基于队列约束的多层无线网状网络容量研究

在常规和随机多层无线网络中,有研究已经得到了网络可以达到的吞吐量数量级,然而对单个节点吞吐量和端到端时延的研究却很少.为了解决这一问题,对多跳常规多层无线网状网络中的单节点最大吞吐量和端到端时延进行了研究.推导了网络中的分组吸收概率;根据网络的排队模型,在这个分组吸收概率的基础上,使用扩散近似法得到了单节点可达吞吐量和端到端时延.仿真分析了业务模式和网络拓扑对端到端时延和单节点最大吞吐量的影响,通过仿真结果可以发现使网络处理大量节点的最优辅助节点数,这有助于优化网络资源分配,且减少网络的拥塞.     

Ad hoc网络中的自适应退避算法的研究

本文提出了一种适用于高等级节点的二进制负指数退避算法BNEB,验证了竞争窗口平均值较小的节点信道竞争能力较强的结论,并根据此结论,针对多跳Ad hoc网络中由于MAC层竞争导致的拥塞问题提出了两种具有拥塞控制功能的退避算法RBAB和CABEB,在节点发生拥塞时调整其分组进入速率和转发速率,能够提高网络的端到端吞吐量并有效缓解网络拥塞。     

基于最低开销模型的传感网络通信优化仿真

当传感网络中某条链路发生变化时,需要重新计算最短路径树,一旦传感网络规模较大,传统的算法采用抑制链路改变的方法提高传感网络通信容量,但这大幅抑制通信节点周期内路径选择灵活性,通信延迟明显.提出一种改进的A-OSPF算法并应用到传感网络通信优化中,该算法在原始的OSPF基础上融人了最低开销节点机制,增强了传感网络中节点构建的概率,考虑了节点移动性,将更加平稳的链路当成节点,按照链路代价原理得到源节点到目标节点的最佳路径,确保数据包可在链路质量最高的路径上进行传递,降低传感网络数据传送的平均端到端延时.仿真结果表明改进算法在传感网络生存周期以及平均端到端延时方法优于原始的OSPF算法,实现了延长传感网络生存周期以及能量均衡的目标.     

一种速率自调节可用带宽测量算法

可用带宽是网络路由、网络服务质量、流量工程等方面的一个关键参数。目前很多研究方法都基于PGM模型和PRM模型,但这两种方法大都假设背景流量速率为固定比特流,不适用于低带宽的测试。提出一种端值自调节可用带宽测量算法,该算法充分考虑了低链路带宽的情况。通过对排队延时的处理、探测分组列速率端值自适应调节,实现了端到端可用带宽快速准确的测量。实验结果表明,该算法具有良好的测量效果,尤其在低带宽条件下较其它同类算法提高了测量准确性,加快了测量速度并减小了对网络的影响。     

基于网络演算计算保证服务端到端延迟上界

归纳总结了网络演算,阐明了网络演算的两个基本工具——进入曲线和服务曲线,得出了服务曲线存在瓶颈效应、端到端延迟的理想与近似确定性上界、提供保证服务网络节点的服务曲线需求等结论,计算了服务曲线以速率等待时间及PGPS(packetizedgeneralizedprocessorsharing)形式表示的保证服务端到端延迟确定性上界.     

基于ZigBee的森林火灾预警系统的设计与实现

设计了一个基于ZigBee森林防火移动预警系统.使用ZigBee技术构建一个数字传感器无线通信网络,该网络主要由终端节点、协调节点和发送节点构成.数字传感器采集的环境参数主要有温度、相对湿度、光照强度等,把各个不同功能的节点按要求散布在森林中采集现场环境参数,采集的数据经处理后通过无线通信网络发送至监测端和服务器控制端.该系统采用射频芯片CC2431作为传感器网络节点,中心节点和GPRS模块DTP-S05 Ci相连,借助移动通信网络将终端节点采集到的数据转发到手机终端或互联网中的服务器,手机客户端采用J2ME实现界面设计,服务器端采用MVC模式实现应用程序的开发,实现了森林火灾移动预警的功能.本系统具有结构精简、技术复杂度低、成本低、实时和准确的特点.     

