Multiprotocol Label Switching (MPLS) fuses the intelligence of routing with the performance of switching and provides significant benefits to networks with a pure IP architecture as well as those with IP and ATM or a mix of ther Layer 2 technologies. MPLS technology is key to scalable virtual private networks (VPNs) and end-to-end quality of service (QoS), enabling efficient utilization of existing networks to meet future growth. The technology also helps to deliver highly scalable, differentiated end-to-end IP services with simpler configuration, management, and provisioning for both Internet providers and end-users. However, MPLS is a connection-oriented architecture. In case of failure MPLS first has to establish a new label switched path (LSP) and then forward the packets to the newly established LSP. For this reason MPLS has a slow restoration response to a link or node failure on the LSP.The thesis provides a description of MPLS-based architecture as a preferred technology for integrating ATM and IP technologies, followed by a discussion of the motivation for the fast and reliable restoration mechanism in an MPLS network. In this thesis first we address the fast rerouting mechanisms for MPLS networks and then we focus on the problem of packet loss, packet reordering and packet delay for protected LSP in MPLS-based network for a single node/link failure. In order to deliver true service assurance for guaranteed traffic on a protected LSP we use the fast rerouting mechanism with a preplanned alternative LSP. We propose enhancements to current proposals described in extant literature. Our fast rerouting mechanism avoids packet disorder and significantly reduces packet delay during the restoration period.An extension of the Fast Rerouting proposal, called Reliable and Fast Rerouting (RFR), provides some preventive actions for the protected LSP against packet loss during a failure. RFR maintains the same advantages of Fast Rerouting while eliminating packet losses, including those packet losses due to link or node failure (circulating on the failed links), which were considered to be "inevitable" up to now.For the purpose of validating and evaluating the behavior of these proposals a simulation tool was developed. It is based on the NS, a well-known network simulator that is being used extensively in research work. An extension featuring the basic functionality of MPLS (MNS) is also available for the NS, and this is the basis of the developed simulation tool.Simulation results allow the comparison of Fast Rerouting and RFR with previous rerouting proposals.In addition to this we propose a mechanism for multiple failure recovery in an LSP. This proposal combines the path protection, segment protection and local repair methods. In addition to the multiple link/node failure protection, the multiple fault tolerance proposal provides a significant reduction of delay that the rerouted traffic can experience after a link failure, because the repair action is taken close to the point of failure.Then we proceed to address an inherent problem of the preplanned alternative LSP. As alternative LSPs are established together with the protected LSP it may happen that the alternative is not the optimal LSP at the time the failure occurs. To overcome this undesired behavior, we propose the Optimal and Guaranteed Alternative Path (OGAP). The proposal uses a hybrid of fast-rerouting and a dynamic approach to establish the optimal alternative LSP while rerouting the affected traffic using the preplanned alternative LSP. This hybrid approach provides the best of the fast rerouting and the dynamic approaches.At the same time we observed that the protection path becomes in fact unprotected from additional failures after the traffic is rerouted onto it.To address this we propose a guarantee mechanism for protection of the new protected LSP carrying the affected traffic, by establishing an alternative LSP for the rerouted traffic after a failure, avoiding the vulnerability problem for the protected traffic.Finally, we present a further optimization mechanism, adaptive LSP, to enhance the existing traffic engineering for Quality of Services (QoS)provision and improve network resource utilization. The adaptive LSP proposal allows more flexibility in network resource allocation and utilization by adapting the LSP to variations in all network loads,resulting in an enhancement of existing MPLS traffic engineering.

