El desarrollo y las avances recientes de la tecnología de los ordenadores y de la tecnología de alta velocidad de redes han hecho posible que las aplicaciones de video bajo demanda estén conectadas a "shared-computing" servidores reemplazando los sistemas tradicionales donde cada aplicación tenía su propia máquina dedicada para servirla. La aplicación de video bajo demanda permite a los usuarios seleccionar de una lista de videos su película favorita y ver su reproducción a su gusto.Sin embargo, la aplicación de video bajo demanda se considera como una de las aplicaciones que debería soportar largos "video streams", que consumen muchos recursos como el anch de banda de red y I/O, a gran número de clientes. Por eso, el servidor de video debería asegurar los recursos necesarios para cada "stream" durante un periodo de tiempo largo (e.g. 7200 segundos) para que los clientes reproduzcan el video sin "jitter" y "starvation" en sus búferes.Esta tesis presenta el diseño y la implementación de un Servidor Proxy de Video (VPS) que puede proveer video bajo demanda interactiva. El VPS consiste de tres componentes (partes) principales. La primera parte es el Modulo de Control de Admisión (ACM) que recibe las peticiones de los clientes, negocia los recursos requeridos, y decide si la petición puede ser aceptada o rechazada basado en la disponibilidad de los recursos. La segunda parte es el Modulo de Manejo de los Recursos (RMM) que maneja los recursos del sistema como el CPU, la Memoria, la Red, y el Disco. Este consta de cuatro "brokers" que reservan a los recursos necesarios basado en una política predefinida. La tercera parte es el algoritmo CB_MDA "Credit_Based Media Delivery Algorithm" que controla y regula el flujo de los "streams" del video. La CB_MDA utiliza una combinación de canales unicast y "multicast" para transmitir el video. Los "streams" de "multicast" se inician para empezar a emitir el video desde el principio, mientras los canales unicast se usan para juntar los llegados tardes a un "stream multicast" apropiado. En la implementación, el CB_MDA detecta los momentos cuando el servidor tiene disponibilidad de recursos y les asigna a los usuarios apropiados para crear un trabajo en adelanto. / The recent advances and development of inexpensive computers and high speed networking technology have enabled the Video on Demand (VoD) application to connect to shared-computing servers, replacing the traditional computing environments where each application was having its own dedicated special purpose computing hardware. The VoD application enables the viewer to select, from a list of video files, his favourite video file and watch its reproduction at will.However, the VoD application is known as one of the applications that must provide long-lived video streams which consume high resources such as I/O and network bandwidth to a large number of clients. Therefore, a video server must secure the necessary resources for each stream during a long period of time (e.g. 7200 seconds) so that the clients can reproduce (play) the video data without witnessing jitter or starvation in their buffers.This thesis presents the design and implementation for a video proxy server (VPS) which can provide interactive video on demand. The VPS consists of three main parts. The first part is the Admission Control Module which receives the clients' requests, negotiates the required resources, and decides whether to accept or reject a client based on the available resources. The second part is the Resources Management Module which manages several shared resources such as the CPU, the Memory, the Network and the Disk It consists of four brokers that can reserve the necessary resources based on a predefined policy. The third part is the CB_MDA algorithm which is responsible for regulating the resources assignment and scheduling the video streams. The CB_MDA uses a combination of multicast and unicast channels for transmitting the video data. The multicast streams are initiated to start a video file from the beginning while the unicast channels are used to join the later arrivals to the appropriate multicast stream. In the implementation, the CB_MDA discovers the period of time when the server has plenty of resources an assigns them to appropriate clients in order to create work-ahead video data.The thesis further goes beyond the design of the VPS and presents a video client architecture that can synchronize with the server and work as a plug-in for producing the video data on different players such as MPEG-Berkely player, Xine.etc.

