自相似网络的时延抖动性能仿真分析

自相似性对网络性能产生了影响是当前的研究热点。建立了一种基于FBM的自相似网络排队时延抖动分析模型,重点讨论了自相似流量作为输入时对排队系统的时延抖动的影响。对理论分形流量和实际测量流量进行了仿真实验,验证了结果的正确性和有效性。实验结果表明：自相似流量长相关强弱的程度对排队系统时延抖动特性具有非常不同的影响,尤其是在缓存较大的情况下。同时,还发现网络流量中长相关发生作用时状态转变与排队系统本身的参数也有关,这是新的发现,对实时业务的网络性能评价具有重要的参考意义。     

保障连续媒体流用户层QoS 的缓存控制

研究了缓存控制对媒体流用户层QoS 的影响.多媒体系统信宿端通常采用播放缓存来补偿时延抖动,提高媒体流播放的连续性.缓存控制虽然能够降低时延抖动的影响,却增加了端到端时延.时延或时延抖动是用户可感知的QoS 参数,缓存控制对用户层QoS 的影响究竟如何呢?利用已有的应用层向用户层QoS 映射的研究结果,分析缓存控制参数与端到端QoS 参数、应用层QoS 参数的关系,获得了缓存控制参数与用户层QoS 参数的关系.从理论上深入挖掘缓存控制对用户层QoS 参数的作用,给出了提供确定时延和时延抖动保障的缓存容量值,论证了在网络环境一定时存在提供最佳用户层QoS 的缓存容量值.实验结果验证了分析.     

IEEE802.11ax基于时延敏感的上行链路自适应帧聚合传输方案

帧聚合传输技术因其高效的传输效率被广泛使用.针对密集部署环境,下一代无线局域网标准IEEE 802.11ax基于帧聚合传输技术提出多流量标识符技术,允许多种不同流量类型的数据聚合传输,提升了密集环境下帧聚合传输性能.然而,上行链路多用户随机接入机制无法保障诸如视频、语音等高吞吐率低时延的传输业务需求.因此,本文利用帧聚合多流量标识符技术,对密集部署站点流量随机到达过程建模分析,推导出帧聚合时延表达式,并提出自适应时延敏感帧聚合传输方案.该方案根据站点缓存数据包的时延约束,使用二分搜索算法动态地调整帧聚合数目来最大化网络整体吞吐率.仿真结果表明,所提方案能够有效地降低站点时延和丢包率,提升网络吞吐率.     

基于令牌桶算法的卫星数据地面传输流量控制方法研究

该文根据卫星数据的特点,在研究令牌桶算法的基础上,提出了一种改进的令牌桶算法,增加了两个模块:令牌放置模块和令牌请求模块,在模拟环境下实现该算法。在对卫星数据进行流量控制的同时,降低了数据传输过程中的抖动,并保证了卫星实时数据的低时延和非实时数据尽量大的带宽,在模拟环境下进行测试,达到了较好的效果。     

一种可提供绝对服务质量分级的优先调度算法

提出了一种能够提供端到端时延保证和满足丢包率要求的多优先级算法。该算法以分组头中记录的时延、丢包率、保证带宽为权重对分组进行调度,通过对信元的相对优先级及服务质量参数的加权算法,得到一种公平的满足绝对服务质量的算法。还能够使系统避免维护每个流的状态信息以及对单个流进行复杂的队列管理和调度,由此增加了系统的可扩展性。计算机仿真表明该算法具有较高的资源利用率,较低的端到端时延和时延抖动以及较低的分组丢弃率等特点。     

QoS接纳控制算法性能分析与比较

对5种QoS接纳控制算法进行性能分析和比较。论文在NS2中构建了实验用的网络拓扑和流量,在RSVP信令下,对3种不同带宽利用率的网络场景进行了仿真,并对以上算法在各种场景下的丢包率、接入率、传输时延和时延抖动情况进行了分析和比较。     

UGC视频网络QoS模型研究

把握网络QoS的变化对UGC视频QoE的影响程度对优化运营商网络至关重要。在不同地点跟踪通过网络损伤模拟各种QoS参数变化情况下UGC典型事务的事务时延,发现各种网络QoS参数对UGC视频QoE的影响程度不一,8 Mbps下行带宽、2%的丢包率、100 ms时延、64 ms抖动能够满足绝大多数UGC事务的QoE。只有将UGC业务部署优化和网络优化结合起来才能满足特定地区用户的QoE。     

