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.     

FCA-ItswTCM:区分服务中一种公平的拥塞自适应标记算法

 本文提出了一种新的滑动窗口标记算法——公平的拥塞自适应标记算法(FCA-ItswTCM).算法近似识别TCP流和UDP流,适度区分标记,规避拥塞控制机制对公平性的影响;细粒度描述拥塞,预测拥塞,以此自适应调节各流注入黄包比例,兼顾网络拥塞状态对公平性的影响.仿真实验表明,与其他几种滑动窗口标记算法相比,FCA-ItswTCM对确保TCP流和UDP流带宽共享的公平性、提高资源利用率及系统稳定性有较好的效果.     

Efficient congestion control mechanism for flow‐aware networks

Transmission based on flows becomes more and more popular in teleinformatics networks. To guarantee proper quality of service, to enable multipath transmissions, or just to increase transmission effectiveness in a network, traffic should be sent as flows. Flow‐aware networking architecture is one of the possible concepts to realize flow‐based transmissions. In this paper, the efficient congestion control mechanism (ECCM) is proposed to improve transmission in flow‐aware networks (FAN). The mechanism makes it possible to minimize acceptance delay of streaming flows (served with high priority) without deteriorating other transmissions in the network. It is confirmed by simulation experiments that the implementation of FAN with the ECCM mechanism is a promising solution for the Future Internet. Copyright © 2015 John Wiley & Sons, Ltd.     

前向主动网络拥塞控制算法及其性能分析

本文提出了一种基于主动式网络(Active Networks)技术的拥塞控制算法FACC(Forward Active Networks Congestion Control).与传统的TCP(Transport Control Protocol)相比,FACC算法通过在网络结点直接提供拥塞检测和拥塞控制机制,大大缩短源端点的拥塞反应时间,从本质上提高了网络拥塞检测和控制的性能,从而提高了终端用户的平均吞吐量.文中还利用计算机仿真研究了FACC算法在各种网络条件下的性能,并与传统的Tahoe,Reno,NewReno及SACK TCP协议做了对比.结果表明无论网络中存不存在非受控数据流时,FACC控制算法均能明显地提高用户终端的平均吞吐量,并且由于采用FACC控制算法而增加的网络结点运算迟延也很小.     

流量/拥塞控制技术研究

流量/拥塞控制的基本目的是以分布处理的方式有效地控制结点间的数据流,从而避免网络中出现拥塞。拥塞控制相应的控制策略称为拥塞控制算法(协议)。简述了Internet上基于TCP/IP的拥塞控制机制,分析和比较了TCP/IP上具体实现算法的稳定性,讨论了TCP/IP拥塞控制所面临的问题。     

A time-dominated TCP congestion control over heterogeneous networks

The traditional transmission control protocol (TCP) suffers from performance problems such as throughput bias against flows with longer packet roundtrip time (RTT), which leads to burst traffic flows producing high packet loss, long delays, and high delay jitter. This paper proposes a TCP congestion control mechanism, TD-TCP, that the sender increases the congestion window according to time rather than receipt of acknowledgement. Since this mechanism spaces out data sent into the network, data are not sent in bursts. In addition, the proposed mechanism reduces throughput bias because all flows increase the congestion window at the same rate regardless of their packet RTT. The implementation of the mechanism affects only the protocol stack at the sender; hence, neither the receiver nor the routers need modifications. The mechanism has been implemented in the Linux platform and tested in conjunction with various TCP variants in real environments. The experimental result shows that the proposed mechanism improves transmission performance, especially in networks with congestion and/or high packet loss rates. Experiments in real commercial wireless networks have also been conducted to support the proposed mechanism's practical use. Copyright © 2008 John Wiley & Sons, Ltd.     

TCP transmission rate control mechanism based on channel utilization and contention ratio in AD hoc networks

In ad hoc networks, both contention and congestion can severely affect the performance of TCP. In our work, we first show that the over-injection of conventional TCP window mechanism results in severe contentions, and medium contentions cause network congestion. Furthermore, introducing two metrics, channel utilization (CU) and contention ratio (CR), we characterize the network status. Then, based on these two metrics, we propose a new TCP transmission rate control mechanism based on Channel utilization and Contention ratio (TCPCC). In this mechanism, each node collects the information about the network busy status and determines the CU and CR accordingly. The CU and CR values fed back through ACK are ultimately determined by the bottleneck node along the flow. The TCP sender controls its transmission rate based on the feedback information. Simulation results show that the proposed TCPCC mechanism significantly outperforms the conventional TCP mechanism and the TCP contention control mechanism in terms of throughput and end-to-end delay.     

