性减少拥塞响应,TCP Vegas可以影响标准TCP,尽管它能够早期检测拥塞。TCP Nice通过将拥塞窗口减半(与标准TCP一样,每个RTT最多一次)而不是线性减少来消除此问题。为了避免过于保守,只有在一个RTT内出现基于延迟的初始拥塞信号的固定预定义部分时,才可以这样做。否则,如果没有数据包丢失,TCP Nice将返回TCP Vegas的拥塞避免规则,如果数据包丢失,则返回标准TCP。TCP Nice的另一个特性是,通过在多个RTT上发送单个数据包,它能够支持小于一个数据包的拥塞窗口。[Ven02]的作者通过ns-2仿真和使用Linux实现的实际实验表明,TCP Nice实现了有效利用备用容量的目标,同时不干扰标准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

除了TCP Vegas和TCP Nice之外,TCP-LP[Kuz06]仅使用单向延迟(OWD)而不是RTT作为初期拥塞的指标。这样做是为了避免对反向交叉交通引起的延迟波动作出反应。使用TCP时间戳选项[RFC1323],OWD被确定为ACK中接收器的时间戳值与接收器复制到ACK中的原始时间戳值之间的差值。虽然此减法的结果只能精确表示时钟同步时的OWD,但其绝对值与TCP-LP无关,因此不需要时钟同步。TCP-LP使用常数平滑参数计算测量OWD的指数加权移动平均值(EWMA),并检查结果是否超过最小值和最小值范围内的阈值

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.

基于延迟的拥塞控制也是旨在使TCP的拥塞避免适应高速网络的建议的基础。其中一些方案,如复合TCP[Tan06][Sri08]和TCP伊利诺伊[Liu08]是基于丢失和延迟的混合机制,而其他方案(如NewVegas[Dev03]、FAST TCP[Wei06]或代码TCP[Cha10])是主要基于延迟的Vegas的变体。

2.1. Accuracy of Delay-Based Congestion Predictors
2.1. 基于延迟的拥塞预测器的精度

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 瓶颈缓冲区大小:原则上,如果缓冲区“足够大”,可以使基于延迟的机制“比TCP更友好”,以便终端主机能够以合理的精度检测RTT波动和/或与最小RTT的偏差。否则,可能很难区分实际延迟变化和测量噪声。

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 RTT测量问题:原则上,RTT样本的分辨率可能很差,这是因为计时器的粒度相对于延迟波动的规模太粗。此外,在某些情况下(例如,流速远低于链路带宽),流可能由于欠采样而获得非常噪声的rtt估计。对于TCP,其他潜在的测量噪声源包括TCP分段卸载(TSO)和延迟ACK的使用[Hay10]。拥塞的反向路径还可能导致对正向路径的拥塞状态的错误评估。最后,在快速或短距离链路的情况下,大多数测量的延迟实际上可能是由于相关主机中的处理造成的;通常情况下,这种处理延迟并不重要,它可能导致与网络无关的波动。

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 统计多路复用和RTT采样的级别:单个流可能很容易“错过”丢失/队列溢出事件,特别是当共享瓶颈缓冲区的流数量很大时。这一点在[McC08]的图1中得到了很好的说明。

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].

o 无线链路的影响:无线链路的几种典型机制,如链路层调度和错误恢复,可能会在短时间内引起强烈的延迟波动[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.

有趣的是,关于基于延迟的拥塞预测的准确性,Bhandarkar等人[Bha07]的结果似乎描绘了一幅稍有不同的图景。Bhandarkar等人声称,通过采用一些简单的技术(平滑RTT样本,增加RTT采样频率),可以显著提高预测精度。尽管如此,他们承认,即使使用这种技术,也不可能消除检测错误。他们提出的基于延迟的拥塞避免方法PERT(Probabilistic Early Response TCP)通过对拥塞检测事件的概率响应机制减轻了剩余检测错误的影响。

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

2.2. LBE传输中基于延迟的拥塞控制的潜在问题

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
3. 基于非延迟的传输协议

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.

4CP[Liu07]代表“竞争和考虑周到的拥塞控制”,是一种通过改变标准TCP的窗口控制规则来提供LBE服务的协议。维护“虚拟窗口”,在所谓的“坏拥塞阶段”期间,该窗口被减小到小于实际拥塞窗口的预定义最小值。只有当虚拟窗口超过此最小值时,拥塞窗口才会再次增加,并且通过这种方式,虚拟窗口控制发送方以固定的最小速率传输的持续时间。拥塞状态是“坏”还是“好”取决于丢失事件率是否高于或低于阈值(或目标值)。4CP拥塞避免算法允许设置目标平均窗口,并在限制对“前景”流的影响的同时避免“背景”流的不足。它的性能在ns-2仿真和Microsoft Windows Vista内核级实现的实际实验中进行了评估。

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.