As more and more wireless applications/services emerge in the market, the already heavily crowded radio spectrum becomes much scarcer. Meanwhile, however,as it is reported in the recent literature, there is a large amount of radio spectrum that is under-utilized. This motivates the concept of cognitive radio wireless networks that allow the unlicensed secondary-users (SUs) to dynamically use the vacant radio spectrum which is not being used by the licensed primary-users (PUs). In this dissertation, we investigate protocol design for both the synchronous and asynchronous cognitive radio networks with emphasis on the medium access control (MAC) layer. We propose various spectrum sharing schemes, opportunistic packet scheduling schemes, and spectrum sensing schemes in the MAC and physical (PHY) layers for different types of cognitive radio networks, allowing the SUs to opportunistically utilize the licensed spectrum while confining the level of interference to the range the PUs can tolerate. First, we propose the cross-layer based multi-channel MAC protocol, which integrates the cooperative spectrum sensing at PHY layer and the interweave-based spectrum access at MAC layer, for the synchronous cognitive radio networks. Second, we propose the channel-hopping based single-transceiver MAC protocol for the hardware-constrained synchronous cognitive radio networks, under which the SUs can identify and exploit the vacant channels by dynamically switching across the licensed channels with their distinct channel-hopping sequences. Third, we propose the opportunistic multi-channel MAC protocol with the two-threshold sequential spectrum sensing algorithm for asynchronous cognitive radio networks. Fourth, by combining the interweave and underlay spectrum sharing modes, we propose the adaptive spectrum sharing scheme for code division multiple access (CDMA) based cognitive MAC in the uplink communications over the asynchronous cognitive radio networks, where the PUs may have different types of channel usage patterns. Finally, we develop a packet scheduling scheme for the PU MAC protocol in the context of time division multiple access (TDMA)-based cognitive radio wireless networks, which is designed to operate friendly towards the SUs in terms of the vacant-channel probability. We also develop various analytical models, including the Markov chain models, M=GY =1 queuing models, cross-layer optimization models, etc., to rigorously analyze the performance of our proposed MAC protocols in terms of aggregate throughput, access delay, and packet drop rate for both the saturation network case and non-saturation network case. In addition, we conducted extensive simulations to validate our analytical models and evaluate our proposed MAC protocols/schemes. Both the numerical and simulation results show that our proposed MAC protocols/schemes can significantly improve the spectrum utilization efficiency of wireless networks.

In this thesis, we consider optimal control problems arising in high-speed integrated communication networks with Quality of Service (QOS) constraints. Integrated networks are expected to carry a large variety of traffic sources with widely varying traffic characteristics and performance requirements. Broadly, the traffic sources fall into two categories: (a) real-time sources with specified performance criteria, like small end to end delay and loss probability (sources of this type are referred to as Type 1 sources below), and (b) sources that do not have stringent performance criteria and do not demand performance guarantees from the network - the so-called Best Effort Type sources (these are referred to as Type 2 sources below). From the network's point of view, Type 2 sources are much more "controllable" than Type 1 sources, in the sense that the Type 2 sources can be dynamically slowed down, stopped or speeded up depending on traffic congestion in the network, while for Type 1 sources, the only control action available in case of congestion is packet dropping. Carrying sources of both types in the same network concurrently while meeting the performance objectives of Type 1 sources is a challenge and raises the question of equitable sharing of resources. The objective is to carry as much Type 2 traffic as possible without sacrificing the performance requirements of Type 1 traffic. We consider simple models that capture this situation. Consider a network node through which two connections pass, one each of Types 1 and 2. One would like to maximize the throughput of the Type 2 connection while ensuring that the Type 1 connection's performance objectives are met. This can be set up as a constrained optimization problem that, however, is very hard to solve. We introduce a parameter b that represents the "cost" of buffer occupancy by Type 2 traffic. Since buffer space is limited and shared, a queued Type 2 packet means that a buffer position is not available for storing a Type 1 