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.     

无线网络中TCP友好流媒体传输改进机制

为保持无线网络中多媒体业务对TCP的友好性，提出了一种适用于无线网络的动态自适应的流媒体传输速率调节机制。该机制通过在接收端区分网络拥塞丢包和链路错误随机丢包，准确判断网络的拥塞状况结合接收端缓存区占用程度，自适应实施多级速率调节，实现了TCP流友好性和流媒体服务质量（QoS）的折中。由于准确区分出无线链路误码丢包和动态调整流媒体QoS要求，该机制能维持较高的网络利用率。仿真实验结果显示在连接数为2和32，链路误码率从0到0．1变化时TCP,TFRC和吞吐量幅度下降幅度较大，WTFCC幅度下降相对较慢，最大相差达2M；在网络负载重时，尽管链路误码率较低，WTFCC区分链路错误与拥塞丢包，因此，端到端丢包率高于TCP和TFRC，但整体传输吞吐量也高于两者。归一化吞吐量显示WTFCC对TCP流友好。     

异构网络下一种新的TCP技术

由于传统传输控制协议（TCP）拥塞控制主要是为带宽时延乘积较小和信道误码率很低的有线网络环境而设计的,因此很难适用于异构网络下长的往返时延RTT、较大的误码率、以及大带宽等问题。提出了一种新的TCP-selective技术,通过在同一个服务器上使用不同的TCP版本,与实际的链路质量相匹配,根据链路信息来调用与之相适应的TCP来解决单一TCP性能增强方案所不能解决的问题,仿真结果证明这种新的TCP技术是有效可行的。     

TCP友好的速率控制

TCP友好的速率控制(TFRC)主要适用于实时数据传输的一种拥塞控制机制,具有突出的TCP友好性即在相同的环回时间(RTT)下可以和TCP流享有近乎相同的带宽,从而避免了由于UDP等传输层协议缺乏拥塞控制而带来的网络拥塞甚至崩溃.本文简要介绍了它的协议机制并通过一些仿真和试验的结果初步讨论了其性能.     

TCP-H：基于三重判决的TCP拥塞控制

在异构无线网络中存在高误码、切换、信号衰落等链路特性,使传统的TCP拥塞控制机制受到了挑战。在不增加开销的情况下,基于丢包检测、RTT时间和ACK返回速率三重判决,提出了一种新的TCP拥塞控制机制（TCP—H）。仿真结果表明,TCP—H增强了对拥塞和随机差错的区分能力,满足公平性要求,改进的最小RTT计算方法解决了在低延迟向高延迟网络切换的时Vegas,Westwood等算法存在的最小RTT更新问题,有效提高了在异构无鲅网络环埔下TCP的性能．     

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.     

一种提高异构网络传输性能的双向流量控制机制

针对异构网络存在的高误码率和不对称带宽等特点使TCP性能降低的问题,本文提出了一种双向流量控制机制BFC,它根据数据链路的双向通信状况, 在前后向两个信道进行流量控制.在前向信道,我们设计了一种显式的ACK详细指示方法,协调多层控制机制及时调节流量;在后向信道,提出了一个ACK流量控制算法,根据双向拥塞状况,对ACK流进行自适应地控制.实验表明,该机制可以有效地提高异构网络传输性能,尤其是当双向信道的不对称参数较高的情况下,该方法显示出独特的优势.     

MP-Start：基于带宽测量的分阶段TCP慢启动机制

针对TCP慢启动机制门限值参数难于设置以及突发流量引起网络抖动的问题，提出了一种基于带宽测量及分阶段实施的TCP慢启动改进算法。该方法利用在线网络带宽测量技术，探测出网络带宽，从而根据网络状态的动态变化实现慢启动门限值的动态更新，同时，采用分阶段方法调节拥塞窗口的增加，在拥塞窗口大于门限值一半时，拥塞窗口增幅为门限值和拥塞窗口差值的一半，逐步迭代逼近门限值，使拥塞窗口在连接启动和过渡到拥塞避免阶段增加幅度比较小，而在中间阶段窗口增加幅度比较大。有效避免了多个分组丢失现象的发生，实现了连接的平滑接入和过渡到拥塞避免阶段；为提高改进算法的性能，设计了一种灵活的参数化的模型和自适应参数设置方案。大量仿真实验结果表明MP-Start能在多种网络环境下有效改善网络性能，如降低分组丢弃数、减少数据传输时间、降低共享链路分组队列长度、保持连接的传输平滑性。     

