O segmento das comunicações, já há alguns anos, vem passando por significativas transformações exigindo a interação entre ambientes tecnológicos convergentes heterogêneos, com qualidade na continuidade de serviços, para se manter competitivo, pois é este mercado que exige, constantemente, que mais recursos tecnológicos lhe sejam colocados à disposição. O ineditismo da proposta desenvolvida neste trabalho é a elaboração de um modelo para assegurar a continuidade da qualidade de serviço percebida pelo usuário final através de handover (ou handoff) vertical, o que caracteriza-se como uma necessidade de solução global, isto é, o modelo é genérico e independente da tecnologia, o que permite sua adoção em qualquer ambiente de rede existente aproveitando a capilaridade já disponível das redes legadas. É analisada também a situação na qual um usuário final esteja acessando simultaneamente serviços gerenciados por Provedores de Serviço distintos, o que conduz a duas situações possíveis: a) o usuário está acessando serviços distintos contratados a Provedores de Serviço diferentes; b) o usuário está acessando o mesmo serviço contratado a Provedores de Serviço distintos. Nesta última situação pode surgir a necessidade de disparar um processo de handover exigindo a decisão de qual dos Provedores o executará, de acordo com o SPHDA Service Providers Handover Decision Agreement. A metodologia adotada para o desenvolvido do modelo foi a RM-ODP - Reference Model for Open Distributed Processing, por abranger todos os aspectos técnicos e comerciais necessários à sua construção. / The sector of communications, for some years, has undergone significant changes requiring interaction between converging heterogeneous technology environments, with quality and continuity of services to stay competitive, because this market is that requires constantly more technological resources available. The novelty of the proposal developed in this work is the development of a model to ensure the continued quality of service perceived by end users via vertical handover (or handoff), which characterizes itself as a need for a global solution, i.e., the model is generic and technology independent, allowing its adoption in any network environment taking advantage of the capillary already available from legacy networks. It is also analyzed the situation in which an end user is simultaneously accessing services managed by different Service Providers, which leads to two possible situations: a) the user is accessing different services contracted to different Service Providers; b) the user is accessing the same service contracted to different Service Providers. In this last situation may be necessary to trigger a handover process requiring the decision of which the Providers shall execute it in accordance with the SPHDA - Service Providers Handover Decision Agreement. The methodology adopted for the model development was the RM-ODP - Reference Model for Open Distributed Processing, as it includes all technical and commercial aspects necessary for its construction.

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.

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.

L'objectif de ce travail de thèse dans un premier temps est l'évaluation de performances des modèles de routage multi-chemins pour l'ingénierie de trafic et l'équilibrage de charge sur un réseau de type IP/MPLS (MPLS-TE). Nous comparons la capacité de ces modèles à équilibrer la charge du réseau tout en faisant de la différentiation de trafic. Nous les appliquons sur des grandes topologies générées par le générateur automatique des topologies BRITE, qui s'approchent en forme et en complexité du réseau réel. Nous mesurons ainsi l'impact de leur complexité respective et donc la capacité à les déployer sur des réseaux de grande taille (scalabilité). Dans un second temps, l'objectif est de proposer un concept de modélisation générale d'un réseau à commutations par paquets. Ce modèle est établi sur la base de la théorie différentielle de trafic et la théorie des files d'attente, tout en utilisant des approches graphiques. Le but est d'estimer l'état de charge du réseau et de ses composants (routeurs, liens, chemins). Ensuite, en fonction de ça, nous développons des approches de contrôle de congestion et commande sur l'entrée améliorant les techniques de routage adaptatif et l'équilibrage de charge dans les réseaux IP/MPLS / In This work, firstly we present and evaluate the behavior of multipath routing models for the DS-TE (DiffSev aware MPLS traffic Engineering) called PEMS and LBWDP. To clarify network topologies and routing models that are suitable for MPLS Traffic Engineering, we evaluate them from the viewpoint of network scalability and end-to-end quality. Using a network topology generated by BRITE, that has many alternative paths, we applied these models on a huge topology that correspond to real network. This can provide a real simulation for the internet and gives a good evaluation for the end-to-end quality and the network use.Secondly, the aim of this work is to propose a general model for Packet switching networks. This model is established on the traffic differential theory and the Queuing theory, while using graphic approaches. The aim of this model is to calculate the network use state and its components (router, link, path...). Then, we develop control and command approaches in the entry of network to improve an adaptive routing plan and load balancing in IP/MPLS networks