In this research, we propose a framework for measurement-based traffic engineering and connection admission control in radio access networks based on the Internet Protocol (IP). This framework is evaluated by simulation using the popular network simulator ns-2. The framework is adaptive to changes in the network load and can distinguish between different types of service. All traffic engineering decisions are made by edge routers (ERs) at the rim of the network domain. Multiple disjoint paths are configured between those ERs. Network state information is gathered in two different fashions. We evaluate a scheme based on the states of the queues on each alternative path and a scheme based on end-to-end probe packet transmission characteristics on each alternative path. Both schemes are compared to a shortest path first (SPF) routing approach.

We propose a novel call admission control which makes use of ratio-based traffic measurement to estimate the required bandwidth when a new call is issued. Existing approaches fail to estimate properly the required bandwidth. To alleviate the problem, we calculate the ratio between the measured mean rate and the mean rate declared by UPC parameters. The ratio and the target cell loss rate are used to estimate the required bandwidth to make decision if a new call is accepted or rejected. Because of more accurate estimation of required bandwidth, our method can provide a better control on quality of service.

ATM (Asynchronous Transfer Mode) networks were deemed the best choice for multimedia communication. The traditional mode was replaced because ATM can provide varied traffic types and QoS (quality of service). Maintaining QoS, however, requires a flexible traffic control, including call admission control and congestion control. Traditional approaches fail to estimate the required bandwidth and cell loss rate precisely. To alleviate these problems, we employ AI methods to improve the capability of estimated bandwidth and predicted cell loss rate. This thesis aims to apply neural network techniques to ATM traffic control and consists of two parts. The first part concerns a neural-based call admission control, while the second part presents an intelligent congestion control for ATM networks. In the first part, we focus on the improvement of RBF (Radial basis function) networks and the design of a neural-based call admission control. RBF networks have been widely used for modeling a function from given input-output patterns. However, two difficulties are encountered with traditional RBF networks. One is that the initial configuration of a RBF network needs to be determined by a trial-and-error method. The other is that the performance suffers from some difficulties when the desired output has abrupt changes or constant values in certain intervals. We propose a novel approach to overcome these difficulties. New kernel functions are used for hidden nodes, and the number of nodes is determined automatically by an ART-like algorithm. Parameters and weights are initialized appropriately, and then tuned and adjusted by the gradient descent method to improve the performance of the network. Then, we employ ART-RBF networks to design and implement a call admission control. Traditional approaches fail to estimate appropriately the required bandwidth, leading to a waste of bandwidth or a high cell loss rate. To alleviate the problem, we employ ART-RBF networks to estimate the required bandwidth, and thus a new connection request can then be accepted or rejected. Because of the more accurate estimation on the required bandwidth, the proposed method can provide a better control on quality of service for ATM networks. In the second part, we propose a neural-fuzzy rate-based feedback congestion control for ATM networks. Traditional methods perform congestion control by monitoring the queue length. The source rate is decreased by a fixed rate when the queue length is greater than a predefined threshold. However, it is difficult to get a suitable rate according to the degree of traffic congestion. We employ a neural-fuzzy mechanism to control the source rate. Through learning, cell loss can be predicted from the current value and the derivative of the queue length. Then an explicit rate is calculated and the source rate is controlled appropriately. In summary, we have proposed improvements on architecture and performance of neural networks, and applied neural networks to traffic control for ATM networks. We have developed some control mechanisms which, through simulations, have been shown to be more effective than traditional methods.

New technologies are expected to play a major role for wireless cellular systems beyond the existing 4G paradigm. The need for several orders of magnitude increase in system capacity has led to the proliferation of low-powered cellular layers overlaid on the existing macrocell layer. This type of network consisting of different cellular layers, each with their unique characteristics including transmission power and frequency of operation among others is termed as a heterogeneous network (HetNet). The emergence of HetNets leads to several research challenges and calls for a profound rethinking of several existing approaches for mobility management and interference management among other issues.