基于流量自相似性的网络队列管理算法

网络流量的自相似性会导致数据突发状态持续,传统队列管理算法无法对网络流量突发状态进行预测,从而影响网络端到端时延、丢包率和吞吐性能。针对该问题,提出一种基于网络流量预测的主动队列管理算法P-ARED。基于网络流量的均值和方差给出网络流量等级的概念,讨论网络流量等级转移概率与Hurst参数之间的关系,提出基于贝叶斯估计思想的网络流量等级预测方法。在此基础上,在对自相似网络流量环境下的平均队列长度、缓存队列长度最小阈值等参数优化设置的基础上,基于Hurst参数和自相似流量等级预测结果,重新设计ARED算法中分组丢弃概率的计算方法,以提高缓存队列长度的稳定性。仿真结果表明,P-ARED算法与对比的主动队列管理算法相比,降低了网络端到端时延和丢包率,提高了端到端吞吐性能,其中平均吞吐量最高提升7.63%,平均时延最多降低17.52%。     

WLAN与EPON融合接入网上行带宽分配算法

为了提高带宽的利用率,提出一种WLAN与EPON融合接入网上行带宽分配算法。该算法将无线终端接入的业务分为不同的服务等级,以实现不同业务Qo S保证。首先,ONU-AP给各个无线终端STA分配带宽,采用IEEE 802.11E协议的简单调度算法给语音业务和一般数据业务分配带宽,利用视频流的平均速率估算视频业务的传输带宽。其次,光线路终端OLT给各个ONU-AP分配带宽,OLT根据语音业务速率和当前视频业务流量分别估算语音、视频业务在下一个轮询周期的带宽,并将剩余带宽在重负载终端中二次分配,最后给一般数据业务分配带宽。通过仿真实验,结果表明:与传统算法相比,该算法的网络时延和丢包率明显降低,实现了带宽资源的合理分配。     

基于业务安全策略需求的路由模型研究

首先对业务进行分类,不同的业务对网络不同的要求使其具有不同的QoS参数约束.然后研究并提出了基于智能业务识别的QoS路由模型和路由结构,根据动态配置的安全/QoS策略,在业务识别的基础上,标志数据包,根据DiffServ代码点DSCP值选择合适的路由算法.并针对带宽-时延-时延抖动-丢包率限制路由提出了一种改进的启发式路由算法,将丢包率转化为可加性条件,并把带宽限制作为剪枝条件,最后通过实验证明了其可行性.     

自相似背景流量模型下的CBR流延迟抖动分析

实时业务是网络中快速增长的业务类型,但网络中需要传送多种业务的混合流量。实时 业务的性能取决于分组延迟抖动。过大的分组延迟抖动将导致语音的中断,画面的停顿和跳跃。延 迟抖动主要是背景流量在边缘路由器的干扰引起的。以往的延迟抖动分析都是假设背景流量为泊松 过程,研究表明这种假设已经不符合当前网络流量的特性。本文将对自相似背景流量下的CBR流的 延迟抖动进行分析,给出其分布函数,并以仿真结果验证其与泊松流量对CBR流的不同影响。     

JitterPath: Probing Noise Resilient One-Way Delay Jitter-Based Available Bandwidth Estimation

Measurement of end-to-end available bandwidth has received considerable attention due to its potential use in improving QoS. Available bandwidth enables the sending rate to adapt to network conditions, so that packet loss, caused by congestion, can be significantly reduced before error control mechanisms are finally employed. To this end, we propose a probing noise resilient available bandwidth estimation scheme, called JitterPath, which is adaptive to both the fluid and bursty traffic models. Two key factors, one-way delay jitter and accumulated queuing delay, are both exploited to predict the type of queuing region for each packet pair. Then, the bottleneck utilization information included in the joint queuing regions is estimated and used to quantify the captured traffic ratio, which indicates the relationship between the probing rate and available bandwidth. The contributions of our method are as follows: 1) JitterPath can work without being restricted to fluid traffic models; 2) since JitterPath does not directly use the bottleneck link capacity to calculate the available bandwidth, it is feasible for use in a multihop environment with a single bottleneck; and 3) JitterPath inherently reduces the impact of probing noises under the bursty cross traffic model. Extensive simulations, Internet experiments, and comparisons with other methods were conducted to verify the effectiveness of our method under both single-hop and multihop environments     

Reservation arbitrated access — a bandwidth on demand protocol for multiplexing variable bit rate isochronous traffic over dual bus metropolitan area networks