数据流的活动队列管理算法:MBLUE

MBLUE(Modified BLUE)是一种面向数据流的活动队列管理算法.它不是使用平均队列长度指示缓冲区拥塞状态,而是使用数据报丢弃的频率和队列空闲程度来管理网络拥塞.探测瓶颈连接早期的拥塞信息,通过数据报的丢弃和标记避免拥塞.它只维护一个先进先出队列,以较少的数据流状态信息,在不同流之间公平的分配网络带宽.能够适应瞬时的猝发流,能合理控制非TCP数据流,又能够保持较短的平均队列长度,从而控制、减轻网络拥塞.通过TCP/IP网络的模拟,证实算法在公平的分配网络带宽和降低数据报的丢失率上具有较好的鲁棒性.     

EVA: a better TCP version for resource‐insufficient networks

TCP Vegas exhibits unfair congestion avoidance mechanism, which aggravates when there are insufficient network resources to accommodate buffer space of a pipe (bandwidth delay product). To remedy this shortcoming, we propose an Enhanced VegAs (EVA) that employs three auxiliary mechanisms: Δ revision, congestion detection and congestion tendency detection. A 2^k factorial design with replications is used to study the effect of the three mechanisms. Our results show that TCP EVA achieves better performance than Vegas under various network conditions. Furthermore, congestion avoidance schemes, such as TCP EVA, perform much better than congestion control schemes, such as TCP Reno, in resource‐insufficient networks. Copyright © 2002 John Wiley & Sons, Ltd.     

A stochastic model for the behavior of multiple TCP NewReno sources over optical burst switching network

This work proposes a stochastic model to characterize the transmission control protocol (TCP) over optical burst switching (OBS) networks which helps to understand the interaction between the congestion control mechanism of TCP and the characteristic bursty losses in the OBS network. We derive the steady-state throughput of a TCP NewReno source by modeling it as a Markov chain and the OBS network as an open queueing network with rejection blocking. We model all the phases in the evolution of TCP congestion window and evaluate the number of packets sent and time spent in different states of TCP. We model the mixed assembly process, burst assembler and disassembler modules, and the core network using queueing theory and compute the burst loss probability and end-to-end delay in the network. We derive expression for the throughput of a TCP source by solving the models developed for the source and the network with a set of fixed-point equations. To evaluate the impact of a burst loss on each TCP flow accurately, we define the burst as a composition of per-flow-bursts (which is a burst of packets from a single source). Analytical and simulation results validate the model and highlight the importance of accounting for individual phases in the evolution of TCP congestion window.     

一种多播拥塞控制中速率的调节机制

本文针对计算机高速互联网中多播流的速率调节问题,在单点对多点的多播流量模型基础上,提出了一种由发送方驱动的单速率多播拥塞控制器的设计方法。并且运用现代控制理论和方法,讨论如何利用基于多播的单速率拥塞控制方法来对多播发送节点的发送速率进行调节,从而使得发送节点的发送速率趋于稳定。对所提出的拥塞控制方案,本文进行了分类仿真,仿真结果显示,控制方案使网络性能表现良好。     

区分服务中一种拥塞感知的单速三色标记算法

确保服务的实现依赖于在边界路由器执行的数据包标记策略和在核心路由器执行的队列管理策略.TCP流由于其拥塞自适应的特点对丢包很敏感,网络拥塞对其吞吐量影响很大.为此,我们设计了一种拥塞感知的单速三色标记算法CASR3CM.仿真实验表明,该算法不仅提高了AS TCP流的平均吞吐量,而且增强了吞吐量的稳定性.另外该算法也提高了AS TCP流之间占用带宽的公平性.     

Relative loss rate differentiation: performance of short‐lived TCP flows

The relative differentiated service model provides assurances for the relative quality ordering between service classes, rather than for the actual service level in each class. In this paper, we describe a relative loss rate differentiation scheme where packet drop probabilities are determined according to an active queue management (AQM) mechanism based on random early detection (RED) in a first‐in first‐out (FIFO) queue, are weighted in inverse proportion to the price that the network operator assigns to each service class. Basically, we describe a scheme where relative loss rate differentiation is incorporated directly into AQM. Most TCP flows today, particularly Web flows, can be characterized as short‐lived flows. Using simulations with short‐lived TCP flows, we show that the scheme is very effective in ensuring relative loss rate differentiation between service classes during times of network congestion. Copyright © 2004 John Wiley & Sons, Ltd.     

Improving fairness of PGMCC

PGM congestion control (pgmcc), a single‐rate multicast congestion control scheme, is one of the most promising schemes which aim to solve the problem of multicast fairness with TCP. However, by deep investigation, we find that there exist two inappropriate mechanisms in this scheme, which are the fixed slow start threshold (ssthresh) and the mechanism of changing a group representative (acker). Our experiments have revealed the fact that these mechanisms can lead to the unfairness of pgmcc under some network conditions. In this paper, two new mechanisms have been proposed to replace the original ones. One mechanism is to make the sender adapt the value of ssthresh to the network conditions to mimic the action of TCP, and in order to avoid being more aggressive than TCP, the other one is to make the sender reduce the congestion window by half when the acker changes. Our experiment results, parts of which are discussed in this paper, show that the modified pgmcc can achieve better performance than the original one. Copyright © 2005 John Wiley & Sons, Ltd.     

