A survey on TCP-friendly congestion control

New trends in communication, in particular the deployment of multicast and real-time audio/video streaming applications, are likely to increase the percentage of non-TCP traffic in the Internet. These applications rarely perform congestion control in a TCP-friendly manner; they do not share the available bandwidth fairly with applications built on TCP, such as Web browsers, FTP, or e-mail clients. The Internet community strongly fears that the current evolution could lead to congestion collapse and starvation of TCP traffic. For this reason, TCP-friendly protocols are being developed that behave fairly with respect to coexistent TCP flows. We present a survey of current approaches to TCP friendliness and discuss their characteristics. Both unicast and multicast congestion control protocols are examined, and an evaluation of the different approaches is presented     

一种新的逐跳TCP友好的主动分层多播拥塞控制方案

基于Internet的多媒体多播应用的迅猛发展对多播拥塞控制提出了要求.分层多播是适应网络异构性较有效的方案.针对现有分层多播存在的问题,将主动网技术思想引入到分层多播拥塞控制中,提出了一种逐跳TCP友好的主动分层多播拥塞控制方案(HTLMA),采用主动标记分层、逐跳探测TCP友好可用带宽,以及主动速率控制机制.仿真实验表明,HTLMA方案大大改进了分层多播拥塞控制的性能,具有较快的拥塞响应速度、较好的稳定性和TCP友好特性.     

Detecting shared congestion of flows via end-to-end measurement

Current Internet congestion control protocols operate independently on a per-flow basis. Recent work has demonstrated that cooperative congestion control strategies between flows can improve performance for a variety of applications, ranging from aggregated TCP transmissions to multiple-sender multicast applications. However, in order for this cooperation to be effective, one must first identify the flows that are congested at the same set of resources. We present techniques based on loss or delay observations at end hosts to infer whether or not two flows experiencing congestion are congested at the same network resources. Our novel result is that such detection can be achieved for unicast flows, but the techniques can also be applied to multicast flows. We validate these techniques via queueing analysis, simulation and experimentation within the Internet. In addition, we demonstrate preliminary simulation results that show that the delay-based technique can determine whether two TCP flows are congested at the same set of resources. We also propose metrics that can be used as a measure of the amount of congestion sharing between two flows     

新的多速率多播拥塞控制方案

提出了一种新的基于数据包束探测(packet-bunch probe)和TCP吞吐量公式的多速率多播拥塞控制方案PTMCC(packet-bunch probe and TCP-formula based multicast congestion control)。这种接收端驱动的拥塞控制,采用数据包束来探测网络的可用带宽,利用TCP吞吐量公式得到TCP友好速率,并采用了新的速率调节算法。仿真实验表明,PTMCC在收敛性、灵敏性以及TCP友好性上具有较好的性能。     

DCCP Congestion Control with Virtual Recovery to Achieve TCP-Fairness

The Datagram Congestion Control Protocol (DCCP) has been standardized by the IETF as an unreliable transport protocol for real-time streaming applications. DCCP features unreliable data delivery and congestion control to achieve timely transmission and to fairly share bandwidth with TCP respectively. However, we found that unfair bandwidth sharing between DCCP and TCP still exists because DCCP does not utilize fast recovery. This paper proposes a novel congestion control for DCCP, called TCP-friendly DCCP (TF-DCCP), which lets DCCP own "virtual recovery" and thus enter congestion avoidance simultaneously with TCP. Simulation results demonstrate that TF-DCCP can provide excellent TCP-fairness.     

The dynamics of responsiveness and smoothness in heterogeneous networks

Additive increase/multiplicative decrease-based protocols, including transmission control protocol (TCP), TCP-friendly, and a new generation of rate-based protocols, attempt to control the tradeoff of responsiveness and smoothness. Traditionally, smoothness has not been a main concern since it does not impact the performance of regular Internet applications such as the Web, FTP, or e-mail. However, multimedia-driven protocols attempt to favor smoothness at the cost of responsiveness. In general, smoothness and responsiveness constitute a tradeoff; however, we uncover undesirable dynamics of the protocols in the context of wireless/mobile networks with high-error rate or frequent handoffs: low responsiveness is not counterbalanced by gains in smoothness, but instead, produces a conservative behavior that degrades protocol performance with both delay-tolerant and -sensitive applications. Based on our observations, as well as on further analysis of the impact of the bottleneck queue on channel utilization, we seek an alternative strategy for smooth window adjustments. We introduce a new parameter /spl gamma/, which implements a congestion avoidance tactic and reaches better smoothness without damaging responsiveness.     

