基于云模型的Ad-hoc拥塞跨层控制仿真

针对传统网络拥塞控制方法没有考虑到控制丢包率,导致控制效果不佳的问题,提出基于云模型的移动Ad-hoc网络拥塞跨层控制方法.分析移动Ad-hoc网络的分层结构,构建网络拓扑结构模型.运用该模型分析丢包产生的原因,并加以优化.在此基础上为了提升网络链路的使用率、降低网络拥塞程度,在缓冲队列中采用云模型检测网络拥塞情况,调节网络数据的发送速率,从而实现对网络拥塞的跨层控制.通过仿真分析可知,上述方法的控制效果较理想,丢包率较低,有效解决了传统方法中存在较大丢包率的问题,能够为相关工作提供可靠的参考,具有较高的使用价值.     

WSN中一种改进的DCCP拥塞控制机制

由于WSN具有链路质量差、数据冗余度高和"多对一"汇聚等特性,使得传统TCP协议的应用受到了挑战。为此,基于数据报传输协议(DCCP)提出一种改进的拥塞控制机制IM_DCCP。IM_DCCP根据误码丢包率来区分链路误码和网络拥塞引发的丢包,同时引入信道繁忙比率来协作进行网络拥塞检测,并针对不同的拥塞等级选择正确的窗口调节策略实施拥塞控制。NS2仿真表明,所提出的机制与其他机制比较,公平性较好,且能有效的降低丢包率、提高网络吞吐量性能。     

基于自相似流量预测的TCP拥塞控制算法

针对高带宽网络环境下传统TCP Reno的拥塞控制效率低的问题,提出了一种适用于随机丢失网络的TCP拥塞控制改进算法.介绍并分析了网络流量自相似特性和预测的时间尺度,并进行了TCP链路流量预测及可用带宽估计,当网络出现丢包时根据估计的带宽动态设置慢启动阈值.OPNET仿真结果表明,该算法在高随机丢包的情况下吞吐量得到了明显改善.     

一种IP网络拥塞链路丢包率范围推断算法

针对大规模IP网络拥塞链路丢包率范围推断算法中存在的不足,提出一种贪婪启发式拥塞链路丢包率范围推断算法.借助多时隙路径探测,避开单时隙探测对时钟同步的强依赖;通过学习各链路拥塞先验概率,借助贝叶斯最大后验定位拥塞链路;提出聚类拥塞链路相关、性能相近路径集合的策略,通过对聚类路径集合中性能相似系数求解,循环推断拥塞链路丢包率范围.实验验证了本文提出算法的准确性及鲁棒性.     

基于流量调度的SDN数据中心网络拥塞控制算法

针对采用软件定义网络(SDN)的数据中心网络拥塞的问题,提出一种基于流量调度的数据中心网络拥塞控制算法。当链路发生拥塞时,该算法首先判别拥塞链路中 链路上关键度最大的大流,然后对大流进行重路由计算,选择调度开销最小的流,并进行调度代价计算,最后对调度代价最小的流进行调度。实验结果表明,所提算法能够有效缓解网络拥塞,降低丢包率,提高链路利用率,使得网络性能更为稳定。     

V2G网络中基于带宽自适应的拥塞控制协议优化

V2G网络下PLC链路带宽受限、高误码率等特点导致现有的TCP NewReno拥塞控制机制缺乏对丢包类型的有效判断,将链路上由噪声干扰的随机错误丢包与网络拥塞丢包统一当做拥塞事件处理,从而造成不必要的拥塞避免,导致了低吞吐量问题.根据此问题,提出了一种基于带宽自适应的拥塞控制算法.该算法通过分组预测拥塞等级感知网络状态,由此估计可用带宽来判断丢包类型,实现了拥塞窗口自适应调节.仿真结果表明该算法在拥塞窗口的增长、吞吐量、公平性、收敛性和友好性等方面都优于现有算法,V2 G网络的吞吐量得到明显提升.     

基于改进Dijkstra算法的配用电通信网流量调度策略

针对配用电通信网中数据汇聚易产生拥塞的问题,提出了一种复合边权值流量调度路由算法。首先,依据跳数建立节点分层模型;然后,划分配用电业务优先级和节点拥塞等级;最后,以跳数、流量负载率和链路利用率为综合指标计算边权值,对需要流量调度的节点根据改进的Dijkstra算法进行路由选择,同时对重度拥塞节点按照配用电业务优先级进行调度。与最短路径（SPF）算法和贪婪背压算法（GBRA）相比,在数据生成率为80 kb/s时,所提算法紧急型业务丢包率分别减少了81.3%和67.7%,关键型业务丢包率分别减少了79%和63.8%。仿真结果表明,所提算法能有效缓解网络拥塞,提高网络有效吞吐量,降低网络端到端时延和高优先级业务的丢包率。     

