计算机高速互联网中一类基于速率的PD拥塞控制方法

运用现代控制理论和方法,针对计算机高速互联网中最大努力服务交通流即能控交通 流的调节问题,提出了一种基于速率的具有比例加微分(PD)控制器结构的拥塞控制理论和方法． 在单个节点的交通流的模型基础上,运用控制理论中系统稳定性分析方法,讨论如何利用信终端 节点缓冲占有量的比例加微分的反馈形式来调节信源节点的能控交通流的输入速率,从而使被 控网络节点的缓冲占有量趋于稳定.仿真结果显示,在所设计的PD控制方案下,网络的有关性能 较好.     

基于网络节点输出速率调节的非线性拥塞控制及稳定性鲁棒性分析

该文针对计算机高速互联网中最大服务交通流即能控交通流的拥塞调节问题,提出了一种基于网络节点输出速率调节的非线性拥塞控制机制并对其进行了稳定性和鲁棒性分析。克服了传统的基于输入端控制需要信源端和传输通道多重支持的约束。在单个节点的业务流模型基础上,将信源输入视为系统的扰动,通过调节输出速率来控制节点队列稳定。运用OPNET仿真软件对其进行了不同输入和工作条件下的仿真,结果显示,在所设计的非线性模糊PID队列控制机制下,队列稳定性好且具有较好的鲁棒性和抗干扰性。     

一种针对组播的分布式自调节显式速率控制器

拥塞控制是组播服务需要解决的重要问题.当存在大量异质的、传播时延较高的组播接收者时,到达源端的反馈在一定程度上已经过时,因此常常严重影响控制效果.提出了一种分布式、自调节的显式速率比例控制方案,它在源端和中间节点实施控制算法调整各自的发送速率,并且比例控制系数会根据路由器缓冲区占有量偏离理想值的程度自动调节.这种网络参与控制的主动行为比传统路由器尽力服务的工作方式对网络状态的响应更加迅速,自调节的控制系数比固定的控制系数更能及时调节发送速率.仿真实验结果表明,算法具有响应快,链路利用率高和路由器缓冲区占有量比较稳定的特性.     

Smith原则应用于多链路网络的拥塞控制

利用传统控制理论和Smith原则设计了一种计算机网络拥塞控制方案，数学分析和仿真结果表明该拥塞控制方案使得网络中的缓冲队列占有量较小，并且网络受控输出速率的抖动性较小，从而提高了网络的吞吐量，优化了网络的性能。     

一种自组织的二维元胞自动机网络模型及分析

提出了一种自组织的二维元胞自动机网络模型，并研究了网络中的临界特征和长相关特性。模型中，每个元胞包含一个路由器和随机数目的终端，并且能独立地根据探测到的网络拥塞状况调节其终端的数据包发送速率。仿真结果表明，在拥塞控制机制的作用下，网络能自组织地工作在临界状态，虽然节点间呈现出明显的异构特征，但节点数据包排队长度在时间和空间上仍都表现出较强的相关性。     

结合信源特性与网络拥塞控制的可靠性视频传输算法

提出了一种用于在无线网络中传输视频的结合信源特性及网络拥塞控制的鲁棒性算法.通过场景建模以及特性分析,将分级编码产生的所有码流层划分成不同的类型,并根据它们对网络拥塞控制的贡献以及对重建图像质量的贡献不同,将其分成两个不同的队列.系统根据不同的网络丢包状态(即丢包是由网络拥塞引起还是由无线信道的不可靠传输引起)动态地调整信源速率、不等错误保护强度以及拥塞控制策略.仿真结果表明,该方法与MPEG-4信源编码加固定速率Turbo码方法以及动态调整信源、信道编码速率加选择性丢I,B,P包的网络拥塞控制方法相比,能够提供更好的性能.     

网络流量系统的有限拍控制

应用控制理论方法讨论了ATM网络的ABR通信中基于速率反馈的流量控制问题.针对网络的传播时延提出了有限拍控制策略,使系统的响应在有限拍的时间内严格地到达稳态,从而使信源发送速率快速响应网络状态的变化,较好地克服了时延对流量控制快速性和稳定性的影响.该方法能有效地避免拥塞发生,同时降低信元丢失率,且结构简单.仿真结果验证了此算法的有效性.     

