Trafic d’un nombre limité de sources mélangeant attente et perte

We consider a multiserver queueing system carrying traffic from a finite-size population (Engset model). When faced with congestion, some of the requests vanish (lost calls), while the others are allowed to wait for a free server, resulting in a mixed (waiting + loss) system. Grade of service requirements lead to calculate loss probability and waiting time distribution — since standards usually upperbound waiting times.     

Individual Traffic Characteristics Queueing Systems with Multiple Poisson and Overflow Inputs

A theoretical method for analyzing overflow problems in queueing systems is presented. An interrupted Poisson process (IPP) approximation of overflow traffic is employed. An overflow stream is replaced by an IPP using the three-moment matching method. For a three-input model, to which one Poisson and two IPP streams are offered simultaneously, explicit and iterative formulas are derived to calculate the mean waiting time, overflow probability, and moments of overflow traffic intensity from the system for each of the three input streams. This three-input model is a general one, and can be used for analyzing complex problems such as multistage overflow models and individual traffic characteristics for a model with more than three inputs. By setting the capacity of the waiting room of the three-input model to 0, this method can cover loss systems. For both queueing and loss systems, several numerical examples of typical traffic models are shown. Comparisons are made between theoretical values and experimental values by computer simulations, and it is demonstrated that the accuracy of the method is excellent.     

A queuing system approach for the design of coast guard vessel traffic services communications

In order to provide for the safe and expeditious passage of maritime traffic in congested waters, the U.S. Coast Guard is authorized by the Ports and Waterways Safety Act of 1972 to establish, operate, and maintain Vessel Traffic Services (VTS) where needed. In larger areas, a VTS will generally require a communications system to enable the vessel traffic center and the participating vessels to exchange information. In designing such a system, it is necessary to assess the expected communications loading in order to determine frequency requirements and evaluate alternative configurations for the system. Here, VTS communications are viewed as a queuing system. The customers (messages) arrive at the service facility (communications channel) according to some probabilistic process, and are then serviced (transmitted) according to some other probabilistic process. Queues or waiting lines form as arriving messages wait to be transmitted, because the communications channels are busy. Three classes of messages are considered in the arrival process: check in/check out (basic VTS) messages; Vessel Movement Reporting System (VMRS) messages; and bridge-to-bridge messages. Each class is characterized by an independent Poisson distribution, and the resultant arrival process is a well-defined nonhomogeneous Poisson process. The service time is characterized by a general distribution with a known mean and variance. The queuing results, which are developed, include the utilization factor, the expected waiting time, and the expected number of messages waiting to be transmitted. The arrival process and the queuing results vary according to the time of the day, reflecting the varying traffic load throughout the day. A detailed example is given for a preliminary analysis of New York Harbor VTS communications.     

An optimal policy for competitive processing of high- and low-priority arrivals

We consider two mutually independent traffic streams whose arrivals are stored in an infinite capacity buffer. The two arrival traffics are modelled by renewal processes and are competing for service by a single processor, where one of the processes requires a strict upper bound on the total delay per arrival. It is assumed that the processing per arrival, for both processes, is a constant; the value of this constant is different for each of the two traffic streams. Given the above model, we consider policies that operate in a non-interruptive mode. We first find the optimal scheduling policy, which attains throughput one and minimizes the expected delays of both traffic streams. The delays induced by the latter policy are analysed via a methodology based on the regenerative theorem. The theoretical results are supported by numerical examples.     

DYNAMIC BANDWIDTH ALLOCATION IN AN ATM MULTIPLEXER WITH MULTICLASS TRAFFIC AND ITS PERFORMANCE ANALYSIS

This paper introduces two source models: MAP(Markovian arrival process) model for the traffic with correlation and burst, e.g., voice, video, etc. and PAP(Poisson arrival process) model for the traffic with non-correlation, such as data, etc. Then a movable boundary bandwidth access policy is chosen.Basing on above model, the performance measures, e.g., mean waiting time and loss probability,especially the queue length time distribution are obtained. Finally, a number of numerical results are provided and shown through simulation.     