MulTFRC[Dam09]协议是TCP友好速率控制(TFRC)[RFC5348]对多个流的扩展。MulTFRC进一步采纳了MulTCP[Cro98]和类似提案(例如[Hac04]、[Hac08]、[Kuo08])的主要思想。单个MulTCP流尝试模拟(并像模拟)多个N>1的并行TCP流。通过支持0到1之间的N值,MulTFRC可以用作LBE服务的机制。由于MulTFRC不像第2节中描述的协议那样对延迟做出反应,而是像TFRC一样调整其速率,因此MulTFRC可能比TCP Nice或TCP-LP等机制更具攻击性。这也意味着MulTFRC不太可能挨饿,因为它的攻击性在细粒度上是可调的,即使N在0到1之间。

4. Upper-Layer Approaches
4. 上层通道

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.

所谓的后台智能传输服务[BITS]在多个版本的Microsoft Windows中实现。BITS将应用层优先级系统用于文件传输作业,同时监测网络接口(或在更新版本中,连接到终端主机的网络网关)的带宽使用情况,以便给定终端主机上的低优先级传输让位给高优先级传输(前台)来自同一主机上的交互式应用程序的传输和流量。

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
4.1. 面向接收器、基于流量控制的方法

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

[Spr00]中描述的机制是上述技术的一个示例。这种机制通过让接收器智能地操纵标准TCP的接收器窗口来控制带宽。这是可能的,因为作者假设客户端-服务器设置,其中接收方的访问链路通常是瓶颈。该方案结合了瓶颈处预期队列长度的基于延迟的计算,这与上述基于延迟的协议(例如TCP Vegas)中的计算非常相似。使用Linux实现,其中TCP流被分类

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
5. 网络辅助方法

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.

作者认为,这些边缘节点应该是组织的网关,以便使其方案与使用激励保持一致,但如果Harp仅部署在终端主机中,技术解决方案也会起作用。它通过查看延迟来检测即将发生的拥塞,类似于TCP Nice[Ven02],并在不需要对TCP本身进行任何更改的情况下,通过纯单路径解决方案提高TCP的利用率和公平性。

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.

在[Ven08]中,Venkataraman等人提出了一种传输层方法,以利用现有的基于优先级排队的网络层LBE服务。他们称之为PLT(优先级层传输)的传输协议将第4层连接分成两个流,一个高优先级流和一个低优先级流。高优先级流通过更高优先级的排队类(原则上,提供尽力而为的服务)发送,使用AIMD、类似TCP的拥塞控制机制。映射到LBE类的低优先级流使用非TCP友好的拥塞控制算法。因此,PLT的目标是通过积极利用未使用的容量来最大化其总吞吐量,同时保护best effort类所承载的标准TCP流。类似的精神,[Ott03]建议只对TCP的AIMD参数进行简单更改,以便在网络层LBE服务上使用,这样“填充”流量可能会大量消耗未使用的带宽。请注意,[Ven08]还考虑了一种用于检测网络中缺少优先级排队的机制,以便可以抑制非TCP友好流。PLT接收器监测两个流的损失率;如果高优先级流开始出现损失,而低优先级流没有经历100%的损失,这表明没有严格的优先级排队。

6. LEDBAT Considerations
6. 莱德巴特的考虑

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
7. 致谢

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.

作者要感谢Melissa Chavez、Dragana Damjanovic和Yinxia Zhao提供的参考点,以及Jari Arkko、Mayutan Arumaithurai、Elwyn Davies、Wesley Eddy、Stephen Farrell、Mirja Kuehlewind、Tina Tsou和Rolf Winter提供的详细评论和建议。

8. Security Considerations
8. 安全考虑

This document introduces no new security considerations.


9. Informative References
9. 资料性引用

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[ARU10] ARUMAITURURI,M,FU,X,和K. Ramakrishnan,“NF-TCP:一个网络友好的TCP变体,用于背景延迟不敏感的应用”,技术报告编号IF-TB-201005,计算机科学学院,哥廷根大学,德国,2010年9月,<>。

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[Cro98b] Crovella, M. and P. Barford, "The network effects of prefetching", Proceedings of IEEE INFOCOM 1998, April 1998.


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[Dyk02] 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.


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[Hac04] Hacker, T., Noble, B., and B. Athey, "Improving Throughput and Maintaining Fairness using Parallel TCP", Proceedings of IEEE INFOCOM 2004, March 2004.

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[Key04] 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.

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[Mar03] Martin, J., Nilsson, A., and I. Rhee, "Delay-based congestion avoidance for TCP", IEEE/ACM Transactions on Networking, 11(3):356-369, June 2003.