MPLS网络主动式流量和拥塞控制机制及性能分析

MPLS是具有大带宽一时延迟的网络,用传统的TCP解决MPLS拥塞问题显得十分困难,该文结合MPLS的网络特点,提出了一种适合MPLS网络的主动式流量和拥塞控制机制,在网络边缘节点引入拥塞反馈处理,对实验模型进行了性能仿真分析,实验证明,与传统的TCP协议相比,该机制将流量和拥塞控制从用户端点扩展到MPLS边缘路由器,能够更及时地检测和控制网络拥塞,缩短了控制时延,可以进行较精确的流量调节,实现了提高吞吐量和改善缓冲区利用率的目的。     

一种拥塞感知的TFRC协议慢启动算法

 本文分析了TFRC(TCP-Friendly Rate Control)协议在慢启动阶段采用类似TCP协议的倍增发送速率机制存在的问题,提出了一种利用回路响应时间(Round Trip Time,RTT)来自适应调节慢启动阶段速率的算法.通过分析实际RTT值和EWMA(Exponentially Weighted Moving Average)处理后的平均RTT值来感知网络当前的拥塞状况,以调节发送速率的激进程度.仿真实验表明,该方法对TFRC协议具有明显的改进作用,减少了慢启动阶段结束时的报文丢失率,提高了协议的传输平稳度和吞吐量,从而能更有效地适应多媒体流的传输要求.     

Delay-based congestion avoidance for TCP

The set of TCP congestion control algorithms associated with TCP-Reno (e.g., slow-start and congestion avoidance) have been crucial to ensuring the stability of the Internet. Algorithms such as TCP-NewReno (which has been deployed) and TCP-Vegas (which has not been deployed) represent incrementally deployable enhancements to TCP as they have been shown to improve a TCP connection's throughput without degrading performance to competing flows. Our research focuses on delay-based congestion avoidance algorithms (DCA), like TCP-Vegas, which attempt to utilize the congestion information contained in packet round-trip time (RTT) samples. Through measurement and simulation, we show evidence suggesting that a single deployment of DCA (i.e., a TCP connection enhanced with a DCA algorithm) is not a viable enhancement to TCP over high-speed paths. We define several performance metrics that quantify the level of correlation between packet loss and RTT. Based on our measurement analysis, we find that, although there is useful congestion information contained within RTT samples, the level of correlation between an increase in RTT and packet loss is not strong enough to allow a TCP-sender to improve throughput reliably. While DCA is able to reduce the packet loss rate experienced by a connection, in its attempts to avoid packet loss, the algorithm reacts unnecessarily to RTT variation that is not associated with packet loss. The result is degraded throughput as compared to a similar flow that does not support DCA.     

Implicit admission control

Internet protocols currently use packet-level mechanisms to detect and react to congestion. Although these controls are essential to ensure fair sharing of the available resource between multiple flows, in some cases they are insufficient to ensure overall network stability. We believe that it is also necessary to take account of higher level concepts, such as connections, flows, and sessions when controlling network congestion. This becomes of increasing importance as more real-time traffic is carried on the Internet, since this traffic is less elastic in nature than traditional Web traffic. We argue that, in order to achieve better utility of the network as a whole, higher level congestion controls are required. By way of example, we present a simple connection admission control (CAC) scheme which can significantly improve the overall performance. This paper discusses our motivation for the use of admission control in the Internet, focusing specifically on control for TCP flows. The technique is not TCP specific, and can be applied to any type of flow in a modern IP infrastructure. Simulation results are used to show that it can drastically improve the performance of TCP over bottleneck links. We go on to describe an implementation of our algorithm for a router running the Linux 2.2.9 operating system. We show that by giving routers at bottlenecks the ability to intelligently deny admission to TCP connections, the goodput of existing connections can be significantly increased. Furthermore, the fairness of the resource allocation achieved by TCP is improved     

A study on a receiver‐based management scheme of access link resources for QoS‐controllable TCP connections

Although the bandwidth of access networks is rapidly increasing with the latest techniques such as DSL and FTTH, the access link bandwidth remains a bottleneck, especially when users activate multiple network applications simultaneously. Furthermore, since the throughput of a standard TCP connection is dependent on various network parameters, including round‐trip time and packet loss ratio, the access link bandwidth is not shared among the network applications according to the user's demands. In this thesis, we present a new management scheme of access link resources for effective utilization of the access link bandwidth and control of the TCP connection's throughput. Our proposed scheme adjusts the total amount of the receive socket buffer assigned to TCP connections to avoid congestion at the access network, and assigns it to each TCP connection according to characteristics in consideration of QoS. The control objectives of our scheme are (1) to protect short‐lived TCP connections from the bandwidth occupation by long‐lived TCP connections, and (2) to differentiate the throughput of the long‐lived TCP connections according to the upper‐layer application's demands. One of the results obtained from the simulation experiments is that our proposed scheme can reduce the delay of short‐lived document transfer perceived by the receiver host by up to about 90%, while a high utilization of access link bandwidth is maintained. Copyright © 2006 John Wiley & Sons, Ltd.     