TCP拥塞控制算法研究

随着各种网络应用的不断涌现,网络拥塞问题变得越来越严重,解决这一问题也就越来越重要。阐述了TCP拥塞控制的基本原理,然后介绍了现有的几种TCP拥塞控制算法,并对这些算法进行了分析比较,结果表明虽然每种算法都有其优点,但也存在不足,不能从根本上解决拥塞问题。文章最后指出了拥塞控制的问题所在以及解决方法的发展趋势。     

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.     

Comparison of different congestion control strategies for low priority controllable traffic in packet switched backbone networks

It is expected that a significant part of the data flows of future multi‐service packet switched backbone networks will use low priority, non‐real‐time data transmission services of the networks. The common benefit for both user applications and network operators is that the data flows of the low priority services could use the free capacity of the networks, after the load of higher priority data flows. Congestion control methods are needed for these low priority data flows to reach an optimal utilisation level of the networks, high throughput and low packet loss ratios. This kind of low priority data transmission service which adjusts the data rates of the data flows according to the data rate changes of higher priority data flows, but does not guarantee any specific service for these data flows, is called a controlled load service. In this paper, we have compared the performance, efficiency and scalability of four different congestion control methods designed for the controlled load service. Two of these methods were based on very simple congestion control algorithms and the other two used relatively complex control algorithms based on control methods utilising computational intelligence. The principal aim of this study was to research how remarkable were the effects that the different complexities of the congestion control methods had on the achieved level of service. The simulation tests indicate that the complexity of the methods clearly affects the performance and efficiency of the methods. Copyright © 2001 John Wiley & Sons, Ltd.     

Removing redundant TCP functionalities in wired‐cum‐wireless networks with IEEE 802.11e HCCA support

The TCP was originally designed for wired networks, assuming transmission errors were negligible. Actually, any acknowledgment time‐out unconditionally triggers the congestion control mechanism, even in wireless networks in which this assumption is not valid. Consequently, in wireless networks, TCP performance significantly degrades. To avoid this degradation, this paper proposes the so‐called split TCP and UDP. In this approach, the access point splits the TCP connection and uses a customized and lighter transport protocol for the wireless segment. It takes advantage of the IEEE 802.11e Hybrid Coordination Function Controlled Channel Access (HCCA) mechanisms to remove redundant TCP functionalities. Specifically, the HCCA scheduler allows disabling of the congestion control in the wireless link. Similarly, the IEEE 802.11e error control service makes possible to eliminate TCP acknowledgments, therefore reducing the TCP protocol overhead. Finally, the usage of an HCCA scheduler permits providing fairness among the different data flows. The proposed split scheme is evaluated via extensive simulations. Results show that split TCP and User Datagram Protocol outperforms the analyzed TCP flavors—specifically designed for wireless environments—and the split TCP solution, achieving up to 95% of end‐user throughput gain. Furthermore, the proposed solution is TCP friendly because TCP flows are not degraded by the presence of flows by using this approach. Copyright © 2013 John Wiley & Sons, Ltd.     

Congestion control for multimedia applications in the wireless internet

This paper provides a parallel review of two important issues for the next‐generation multimedia networking. Firstly, the emerging multimedia applications require a fresh approach to congestion control in the Internet. Currently, congestion control is performed by TCP; it is optimised for data traffic flows, which are inherently elastic. Audio and video traffic do not find the sudden rate fluctuations imposed by the TCP multiplicative‐decrease control algorithm optimal. The second important issue is the mobility support for multimedia applications. Wireless networks are characterized by a substantial packet loss due to the imperfection of the radio medium. This increased packet loss disturbs the foundation of TCP's loss‐based congestion control. This paper contributes to the ongoing discussion about the Internet congestion control by providing a parallel analysis of these two issues. The paper describes the main challenges, design guidelines, and existing proposals for the Internet congestion control, optimised for the multimedia traffic in the wireless network environment. Copyright © 2004 John Wiley & Sons, Ltd.     

一种基于链路传输延迟的MPTCP拥塞控制算法*

由于传统TCP拥塞控制算法直接应用到MPTCP(Multipath Transport Control Protocol)中存在公平性问题,以及不能有效地发挥多路径传输的优势,因而从公平性方面对MPTCP现有拥塞控制算法进行研究.研究发现,现有的MPTCP拥塞控制算法均受到相同的回路时间限制.提出一种基于链路延迟的RTT补偿算法(Compensating for RTT mismatch,C-RTT ).该算法通过设置网络带宽占用比参数以及对MPTCP连接的子流设置侵略因子,从而保证瓶颈链路处MPTCP数据流和TCP数据流公平地共享可用带宽.最后通过NS3仿真实验证明,该算法能够有效地补偿链路中因RTT不等引起的公平性问题,并避免链路之间数据的非周期抖动,且保证了多路径传输的优越性.