With the advent of multimedia communication services, transport of real-time traffic over metropolitan area networks (MANs) is becoming an important problem. We present a novel reservation arbitrated (RA) access protocol for multiplexing variable bit rate isochronous (VBRI) traffic such as packet voice and video over dual bus MANs in general and IEEE 802.6 MANs in particular. In combination with a cyclic release mechanism, RA access allows variable bit rate traffic sources (VBRSs) to capture and reserve some isochronous channels on a bandwidth on demand basis in a round robin fashion. For a reasonable bus length suitable for metropolitan coverage, it is possible to select operation parameters which enable contention free access in the reservation process. Bandwidth utilization can be further improved by employing a movable boundary option to efficiently integrate VBRI traffic with other traffic. System performances including packet loss ratio, packet delay, delay jitter, probability distribution of consecutive packet loss and channel utilization are analyzed by both theoretical computations and computer simulations for voice, video conference and motion video traffic. Results indicate that the protocol is fair and provides a nearly isochronous transport service while ensuring efficient bandwidth utilization, yielding substantial capacity improvements over pre-arbitrated (PA) access. Compared to queue-arbitrated (QA) access, RA access not only provides variable bit rate isochronous channels but also allows VBRSs to adapt to the reserved bandwidth during network congestion so that performance degradation can be minimized. RA access complements existing PA and QA access methods in 802.6 MANs to provide a complete traffic transport solution for all types of BISDN services.     

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.     

软件定义网络中基于分段路由的多路径调度算法

针对当前软件定义网络(SDN)在应对大量数据流时造成的流表利用率低、转发响应较慢以及当前网络调度算法容易造成网络局部拥塞和负载不均衡等问题,提出一种基于分段路由的多路径调度算法SRMF。首先,SDN控制器根据网络拓扑连接情况下发初始流表;综合考虑网络链路剩余带宽、丢包率和数据流估测带宽需求进行路径权重计算;最后,根据路径权重选择最优路径并构造分段流表下发到边缘交换机。实验结果表明分段路由转发技术在多种网络拓扑下较一般转发技术在流表项开销方面有明显优势,SRMF算法与Hedera、ECMP相比,在业务流端到端时延、端到端时延抖动、网络吞吐率、丢包率等方面有一定的优势。     

基于SDN与NDN的卫星网络多约束路由算法

针对NDN卫星网络内容传输时延高、丢包率高且请求命中率低的问题,提出了一种基于SDN与NDN的卫星网络多约束路由算法,并命名为SNMcRA。基于SDN的集中控制与全局视图,通过建立多约束路由模型,将链路多约束信息与蚁群算法相结合以求解满足时延、带宽、丢包率多约束的代价最小路径,由节点在包转发的过程中动态完成转发表FIB和待定请求表PIT的构建。实验结果表明,该算法与DSP算法相比时延降低了35%,带宽利用率提升了29%,丢包率降低了17%,并且在请求命中率方面也具有显著优势。     

改进的实时流媒体中流量自适应控制机制

为了提高实时多媒体通信的服务质量,在综合考虑网络延迟和网络抖动对实时流媒体的影响下,定义了基于RTT的综合标志量。在此基础上,提出了一种改进的实时流量自适应控制机制。仿真结果表明,与基于丢包率和仅考虑延迟的RTT算法相比,该机制有效提高了数据流的平稳性和带宽的利用率,有更高的自适应性。     

基于平均变化率和空闲长度的丢包概率模型

针对RED队列丢包概率模型在计算丢包概率时精确性不足且未考虑网络流量的自相似性问题,提出了基于数据包入队速率平均变化率和队列空闲长度的队列丢包概率模型（DRED）,给出了相应的实现算法。DRED将网络流量状态引入到丢包概率的计算过程中,丢包概率随着网络流量状态的变化而变化,克服了RED队列丢包概率模型在平均队列长度大于队列最大阈值小于队列最大长度时直接将到达的数据包全部丢弃的弊端。实验结果表明,与RED相比,DRED丢包概率的计算更加精确,丢包率有所降低,吞吐量相对提高,端到端时延虽稍有增大,但时延抖动较小,网络的整体性能有一定提高。     

Investigation of the M2/G2/1/∞,N queue with restricted admission of priority customers and its application to HSDPA mobile systems

This paper investigates a queuing system for QoS optimization of multimedia traffic consisting of aggregated streams with diverse QoS requirements transmitted to a mobile terminal over a common downlink shared channel. The queuing system, proposed for buffer management of aggregated single-user traffic in the base station of High-Speed Downlink Packet Access (HSDPA), allows for optimum loss/delay/jitter performance for end-user multimedia traffic with delay-tolerant non-real-time streams and partially loss tolerant real-time streams. In the queuing system, the real-time stream has non-preemptive priority in service but the number of the packets in the system is restricted by a constant. The non-real-time stream has no service priority but is allowed unlimited access to the system. Both types of packets arrive in the stationary Poisson flow. Service times follow general distribution depending on the packet type. Stability condition for the model is derived. Queue length distribution for both types of customers is calculated at arbitrary epochs and service completion epochs. Loss probability for priority packets is computed. Waiting time distribution in terms of Laplace–Stieltjes transform is obtained for both types of packets. Mean waiting time and jitter are computed. Numerical examples presented demonstrate the effectiveness of the queuing system for QoS optimization of buffered end-user multimedia traffic with aggregated real-time and non-real-time streams.