packet; to discourage the Type 2 connection from hogging the buffer, the cost parameter b is introduced, while a reward for each Type 2 packet coming into the buffer encourages the Type 2 connection to transmit at a high rate. Using standard on-off models for the Type 1 sources, we show how values can be assigned to the parameter b; the value depends on the characteristics of the Type 1 connection passing through the node, i.e., whether it is a Variable Bit Rate (VBR) video connection or a Continuous Bit Rate (CBR) connection etc. Our approach gives concrete networking significance to the parameter b, which has long been considered as an abstract parameter in reward-penalty formulations of flow control problems (for example, [Stidham '85]). Having seen how to assign values to b, we focus on the Type 2 connection next. Since Type 2 connections do not have strict performance requirements, it is possible to defer transmitting a Type 2 packet, if the conditions downstream so warrant. This leads to the question: what is the "best" transmission policy for Type 2 packets? Decisions to transmit or not must be based on congestion conditions downstream; however, the network state that is available at any instant gives information that is old, since feedback latency is an inherent feature of high speed networks. Thus the problem is to identify the best transmission policy under delayed feedback information. We study this problem in the framework of Markov Decision Theory. With appropriate assumptions on the arrivals, service times and scheduling discipline at a network node, we formulate our problem as a Partially Observable Controlled Markov Chain (PO-CMC). We then give an equivalent formulation of the problem in terms of a Completely Observable Controlled Markov Chain (CO-CMC) that is easier to deal with., Using Dynamic Programming and Value Iteration, we identify structural properties of an optimal transmission policy when the delay in obtaining feedback information is one time slot. For both discounted and average cost criteria, we show that the optimal policy has a two-threshold structure, with the threshold on the observed queue length depending, on whether a Type 2 packet was transmitted in the last slot or not. For an observation delay k > 2, the Value Iteration technique does not yield results. We use the structure of the problem to provide computable upper and lower bounds to the optimal value function. A study of these bounds yields information about the structure of the optimal policy for this problem. We show that for appropriate values of the parameters of the problem, depending on the number of transmissions in the last k steps, there is an "upper cut off" number which is a value such that if the observed queue length is greater than or equal to this number, the optimal action is to not transmit. Since the number of transmissions in the last k steps is between 0 and A: both inclusive, we have a stack of (k+1) upper cut off values. We conjecture that these (k + l) values axe thresholds and the optimal policy for this problem has a (k + l)-threshold structure. So far it has been assumed that the parameters of the problem are known at the transmission control point. In reality, this is usually not known and changes over time. Thus, one needs an adaptive transmission policy that keeps track of and adjusts to changing network conditions. We show that the information structure in our problem admits a simple adaptive policy that performs reasonably well in a quasi-static traffic environment. Up to this point, the models we have studied correspond to a single hop in a virtual connection. We consider the multiple hop problem next. A basic matter of interest here is whether one should have end to end or hop by hop controls. We develop a sample path approach to answer this question. It turns out that depending on the relative values of the b parameter in the transmitting node and its downstream neighbour, sometimes end to end controls are preferable while at other times hop by hop controls are preferable. Finally, we consider a routing problem in a high speed network where feedback information is delayed, as usual. As before, we formulate the problem in the framework of Markov Decision Theory and apply Value Iteration to deduce structural properties of an optimal control policy. We show that for both discounted and average cost criteria, the optimal policy for an observation delay of one slot is Join the Shortest Expected Queue (JSEQ) - a natural and intuitively satisfactory extension of the well-known Join the Shortest Queue (JSQ) policy that is optimal when there is no feedback delay (see, for example, [Weber 78]). However, for an observation delay of more than one slot, we show that the JSEQ policy is not optimal. Determining the structure of the optimal policy for a delay k>2 appears to be very difficult using the Value Iteration approach; we explore some likely policies by simulation.