Using object multiplex technique in data broadcast on digital CATV channel

Data broadcast is a new kind of value-add service of DTV broadcasting and some data broadcast protocols have already been established. However, these protocols only describe the method for locating files in data streams, and a method for distribution of a large collection of files in one or more data streams is not provided. Research on this problem mainly focuses on how to decrease the wait time and some methods of allocating files on multiple streams based on access probability are proposed, but how to assign the file with a reasonable bandwidth is ignored. In this paper, we introduced an object multiplex algorithm to optimize the allocation of objects on a DTV channel. This method assigns different bandwidth statistically to a different object according to its size and access probability. In this method, both download time and wait time are considered. It adopts a modified virtual clock (VC) scheduling algorithm to multiplex files accurately and smoothly.     

Approximate performance analysis of real-time traffic over heavilyloaded networks with timed token protocols

This paper studies timed token protocols with respect to real-time packet traffic in local area networks (LANs), such as FDDI and token bus, employed in distributed control systems. Typically, in such systems, three classes of packet traffic are encountered. The first class consists of packets cyclically generated by data acquisition tasks. The second traffic class is represented by packets generated in a random manner by control tasks and sporadic events. Finally, the third traffic class represents nonreal-time packet streams such as, for example, file transfers. To evaluate protocol performance, three performance measures are taken into account with respect to randomly generated real-time traffic: the mean waiting time, the blocking probability, and the probability that accepted packets will wait for service no longer than a specified time limit. In order to determine the last performance measure, a two-moment approximation of the waiting time distribution is applied. All three performance measures are evaluated at the beginning of the heavy network load region. Two examples of numerical calculations compared with computer simulations done for FDDI-II and token bus networks are given     

Aggregate bandwidth estimation in stored video distribution systems

Multimedia applications like video on demand, distance learning, internet video broadcast, etc. will play a fundamental role in future broadband networks. A common aspect of such applications is the transmission of video streams that require a sustained relatively high bandwidth with stringent requirements of quality of service. In this paper various original algorithms for evaluating, in a video distribution system, a statistical estimation of aggregate bandwidth needed by a given number of smoothed video streams are proposed and discussed. The variable bit rate traffic generated by each video stream is characterized by its marginal distribution and by conditional probabilities between rates of temporary closed streams. The developed iterative algorithms evaluate an upper and lower bound of needed bandwidth for guaranteeing a given loss probability. The obtained results are compared with simulations and with other results, based on similar assumptions, already presented in the literature. Some considerations on the developed algorithms are made, in order to evaluate the effectiveness of the proposed methods. Copyright © 2002 John Wiley & Sons, Ltd.     

An efficient modeling and dimensioning approach for integrated streaming and elastic traffic

Recent studies show that both data traffic and real-time traffic grow very fast in wired and wireless networks. To provide better performance guarantee, these applications need efficient network modeling and planning. In this paper, the problem where the total bandwidth of a link is shared by streaming traffic (real time traffic such as voice or video etc.) and elastic traffic (such as data) is studied. Integrating streaming traffic and elastic traffic presents a unique dimensioning problem. This paper considers dimensioning a link to satisfy both quality of service (QoS) requirements for streaming traffic, such as loss probability, and elastic traffic, such as mean waiting (delay) time. The Erlang loss model is applied to streaming traffic and a bursty traffic model is applied to the elastic traffic. Efficient dimensioning algorithms based on classical Markovian models and time-scale decomposition are then proposed. Numerical results show that the proposed methods have good accuracy.     

Convexity in queues with general inputs

In this correspondence, we develop fundamental convexity properties of unfinished work and packet waiting time in a queue serving general stochastic traffic. The queue input consists of an uncontrollable background process and a rate-controllable input stream. We show that any moment of unfinished work is a convex function of the controllable input rate. The convexity properties are then extended to address the problem of optimally routing arbitrary input streams over a collection of K queues in parallel with different (possibly time-varying) server rates (/spl mu//sub 1/(t),...,/spl mu//sub K/(t)). Our convexity results hold for stream-based routing (where individual packet streams must be routed to the same queue) as well as for packet-based routing where each packet is routed to a queue by probabilistic splitting. Our analysis uses a novel technique that combines sample path observations with stochastic equivalence relationships.     

