In recent years, wireless mesh networking has attracted a growing interest due to its inherent flexibility, scalability, and reliability. The IEEE 802.16 standard, commonly known as worldwide interoperability for microwave access (WiMAX), is the latest technology that enables broadband wireless access over long distances. WiMAX, which emerges as a wireless alternative to cable and digital subscriber line (DSL), is an ideal candidate to serve as the infrastructure for large scale wireless mesh networks. This thesis focuses on the quality of service (QoS) provisioning techniques in WiMAX-based metropolitan area mesh networks. We study the connection admission control (CAC) and routing issues in the design and operation of wireless multihop mesh networks. We propose a joint CAC and routing scheme for multiple service classes with the objective to maximize the overall revenue from all carried connections. Connection-level QoS constraints such as handoff connection dropping probability can be guaranteed within a threshold. Multiple service classes can be prioritized by imposing different reward rates. We apply optimization techniques to obtain the optimal CAC policies. The optimality criterion is the long-run average reward. We demonstrate that the proposed scheme can the maximum revenue obtainable by the system under QoS constraints. We show that the optimal joint policy is a randomized policy, i.e., connections are admitted to the system with some probabilities when the system is in certain states. Simulation results illustrate that the proposed scheme meets our design goals and outperforms the existing scheme. / Applied Science, Faculty of / Electrical and Computer Engineering, Department of / Graduate

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. VoDSL employs packet voice technology instead of the traditional circuit switched voice. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice packets over DSL. ATM is currently the preferred technology, since it offers the advantage of ATM's built-in Quality of Service (QoS) mechanisms. IP, on the other hand, cannot provide QoS guarantees in its traditional form. IP QoS mechanisms have been evolved only in the recent years. VoIP has gained popularity in the core networks. If it could replace VoATM in the access networks, it would open the door for end-to-end IP telephony that would result in major cost savings. In this thesis, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. Voice and data packets are placed in separate output queues at the bottleneck link. The Weighted Fair Queuing algorithm in used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. The router can infer the resource requirements for the connection by looking at certain field in the IP header of the application layer signaling packets. This eliminates the need for an explicit signaling protocol. We evaluate the performance of our QoS architecture by means of a simulation study. Our primary metrics are the end-to-end delay of voice packets across the access network and the bandwidth consumed by a voice call. Our results show that the end-to-end delays of voice packets in our VoIP architecture are comparable to that in the VoATM architecture. ACIS limits the number of voice calls admitted into the premium service class and provides guaranteed service to those calls under all loads. It also provides acceptable service to regular calls under light loads. We also show that PPP is a better choice than ATM as a Layer 2 protocol for our VoIP architecture. PPP offers the advantages of low bandwidth requirement and interleaving of voice packets in between fragments of large data packets during transmission over the bottleneck link. We conclude that our VoIP architecture would be suitable for future VoDSL deployments. / Master of Science

An overwhelming number of models in the literature use average inter-cell interference for the calculation of capacity of a Code Division Multiple Access (CDMA) network. The advantage gained in terms of simplicity by using such models comes at the cost of rendering the exact location of a user within a cell irrelevant. We calculate the actual per-user interference and analyze the effect of user-distribution within a cell on the capacity of a CDMA network. We show that even though the capacity obtained using average interference is a good approximation to the capacity calculated using actual interference for a uniform user distribution, the deviation can be tremendously large for non-uniform user distributions. Call admission control (CAC) algorithms are responsible for efficient management of a network's resources while guaranteeing the quality of service and grade of service, i.e., accepting the maximum number of calls without affecting the quality of service of calls already present in the network. We design and implement global and local CAC algorithms, and through simulations compare their network throughput and blocking probabilities for varying mobility scenarios. We show that even though our global CAC is better at resource management, the lack of substantial gain in network throughput and exponential increase in complexity makes our optimized local CAC algorithm a much better choice for a given traffic distribution profile.

In this thesis, optimization techniques for a joint admission control and radio resource allocation are developed for multicasting over high altitude platforms. First, a primary system model in a multicellular high altitude platform system is considered, in which each user can receive any requested multicast session in its cell from no more than only one HAP antenna simultaneously. All the users have equal priority for admission. The users are selected to join the respective multicast groups and the power, subchannels and time slots are allocated such that the spectrum utilization is maximized while satisfying the quality of service requirements. Lagrangian relaxation and the subgradient algorithm are used to obtain solution bounds for the primary system model problem formulation. These bounds were then used in the branch and bound algorithm for pruning of nodes. The numerical results illustrate the goodness of the bounds for different constraint set dualizations and for different subgradient step size rules. The system model is then extended to allow the multicast group users to receive a session's transmission from more than one antenna simultaneously at different frequencies. This also allows the user to receive multicast sessions transmitted in neighboring cells too, not just those transmitted in the cell which the user resides in. The users have different priority levels of admission and the objective is to maximize the admission of highest priority users to the system. A much efficient formulation is obtained for the extended model in terms of size, as compared to the primary model. Linear outer approximation using McCormick underestimators are used for the relaxation of the mixed binary quadratically constrained problem. The solution method is based on branch and cut scheme in which cutting planes, domain propagation and heuristics are integrated. Various branching schemes are considered and a presolving reformulation linearization scheme for a specific set of quadratic constraints is considered. The numerical experiments compare the performances in terms of the duality gap, number of nodes, number of iterations, the number of iterations per node, the time needed to obtain the first feasible solution and the percentage of instances a feasible solution was found. / October 2016

The thesis reviews current admission control schemes in cellular IP networks. It proposes an improved version of Threshold Access Sharing and a new scheme: weight-based scheme. Finally, an admission control scheme for hierarchical cellular network is introduced.

A Mobile Ad Hoc Network (MANET) is a shared wireless network without any infrastructure, consisting of mobile nodes connected by wireless links. The nodes are free to move and organize themselves arbitrarily. The nodes in the network are therefore depending on each other in order to communicate over multiple hops. Due to the physical characteristics of wireless networks, the channel is time-varying, which makes it hard to both predict and sustain a bit rate level. The nodes’ mobility causes topology changes, and further load and capacity variations. Traditional usage areas are battlefield and disaster areas, while new areas like extended network coverage and gaming are emerging. Quality of Service (QoS) is needed in every network in order to differentiate traffic with different performance requirements, e.g. voice and e-mail applications. Providing QoS in wireless environments with varying conditions is complex, and hard guarantees can not be given. Consequently, the aim is to give differentiated treatment to traffic with different performance requirements. In addition, we can not study the MANET without considering fixed networks. Communication with fixed networks is important, for example by accessing the Internet. In this thesis the Differentiated Services (DiffServ) architecture is applied and adapted to MANETs. Using the same QoS architecture will ease the transition between the wireless and wired domain. But the special characteristics of wireless networks require modifications to the original DiffServ architecture. In investigations there was found restrictions on the number of classes to use, and this number was dependent on the type of traffic in the network. A QoS architecture based on the DiffServ framework is proposed, with an admission control based on the concept of shadow classes, and Explicit Congestion Notification (ECN) to avoid congestion. New flows are tested in a shadow class before getting admission to the network and its designated class. The shadow class has the same scheduling properties as the designated class, but is differentiated by a higher drop probability in the buffers. Both the admission control and ECN are thus build on the same principle by controlling the load from probabilistic functions in the buffers, and are studied to find their individual and combined effects. In wireless environments the probability of a packet loss increases with the number of hops, which gives services an unpredictable performance for users. A predictable service, independent of number of hops, is provided by scheduling based on the path information; the packets are differentiated based on the number of hops made or left to make, increasing the predictability at the cost of performance.

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.