一种丢包区分方法及其在Linux下的实现

针对传统拥塞控制机制在无线链路丢包的情况下出现的盲目降低拥塞窗口、TCP吞吐量下降的问题,研究无线/有线混合环境下基于该机制的一种丢包区分方法——WECN,在Linux操作系统下进行实现。搭建混合网络实验床仿真WECN与TCP Reno, TCP Westwood 2种协议相结合的网络模型。测试结果表明,WECN能扩展到已有的TCP协议中,提高含无线链路网络中TCP的吞吐量。     

一种新的被动队列管理算法

针对因网络拥塞而产生的丢包问题,基于弃头方式提出了一种新的被动队列管理算法(Drop Front n based on Wavelet and Shuffled frog leaping,DFWS)。该算法首先利用小波变换对实际流量的长相关特性进行刻画,并利用混合蛙跳方法获取网络队长,同时结合弃头方式提出丢包策略和丢包概率。最后,以实际数据进行仿真实验,深入研究了网络队长、丢包数量和服务率之间的关系。结果发现,与其他算法相比,DFWS具有较好的适应性。     

基于主动网络的多路径流量均衡模型研究

现有网络中到达某一目的地址的流量一般集中在最优路径上,这使得网络资源利用不均衡,拥塞不可避免,而传统网络体系结构很难实现流量均衡。利用主动网络节点的协作分布式计算能力给出了一种多路径的流量均衡模型,该模型包括多路径输出流量优化分配、输入带宽优化分配、拥塞抑制等算法,实现了流量与资源的均衡配置。实验证明该模型提高了网络链路的利用率,避免或减少了网络拥塞的发生,整个网络性能得到了提高。     

基于Pareto分布的IP骨干节点输入通信量模型

输入通信量的行为特性对于IP骨干网节点的性能和设计有重要的影响.本文从IP骨干网络节点输入通信量角度出发,提出了一个基于Pareto和指数分布的混合通信量模型,其中数据包到达间隔为Pareto分布,包大小为指数分布.该模型克服了传统网络通信量模型中没有显式考虑数据包大小的缺点,从数据包级对网络通信量进行了精确的描述,从而能够更好地进行路由器结构设计和性能分析.仿真实验验证了我们模型的有效性.     

针对逆向流的Reno和Vegas算法性能的仿真分析

网络拥塞已经成为制约网络发展和应用的一个瓶颈.在TCP中已经实现的Reno和Vegas均是基于端到端的拥塞控制算法,它们虽然减轻了前向路径上的拥塞程度,但不能改善后向路径上数据流拥塞所导致的诸如吞吐率、丢包率等网络性能降低这一状况.基于"包丢失是由于网络拥塞引起"的假定,利用OPNET网络仿真软件,建立了单个瓶颈网络,设定了吞吐量、丢包率以及拥塞窗口(cwnd)等参数,对Reno和Vegas两种算法在逆向流中的性能进行比较,分析影响网络吞吐量的因素.仿真结果表明,在非对称网络中,Reno比Vegas更灵活且更实用,后向路径上逆向流的拥塞会极大地降低Vegas的吞吐量,使得Vegas和Reno不能很好地兼容.     

TPDCM:一种新型的分布式拥塞管理方案

提出一种新型的面向区分服务网络的分布式拥塞管理方案,基本思想是利用拥塞状态反馈信息在边缘节点或主机上实施拥塞管理。该方案主要包括三个组成部分:流量控制信息协议、早期拥塞检测和流量控制算法。该方案具有以下特点:(1)在核心网络中,拥塞控制机制独立于特定的传输层协议;(2)对将要发生的拥塞做出快速响应;(3)降低网络的分组丢失率;(4)不需要在网络核心节点中保存每个传输流的状态信息。实验结果表明,与标准的区分服务网络相比,该方案能在TCP和UDP聚集之间公平地分配带宽并能显著地降低分组丢失率。     

Congestion Adaptive Routing in Mobile Ad Hoc Networks

Mobility, channel error, and congestion are the main causes for packet loss in mobile ad hoc networks. Reducing packet loss typically involves congestion control operating on top of a mobility and failure adaptive routing protocol at the network layer. In the current designs, routing is not congestion-adaptive. Routing may let a congestion happen which is detected by congestion control, but dealing with congestion in this reactive manner results in longer delay and unnecessary packet loss and requires significant overhead if a new route is needed. This problem becomes more visible especially in large-scale transmission of heavy traffic such as multimedia data, where congestion is more probable and the negative impact of packet loss on the service quality is of more significance. We argue that routing should not only be aware of, but also be adaptive to, network congestion. Hence, we propose a routing protocol (CRP) with such properties. Our ns-2 simulation results confirm that CRP improves the packet loss rate and end-to-end delay while enjoying significantly smaller protocol overhead and higher energy efficiency as compared to AODV and DSR     