QoS routing is one of the major building blocks for supporting QoS in communication networks and, hence, a necessary component of future communication networks. Bandwidth- Constrained Routing Algorithm (BCRA) may help to satisfy QoS requirements such as end-to-end delay, delay-jitter etc when WFQ-like (Weighted Fair Queuing) scheduling mechanisms are deployed. The existing algorithms for bandwidth constrained routing suffer from high message overhead and have a high computational and space complexity. The work presented in the thesis, therefore, focuses on the different techniques that an be used to reserve bandwidth for a unicast connection with low protocol overhead in terms of number of messages. We have compared the performance of the proposed routing algorithms using simulation studies with other bandwidth constrained routing algorithms. The call blocking ratio and message overhead have been used as the performance metric to compare the proposed algorithm with the existing ones. We present three source routing algorithms for unicast connections satisfying the band- width requirement. The first two routing algorithms are based on the partitioning of the network. The link-state broadcasts are limited to the partition. In the first algorithm, the source node queries the other partitions for the state information on a connection request and computes the path based on the information received from the other partitions. The second algorithm is based on state aggregation. The aggregated state of other partitions is maintained at every node. The source node finds a feasible path based on the aggregated information. The path is expanded in every partition, if required, at the time of resource reservation. The third QoS routing algorithm uses the Distance Vector Tables to find a route for a connection. If the shortest path satisfies the bandwidth requirement, then it is selected; otherwise a random deviation is taken at the point where bandwidth requirement is not satisfied and shortest path algorithm is again followed. In all the three algorithms presented, the packets carry the entire path information to the destination node. Therefore, no per connection information is required to be maintained at the intermediate nodes. Simulation results indicate that the proposed algorithms indeed help educing the protocol overhead considerably, and at the same time they give comparable or better performance in terms of resource utilization across a wide range of workloads.

In this thesis we address the problem of providing end-to-end quality of service (QoS) to real-time and data connections in a third generation (3G) cellular network based on the Universal Mobile Telecommunication System (UMTS) standard. Data applications usually use TCP (Transmission Control Protocol) and the QoS is a minimum guaranteed mean throughput. For this one first needs to compute the throughput of a TCP connection sending its traffic through the UMTS network (possibly also through the wired part of the Internet). Thus we obtain closed form expressions for a TCP throughput in a UMTS environment. For downloading data at a mobile terminal, the packets of each TCP connection are stored in separate queues at the base station (node B). These are fragmented into Protocol Data Units (PDU). The link layer uses ARQ (Automatic Repeat Request). Thus there can be significant random transmission/queueing delays of TCP packets at the node B. On the other hand the link may not be fully utilized due to the delays of the TCP packets in the rest of the network. In such a scenario the existing models of TCP may not be sufficient. Thus we provide new approximate models for TCP and also obtain new closed form expressions of mean window size. Using these we obtain the throughput of a TCP connection for the scenario where the queueing delays are non-negligible compared to the overall Round Trip Time (RTT) and also the link utilization is less than one. Our approximate models can be useful not only in the UMTS context but also else where. In the second half of the thesis, we use these approximate models of TCP to provide minimum mean throughput to data connections in UMTS. We also consider real-time applications such as voice and video. These can tolerate a little packet loss (~1%) but require an upper Bound on the delay and delay jitter (≤ 150 ms). Thus if the network provides a constant bandwidth and the received SINR is above a specified threshold ( with a certain probability), QoS for the real-time traffic will be satisfied. The 3G cellular systems are interference limited. Thus wise allocation of power is critical in these systems. Hence we consider the problem of providing end-to-end QoS to different users along with the minimization of the downlink power allocation.

Les réseaux maillés sans fil (RMSF), grâce à leurs caractéristiques avantageuses, sont considérés comme une solution efficace pour le support des services de voix, vidéo et de données dans les réseaux de prochaine génération. Le standard IEEE 802.16-d a spécifié pour les RMSF, à travers son mode maillé, deux mécanismes de planifications de transmission de données; à savoir la planification centralisée et la planification distribuée. Dans ce travail, on a évalué le support de la qualité de service (QdS) du standard en se focalisant sur la planification distribuée. Les problèmes du système dans le support du trafic de voix ont été identifiés. Pour résoudre ces problèmes, on a proposé un protocole pour le support de VoIP (AVSP) en tant qu’extension au standard original pour permettre le support de QdS au VoIP. Nos résultats préliminaires de simulation montrent qu’AVSP offre une bonne amélioration au support de VoIP. / Wireless mesh networks (WMNs), because of their advantageous characteristics, are considered as an effective solution to support voice services, video and data in next generation networks. The IEEE 802.16-d specified for WMNs, through its mesh mode, two mechanisms of scheduling data transmissions; namely centralized scheduling and distributed scheduling. In this work, we evaluated the support of the quality of service (QoS) of the standard by focusing on distributed scheduling. System problems in the support of voice traffic have been identified. To solve these problems, we proposed a protocol for supporting VoIP, called Assisted VoIP Scheduling Protocol (AVSP), as an extension to the original standard to support high QoS to VoIP. Our preliminary simulation results show that AVSP provides a good improvement to support VoIP.

