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结合Markov调制泊松过程(MMPP)和混合泊松流,为突发性分组业务设计了Gamma调制泊松过程(GMPP)业务流模型。面向开源仿真器NS2的功能扩展,给出了纯泊松流、MMPP流和GMPP流的NS2扩展设计和实现方案,并经仿真实验进行了验证。分析表明GMPP更适用于实际业务流。 相似文献
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利用泊松噪声分布与图像灰度值相关这一特性,结合图像的水平集曲线对图像灰度值的刻画能力,在Bayesian-MAP框架下,提出了欧拉弹性正则与泊松似然保真的图像泊松去噪变分正则化模型.利用交替方向乘子法,将原问题转化为几个不同低阶子问题的求解.对于子问题中出现的高阶非线性项,利用滞后扩散不动点迭代进行线性化,从而得到模型的快速迭代求解算法.通过数值模拟实验,证明了当图像受不同强度泊松噪声影响时,所提出的泊松去噪方法都能够有效的抑制泊松噪声,同时具有良好的结构保持性能. 相似文献
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本文生成分形布朗运动(FGN)过程产生动态的自相似流量的业务。在NSF网络拓扑下,结合P圈算法对自相似业务的WDM网络阻塞率和泊松业务的WDM网络阻塞率进行比较。仿真结果表明,泊松业务较平稳,自相似业务波动较大,较能表现出业务量的突发性,采用基于自相似业务流量的阻塞率高于差于基于泊松业务流量的阻塞率。 相似文献
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本文分析快速分组交换中分组流的概率特性与输出排队。在输入分组流为复合泊松流的假设之下,论证了复合泊松流在分组交换过程中的叠加性、分解性、泊松性、马尔柯夫性、输出与输入的不变性等。然后,将输出分组流转换为连续时间的马尔柯夫链,分析了输出分组流的概率特性,并得到了输出排队长分布和充满缓冲器的概率。 相似文献
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根据移动因特网中基本的硬切换和改进的半软切换算法时间流程仿真研究了 2种切换在泊松和自相似流量下的切换损失率。仿真数据表明 ,在网络负载均值相同的情况下 ,对于硬切换和半软切换 ,自相似流量的切换损失率要明显低于传统泊松流量 ,并通过分析泊松和自相似流量的概率密度函数尝试给出产生这一区别的原因。 相似文献
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有限尺度汇聚流带宽规划研究 总被引:2,自引:0,他引:2
互联网分组传输具有尽力而为的服务特性,基于目前复杂的接入网结构难以满足中国下一代广播电视网中具有长相关特性实时互动流媒体流量传输的要求。该文根据统计复用框架下服务质量保障策略,分析了有限时间尺度集汇聚流媒体流量带宽规划的有效性;并根据泊松帕雷多突发过程流量模型,给出了基于链路瓶颈造成流量突发程度粗糙度的带宽规划方法。进一步分析表明汇聚流量自相似程度由具有较强粗糙度的子流决定,采用大规模接入汇聚的方式可以有效地避免由于链路瓶颈造成的突发,基于流量粗糙度的有限尺度汇聚流带宽规划方法可为下一代广播电视网的部署实施提供技术支撑。 相似文献
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First, we propose a new modeling method for superposed ATM traffic by the MMPP(2), which is a special case of the MAP(2). In this new method, we measure the mean and autocorrelation of cell interarrival times, and the histogram of the number of arrivals during measurement windows of fixed size. The MMPP(2) has interarrival times with a second-order hyper-exponential distribution with coefficient of variation cν > 1. However, superposed traffic is often observed to have cν < 1. To cover this situation, we extend the MMPP(2) to a MAP(3) by adding a new state with inter-state transition accompanied by an arrival. For the MAP(3) model, we take into account the second moment of the interarrival times. From numerical examples, we observe that both the proposed MMPP(2) and MAP(3) yields very good estimation of the cell loss ratio (CLR) for usual superpositions of voice and/or VBR video sources. However, when we have superpositions from CBR video sources together with other VBR sources, c ν. is much less than 1, and the MAP(3) outperform the MMPP(2), as expected. The proposed MAP(3) well characterizes the cell scale component as well as the burst scale component of superposed traffic streams 相似文献
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Improved loss calculations at an ATM multiplexer 总被引:1,自引:0,他引:1
In this paper we develop a simple and accurate analytical technique to determine the loss probability at an access node to an asynchronous transfer mode (ATM) network. This is an important problem from the point of view of admission control and network design. The arrival processes we analyze are the Markov-modulated Poisson process (MMPP) and the Markov-modulated fluid (MMF) process. These arrival processes have been shown to model various traffic types, such as voice, video, and still images, that are expected to be transmitted by ATM networks. Our hybrid analytical technique combines results from large buffer theories and quasi-stationary approaches to analyze the loss probability of a finite-buffer queue being fed by Markov-modulated sources such as the MMPP and MMF. Our technique is shown to be valid for both heterogeneous and homogeneous sources. We also show that capacity allocation based on the popular effective-bandwidth scheme can lead to considerable under-utilization of the network and that allocating bandwidth based on our model can improve the utilization significantly. We provide numerical results for different types of traffic and validate our model via simulations 相似文献