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[Mo99]Mo,J.,La,R.,Anantharam,V.,和J.Walrand,“TCP雷诺和TCP维加斯的分析和比较”,IEEE INFOCOM 1999年会议记录,1999年3月。

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[Pra04] Prasad, R., Jain, M., and C. Dovrolis, "On the effectiveness of delay-based congestion avoidance", Proceedings of PFLDnet, 2004.


[RFC1323] Jacobson, V., Braden, B., and D. Borman, "TCP Extensions for High Performance", RFC 1323, May 1992.

[RFC1323]Jacobson,V.,Braden,B.,和D.Borman,“高性能TCP扩展”,RFC 1323,1992年5月。

[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.

[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.,和L.Zhang,“关于互联网中队列管理和拥塞避免的建议”,RFC 2309,1998年4月。

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

[RFC3168]Ramakrishnan,K.,Floyd,S.,和D.Black,“向IP添加显式拥塞通知(ECN)”,RFC 3168,2001年9月。

[RFC3662] Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort Per-Domain Behavior (PDB) for Differentiated Services", RFC 3662, December 2003.

[RFC3662]Bless,R.,Nichols,K.,和K.Wehrle,“区分服务的低域行为(PDB)”,RFC 36622003年12月。

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

[RFC5348]Floyd,S.,Handley,M.,Padhye,J.,和J.Widmer,“TCP友好速率控制(TFRC):协议规范”,RFC 5348,2008年9月。

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

[RFC5681]Allman,M.,Paxson,V.和E.Blanton,“TCP拥塞控制”,RFC 56812009年9月。

[Rew06] 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.

[ReW06] Reaskar,S.,Kaur,J.和D. Smith,“为什么不基于延迟的拥塞估计在现实世界中工作?”,北卡罗来纳大学教堂山分校计算机科学系,技术报告Tr06-01,2006年1月。

[Sch10] Schneider, J., Wagner, J., Winter, R., and H. Kolbe, "Out of my Way -- Evaluating Low Extra Delay Background Transport in an ADSL Access Network", Proceedings of the 22nd International Teletraffic Congress ITC22, 2010.


[Sha05] 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.


[Sha11] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, "Low Extra Delay Background Transport (LEDBAT)", Work in Progress, May 2011.


[Spr00] Spring, N., Chesire, M., Berryman, M., Sahasranaman, V., Anderson, T., and B. Bershad, "Receiver based management of low bandwidth access links", Proceedings of IEEE INFOCOM 2000, pp. 245-254, vol. 1, 2000.

[Spr00]Spring,N.,Chesire,M.,Berryman,M.,Sahasranaman,V.,Anderson,T.,和B.Bershad,“低带宽接入链路的基于接收器的管理”,IEEE INFOCOM 2000会议录,第245-254页,第1卷,2000年。

[Sri08] Sridharan, M., Tan, K., Bansala, D., and D. Thaler, "Compound TCP: A New TCP Congestion Control for High-Speed and Long Distance Networks", Work in Progress, November 2008.


[Tan06] 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.

[Tan06]Tan,K.,Song,J.,Zhang,Q.,和M.Sridharan,“用于高速和远程网络的复合TCP方法”,IEEE INFOCOM 2006年会议记录,西班牙巴塞罗那,2008年4月。

[Ven02] Venkataramani, A., Kokku, R., and M. Dahlin, "TCP Nice: a mechanism for background transfers", Proceedings of OSDI '02, 2002.

[Ven02]Venkataramani,A.,Kokku,R.,和M.Dahlin,“TCP Nice:后台传输机制”,OSDI'02会议录,2002年。

[Ven08] 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.

[Ven08]Venkataraman,V.,Francis,P.,Kodialam,M.,和T.Lakshman,“高带宽延迟产品网络传输的优先级分层方法”,ACM CoNEXT会议记录,马德里,2008年12月。

[Wan91] Wang, Z. and J. Crowcroft, "A new congestion control scheme: slow start and search (Tri-S)", ACM Computer Communication Review, 21(1):56-71, January 1991.


[Wan92] Wang, Z. and J. Crowcroft, "Eliminating periodic packet losses in the 4.3-Tahoe BSD TCP congestion control algorithm", ACM Computer Communication Review, 22(2):9-16, January 1992.

[Wan92]Wang,Z.和J.Crowcroft,“消除4.3-Tahoe BSD TCP拥塞控制算法中的周期性数据包丢失”,ACM计算机通信评论,22(2):9-16,1992年1月。

[Wei05] 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.


[Wei06] Wei, D., Jin, C., Low, S., and S. Hegde, "FAST TCP: Motivation, architecture, algorithms, performance", IEEE/ 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: +47 22 85 24 20
   Phone: +47 22 85 24 20

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

David Ros Institut Telecom/Telecom Bretagne Rue de la Chataignerie,CS 17607 35576塞森塞维尼塞德克斯法国

   Phone: +33 2 99 12 70 46
   Phone: +33 2 99 12 70 46