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.     

Fair and efficient Transmission Control Protocol access in the IEEE 802.11 infrastructure basic service set

When the stations in an IEEE 802.11 infrastructure basic service set employ Transmission Control Protocol (TCP), this exacerbates per‐flow unfair access problem. We propose a novel analytical model to approximately calculate the maximum per‐flow TCP congestion window limit that prevents packet losses at the access point buffer and therefore provides fair TCP access both in the downlink and uplink. The proposed analysis is unique in considering the effects of varying number of uplink and downlink TCP flows, differing round trip times among TCP connections and the use of delayed TCP acknowledgment (ACK) mechanism. Motivated by the findings of this theoretical analysis and simulations, we design a link layer access control block to be employed only at the access point in order to resolve the unfair access problem. The proposed link layer access control block uses congestion control and ACK filtering approach by prioritizing the access of TCP data packets of downlink flows over TCP ACK packets of uplink flows. Via simulations, we show that the proposed algorithm can provide both short‐term and long‐term fair accesses while improving channel utilization and access delay. Copyright © 2013 John Wiley & Sons, Ltd.     

Protecting TCP from a misbehaving receiver

This study presents the design and implementation of a robust TCP congestion control algorithm. TCP was originally designed for cooperative environments, and its evolution over the years has been built on the same basis. TCP expects the end hosts to cooperate with the TCP senders in implementing end‐to‐end congestion control. Therefore, misbehavior of a TCP receiver may result in an unfair division of the available bandwidth between the conforming flows and the irresponsible flows. Accordingly, this study examines the issues arising when conforming TCP connections are obliged to coexist with misbehaving connections. A modification to the TCP protocol is proposed to deal with various types of TCP misbehavior. Copyright © 2006 John Wiley & Sons, Ltd.     

A Novel Congestion Detection Scheme in TCP Over OBS Networks

This paper introduces a novel congestion detection scheme for high-bandwidth TCP flows over optical burst switching (OBS) networks, called statistical additive increase multiplicative decrease (SAIMD). SAIMD maintains and analyzes a number of previous round-trip time (RTTs) at the TCP senders in order to identify the confidence with which a packet loss event is due to network congestion. The confidence is derived by positioning short-term RTT in the spectrum of long-term historical RTTs. The derived confidence corresponding to the packet loss is then taken in the developed policy for TCP congestion window adjustment. We will show through extensive simulation that the proposed scheme can effectively solve the false congestion detection problem and significantly outperform the conventional TCP counterparts without losing fairness. The advantages gained in our scheme are at the expense of introducing more overhead in the SAIMD TCP senders. Based on the proposed congestion control algorithm, a throughput model is formulated, and is further verified by simulation results.      

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.     

Queue length‐based load balancing in data center networks

Traffic load balancing in data centers is an important requirement. Traffic dynamics and possibilities of changes in the topology (e.g., failures and asymmetries) make load balancing a challenging task. Existing end‐host–based schemes either employ the predominantly used ECN or combine it with RTT to get congestion information of paths. Both congestion signals, ECN and RTT, have limitations; ECN only tells whether the queue length is above or below a threshold value but does not inform about the extent of congestion; similarly, RTT in data center networks is on the scale of up to few hundreds of microseconds, and current data center operating systems lack fine‐grained microsecond‐level timers. Therefore, there is a need of a new congestion signal which should give accurate information of congestion along the path. Furthermore, in end‐host–based schemes, detecting asymmetries in the topology is challenging due to the inability to accurately measure RTT on the scale of microseconds. This paper presents QLLB, an end‐host–based, queue length–based load balancing scheme. QLLB employs a new queue length–based congestion signal that gives an exact measure of congestion along the paths. Furthermore, QLLB uses relative‐RTT to detect asymmetries in the topology. QLLB is implemented in ns‐3 and compared with ECMP, CONGA, and Hermes. The results show that QLLB significantly improves performance of short flows over the other schemes and performs within acceptable level, of CONGA and Hermes, for long flows. In addition, QLLB effectively detects asymmetric paths and performs better than Hermes under high loads.