Leveraging single rate schemes in multiple rate multicast congestion control design

A significant impediment to deployment of multicast services is the daunting technical complexity of developing, testing and validating congestion control protocols fit for wide-area deployment. Protocols such as pragmatic general multicast congestion control (pgmcc) and TCP-friendly multicast congestion control (TFMCC) have recently made considerable progress on the single rate case, i.e., where one dynamic reception rate is maintained for all receivers in the session. However, these protocols have limited applicability, since scaling to session sizes beyond tens of participants with heterogeneous available bandwidth necessitates the use of multiple rate protocols. Unfortunately, while existing multiple rate protocols exhibit better scalability, they are both less mature than single rate protocols and suffer from high complexity. We propose a new approach to multiple rate congestion control that leverages proven single rate congestion control methods by orchestrating an ensemble of independently controlled single rate sessions. We describe a new multiple rate congestion control algorithm for layered multicast sessions that employs a single rate multicast congestion control as the primary underlying control mechanism for each layer. Our new scheme combines the benefits of single rate congestion control with the scalability and flexibility of multiple rates to provide a sound multiple rate multicast congestion control policy.     

Multicasting with Localized Control in Wireless Ad Hoc Networks

This paper investigates how to support multicasting in wireless ad hoc networks without throttling the dominant unicast flows. Unicast flows are usually congestion-controlled with protocols like TCP. However, there are no such protocols for multicast flows in wireless ad hoc networks and multicast flows can therefore cause severe congestion and throttle TCP-like flows in these environments. Based on a cross-layer approach, this paper proposes a completely-localized scheme to prevent multicast flows from causing severe congestion and the associated deleterious effects on other flows in wireless ad hoc networks. The proposed scheme combines the layered multicast concept with the routing-based congestion avoidance idea to reduce the aggregated rate of multicast flows when they use excessive bandwidth on a wireless link. Our analysis and extensive simulations show that the fully-localized scheme proposed in this paper is effective in ensuring the fairness of bandwidth sharing between multicast and unicast flows in wireless ad hoc networks.     

A spectrum of TCP-friendly window-based congestion control algorithms

The increasing diversity of Internet application requirements has spurred recent interest in transport protocols with flexible transmission controls. In window-based congestion control schemes, increase rules determine how to probe available bandwidth, whereas decrease rules determine how to back off when losses due to congestion are detected. The control rules are parameterized so as to ensure that the resulting protocol is TCP-friendly in terms of the relationship between throughput and loss rate. This paper presents a comprehensive study of a new spectrum of window-based congestion controls, which are TCP-friendly as well as TCP-compatible under RED. Our controls utilize history information in their control rules, and by doing so, they improve the transient behavior. We demonstrate analytically, and through extensive ns simulations, the steady-state and transient behavior of several instances of this new spectrum.     

Equation-based end-to-end single-rate multicast congestion control

Among the recently proposed single-rate multicast congestion control protocols is transmission control protocol-friendly multicast congestion control (TFMCC; Widmer and Handley 2001; Floyd et al. 2000; Widmer et al. IEEE Netw 15:28–37, 2001), which is an equation-based single-rate protocol that extends the mechanisms of the unicast TCP-friendly rate control (TFRC) protocol into the multicast domain. In TFMCC, each receiver estimates its throughput using an equation that estimates the steady-state throughput of a TCP source. The source then adjusts its sending rate according to the slowest receiver within the session (a.k.a., current-limiting receiver, CLR). TFMCC is a relatively simple, scalable, and TCP-friendly multicast congestion control protocol. However, TFMCC is hindering its throughput performance by adopting an equation derived from the unicast TFRC protocol. Further, TFMCC is slow to react to congestion conditions that usually result in a change of the CLR. This paper is motivated by these two observations and proposes an improved version of TFMCC, which we refer to as hybrid-TFMCC (or H-TFMCC for short). First, each receiver estimates its throughput using an equation that models the steady-state throughput of a multicast source controlled according to the additive increase multiplicative decrease (AIMD) approach. The second modification consists of adopting a hybrid sender/receiver-based rate control strategy, where the sending rate can be adjusted by the source or initiated by the current or a new CLR. The source monitors RTT variations on the CLR path, in order to rapidly adjust the sending rate to network conditions. Simulation results show that these modifications result in remarkable performance improvement with respect to throughput, time to react, and magnitude of oscillations. We also show that H-TFMCC remains TCP-friendly and achieves a higher fairness index than that achieved by TFMCC.     