Policing of realistic VBR video traffic in an ATM network

ATM has been accepted by CCITT as the transport mechanism for the future BISDN and will also be widely used in future customer premises networks. Networks based on the ATM principle are expected to provide a very flexible communications infrastructure allowing customers to make effective use of a wide variety of offered services. To provide this flexibility with an acceptable quality of service while operating the network in an economic way, elaborate traffic management functions will be necessary to control the traffic flows within the network. This paper will study one of these functions—the so-called ‘usage parameter control’ or ‘policing’ function—in some detail to illustrate some of the problems that arise and point out possible solutions. The mechanisms chosen to implement the policing function will be the ‘leaky bucket’ mechanism, the ‘jumping window’ mechanism and the ‘moving window’ mechanism. The input streams used to assess the mechanisms represent different types of video communication—videophone, video conference and entertainment video—coded according to different variable bit-rate (VBR) algorithms. In contrast to most of the previous studies, where artificial, statistical traffic sources have been used, the sources used in this paper are directly based on measured ‘real-life’ video data. This ensures that all the statistical properties of the actual traffic stream are preserved and allows identification of the different factors that influence the dimensioning and the performance of the policing mechanism. The results of this study show that the uncertainty about the key parameters at call set-up and the considerable impact of single scenes make the proper dimensioning of policing mechanisms difficult. Furthermore, it seems not to be practical to use the long term mean bit-rate as the key traffic control parameter for these sources. Results indicating that the long-term cell loss ratio is not a sufficient measure for the quality of service are also presented. A comparison of the mechanisms shows that from a performance perspective, the ‘leaky bucket’ mechanism is superior to the two window mechanisms. This work is relevant to evolving standards for both BISDN traffic management and variable bit-rate video coding.     

Modeling and Simulation of Mixed Queueing and Loss Systems

In this paper, we investigate a multi-rate network in which wide-band calls are allowed to wait if insufficient resources are available at the time of the call arrival. On the link level, an analytical model is presented and simulations have been carried out on the network level. The results indicate that allowing a few wide-band calls to queue can give a significant improvement in performance in terms of network revenue , as well as a means to level out the blocking probabilities of the different traffic classes. This improvement becomes significant when the service discipline of the waiting calls (of different bandwidth requirements) is adaptive in the sense that longer queues get served first. This observation motivates the investigation of the impact of various buffer space assignment and queueing disciplines on network revenue and call blocking probabilities. The study of such mixed delay and queueing networks is motivated by its possible applications to traffic problems in future Broadband Integrated Services Digital Networks as well as in multi-rate cellular radio networks.     

The extended connection‐dependent threshold model for call‐level performance analysis of multi‐rate loss systems under the bandwidth reservation policy

Firstly, we reviewed two extensions of the Erlang multi‐rate loss model, whereby we can assess the call‐level QoS of telecom networks supporting elastic traffic: (i) the extended Erlang multi‐rate loss model, where random arriving calls of certain bandwidth requirements at call setup can tolerate bandwidth compression while in service; and (ii) the connection‐dependent threshold model, where arriving calls may have several contingency bandwidth requirements, whereas in‐service calls cannot tolerate bandwidth compression. Secondly, we proposed a new model, the extended connection‐dependent threshold model. Calls may have alternative bandwidth requirements at call setup and can tolerate bandwidth compression while in service. We proposed a recurrent formula for the efficient calculation of link occupancy distribution and consequently call blocking probabilities, link utilization, and throughput per service class. Furthermore, in the proposed model, we incorporated the bandwidth reservation policy, whereby we can (i) equalize the call blocking probabilities of different service classes, (ii) guarantee specific QoS per service class, and (iii) implement different maximum bandwidth compression/expansion rate per service class so that the network supports both elastic and stream traffic. The accuracy of the new model is verified by simulation. Moreover, the proposed model performs better than the existing models. Finally, we generalize the proposed model by incorporating service classes with either random or quasi‐random arrivals. Copyright © 2011 John Wiley & Sons, Ltd.     

Bandwidth Allocation Strategies in Wide-Band Integrated Networks

A common digital transmission facility in a wide-band integrated service digital network (ISDN) provides shared access to a community of heterogeneous users. Traffic demands from these users vary in their arrival rate, their service time, and their bit rate. In order for this type of communication system to handle its traffic demands with high efficiency and flexibility, a close control of access to the shared bandwidth is required. We model the system by a general multiserver queueing system where customers demand service from a random number of servers. If no waiting is allowed, this queueing model is readily analyzed, and various server allocation strategies can be studied. If the various access requests are queued for service, then the system calls for efficient strategies for allocating servers to waiting customers. In this case, exact analysis of the underlying queueing model becomes quite difficult. For this case, we present some analytic and simulation results of the performance of the system under several server allocation policies.     