Made available in DSpace on 2014-12-17T14:55:52Z (GMT). No. of bitstreams: 1 RenataPB_DISSERT.pdf: 1489254 bytes, checksum: 88fdf1027875fb6b83dbe203da3c24f7 (MD5) Previous issue date: 2011-11-14 / It s notorious the advance of computer networks in recent decades, whether in relation to transmission rates, the number of interconnected devices or the existing applications. In parallel, it s also visible this progress in various sectors of the automation, such as: industrial, commercial and residential. In one of its branches, we find the hospital networks, which can make the use of a range of services, ranging from the simple registration of patients to a surgery by a robot under the supervision of a physician. In the context of both worlds, appear the applications in Telemedicine and Telehealth, which work with the transfer in real time of high resolution images, sound, video and patient data. Then comes a problem, since the computer networks, originally developed for the transfer of less complex data, is now being used by a service that involves high transfer rates and needs requirements for quality of service (QoS) offered by the network . Thus, this work aims to do the analysis and comparison of performance of a network when subjected to this type of application, for two different situations: the first without the use of QoS policies, and the second with the application of such policies, using as scenario for testing, the Metropolitan Health Network of the Federal University of Rio Grande do Norte (UFRN) / ? not?rio o avan?o das redes de computadores nas ?ltimas d?cadas, seja em rela??o ?s taxas de transmiss?o, ao n?mero de dispositivos interconectados ou mesmo ?s aplica??es existentes. Em paralelo, percebemos tamb?m este avan?o nos diversos segmentos da ?rea de automa??o, tais como: industrial, comercial e residencial. Em uma de suas ramifica??es, encontram-se as redes hospitalares, que podem fazer uso de uma gama de servi?os, que v?o desde o simples cadastro de pacientes at? uma cirurgia feita por um rob? sob a supervis?o de um m?dico especialista. No contexto dos dois universos, aparecem as aplica??es em Telemedicina e Telessa?de, que trabalham com a transfer?ncia, em tempo real, de imagens de alta resolu??o, som, v?deo e dados de pacientes. Surge ent?o um problema, visto que as redes de computadores, inicialmente criadas para a transfer?ncia de dados menos complexos, est? sendo agora usada por um servi?o que envolve altas taxas de transfer?ncia e apresenta requisitos em rela??o ? qualidade do servi?o (QoS) oferecido pela rede. Desta forma, este trabalho realiza uma an?lise e compara??o de desempenho de uma rede quando submetida a esse tipo de aplica??o, para duas situa??es distintas: a primeira sem o uso de pol?ticas de QoS, e a segunda com a aplica??o de tais pol?ticas, usando como cen?rio para os testes, a Rede Metropolitana de Sa?de da Universidade Federal do Rio Grande do Norte (UFRN)