Promoting the use of end-to-end congestion control in the Internet

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, “not TCP-friendly”, or simply using disproportionate bandwidth. A flow that is not “TCP-friendly” is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion     

CYRF: a theory of window-based unicast congestion control

This work presents a comprehensive theoretical framework for memoryless window-based congestion control protocols that are designed to converge to fairness and efficiency. We first derive a necessary and sufficient condition for stepwise convergence to fairness. Using this, we show how fair window increase/decrease policies can be constructed from suitable pairs of monotonically nondecreasing functions. We generalize this to smooth protocols that converge over each congestion epoch. The framework also includes a simple method for incorporating TCP-friendliness. Well-studied congestion control protocols such as TCP, GAIMD, and Binomial congestion control can be constructed using this method. Thus, we provide a common framework for the analysis of such window-based protocols. We also present two new congestion control protocols for streaming media-like applications as examples of protocol design in this framework: The first protocol, LOG, has the objective of reconciling the smoothness requirement of an application with the need for a fast dynamic response to congestion. The second protocol, SIGMOID, guarantees a minimum bandwidth for an application but behaves exactly like TCP for large windows.     

VIRAL: coupling congestion control with fair video quality metric

Video streaming is often carried out by congestion controlled transport protocols to preserve network sustainability. However, the success of the growth of such non-live video flows is linked to the user quality of experience. Thus, one possible solution is to deploy complex quality of service systems inside the core network. Another possibility would be to keep the end-to-end principle while making aware transport protocols of video quality rather than throughput. The objective of this article is to investigate the latter by proposing a novel transport mechanism which targets video quality fairness among video flows. Our proposal, called VIRAL for virtual rate-quality curve, allows congestion controlled transport protocols to provide fairness in terms of both throughput and video quality. VIRAL is compliant with any rate-based congestion control mechanisms that enable a smooth sending rate for multimedia applications. Implemented inside TFRC a TCP-friendly protocol, we show that VIRAL enables both intra-fairness between video flows in terms of video quality and inter-fairness in terms of throughput between TCP and video flows.     

一种支持多媒体通信QoS的拥塞控制机制

本文针对Internet传输协议TCP的和式增加积式减少(AIMD)拥塞控制机制不适应多媒体通信,而目前拥塞控制的研究又大多集中在尽量做好(Best-effort)服务上的问题,结合Internet上多媒体通信的特点及其对QoS的要求,提出了一种将多媒体通信服务质量(QoS)控制和基于速率拥塞控制结合起来的拥塞控制的新机制.本文详细地研究了这一机制,并提出了源端多媒体数据流的带宽控制策略、基于动态部分缓存共享(DPBS)的数据包丢失控制方案和接收端计算包丢失率p的方法.最后给出了整个拥塞控制机制的系统结构.     

MRF: A Framework for Source and Destination Based Bandwidth Differentiation Service

In this paper, we shall generalize the concepts of fairness, TCP-friendliness and TCP-compatibility such that more source adaptation schemes can be designed to support diverse applications over the Internet. A simple but efficient framework, in the form of a monotonic response function (MRF), is proposed for the analysis and the design of memoryless window-based source adaptation protocols by using these concepts. We first derive a necessary and sufficient condition for step-wise convergence to the weighted fairness. It is then used to construct increase-decrease policies. The requirements of our increase-decrease policy are less conservative than those of the CYRF (Choose Your Response Function) that was proposed in . Our MRF is suitable for transmission control protocol (TCP) and user datagram protocol (UDP), and can be used to design TCP-friendly and multimedia-friendly source adaptation schemes. Meanwhile, our MRF can be applied to provide bandwidth differentiation service without any change to the router of the existing Internet.     

Limitations of Equation-Based Congestion Control

We study limitations of an equation-based congestion control protocol, called TCP-friendly rate control (TFRC). It examines how the three main factors that determine TFRC throughput, namely, the TCP-friendly equation, loss event rate estimation, and delay estimation, can influence the long-term throughput imbalance between TFRC and TCP. Especially, we show that different sending rates of competing flows cause these flows to experience different loss event rates. There are several fundamental reasons why TFRC and TCP flows have different average sending rates, from the first place. Earlier work shows that the convexity of the TCP-friendly equation used in TFRC causes the sending rate difference. We report two additional reasons in this paper: 1) the convexity of where is a loss event period and 2) different retransmission timeout period (RTO) estimations of TCP and TFRC. These factors can be the reasons for TCP and TFRC to experience initially different sending rates. But we find that the loss event rate difference due to the differing sending rates greatly amplifies the initial throughput difference; in some extreme cases, TFRC uses around 20 times more, or sometimes 10 times less, bandwidth than TCP. Despite these factors influencing the throughput difference, we also find that simple heuristics can greatly mitigate the problem.     