Durante los últimos años las redes móviles celulares han experimentado un considerable crecimiento y desarrollo debido al nuevo modo en que la sociedad crea, comparte y consume información. El enorme impacto y penetración de los servicios móviles telefónicos en la sociedad actual, así como la introducción de un amplio abanico de nuevos servicios de datos y servicios multimedia, han llevado a un crecimiento espectacular del volumen de tráfico transportado por este tipo de redes. Esta tendencia se mantendrá en los próximos años ya que constantemente van apareciendo nuevas aplicaciones que demandan mayor calidad de servicio y ancho de banda. Sin embargo, lasredesmóviles celulares actualestienen fuerteslimitaciones de ancho de banda debido a la escasez de frecuencias en el espectro radioeléctrico. Así, estas nuevas necesidades establecen nuevosretos para la industria de las telecomunicaciones. Es necesario gestionar un creciente número de servicios que demandan elevadas cantidades de recursos, conjuntamente con la escasez del espectro radioeléctrico, para ofrecer servicios que satisfagan las necesidades de los usuarios de un modo eficaz y económico. Dentro de este contexto, la gestión de los recursos radio se presenta como una estrategia clave para hacer frente a las características especiales de estasredes. Concretamente, el control de admisión es un mecanismo clave para utilizar eficientemente los recursos radio disponibles, proporcionando al mismo tiempo las garantías de calidad de servicio requeridas para todoslos usuarios. El presente trabajo se centra en el diseño y evaluación de políticas de control de admisión implementadas en redes móviles celulares multiservicio que transportan diferentes servicios portadores. Además, la presente tesis es una contribución al desarrollo de modelos con los que evaluar diferentes políticas de control de admisión en el desafiante contexto introducido por las redes 4G de próxima aparición. En la primera parte de esta tesis se trata el desarrollo de políticas de control de admisión para redes utilizadas actualmente, mientras que en la segunda parte de esta tesis se proponen políticas de control de admisión novedosas con el objetivo de hacerfrente a los retos introducidos por las redes de próxima aparición, tales como las redes Long TermEvolution o las de tecnología Cognitive Radio / Bernal Mor, E. (2013). Design and evaluation of admission control policies in mobile cellular networks [Tesis doctoral]. Editorial Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/27714 / Alfresco

In this work we introduce a novel queueing model with two classes of users in which, instead of accessing a resource, users wait in the system to match with a candidate from the other class. The users are selective and the matchings occur probabilistically. This new model is useful for analysing the traffic in web portals that match people who provide a service with people who demand the same service, e.g. employment portals, matrimonial and dating sites and rental portals. We first provide a Markov chain model for these systems and derive the probability distribution of the number of matches up to some finite time given the number of arrivals. We then prove that if no control mechanism is employed these systems are unstable for any set of parameters. We suggest four different classes of control policies to assure stability and conduct analysis on performance measures under the control policies. Contrary to the intuition that the rejection rate should decrease as the users become more likely to be matched, we show that for certain control policies the rejection rate is insensitive to the matching probability. Even more surprisingly, we show that for reasonable policies the rejection rate may be an increasing function of the matching probability. We also prove insensitivity results related to the average queue lengths and waiting times. Further, to gain more insight into the behaviour of probabilistic matching systems, we propose approximation methods based on fluid and diffusion limits using different scalings. We analyse the basic properties of these approximations and show that some performance measures are insensitive to the matching probability agreeing with the results found by the exact analysis. Finally we study the optimal control and revenue management for the systems with the objective of profit maximization. We formulate mathematical models for both unobservable and observable systems. For an unobservable system we suggest a deterministic optimal control, while for an observable system we develop an optimal myopic state dependent pricing.