An ever increasing demand for instant and reliable information on modern communication networks forces codewords to operate in a non-asymptotic regime. To achieve reliability for imperfect channels in this regime, codewords need to be retransmitted from receiver to the transmit buffer, aided by a fast feedback mechanism. Large occupancy of this buffer results in longer communication delays. Therefore, codewords need to be designed carefully to reduce transmit queue-length and thus the delay experienced in this buffer. We first study the consequences of physical layer decisions on the transmit buffer occupancy. We develop an analytical framework to relate physical layer channel to the transmit buffer occupancy. We compute the optimal code-rate for finite-length codewords operating over a correlated channel, under certain communication service guarantees. We show that channel memory has a significant impact on this optimal code-rate. Next, we study the delay in small ad-hoc networks. In particular, we find out what rates can be supported on a small network, when each flow has a certain end-to-end service guarantee. To this end, service guarantee at each intermediate link is characterized. These results are applied to study the potential benefits of setting up a network suitable for network coding in multicast. In particular, we quantify the gains of network coding over classic routing for service provisioned multicast communication over butterfly networks. In the wireless setting, we study the trade-off between communications gains achieved by network coding and the cost to set-up a network enabling network coding. In particular, we show existence of scenarios where one should not attempt to create a network suitable for coding. Insights obtained from these studies are applied to design a distributed rate control algorithm in a large network. This algorithm maximizes sum-utility of all flows, while satisfying per-flow end-to-end service guarantees. We introduce a notion of effective-capacity per communication link that captures the service requirements of flows sharing this link. Each link maintains a price and effective-capacity, and each flow maintains rate and dissatisfaction. Flows and links update their respective variables locally, and we show that their decisions drive the system to an optimal point. We implemented our algorithm on a network simulator and studied its convergence behavior on few networks of practical interest.

In the wireless channel, cooperative communications allow one or many relays to assist the communication between the source and the destination. The aim of this thesis is the development of tools for the analysis of cooperative systems, when HARQ techniques are employed to provide cross-layer error protection. The first chapter of the thesis gives background information on network coding in cooperative relay networks, and introduces the motivation for this work. The second chapter is devoted to the analysis of the energetic-fair performance evaluations of FEC, ARQ-STBC and HARQ schemes at the MAC and IP layers. New analytical framework is derived and applied to a point-to-point network scenario. This framework allows to make energetic fair comparisons between the schemes with and without retransmissions. We determine under which channel conditions the cross-layer error protection is energetically more efficient than the simple channel coding. In the third chapter of this thesis we study the cooperative deterministic protocols. The protocols that we consider differ based on the behaviour of the relay(s), source(s), and destination. We consider two major types of cooperative protocols: decode-and forward (DCF), and demodulate-and-forward (DMF). Each of these protocols in its turn is analysed with and without combining mechanisms at the destination. We derive the soft decoders at the destination side for each respective case, and compare the performances of these protocols at the MAC layer. The following quality of service metrics are evaluated: frame error rate, delay, efficiency, goodput. The analysis is done evaluating the steady-state, using finite state Markov chains and a combinatorial approach. The analysis, however, becomes very complex as the number of transmissions and/or nodes in the network increases. The fourth chapter introduces a class of probabilistic communication protocols, where the devices retransmit with a given probability. We prove the existence of an equivalent class of protocols, with the same performances as the deterministic class. Using proofs of concept it is shown that the probabilistic protocol class allows for tractable steady-state analysis, even for many nodes in the network. Based on this, we then derive the QoS metrics and evalute them also by simulations. The derived performance evaluation metrics are then optimized by constraining the frame error rate, and trying to find the most optimal transmissions number and code rates which maximize the goodput. It is furthermore shown, that the equivalent protocol has larger optimal region than the deterministic one.

