随机延迟污染对IP网络坐标系统的影响及其抑制方法研究

针对实际网络中存在各种随机延迟污染严重影响IP网络坐标系统的时延预测性能和稳定性问题,从侧重于维持实测时延统计特征无损失和增强抑制时延随机波动2个角度,分别提出t-Filter和Δ-Filter 2种随机延迟污染抑制方法.通过PlanetLab数据集的仿真结果表明:与MP-Filter相比,它们能更有效地提高IP网络坐标系统的时延预测准确性、系统稳定性和收敛性.     

一种时延受限的低代价移动IP组播路由算法

基于固定和移动IP混合网络，针对时延敏感的实时通信业务，建立了网络模型，提出了有时延约束的低代价组播路由问题，给出了一种分布启发式组播路由算法，证明了算法的正确性，分析了算法的复杂度。仿真结果表明，算法是有效的、稳定的。     

基于UMTS的PoC业务QoS研究

文章首先介绍了UMTS系统端到端的QoS架构及业务的QoS等级分类,然后重点分析了基于UMTS系统的PoC业务的端到端QS保证问题,最后针对PoC业务无线链路传输的时延问题提出了一种降低时延的方法,仿真表明:对于支持IP包压缩的网络来说,该方法可有效优化PoC业务用户体验.     

基于接收端的TCP流量控制及优化研究

文中分析了发送方拥塞窗口、接收方缓存和网络容量三方面对网络流量的限制,并通过仿真得到接收端的缓存设置对网络吞吐量和时延影响的最优值,从而网络业务可以灵活调整缓存来对时延要求进行一定的优化。     

基于UMTS的P0C业务QoS研究

文章首先介绍了UMTS系统端到端的QoS架构及业务的QoS等级分类，然后重点分析了基于UMTS系统的PoC业务的端到端QoS保证问题，最后针对PoC业务无线链路传输的时延问题提出了一种降低时延的方法，仿真表明：对于支持IP包压缩的网络来说，该方法可有效优化PoC业务用户体验。     

基于OPNET的MPLS实用仿真分析

基于OPNET网络仿真环境对单播的数据传输模式进行了仿真,研究TMPLS技术在网络中的应用情况,通过对比分析仿真结果,表明将MPLS技术应用于传统IP网络可以有效地提高网络负载均衡、时延以及网络整体带宽的使用率等性能,尤其是对于复杂的网络更能使QoS得到优化。     

基于SIP协议的多媒体视讯业务承载质量研究——视频电话业务

视频电话业务是在传统语音电话业务上发展起来的新兴业务,在NGB网络中采用RTPoverIP协议的方式承载。IP网络具有的统计时分复用的特性,当网络负载状态发生变化时会导致在其中复用传输的业务报文复用受到时延变化、报文丢失、报文差错的影响,继而影响用户的业务体验。本文将在分析ITU-T和IETF关于业务承载技术和质量评估的标准建议基础上,结合业务仿真实验,就IP网络的时延、时延变化、丢包和错包等性能指标对视频电话业务影响进行分析,并给出网络承载指标的建议。     

保障流媒体时间结构的网络时延分析

流媒体同步对端到端时延和时延抖动提出了确定的要求,而终端抖动缓存一方面能消除时延抖动的影响,一方面却增加了端到端时延,流媒体同步保障对网络时延的要求不明确。论文从概率保障流媒体同步的角度,确定了保障流媒体同步的抖动缓存容量范围,提出了流媒体同步网络保障的充分条件,针对基于Internet VoIP(Voice over IP)业务的实际网络测试结果,给出了应用流媒体同步网络保障充分条件进行同步保障评价的应用实例并验证了其正确性。     

基于OPNET仿真的EIGRP和OSPF路由协议性能分析

路由协议是影响现代通信网络质量的关键。对于承载实时应用系统的IP通信网络,要求路由协议具备快速收敛、较小的端到端时延、高可靠性等能力。本文利用OPNET仿真工具,分析当前IP通信网络中应用最为广泛的OSPF和EIGRP路由协议,分别设计了采用OSPF、EIGRP以及同时存在OSPF与EIGRP的三个仿真实验场景,并模拟设备出现故障的情形,比较OSPF和EIGRP路由协议在这些仿真场景下的收敛性、端到端时延、可靠性等方面的差别。仿真结果显示,EIGRP的收敛时间比OSPF快5倍以上,EIGRP和OSPF同时存在时的时延抖动优于EIGRP或OSPF。     