In recent years, there has been an exponential growth in the use of multimedia applications. A satellite system offers great potential for multimedia applications with its ability to broadcast and multicast a large amount of data over a very large area as compared to a terrestrial system. However, the limited transmission capacity along with the dynamically varying channel conditions impedes the delivery of good quality multimedia service in a satellite system which has resulted in research efforts for deriving efficient radio resource management techniques. This issue is addressed in this thesis, where the main emphasis is to design a CAC framework which maximizes the utilization of the scarce radio resources available in the satellite and at the same time increases the performance of the system for a UMTS based satellite system supporting unicast and multicast traffic. The design of the system architecture for a UMTS based satellite system is presented. Based on this architecture, a CAC framework is designed consisting of three different functionalities: the admission control procedure, the retune procedure and the pre-emption procedure. The joint use of these functionalities is proposed to allow the performance of the system to be maintained under congestion. Different algorithms are proposed for different functionalities; an adaptive admission control algorithm, a greedy retune algorithm and three pre-emption algorithms (Greedy, SubSetSum, and Fuzzy). A MATLAB simulation model is developed to study the performance of the proposed CAC framework. A GUI is created to provide the user with the flexibility to configure the system settings before starting a simulation. The configuration settings allow the system to be analysed under different conditions. The performance of the system is measured under different simulation settings such as enabling and disabling of the two functionalities of the CAC framework; retune procedure and the pre-emption procedure. The simulation results indicate the CAC framework as a whole with all the functionalities performs better than the other simulation settings.

General purpose operating systems have been designed to provide fast, loss-free disk service to all applications. However, multimedia applications are capable of tolerating some data loss, but are very sensitive to variation in disk service timing. Present research efforts to handle multimedia applications assume pessimistic disk behaviour when deciding to admit new multimedia connections so as not to violate the real-time application constraints. However, since multimedia applications are ``soft' real-time applications that can tolerate some loss, we propose an optimistic scheme for admission control which uses average case values for disk access. Typically, disk scheduling mechanisms for multimedia applications reduce disk access times by only trying to minimize movement to subsequent blocks after sequencing based on Earliest Deadline First. We propose to implement a disk scheduling algorithm that uses knowledge of the media stored and permissible loss and jitter for each client, in addition to the physical parameters used by the other scheduling algorithms. We will evaluate our approach by implementing our admission control policy and disk scheduling algorithm in Linux and measuring the quality of various multimedia streams. If successful, the contributions of this thesis are the development of new admission control and flexible disk scheduling algorithm for improved multimedia quality of service.

In this thesis we explore precision in measurement of buffer overflow and loss probability. We see how buffer overflow probability compares with queuing delay measurements covered in the literature. More specifically, we measure the overflow probability of a packet buffer for various sampling rates to see the effect of sampling rate on the estimation. There are various reasons for measurement in networks; one key context assumed here is Measurement Based Admission Control. We conduct simulation experiments with analytically derived VoIP and bursty traffic parameters, in Matlab, while treating the buffer under consideration as a two-state Markov Chain. We note that estimation error decreases with increase in sampling gap (or in other words precision improves/variance decreases with decrease in sampling rate). We then perform experiments for VoIP and bursty data using NS-2 simulator and record the buffer states generated therein. We see a similar trend of increase in precision with increase in sampling gap. In our simulations, we have mainly considered static traffic passing through the buffer, and we use elastic traffic (TCP) for comparison. We see from our results that the sampling error becomes constant beyond certain asymptotic level. We thus look into asymptotic error in estimation,for the lowest sampling gap, to establish a lower bound on estimation error for buffer loss probability measurement. We use formulae given in recent literature for computing the experimental and theoretic asymptotic variance of the buffer state traces in our scenarios. We find that the theoretical and experimental asymptotic variance of overflow probability match when sampling a trace of buffer states modelled as a two-state Markov Chain in Matlab. We claim that this is a new approach to computing the lower bound on the measurement of buffer overflow probability, when the buffer states are modelled as a Markov process. Using Markov Chain modelling for buffer overflow we further explore the relationship between sampling rate and accuracy. We find that there is no relationship between sampling gap and bias of estimation. Crucially we go on to show that a more realistic simulation of a packet buffer reveals that the distribution of buffer overflow periods is not always such as to allow simple Markov modelling of the buffer states; while the sojourn periods are exponential for the smaller burst periods, the tail of the distribution does not fit to the same exponential fitting. While our work validates the use of a two-state Markov model for a useful approximation modelling the overflow of a buffer, we have established that earlier work which relies on simple Markovian assumptions will thereby underestimate the error in the measured overflow probabilities.