基于多条件约束的网络信息传输路径动态选择

针对传统网络信息传输路径动态选择速率低的问题,提出基于多条件约束的网络信息传输路径动态选择。笔者在原有技术基础上增加传输路径时延和丢失率两个参数条件,约束传输路径动态选择,建立新的传输路径动态选择模型,利用Qos算法将模型通过线性整数规划方式对其求解。经实例验证,该方法能够快速地选择出距离短、时延最低且丢失率较低。     

一种片上网络的低延迟容错算法

为解决片上网络容错问题,利用端到端模式设计一种低延迟可靠传输算法。该算法利用发送端主动发送冗余数据包获得较小的延迟,将数据分成包集以提高链路利用率,进一步降低延迟。发送端只在收到接收端对当前包集的确认后才发送下一个包集的数据,由此保证高可靠性。通过不断发送数据包及端到端的反馈保证传输的正确性。NS-2仿真实验结果证明,该算法延迟低,片上通信可靠性高,可以有效处理传输过程中的瞬时错误。     

蓝牙Mesh网络AODV路由改进算法

蓝牙Mesh网络使用泛洪进行多跳通信,在没有路由机制的情况下,由于消息的连续广播,原有泛洪机制会导致网络开销增大和通信延迟。本文基于能量有效的AODV改进算法E-AODV使用MATLAB进行仿真,根据跳数、节点剩余能量、链路质量来选择最优节点进行数据包的转发。仿真结果表明,E-AODV算法可减小蓝牙Mesh网络中RREQ数据包传输数量,通过与传统泛洪、AODV算法比较,该算法能够有效降低数据包传输时延,降低网络能耗,提高网络性能。     

具有数据包丢失的奇异网络控制系统指数稳定性

考虑存在时延和数据包丢失的情况,研究了奇异被控对象的网络控制系统建模与指数稳定性问题.当时延不大于一个采样周期且数据包丢失率一定时,将正则、无脉冲的奇异网络控制系统建模为数据包丢失率约束的异步动态切换系统,给出了状态反馈和动态输出反馈的统一数学模型;推导出数据包丢失率约束的系统指数稳定的充分条件,给出了使系统指数稳定的最大允许数据包丢失率,仿真结果表明了该方法的有效性和可行性.     

网络相关媒体流缓存管理策略NRC

Internet网络上媒体流的应用受限于应用时的网络状况，如时延、包丢失率等．本文研究通过在网络边缘处设置缓存代理来减小这些影响，提出一种新的缓存管理算法NRC，即接入媒体流服务时，用户以两种方式获取媒体流对象：一部分对象内容从代理缓存中获取，而另一部分对象内容则直接从流媒体源服务器处传输而来；从而加速媒体流接入服务，提高媒体流服务质量,算法同网络特性和媒体流特性相关．最后仿真实验证实同网络和流媒体特性相关的缓存管理算法NRC可以很好地减少服务延迟和提高媒体流的总体服务质量．     

一种异构片上网络路由算法的研究

随着集成电路工艺的迅速发展,传统的片上网络由于缓存引起芯片面积开销和能耗增加,从而使得无缓存路由技术得到了广泛关注。通过消除缓存, 整体的流水线进程大大得到简化,性能得到提高。但当网络负载量较大时,数据包被多次偏转或误传,导致网络的延迟增加,系统健壮性较差。针对片上网络运行应用的多样性,异构网络作为一种相对灵活的网络结构,能有效地降低网络的传输时延,提高系统性能。文中设计了无缓存NoC和带缓存NoC两种路由方式相结合的异构片上网络,并匹配静态路由算法和动态的自适应路由算法(AFC)进行数据包的传输。同时,还提出了一种针对AFC的优化算法(AFC-LP),其通过对无缓存路由计算的二次仲裁,进一步降低了通信的平均时延,提高了网络性能。实验表明,AFC-LP算法相比于传统带缓存的维序X-Y路由算法,片上网络的平均延迟降低了28.4%,CPU每一时钟周期内所执行的指令数IPC(Instruction Per Cycle)提升了10.4%。     

基于滑动缓存的WSN可靠协议及其能耗分析

数据如何以较低的能耗进行可靠传输是无线传感器网络中数据采集亟需解决的问题。基于此,提出一种缓存位置滑动调整的可靠传输协议,该协议基于通信的中间节点缓存数据包,按照通信距离将链路上的节点划分为近源节点和近汇聚节点,根据链路质量动态调整缓存数据包的节点区域,数据包在相应节点区域的缓存呈正态分布。NS2仿真结果与基于马尔科夫链能耗分析结果表明,该协议中数据包的传输时延小于HHRA协议,吞吐量有较大程度的提高。     

基于路径选择的车辆网络路由算法

车辆网络作为一个新兴的研究领域,受到广泛的关注.已有的车辆网络路由协议通常选择一条路径来获得最大的传输成功率,而不考虑传输延迟.本文提出一种算法来找到一条路径转发数据包,目标是得到最大的传输成功率,并且传输时延要控制在合理的范围内.详尽的实验证明该方法是有效的,结果显示该算法能获得最大的传输成功率和合理的传榆时延.     

有时延和数据包丢失的网络控制系统控制器设计

在同时考虑网络诱导时延和数据包丢失的基础上,研究了动态输出反馈网络控制系统指数稳定性和控制器设计问题.基于一定的数据包丢失率和不大于一个采样周期的时延,将系统建模为结构事件率约束的异步动态系统.利用线性矩阵不等式方法推导出网络接通率约束的系统指数稳定的充要条件,给出了确保系统稳定的控制器设计方法.Matlab数值算例说明了研究结果是有效可行的.     