Towards end-host-based identification of competing protocols against TCP in a bottleneck link

Classical Transmission Control Protocol (TCP) designs have never considered the identity of the competing transport protocol as useful information to TCP sources in congestion control mechanisms. When competing against a TCP flow on a bottleneck link, a User Datagram Protocol (UDP) flow can unfairly occupy the entire link bandwidth and suffocate all TCP flows on the link. If it were possible for a TCP source to know the type of transport protocol that deprives it of link access, perhaps it would be better for the TCP source to react in a way which prevents total starvation. In this paper, we use coefficient of variation and power spectral density of throughput traces to identify the presence of UDP transport protocols that compete against TCP flows on bottleneck links. Our results show clear traits that differentiate the presence of competing UDP flows from TCP flows independent of round-trip times variations. Signatures that we identified include an increase in coefficient of variation whenever a competing UDP flow joins the bottleneck link for the first time, noisy spectral density representation of a TCP flow when competing against a UDP flow in the bottleneck link, and a dominant frequency with outstanding power in the presence of TCP competition only. In addition, the results show that signatures for congestion caused by competing UDP flows are different from signatures due to congestion caused by competing TCP flows regardless of their round-trip times. The results in this paper present the first steps towards development of more ’intelligent’ congestion control algorithms with added capability of knowing the identity of aggressor protocols against TCP, and subsequently using this additional information for rate control.     

End-to-end rate-based congestion control: convergence properties and scalability analysis

We study several properties of binary-feedback congestion control in rate-based applications. We first derive necessary conditions for generic binary-feedback congestion control to converge to fairness monotonically (which guarantees asymptotic stability of the fairness point) and show that AIMD is the only TCP-friendly binomial control with monotonic convergence to fairness. We then study the steady-state behavior of binomial controls with n competing flows on a single bottleneck. Our main result here shows that combined probing for new bandwidth by all flows results in significant overshoot of the available bandwidth and rapid (often super-linear as a function of n) increase in packet loss. We also show that AIMD has the best scalability and lowest packet-loss increase among all TCP-friendly binomial schemes. We conclude the paper by deriving the conditions necessary to achieve constant packet loss regardless of the number of competing flows, n, and, in both simulation and streaming experiments, examine one new scheme, called ideally scalable congestion control, with such constant packet loss.     

SLM-一种安全分层组播协议

流媒体分发的一种典型实现方法是采用具有接收方驱动拥塞控制机制的分层组播.由于目前分层组播拥塞控制协议缺乏对用户行为的限制,接收方可违规订阅上层组播组发起自利型攻击,导致非公平的带宽利用.本文提出了一种较通用的安全分层组播协议SLM(Secure Layered Multicast).在路由器辅助拥塞控制条件下,在边界路由器采用基于Shamir秘密共享体制的拥塞状态相关访问控制(CR-AC,Congestion state Related Access Control)算法,管理用户组订阅行为,避免了用户自利型攻击,并使服务提供商可根据其与用户的协约限定不同用户的最高订阅级别.分析和仿真实验表明,该协议可实时保证网络流量安全共享带宽并具有较好的可扩展性.     

Multilayer multicast congestion control in satellite environments

It is well known that long and variable link delays, link errors, and handoffs in satellite environments seriously interfere with transmission control protocol's (TCP's) congestion control mechanisms. These channel characteristics also adversely affect existing multilayer multicast congestion control schemes when they are used in satellite environments. In addition, these schemes still have problems with fairly sharing bandwidth with TCP flows, controlling the overhead of frequent grafting and pruning, and handling misbehaving receivers. In this paper, we present a new multilayer multicast congestion control scheme that is suitable for satellite environments and overcomes most of the disadvantages of existing schemes. Our scheme is not affected by the long and variable delays of satellite links. Link errors also do not decrease the performance of our scheme. Further, our scheme has very limited control overhead. In addition to these advantages specific to satellite environments, our scheme achieves good fairness in sharing bandwidth with TCP sessions and is not sensitive to misbehaving receivers.