Performance evaluation of bandwidth allocation methods in a geostationary satellite channel in the presence of internet traffic

Resource allocation represents an important issue for the next generation TCP/IP Quality of Service-based satellite networks. Many schemes, proposed in the recent literature, consider Internet traffic as the superposition of traffic sources without distinguishing between User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) flows, even if UDP and TCP imply very different traffic characteristics. The basic idea of this work is that a resource allocation algorithm which is conscious of the difference may be more efficient because it can make use of the different behaviour of TCP and UDP in the presence of network congestion. Actually TCP reduces the source flow rate and, as a consequence, also the bandwidth occupancy when there is network congestion. The use of this feature within the bandwidth allocation scheme allows reducing the bandwidth waste due to over provisioning and using the residual bandwidth for other sources. The advantage is particularly clear over satellite channels where fading often affects the communication: having some residual bandwidth available for stations which have experienced fading can improve the satellite network performance.This paper presents a detailed performance evaluation of a bandwidth allocation scheme, called E-CAP-ABASC and studied for the satellite environment. The bandwidth is assigned to the earth stations that compose the network by a master station on the basis of a cost function whose main part is represented by a closed-form of the packet loss probabilities for the TCP and UDP traffic. The use of two different packet loss probability models for TCP and UDP allows exploiting the different features of the two traffic types, so improving the overall performance either in terms of packet loss or, on the other hand, in terms of the traffic admitted.The performance evaluation is carried out by varying the link degradation due to fading, the traffic load, and the flow balance between UDP and TCP. The results show a good performance of E-CAP-ABASC, compared with two other schemes. Advantages and drawbacks are discussed.     

Multi-indicator Active Queue Management Method

A considerable number of applications are running over IP networks. This increased the contention on the network resource, which ultimately results in congestion. Active queue management (AQM) aims to reduce the serious consequences of network congestion in the router buffer and its negative effects on network performance. AQM methods implement different techniques in accordance with congestion indicators, such as queue length and average queue length. The performance of the network is evaluated using delay, loss, and throughput. The gap between congestion indicators and network performance measurements leads to the decline in network performance. In this study, delay and loss predictions are used as congestion indicators in a novel stochastic approach for AQM. The proposed method estimates the congestion in the router buffer and then uses the indicators to calculate the dropping probability, which is responsible for managing the router buffer. The experimental results, based on two sets of experiments, have shown that the proposed method outperformed the existing benchmark algorithms including RED, ERED and BLUE algorithms. For instance, in the first experiment, the proposed method resides in the third-place in terms of delay when compared to the benchmark algorithms. In addition, the proposed method outperformed the benchmark algorithms in terms of packet loss, packet dropping, and packet retransmission. Overall, the proposed method outperformed the benchmark algorithms because it preserves packet loss while maintaining reasonable queuing delay.     

一种新的流媒体拥塞控制算法

提出一种新的拥塞控制算法(TCP MS).该算法更适用于流媒体应用,有更高的带宽利用率、公平性,传输速率也更平滑.不同于传统的利用丢包率和排队延迟来探测拥塞的TCP拥塞控制算法,该算法通过确认数据包的速率来探测拥塞,并在每一轮往返时间内及时调整窗口.该算法提供的拥塞窗口变化更准确,传输速率抖动更小.因此,提高了网络带宽的利用率以及传输速率的平滑性.最后,文章将TCP MS与典型的基于丢包率的TCP Reno算法和基于排队延迟的TCP Vegas算法在带宽利用率、速率抖动以及公平性等方面分别做了比较,仿真结果表明TCP MS是一种理想的流媒体拥塞控制算法.     

组播协议在OPNET中的建模与仿真

该文以IP组播技术为重点，结合网络仿真软件OPENT Modeler，分析该软件环境下IP组播网络的建模机制，包括参考标准、组的管理、支持的应用、组播路由协议的选择，节点加入组播组与发送源发送组播数据的流程。以校园网视频会议和FTP传输应用为例，构建网络仿真模型，一方面比较单播与组播方式下的网络性能，分析了视频流量的发送情况、视频会议分组的端到端延时，FTP传输的响应时间，骨干网络点到点链路吞吐量；另一方面比较了组播方式采用共享树机制和由共享树切换到最短路径树在网络性能上的改进，包括分组延迟的降低、汇合点路由器上拥塞发生的减少等。同时，也对无线移动通信网络环境下的组播技术提出更多需要考虑的因素。     

针对视频会议的一种传输控制方案

为满足视频会议中网络流量的平滑稳定性以及公平竞争性,设计了一种新的传输拥塞控制方案——CTP( Conference Transmission Protocol).CTP在逻辑上分为DCC(Dynamic Congestion Control Layer)和RAC( Rate Amended Control Layer...