吸收协议-一个选播路由协议

选播是一种很有用的通信模式．由于在复制服务器、移动IP等多个应用领域的需求．使越来越多的人关注怎样更好的实现选播．提出了一个选播路由协议一吸收协议．协议在提供最小端到端延迟路由前提下支持多路路由．以平衡网络流量．改善网络链路利用率．协议通过从选播成员开始的吸收过程．使网络中每个节点都有一条或者多条到选播地址的路由指向到该节点端到端延迟最小的选播成员．吸收协议原理简单，开销小．易于实现．不依赖其他路由协议．修改、升级不会影响其他路由协议．     

基于视频发送端速率控制的QoS技术研究

基于尽力而为的网络模式不能提供QoS保证,网络拥塞和分组丢失不可避免。在端到端视频单播结构下,论文提出了一个发送端速率控制框架SRCF,在此框架下首先利用RTCP报文中的字段提出了一种网络参数测量方法,然后设计了一个自适应速率算法SRCA,SRCA利用已得到的网络传输延迟和分组丢失率参数作为初始参数,来调整编码速率,达到充分利用带宽的目的,避免了视频质量由于调整参数带来的剧烈抖动。仿真结果表明,该算法在网络出现一定拥塞的条件下,能跟踪带宽的变化,网络和媒体QoS能保证视频质量较好。     

基于信息熵的IP网端到端行为分析与建模

具有开放、分布式、不协作、异构、无中心控制等特点的Internet复杂巨系统的管理、容量规划、新一代网络体系结构设计与分析和性能预测都离不开对网络行为的充分理解。而端到端行为作为网络行为的一个重要组成部分,具有一定的研究价值。该文利用信息熵原理建立用于分析端到端整体宏观行为的信息熵模型,该模型能很好地反映端到端整体宏观行为与链路上各节点的状态概率之间关系,根据该模型可以分析端到端链路上各节点之间的相互作用关系以及它们是如何引起端到端整体宏观行为的。最后,给出了该模型的有效性和稳定性定量分析的判别式。     

基于网络演算的无线Mesh网时延上界

无线Mesh网是一种新型宽带接入通信网络，端到端时延上界分析的准确性会直接影响网络的QoS保障。针对无线Mesh网传统时延边界分析方法没有考虑当前节点处理时延对后一个节点到达曲线的影响，造成时延上界计算不紧致的问题，利用(ρ,σ)模型作为数据流到达曲线，延迟-速率函数LR作为数据流服务曲线，同时考虑前一个节点的处理时延对后一个节点到达曲线的影响，推导出无线Mesh网中单节点、单路径和多路径传输时延上界，并利用数值分析方法得到网络服务速率、流量分配方式、多路径数目等因素对时延上界的影响。仿真结果表明，与传统方法相比，以上计算得到的无线Mesh网时延上界更接近实际仿真值。     

Universal Timestamp-Scheduling for real-time networks

Consider a network of computers interconnected by point-to-point communication channels. For each flow of packets through the network, the network reserves a fraction of the packet rate of each channel along the path of the flow. We define a family of scheduling protocols, called Universal Timestamp-Scheduling, to forward packets in this network, such that all members of the protocol family provide the same upper bound on packet delay as the well-known packet delay of Virtual Clock scheduling. The protocol family is called universal because it encompasses a wide variety of protocols. To show this, we prove that many scheduling protocols in the literature are members of the protocol family, and thus provide the above guarantee. In addition, we show that the protocols in the literature have only considered one side of the spectrum of possible scheduling protocols, and that there is another side of the spectrum that deserves attention and remains to be investigated.     

无线传感器网络机会路由协议—MORE的优化

无线传感器网络中的机会路由协议—MORE协议,即独立于MAC层的机会路由和编码协议,从对MORE协议的学习中,得知MORE协议在源节点准备发送数据包时,首先将K个数据包随机地分成批(batch)进行流内随机网络编码,再以广播的形式发送出去,这种提前设定好批的大小的方法不能满足实时的要求,对信道利用率和时延造成影响。针对MORE协议的这种弊端引入一种根据块时延的大小自主适应选择数据块大小的算法对MORE协议进行优化。     

On the modeling and optimal flow control of the Jacksonian network

The modeling and optimal flow control of a Jacksonian network in equilibrium is investigated. The model employed consists of a controller node cascaded with the Jacksonian network. Input packets arrive at the controller node with a Poissonian rate δ. For a blocking type strategy for accessing the network it is shown that the control which maximizes the average throughput of the network subject to a bounded average time delay constraint is a window flow control mechansim. The window size depends on the offered load δ, the maximum service rate of the controlling queueing system, c, and the Norton equivalent service rate of the network μ. The dependence of the average throughput and the average time delay on the control is also analyzed.     