一种支持HFC网中QoS业务的调度方法

为了在HFC中支持不同QoS的业务,必须仔细研究CMTS的上行业务调度算法。但是在当前最重要的标准 MCNS DOCSIS1.1中,没有规定任何针对 QoS设计的调度算法,而很多文献都是以改进的先到先服务(FCFS)为基础讨论的。该文提出了一种可支持 QoS的调度算法,通过将业务分类,按照不同类型业务的带宽占用动态分配业务的优先级,同时还考虑了回带(piggyback)对网络吞吐量的影响,给出了仿真结果,并针对典型的时延敏感业务IP Telephony提出了改善网络通道利用率的方法。     

Quality of service concerns in IP-based control systems

The popularity of network-based control systems (NBCS) is continuously growing. One of the most intriguing aspects is the transportation of control network data over IP-based networks using accepted standards such as EIA-852. To a large extent the actual quality of control (QoC) in such systems depends on the network timing such as delay and delay jitter. This paper presents a classification of relevant quality of service parameters and identifies application classes. Subsequently, the paper focuses on the effect of delay jitter at a fixed mean delay on the QoC. Two sources of delay jitter are identified in IP-based control systems: 1) network traffic induced and 2) protocol induced. As an example of a simple control loop implemented over an EIA-852-based system we investigate how the induced jitter affects the QoC using a time-discrete simulation model. Conclusions are drawn as to how the findings in the EIA-852 system can be interpreted and extended to a generalized NBCS.     

QoS Provisioning with New Effective Bandwidth/Buffer Calculation Scheme in Wireless IP Networks

The huge commercial success of mobile telephony, the phenomenal growth of Internet users, the popularity of IP-based multimedia applications are the major driving forces behind third-generation (3G), ongoing Byond 3G (B3G), and forth-genertion (4G) evolution. 3G brought wired applications, both data and multimedia, into wireless environments. It operates on IP-based infrastructures to provide wider service access capability. To support and satisfy QoS (Quality of Service) of diverse IP-based multimedia applications, traffic management, such as Connection Admission Control (CAC) and resource allocation, becomes essential. CAC and resource allocation are computationally complex when combined with QoS guarantee for traffic with different characteristics. However, CAC and resource allocation are real-time traffic control procedures. Hence, processing load should be minimized to reduce delay. At the same time, network resources should be utilized efficiently to accommodate more users. However, reducing processing load and obtaining high resource utilization efficiency has been considered to be contradictory matter. In addition, CAC and resource allocation schemes which consider multiple QoS criteria – loss and delay – simultaneously have not been adequately studied. Simultaneous QoS consideration is important to satisfy stringent and diverse QoS requirements of multimedia traffic. In this paper, we propose a nobel effective bandwidth/buffer calculation method based on a virtual channel/buffer analysis scheme. We show that our method can achieve high resource utilization efficiency with reduced processing load. Moreover, we show that our scheme allows for simultaneous consideration of multiple QoS criteria, loss and delay.     

Network queue scheduling algorithm based on self-similar traffic level grading prediction

Self-similarity characteristic of network traffic will lead to the continuous burstness of data in the network.In order to effectively reduce the queue delay and packet loss rate caused by network traffic burst,improve the transmission capacity of different priority services,and guarantee the service quality requirements,a queue scheduling algorithm P-DWRR based on the self-similarity of network traffic was proposed.A dynamic weight allocation method and a service quantum update method based on the self-similar traffic level grading prediction results were designed,and the service order of the queue according was determined to the service priority and queue waiting time,so as to reduce the queuing delay and packet loss rate.The simulation results show that the P-DWRR algorithm can reduce the queueing delay,delay jitter and packet loss rate on the basis of satisfying the different service priority requirements of the network,and its performance is better than that of DWRR and VDWRR.     

Traffic engineering standards in IP-networks using MPLS

The explosive growth of the Internet over the last few years has made the IP protocol suite the most predominant networking technology. Furthermore, the convergence of voice and data communications over a single network infrastructure is expected to happen over IP-based networks. Traditional IP-networks offer little predictability of service, which is often unacceptable for applications such as telephony, as well as for emerging and future real-time applications such as telemedicine. One of the primary goals of traffic engineering is to enable networks to offer predictable performance. Due to the need for better traffic control by network service providers, there has been considerable activity in the Internet Engineering Task Force to develop standards for traffic engineering in IP-based networks. This article discusses the direction taken by the IETF and some of the recent standardization efforts for traffic engineering using multiprotocol label switching (MPLS). Our primary focus is on the signaling protocols developed for this purpose     