Les travaux de cette thèse s'inscrivent dans le contexte des réseaux sans fil et des réseaux d'accès par Satellite en particulier, qui facilitent l'installation d'infrastructures réseau dans les zones géographiquement reculées et faiblement peuplées. Cependant, ces derniers présentent certains inconvénients lorsqu'il s'agit de déployer des applications multimédia interactives. En effet, de telles applications requièrent un délai de bout en bout aussi faible que possible et plus généralement exigent une meilleure Qualité de Service (QdS) du système de communication que le classique Meilleur-Effort (BE) afin de fonctionner correctement. Or, les réseaux d'accès par satellite géostationnaires souffrent déjà d'un délai de propagation non négligeable d'autant plus accru que la transmission des données est assurée par des mécanismes d'allocation dynamique, par exemple dans un système DVB-S2/RCS. Dans ces travaux de thèse, nous proposons d'utiliser les informations de signalisation de session des applications multimédia basées sur le protocole de session SIP afin d'ajuster le paramétrage du système de communication selon une approche " cross-layer " qui permet alors d'améliorer de façon significative la réactivité du système. Nous avons proposé plusieurs solutions pour, d'abord réduire le temps entre la demande de communication et le démarrage effectif du transfert des flux multimédia, ensuite réduire le délai de transmission des données multimédia durant la communication (tout en utilisant de manière optimale les ressources réseau disponibles sur la voie retour), et enfin accroître le nombre de flux multimédia admissibles dans le réseau satellite tout en leur garantissant un niveau de QdS satisfaisant. Afin de faciliter l'intégration et l'implémentation des solutions proposées dans un système de communication réel, un mécanisme de communication inter-couches d'optimisation est proposé et développé. De plus, une architecture orientée web services est utilisée afin de faciliter la découverte et l'invocation des différentes niveaux de services de communication présents dans de tels réseaux d'accès. Les solutions proposées ont été évaluées dans des environnements sans fil émulés et réels

L’émergence de nouvelles applications et de nouveaux services (tels que les applications multimédias, la voix-sur-IP, la télévision-sur-IP, la vidéo-sur-demande, etc.) et le besoin croissant de mobilité des utilisateurs entrainent une demande de bande passante de plus en plus croissante et une difficulté dans sa gestion dans les réseaux cellulaires sans fil (WCNs), causant une dégradation de la qualité de service. Ainsi, dans cette thèse, nous nous intéressons à la gestion des ressources, plus précisément à la bande passante, dans les WCNs. Dans une première partie de la thèse, nous nous concentrons sur la prédiction de la mobilité des utilisateurs des WCNs. Dans ce contexte, nous proposons un modèle de prédiction de la mobilité, relativement précis qui permet de prédire la destination finale ou intermédiaire et, par la suite, les chemins des utilisateurs mobiles vers leur destination prédite. Ce modèle se base sur : (a) les habitudes de l’utilisateur en terme de déplacements (filtrées selon le type de jour et le moment de la journée) ; (b) le déplacement courant de l’utilisateur ; (c) la connaissance de l’utilisateur ; (d) la direction vers une destination estimée ; et (e) la structure spatiale de la zone de déplacement. Les résultats de simulation montrent que ce modèle donne une précision largement meilleure aux approches existantes. Dans la deuxième partie de cette thèse, nous nous intéressons au contrôle d’admission et à la gestion de la bande passante dans les WCNs. En effet, nous proposons une approche de gestion de la bande passante comprenant : (1) une approche d’estimation du temps de transfert intercellulaire prenant en compte la densité de la zone de déplacement en terme d’utilisateurs, les caractéristiques de mobilité des utilisateurs et les feux tricolores ; (2) une approche d’estimation de la bande passante disponible à l’avance dans les cellules prenant en compte les exigences en bande passante et la durée de vie des sessions en cours ; et (3) une approche de réservation passive de bande passante dans les cellules qui seront visitées pour les sessions en cours et de contrôle d’admission des demandes de nouvelles sessions prenant en compte la mobilité des