Méthode à trois moments fondée sur un processus de Cox pour analyser le trafic dans un réseau téléphonique

The authors analyse the stochastic behaviour of a trunk group offered several streams of non-Poissonian traffic. Offered streams are modelled using Cox functions and we show how to pass from this representation to a classical representation in terms of the moments of the traffic process. Using the Laplace transforms of the inter arrival time distributions of the different traffic streams, we obtain exact values for the moments of the overflow and carried streams, even when more than one traffic stream is offered to the trunk group.     

修正开关泊松过程（RSPP）和RSPP／M／1排队

在通信网互连中，若被连子网具有不同的最大允许分组长度，那么有信关中一个较长的分组就可能要被拆分为多个较小的分组，这就是公组再分问题，已经证明，在某些情况下。再分后的公组流可以用一个修正的开关泊松过程来，本文ＲＳＰＰ和ＲＳＰＰ／Ｍ／１排队。文中推导出了ＲＳＰＰ到达间隔分布的表达式，并给出了平均到达率。文中还给出了队长分布，平均等候时间的表达式；信关输出流的特性对于全网的性能分析是必需的，因此本文着重     

FEBA: A Bandwidth Allocation Algorithm for Service Differentiation in IEEE 802.16 Mesh Networks

In wireless mesh networks, the end-to-end throughput of traffic flows depends on the path length, i.e., the higher the number of hops, the lower becomes the throughput. In this paper, a fair end-to-end bandwidth allocation (FEBA) algorithm is introduced to solve this problem. FEBA is implemented at the medium access control (MAC) layer of single-radio, multiple channels IEEE 802.16 mesh nodes, operated in a distributed coordinated scheduling mode. FEBA negotiates bandwidth among neighbors to assign a fair share proportional to a specified weight to each end-to-end traffic flow. This way traffic flows are served in a differentiated manner, with higher priority traffic flows being allocated more bandwidth on the average than the lower priority traffic flows. In fact, a node requests/grants bandwidth from/to its neighbors in a round-robin fashion where the amount of service depends on both the load on its different links and the priority of currently active traffic flows. If multiple channels are available, they are all shared evenly in order to increase the network capacity due to frequency reuse. The performance of FEBA is evaluated by extensive simulations. It is shown that wireless resources are shared fairly among best-effort traffic flows, while multimedia streams are provided with a differentiated service that enables quality of service.     

Buffer-size and waiting-time calculations for random arrivals and periodically regular service

The problem of choosing the buffer size for a given waiting time or fractional loss in a situation where the input rate is Poissonian and the output rate is periodically regular is considered. The value of the results lies in that they give a practical way to a compromise between buffer fractional loss and buffer waiting time in those applications where both quantities are of interest. The periodically regular service-time distribution allows for more than one stream of queuing customers to be served by only one servicing channel.     

A new satellite communication bandwidth allocation combined services model and network performance optimization

With the rapid development of online to offline economy, new services compositions would take up a great part in the satellite communication. More and more new services compositions request more bandwidth and network resources, which lead to serious traffic congestion and low channel utilization. Suffering from isolated link connection and changeable delay under the satellite environment, current bandwidth allocation schemes could not satisfy with the demand of low delay and high assess rate for new satellite services. This paper focuses on bandwidth allocation method for satellite communication services compositions. The novel models of services compositions with single‐hop Poisson distribution are designed to simulate original traffic arrival. Isolated independent coefficients take an original distribution to adapt to isolated disconnections. Services queue waiting time would be judged by acceptable delay threshold. Models provide new services compositions with more precise arrival distributions. In order to improve traffic congestion, the method combined services models, and a network performance is proposed. Optimal reserved bandwidth is set according to the priority and arrival distribution of different services compositions, which classify services with feedback transmission performance. We design minimum fuzzy delay tolerant intervals to calculate delay tolerant threshold, which adapt random delay changes in the services network with delay tolerant features. The simulation in OPNET demonstrates that the proposed method has a better performance of queuing delay by 16.3%, end‐to‐end delay by 18.7%, and bandwidth utilization by 13.2%. Copyright © 2016 John Wiley & Sons, Ltd.