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A statistical multiplexer supporting a number of bursty sources is modeled as a discrete time, single server queueing system with an infinite buffer. The probability generating function (PGF) method is used to analyze the queueing behavior. The PGF method requires the determination of a large number of boundary values and, hence, the roots of the characteristic equation. An iterative algorithm to evaluate the characteristic roots is proposed. When the arrival process is a superposition of independent processes, a decomposition approach is used to reduce the state space involved in the computational algorithm. Additionally, the computational algorithm is made efficient through the establishment of conditions under which all the roots are either real or nonnegative real numbers. A set of equations to recursively compute the moments of the queue length are established. Sample applications of the computational methods to evaluate the performance of a multiplexer supporting voice and video sources, modeled by two-state Markov and L-state MMPP processes, respectively, demonstrate the viability of the proposed methods 相似文献
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该文研究在ATM虚通路带宽利用率一定的条件下,AAL2分组话音复接器性能随ATM虚通路输出速率的增加而变化的情况。得出结论:当ATM虚通路带宽利用率一定时,ATM虚通路输出速率越高,AAL2分组话音复接器的分组丢弃概率和平均分组排队时延越小。并提出了一种AAL2分组话音复接器的实现方案。该方案可以随着ATM虚通路输出速率的增加,方便地复接多个E1话音电路上的话音数据。 相似文献
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A lot of studies have been made to characterize and model sources of ATM traffic (voice, data, video) and to evaluate the performance of a multiplexer whose input is a superposition of these sources, using different methods and techniques (fluid flow, matrix-analytic, etc.). However, in order to better understand the end-to-end performance of ATM connections, characterizations and models of ATM traffic inside the network (i.e. after passage through one or more network elements) are needed. In this paper we intend to study the following problems: (i) Traffic profile of an ATM connection after being policed, in particular worst case traffic, and evaluation of the performance of the related statistical multiplexer. (ii) Traffic profile of the output of a multiplexer (characterized by means of the interdeparture time distribution and the index of dispersion for counts and the index of dispersion for interarrival times). (iii) Traffic profile of a single connection after passing a multiplexer. The aim is to obtain useful characterizations and models of ATM traffic in order to evaluate the performance and the efficiency of ATM network elements and of traffic control functions.This work was supported in part by the Commission of the European Communities, under project RACE R2024 (Research and Development on Advanced Communications in Europe) on Broadband Access Facilities. 相似文献
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Ganesh Babu T.V.J. Le-Ngoc T. Hayes J.F. 《Selected Areas in Communications, IEEE Journal on》2001,19(2):355-369
The performance of a priority-based dynamic capacity allocation suitable for wireless ATM systems is presented. The scheduling of ATM cell transmission in each uplink TDMA frame is based on a priority scheme with priority given to real-time traffic over nonreal-time traffic. Real-time traffic exceeding the uplink capacity is lost while nonreal-time traffic that cannot be served is stored in a first-in first-out (FIFO) queue. An analytical model is developed to evaluate the cell loss ratio (CLR) of both real-time and nonreal-time traffic. Aggregate voice, video, and data traffic is modeled by three two-state Markov-modulated Poisson processes (MMPPs). Analytical results for different system capacities and various traffic loads and scenarios are discussed. Simulation results with on-off sources and approximating MMPP sources are also presented 相似文献
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本文根据分组话音业务的特点,结合分组话音业务服务质量的要求,特别是分组丢弃概率和平均分组排队时延的要求,研究AAL2分组话音复接器带宽分配算法.得出结论:对于无比特丢弃的AAL2分组话音复接器,按平均速率分配带宽基本上可以满足分组话音业务服务质量的要求;如果适当降低ATM VC的带宽利用率ρ(例如:令ρ=0.9),则可以进一步提高话音质量,获得令人满意的话音;对于带比特丢弃的AAL2分组话音复接器,按平均速率分配带宽,可以很好地满足分组话音业务服务质量要求,获得较高质量的话音.计算机仿真证实了上述结论是正确的. 相似文献
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Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described 相似文献
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In this paper we study the performance of ATM multiplexing of homogeneous MPEG video sources. A source scheduling method is developed to improve the performance of ATM multiplexer for MPEG video sources. Simulation results show that the level of burstiness for the aggregated MPEG traffic is reduced and the network performance is enhanced. Based on the rationale of the source scheduling method, a simple but efficient bandwidth allocation algorithm is also derived for connection admission of MPEG video in an ATM multiplexer. Copyright © 2001 John Wiley & Sons, Ltd. 相似文献