utilisateurs et le comportement des cellules. Les résultats de simulation indiquent que cette approche réduit largement les ruptures abruptes de sessions en cours, offre un taux de refus de nouvelles demandes de connexion acceptable et un taux élevé d’utilisation de la bande passante. Dans la troisième partie de la thèse, nous nous penchons sur la principale limite de la première et deuxième parties de la thèse, à savoir l’évolutivité (selon le nombre d’utilisateurs) et proposons une plateforme qui intègre des modèles de prédiction de mobilité avec des modèles de prédiction de la bande passante disponible. En effet, dans les deux parties précédentes de la thèse, les prédictions de la mobilité sont effectuées pour chaque utilisateur. Ainsi, pour rendre notre proposition de plateforme évolutive, nous proposons des modèles de prédiction de mobilité par groupe d’utilisateurs en nous basant sur : (a) les profils des utilisateurs (c’est-à-dire leur préférence en termes de caractéristiques de route) ; (b) l’état du trafic routier et le comportement des utilisateurs ; et (c) la structure spatiale de la zone de déplacement. Les résultats de simulation montrent que la plateforme proposée améliore la performance du réseau comparée aux plateformes existantes qui proposent des modèles de prédiction de la mobilité par groupe d’utilisateurs pour la réservation de bande passante. / The emergence of new applications and services (e.g., multimedia applications, voice over IP and IPTV) and the growing need for mobility of users cause more and more growth of bandwidth demand and a difficulty of its management in Wireless Cellular Networks (WCNs). In this thesis, we are interested in resources management, specifically the bandwidth, in WCNs. In the first part of the thesis, we study the user mobility prediction that is one of key to guarantee efficient management of available bandwidth. In this context, we propose a relatively accurate mobility prediction model that allows predicting final or intermediate destinations and subsequently mobility paths of mobile users to reach these predicted destinations. This model takes into account (a) user’s habits in terms of movements (filtered according to the type of day and the time of the day); (b) user's current movement; (c) user’s contextual knowledge; (d) direction from current location to estimated destination; and (e) spatial conceptual maps. Simulation results show that the proposed model provides good accuracy compared to existing models in the literature. In the second part of the thesis, we focus on call admission control and bandwidth management in WCNs. Indeed, we propose an efficient bandwidth utilization scheme that consists of three schemes: (1) handoff time estimation scheme that considers navigation zone density in term of users, users’ mobility characteristics and traffic light scheduling; (2) available bandwidth estimation scheme that estimates bandwidth available in the cells that considers required bandwidth and lifetime of ongoing sessions; and (3) passive bandwidth reservation scheme that passively reserves bandwidth in cells expected to be visited by ongoing sessions and call admission control scheme for new call requests that considers the behavior of an individual user and the behavior of cells. Simulation results show that the proposed scheme reduces considerably the handoff call dropping rate while maintaining acceptable new call blocking rate and provides high bandwidth utilization rate. In the third part of the thesis, we focus on the main limitation of the first and second part of the thesis which is the scalability (with the number of users) and propose a framework, together with schemes, that integrates mobility prediction models with bandwidth availability prediction models. Indeed, in the two first contributions of the thesis, mobility prediction schemes process individual user requests. Thus, to make the proposed framework scalable, we propose group-based mobility prediction schemes that predict mobility for a group of users (not only for a single user) based on users’ profiles (i.e., their preference in terms of road characteristics), state of road traffic and users behaviors on roads and spatial conceptual maps. Simulation results show that the proposed framework improves the network performance compared to existing schemes which propose aggregate mobility prediction bandwidth reservation models.