Adaptive Input Traffic Prediction Scheme for Absolute and Proportional Delay Differentiated Services in Broadband Convergence Network

In this paper, an algorithm that provides absolute and proportional differentiation of packet delays is proposed with the objective of enhancing quality of service in future packet networks. It features an adaptive scheme that adjusts the target delay for every time slot to compensate the deviation from the target delay, which is caused by prediction error on the traffic to arrive at the next time slot. It predicts the traffic to arrive at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot. The difference between them is utilized by the delay control operation for the next time slot to offset it. Because the proposed algorithm compensates the prediction error continuously, it shows superior adaptability to bursty traffic and exponential traffic. Through simulations we demonstrate that the algorithm meets the quantitative delay bounds and is robust to traffic fluctuation in comparison with the conventional non‐adaptive mechanism. The algorithm is implemented with VHDL on a Xilinx Spartan XC3S1500 FPGA, and the performance is verified under the test board based on the XPC860P CPU.     

基于卫星时变网络的时延受限最小费用路由算法

卫星网络是一种特殊的时变网络,其QoS路由问题比地面网络QoS路由问题更为复杂.本文利用卫星拓扑的可预测性,通过引入业务的中断概率和费用模型来描述链路切换对业务QoS以及网络的影响,并给出时延受限最小费用路由(SDCMCR)算法来解决在卫星链路时延长的情况下寻找满足时延限制条件且受切换影响最小的路径的问题,该算法可兼顾网络业务中断率和业务阻塞率等性能,且计算复杂度低,可适应星上设备处理能力的要求.仿真表明算法在计算复杂性、业务请求被阻塞的概率等方面都优于同类算法.     

IP电话的技术障碍和未来发展

ＩＰ电话价格低廉,却因为存在时延、无ＱｏＳ保证、喧宽及标准等技术问题影响了其推广应用,但这些障碍将被逐一清除。以发展的眼光看,ＩＰ领域有４个最新发展方向：赋予普通电话ＩＰ功能;赋予呼叫中心网络能力;统一传信;ＩＰ电视电话。人们也将寻求更为廉价的ＩＰ电话装置。     

Mechanisms and hierarchical topology for fast handover in wireless IP networks

We propose a mechanism to perform fast handover in IP-based wireless networks for real-time applications such as Internet telephony and videoconferencing. Our proposal is designed to reestablish the communication session traffic flow quickly and to minimize the service disruption delay that occurs during mobile IP handover. In this scheme, we propose two different mechanisms to handle micromobility and inter-subdomain mobility, respectively. Micromobility handover handles movements within the same subdomain. Inter-subdomain handover supports handovers between two adjacent subdomains. The reason for having several subdomains is to deploy the network over a wider area to keep the mobile user in the same network for as long as possible. The novelty of the scheme is to retransmit the buffered packets during micromobility handover and to use multicasting to reestablish traffic flow during inter-subdomain movement. The entire scheme is performed within a hierarchical topology based on next-generation IP networks. We analyze both micromobility and inter-subdomain mobility handovers, and display simulation results for both voice and video over IP for micromobility handover.     

大型IP网络时延的主成分分析

网络时延是评估网络性能的关键指标之一。主成分分析(PCA)是数据挖掘领域常用的一种多变量分析和降维算法。通过对大型IP网络时延进行PCA分析,旨在挖掘网络时延的深层原因及网络各节点间的相互依赖关系,并搭建一个科学合理的网络时延评价体系,最终得到IP网络建设、优化改造的有效建议。对历史网络时延进行离线分析只是主成分分析方法的一种初步应用,今后可结合网络拓扑结构、现网流量流向、路由、距离等相关因素,将主成分分析方法应用到针对网络流量、网络时延、网络丢包等网络性能的实时在线监测分析中,进一步提升网络运营的效率和质量。     

DiffServ resource allocation for fast handoff in wireless mobileInternet

The next-generation wireless networks are evolving toward a versatile IP-based network that can provide various real-time multimedia services to mobile users. Two major challenges in establishing such a wireless mobile Internet are support of fast handoff and provision of quality of service (QoS) over IP-based wireless access networks. In this article, a DiffServ resource allocation architecture is proposed for the evolving wireless mobile Internet. The registration-domain-based scheme supports fast handoff by significantly reducing mobility management signaling. The registration domain is integrated with the DiffServ mechanism and provisions QoS guarantee for each service class by domain-based admission control. Furthermore, an adaptive assured service is presented for the stream class of traffic, where resource allocation is adjusted according to the network condition in order to minimize handoff call dropping and new call blocking probabilities