Ad Hoc网络中支持多速率传输通信的仿真研究

本文提出了一种协同媒体接入控制方法(CoopMAC-CTS),该方法通过利用高速率节点"帮助"低速率节点完成通信,并通过设计合理的握手顺序来减小隐藏终端对网络的影响。为了更好的运用协同通信技术,有效的支持多速率传输,我们提出了一种Ad Hoc网络中可支持多速率传输的跨层协同路由方法,通过仿真验证其优越性。     

基于Kalman算法及神经网络预测的网络流量控制

针对通信网络的传播时延会给基于速率反馈的流量控制带来极大的不利影响，提出了基于Kalman算法的反馈控制和神经网络在线预测补偿相结合的复合控制，对ATM网络的ABR流量进行控制，较好地克服了时延对流量控制的快速性和稳定性所产生的不利影响．仿真研究表明：本方案能使信源的发送速率快速响应网络状态的变化，有效地避免拥塞的发生，并使链路带宽得以充分利用．与PID控制方法相比，信元的丢失率更低、链路的利用率更高以及所需的缓冲容量更小．     

一种无线传感器网络的优化速率控制方法

拥塞控制是无线传感器网络中的一个关键性问题。从解决拥塞问题出发,提出了一个基于优化速率的拥塞控制算法ORCC。算法构建了一个分布式的分簇网络结构,利用缓冲的占用情况进行拥塞检测,并使用基于优化理论的速率调节策略来保证网络吞吐量的稳定,从而通过求解出的最优解来实现簇内节点效用的最大化。仿真实验表明,ORCC算法不仅能有效缓解网络拥塞,降低平均延迟,还具有较好的网络传输公平性。     

A hop-by-hop flow controller for a virtual path

In this paper, we propose a hop-by-hop robust controller for the flow regulation of the best-effort (ABR) traffic along a virtual path (VP) in high-speed networks. The local controller at each VP switch can regulate the transmission rate of virtual connections (VCs) based on the knowledge of the buffer occupancy in downstream VP switches. The fluid-flow model is used to design the hop-by-hop congestion controller, and used to study dynamic and steady behaviors of the network. A condition for the network's stability is derived. It is shown that with this controller applied, the network is asymptotically stable without oscillation. It is also shown that the proposed controller can achieve fairness and high utilization, and reduce the overhead traffic.     

An expert system for real-time traffic management in wireless local area networks

The paper explores delay-based congestion and flow control and the offloading of real-time traffic from wireless local area networks (WLANs) to mobile cellular networks (MCNs) in multihomed devices. The control system developed is based on an embedded hierarchical expert system. It adjusts transceivers’ traffic flow(s) for prevailing network conditions to achieve application-dependent delay and throughput limits. In wireless networks, delay and throughput depend on the packet size, packet transmission interval, and node connection density. Therefore, the controller on the destination node monitors average one-way delay and the change of one-way delay of the incoming traffic. On this basis, it adjusts the packet size and transmission interval of the source node by transmitting a control command to the source. If the prevailing level of traffic in the network exceeds its capacity despite of the control actions taken, devices prepare for developed asynchronous offloading of traffic to another access network.The control model was validated via simulation of Voice over Internet Protocol (VoIP) traffic in the OMNeT++ network simulator. The results demonstrate that the expert system developed is able to regulate packet sizes to match the prevailing application-dependent optimum and transfer traffic to another network if the network exceed its capacity no matter the control actions taken. Although this work is motivated mainly by issues of congestion and flow control of WLAN systems and the simulations and results were prepared for the IEEE 802.11b system, the approach and techniques are not limited to these systems, but they are applicable for other packet switched access networks (PSANs), too.     

多媒体播放中的动态双门限自适应调整算法

针对无线网络存在的自相似特性会影响视频流的播放质量问题,提出了基于滑动窗口的接收端播放缓存调整算法,根据网络流量的变化,动态地调整双门限,并利用播放缓存的占用率来控制视频流的播放速度,平滑时延抖动.仿真实验证明,无论网络流量处于平稳状态还是处于突发状态,本文设计的算法都能够较好地保证视频流的连续播放,提高视频流的播放质量,为用户提供良好的视觉效果.     

Dynamic resource-allocation for congestion-control in high-speed LAN Interconnection

When an ATM network is used as a backbone interconnection medium for a set of legacy low-speed LANs, a congestion control scheme is required to protect destination LANs from the risk of congestion that might be created when multiple users attempt to simultaneously communicate with the same LAN. In this paper, we present a novel congestion-control scheme in which the rate-based and bandwidth advertisement approaches inter-operate to efficiently manage and protect destination resources. In this scheme, each destination LAN is connected to source LANs through a tree-based virtual path (VP) layout that is rooted at the destination and has the source LANs as leaves. Under the rate-based framework, fractions (quotas) of the destination bandwidth are periodically computed on a hop-by-hop basis for each VP. Quotas are based on the estimated average load on each VP link. At source LANs, these quotas are used to regulate the instantaneous transmission rate into the ATM network. The bandwidth advertisement framework on the other hand works end-to-end. It uses the destination LAN buffer to regulate quota migration from low activity LANs to the high activity ones depending on the total buffer occupancy, as well as the individual buffer occupancy per source LAN at the destination. The scheme uses simple calculations, which makes it suitable for real time implementation. The performance of the scheme is evaluated through computer simulation under several network parameters and traffic patterns. Results show that a near zero packet loss probability at the destination is achieved while attaining high resource utilization. The results also demonstrate the fairness of the proposed scheme.     