Enhanced bulk scheduling for supporting delay sensitive streaming applications

Providing end-to-end delay guarantees for delay sensitive applications is an important packet scheduling issue with routers. In this paper, to support end-to-end delay requirements, we propose a novel network scheduling scheme, called the bulk scheduling scheme (BSS), which is built on top of existing schedulers of intermediate nodes without modifying transmission protocols on either the sender or receiver sides. By inserting special control packets, which called TED (Traffic Specification with End-to-end Deadline) packets, into packet flows at the ingress router periodically, the BSS schedulers of the intermediate nodes can dynamically allocate the necessary bandwidth to each flow to enforce the end-to-end delay, according to the information in the TED packets. The introduction of TED packets incurs less overhead than the per-packet marking approaches. Three flow bandwidth estimation methods are presented, and their performance properties are analyzed. BSS also provides a dropping policy for discarding late packets and a feedback mechanism for discovering and resolving bottlenecks. The simulation results show that BSS performs efficiently as expected.     

Throttle and preempt: A flow control policy for real-time traffic in wormhole networks

In this paper, we study wormhole routed networks and envision their suitability for real-time traffic in a priority-driven paradigm. A traditional blocking flow control in wormhole routing may lead to a priority inversion in the sense that high priority packets are blocked by low priority packets for unlimited time. This uncontrolled priority inversion causes the frequent deadline missing even at a low network load. This paper therefore proposes a new flow control called throttle and preempt flow control, where high priority packets can preempt network resources held by low priority packets, if necessary. As a result, this flow control does not cause priority inversion. Our simulations show that the throttle and preempt flow control dramatically reduces deadline miss ratio for various real-time traffic configurations without extra virtual channels. It is also observed that the throttle and preempt flow control offers shorter delay for non-real-time traffic than the existing real-time flow control does.     

HMM delay prediction technique for VoIP

This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.     

基于累积时延的流媒体传输模糊拥塞控制

为保证网络流媒体传输质量,在流媒体的传输中需要采用有效的拥塞控制策略.结合流媒体数据对时延敏感的特点,提出了一种基于累积时延的模糊拥塞控制算法,该算法在流媒体数据流传输过程中检测和跟踪其时延,在转发分组数据前,根据容忍时延阈值,丢弃超时数据包,减少不必要的带宽浪费,并且对所到达的数据流按照累积时延进行优先级分类,把全局性缓冲区和各队列的局部性缓冲区按照正常、拥塞避免和拥塞的规则划分为3个具有交叉过渡域的阶段,然后采用整体和局部相结合的拥塞控制方法,实现队列调度过程中的模糊处理,从而对网络拥塞进行有效的控制.理论分析和实验结果表明,使用基于累积时延的模糊拥塞控制算法,能有效改善流媒体的传输性能,是解决流媒体传输拥塞控制的有效途径,并能对提高网络性能起到重要作用.     

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。     

基于终端差异化与D2D网络的IDNC协作重传方案

为了满足终端用户的个性化需求并且降低D2D网络的传输时延,提出了一种基于终端差异化的立即可解网络编码（IDNC）协作重传方案。首先,针对PC-D2D网络存在的解码冲突以及传输冲突问题提出一种新的IDNC算法框架并且在此框架的基础上搜索极大独立集（MIS）,综合考虑数据包的接收情况、终端用户需求以及链路丢包率情况设计权重,衡量权重选取一次重传时延增量最小的并发协作重传终端以及数据包组合生成编码包;同时,考虑不需要数据包提供的未来解码机会,优化终端不需要的数据包,进一步降低传输时延。仿真结果表明,所提方案在满足终端个性化需求的同时能够有效地降低解码时延和完成时间。     

Impact of out-of-sequence processing on the performance of data transmission

Application Level Framing (ALF) was proposed by Clark and Tennenhouse as an important design principle for developing high performance applications. ALF relies in part on the ability of applications and protocols to process packets independently one from the other. Thus, performance gains one might expect from the use of ALF are clearly related to performance gains one might expect from applications that can handle and process packets received out-of-sequence, as compared to application that require in-sequence delivery (FTP, TELNET, etc.). In this paper, we examine how the ability to process out-of-sequence packets impacts the efficiency of data transmission. We consider both the impact of application parameters such as the time to process a packet by the application, as well as network parameters such as network transmission delay, network loss rate and flow and congestion control characteristics. The performance measure of interest are total latency, buffer requirements, and jitter. We show, using experimental and simulation results, that out-of-sequence processing is beneficial only for very limited ranges of transmission delays and application processing time. We discuss the impact of this on the architecture of communication systems dedicated to distributed multimedia applications.