Les réseaux de capteurs sans-fil sont composés de dispositifs électroniques conçus pour mesurer une grandeur physique de l’environnement dans lequel ils sont déployés et pour acheminer ces mesures à un système d’information. Dans nos travaux, nous étudions les architectures de réseau et les protocoles de communication associés afin de déterminer les configurations adéquates à un scénario de réseau de capteurs pour la Ville Intelligente. Après avoir recensé les applications et leurs exigences de Qualité de Service attendue, nous avons construit des modèles analytiques permettant de comparer les différentes familles de protocoles MAC en terme de taux de livraison et de consommation d’énergie. Ces travaux permettent ainsi de motiver le choix d’un protocole synchrone ou asynchrone, à contention ou à accès direct, en fonction du scénario applicatif et du déploiement de réseau considéré : longue portée ou multi-sauts. Sur la base des résultats de cette étude, nous proposons ensuite un ensemble d’optimisation des protocoles de communication permettant d’améliorer la Qualité de Service et la dépense énergétique des compteurs. Nous proposons une méthode d’accès au médium de communication permettant l’allocation de plusieurs instants de transmission en une phase unique de tournoi. Cette optimisation vise à réduire le pôle de consommation majoritaire des réseaux multi-sauts, tel qu’identifié dans l’étude précédente. Le protocole résultant, CT-MAC, résout l’allocation de plusieurs instants de transmission en une unique phase de compétition et de manière distribuée. CT-MAC réduit ainsi la consommation d’énergie de l’arbitrage du médium. Nous proposons ensuite un mécanisme de relayage adaptatif pour l’architecture de réseau longue-portée. Le protocole, nommé SARI-MAC, est conçu de manière à pallier aux ‘trous’ de couverture que présentent les systèmes radio longue-portée lorsqu’ils sont déployés dans un habitat urbain dense. Pour cela, SARI-MAC propose d’exploiter certains nœuds du réseau, choisis pour leurs ressources en énergie, en tant que relais pour leurs voisins dont le bilan radio est insuffisant pour respecter les exigences de Qualité de Service de l’application. SARI-MAC est un protocole MAC asynchrone, initié récepteur dont les paramètres s’adaptent automatiquement aux conditions de trafic et aux contraintes de cycle d’activité imposées par la durée de vie souhaitée des nœuds capteurs et par la législation. Finalement, nous proposons un schéma de routage opportuniste appelé QOR. QOR est un protocole de routage qui tire profit des liens radio longue portée, opportunistes afin d’améliorer la fiabilité et la latence des transmissions de données dans un réseau de collecte. Pour cela, QOR propose une structure de routage combinée à un schéma d’adressage permettant d’identifier un ensemble de nœuds relais entre une source et la passerelle de collecte. / Wireless Sensors are small electronic devices made for measuring physical properties of their environment and communicating them wirelessly to an information system. In this thesis, we study existing network architectures and to devise the best-suited configuration for typical urban wireless Sensor Network use-cases. To that effect, we provide comprehensive analytical models to compare the different families of MAC protocols in terms of Delivery Rate and Energy Consumption, e.g. synchronous vs asynchronous, contention-based vs direct access etc. Headlines results a mathematical framework to devise the least energy-cost contention algorithm for a given Delivery Rate and closed-form expressions of the Energy Consumption and Delivery Rate for popular access control protocols. These results are then synthesised in a comparison study of the two prevailing urban sensors network architectures, i.e. long-range and multihop. We show that long-range sensor networks are best-suited for low-traffic and sparser network topologies while higher traffic loads and denser network topologies demand switching to a multihop network operating a synchronous MAC protocol on higher bitrate radios. Based on the analysis of the architectures best suited for each use-case scenario, i.e. low traffic loads/sparse network and high traffic loads/dense network, we identify suitable optimisations to improve the QoS performance and energy efficiency of said architectures. First, we improve on the energy efficiency of the arbitration of the medium access by defining a cascading tournament contention algorithm. This protocol, CT-MAC, resolves multiple timeslots allocation in a single, energy efficient contention tournament. Second, we propose an adaptive relaying scheme for the long-range network architecture named SARI-MAC. This scheme is an attempt to cope with coverage holes that occurs when using long-range in a dense urban habitat by letting sensor nodes relay communications of nodes whose link budgets are incompatible with the QoS requirements of the network. To that effect, we propose a receiver-initiated MAC protocol that self-adapts to the traffic condition so that the duty-cycle of relayers is kept as low as possible with respect to the load of frames to relay. Finally, we propose an opportunistic relaying scheme named QOR. QOR is a routing protocol that exploits long-range, opportunistic radio links to provide faster and more reliable transmissions. To that effect, QOR proposes a joint routing structure and addressing scheme that allows identifying a limited set of nodes than can become opportunistic relayers between a source sensor and the sink. Those nodes then follow an original cascaded acknowledgement mechanism that brings reliable acknowledgment and ensures a replication-free forwarding of the data frames.