Fuzzy-based rate control for real-time MPEG video

We propose a fuzzy logic-based control scheme for real-time motion picture expert group (MPEG) video to avoid long delay or excessive loss at the user-network interface (UNI) in an asynchronous transfer mode (ATM) network. The system consists of a shaper whose role is to smooth the MPEG output traffic to reduce the burstiness of the video stream. The input and output rates of the shaper buffer are controlled by two fuzzy logic-based controllers. To avoid a long delay at the shaper, the first controller aims to tune the output rate of the shaper in the video frame time scale based on the number of available transmission credits at the UNI and the occupancy of the shaper's buffer. Based on the average occupancy of the shaper's buffer and its variance, the second controller tunes the input rate to the shaper over a much larger time scale by applying a closed-loop MPEG encoding scheme. With this approach, the traffic enters the network at an almost constant bit rate (with a very small variation) allowing simple network management functions such as admission control and bandwidth allocation, while guaranteeing a relatively constant video quality since the encoding rate is changed only in critical periods when the shaper buffer “threatens” to overflow. The performance of the proposed scheme is evaluated through numerical tests on real video sequences     

Approximate bounds and expressions for the link utilization of shortest-path multicast network traffic

Multicast network traffic is information with one source node, but many destination nodes. Rather than setting up individual connections between the source node and each destination node, or broadcasting the information to the entire network, multicasting efficiently exploits link capacity by allowing the source node to transmit a small number of copies of the information to mutually-exclusive groups of destination nodes. Multicasting is an important topic in the fields of networking (video and audio conferencing, video on demand, local-area network interconnection) and computer architecture (cache coherency, multiprocessor message passing). In this paper, we derive approximate expressions for the minimum cost (in terms of link utilization) of shortest-path multicast traffic in arbitrary tree networks. Our results provide a theoretical best-case scenario for link utilization of multicast distribution in tree topologies overlaid onto arbitrary graphs. In real networks such as the Internet MBONE, multicast distribution paths are often tree-like, but contain some cycles for purposes of fault tolerance. We find that even for richly-connected graphs such as the shufflenet and the hypercube, our expression provides a good prediction of the cost (in terms of link utilization) of multicast communication. Thus, this theoretical result has two applications: (1) a lower bound on the link capacity required for multicasting in random tree topologies, and (2) an approximation of the cost of multicasting in regular LAN and MAN topologies.     

面向非全互连3D NoC的自适应单播路由算法

针对在非全互连三维片上网络（3D NoC）架构中的硅通孔（TSV）表只存储TSV地址信息,导致网络拥塞的问题,提出了记录表结构。该表不仅可以存储距离路由器最近的4个TSV地址,也可存储相应路由器输入缓存的占用和故障信息。在此基础上,又提出最短传输路径的自适应单播路由算法。首先,计算当前节点与目的节点的坐标确定数据包的传输方式;其次,检测传输路径是否故障,同时获取端口缓存占用信息;最后,确定最佳的传输端口,传输数据包到邻近路由器。两种网络规模下的实验结果表明,与Elevator-First算法相比,所提算法在平均延时和吞吐率性能指标上有明显的优势,且在网络故障率为50%时,Random和Shuffle流量模型下的丢包率分别为25.5%和29.5%。     

Impact of mobility on the performance of relaying in ad hoc networks – Extended version

We consider a mobile ad hoc network consisting of three types of nodes: source, destination, and relay nodes. All the nodes are moving over a bounded region with possibly different mobility patterns. We introduce and study the notion of relay throughput, i.e. the maximum rate at which a node can relay data from the source to the destination. Our findings include the results that (a) the relay throughput depends on the node mobility pattern only via its (stationary) node position distribution, and (b) that a node mobility pattern that results in a uniform steady-state distribution for all nodes achieves the lowest relay throughput. Random waypoint and random direction mobility models in both one and in two dimensions are studied and approximate simple expressions for the relay throughput are provided. Finally, the behavior of the relay buffer occupancy is examined for the random walk and random direction mobility models. For both models, the explicit form of the mean buffer is provided in the heavy-traffic case.     

Delay-jitter control in multimedia applications

The growing needs of multimedia communications are leading to new developments in providing real-time communication with guarantees. Several extensions have been proposed for different layers of the Open Systems Interconnection Reference Model to accomodate these needs. In this paper, we study methods for guaranteeing delay jitter bounds for high-speed networks in the network and application layers of this model. The method proposed for the network layer provides distributed jitter control. The method proposed for the application layer allows the destination application to control delay jitter. We use a simulation to compare the effects on delay jitter in each method for various scenarios, such as constant bit rate, cross traffic, and bursty data. In addition, the buffer space requirements for accommodating real-time channels are monitored at each node in the network.