TinyOS2.x下基于蚁群算法的WSNs路由协议设计

为了增强无线传感器网络的动态适应性和实现数据包的多路径传输,根据蚁群算法的原理,使用NesC语言在TinyOS2.x下设计了路由协议Ant-PDRP.该协议采用信息素浓度指引路由包和数据包传输,并在数据包传输过程中引入惩罚机制以实现动态均衡网络能耗.TOSSIM仿真和Micaz节点的真实实验表明,改进后的路由协议能够有效减少传输时延,延长网络寿命,保证数据可靠传输.     

Packet audio playout delay adjustment: performance bounds and algorithms

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays. In this paper, we consider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive “loss” due to the arrival of packets at the receiver after their playout time has already passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This information, together with a “delay spike” detection algorithm based on (but extending) our earlier work, is used to dynamically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.     

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.     

Adaptive playout scheduling and loss concealment for voice communication over IP networks

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.     

Delay reduction techniques for playout buffering

Receiver synchronization of continuous media streams is required to deal with delay differences and variations resulting from delivery over packet networks such as the Internet. This function is commonly provided using per-stream playout buffers which introduce additional delay in order to produce a playout schedule which meets the synchronization requirements. Packets which arrive after their scheduled playout time are considered late and are discarded. In this paper, we present the Concord algorithm, which provides a delay-sensitive solution for playout buffering. It records historical information and uses it to make short-term predictions about network delay with the aim of not reacting too quickly to short-lived delay variations. This allows an application-controlled tradeoff of packet lateness against buffering delay, suitable for applications which demand low delay but can tolerate or conceal a small amount of late packets. We present a selection of results from an extensive evaluation of Concord using Internet traffic traces. We explore the use of aging techniques to improve the effectiveness of the historical information and hence, the delay predictions. The results show that Concord can produce significant reductions in buffering delay and delay variations at the expense of packet lateness values of less than 1%     

Design and Experimental Evaluation of an Adaptive Playout Delay Control Mechanism for Packetized Audio for Use over the Internet

We describe the design and the experimental evaluation of a playout delay control mechanism we have developed in order to support unicast, voice-based audio communications over the Internet. The proposed mechanism was designed to dynamically adjust the talkspurt playout delays to the traffic conditions of the underlying network without assuming either the existence of an external mechanism for maintaining an accurate clock synchronization between the sender and the receiver during the audio communication, or a specific distribution of the audio packet transmission delays. Performance figures derived from several experiments are reported that illustrate the adequacy of the proposed mechanism in dynamically adjusting the audio packet playout delay to the network traffic conditions while maintaining a small percentage of packet loss.     

Adaptive playout scheduling algorithm tailored for real-time packet-based voice conversations over wireless ad-hoc networks

The effective provision of real-time, packet-based voice conversations over multi-hop wireless ad-hoc networks faces several stringent constraints not found in conventional packet-based networks. Indeed, MANETs (mobile ad-hoc networks) are characterized by mobility of all nodes, bandwidth-limited channel, unreliable wireless transmission medium, etc. This environment will surely induce a high delay variation and packet loss rate impairing dramatically the user experienced quality of conversational services such as VoIP. Indeed, such services require the reception of each media unit before its deadline to guarantee a synchronous playback process. This requirement is typically achieved by artificially delaying received packets inside a de-jitter buffer. To enhance the perceptual quality the buffering delay should be adjusted dynamically throughout the vocal conversation.In this work, we describe the design of a playout algorithm tailored for real-time, packet-based voice conversations delivered over multi-hop wireless ad-hoc networks. The designed playout algorithm, which is denoted MAPA (mobility aware playout algorithm), adjusts the playout delay according to node mobility, which characterizes mobile ad-hoc networks, and talk-spurt, which is an intrinsic feature of voice signals. The detection of mobility is done in service passively at the receiver using several metrics gathered at the application layer. The perceptual quality is estimated using an augmented assessment approach relying on the ITU-T E-Model paradigm while including the time varying impairments observed by users throughout a packet-based voice conversation. Simulation results show that the tailored playout algorithm significantly outperforms conventional playout algorithms, specifically over a MANET with a high degree of mobility.     

Neural and fuzzy computation techniques for playout delay adaptation in VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays.     

基于节点网络深度的缓冲延时算法研究

基于ZigBee技术的语音导游系统,语音数据包在游客接收端会产生失序问题。为了消除这种语音抖动,提出了一种基于游客节点网络深度的缓冲延时算法,采用E-Model语音预测模型,用客观的语音预测值去表示主观的MOS（Mean Opinion Score,平均意见值）评分值,通过计算最高MOS值得到对应的最优延时变量。通过简化算法并软件仿真,可以看出该算法在语音导游系统中的优越性。     

Formal Performance Modelling and Evaluation of an Adaptive Mechanism for Packetised Audio over the Internet

A case study is presented which concerns the design of an adaptive mechanism for packetised audio for use over the Internet. During the design process, the audio mechanism was modelled with the stochastically timed process algebra EMPA and analysed via simulation by the EMPA based software tool TwoTowers in order to predict the percentage of packets that are received in time for being played out. The predicted performance figures obtained from the algebraic model illustrated in advance the adequacy of the approach adopted in the design of the audio playout delay control mechanism. Based on these performance figures, it was possible to implement and develop the complete mechanism without incurring additional costs due to the late discovery of unexpected errors or inefficiency. Performance results obtained from experiments conducted on the field confirmed the predictive simulative results. Received March 1997 / Accepted in revised form July 1998     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。