The routing strategy used in today's Internet is best-effort service, where all data packets are treated equally. This type of service is not suited for applications such as video conferencing, and video on demand, that requires the availability of certain resources (such as bandwidth) to be guaranteed for them to function properly. The routing in this context, called Quality-of-Service (QoS) Routing, is the problem of finding suitable paths that meet the application's resource requirements. The majority of proposed QoS routing schemes operate by maintaining the global state of the network, and using this knowledge to compute the QoS route. However, all these schemes suffer from the inherent drawback of scalability, because of the need for each node to collect state information about the complete network. The other type of QoS routing schemes do not maintain network state information, but instead flood the network with QoS connection establishment requests. This type of scheme suffers from excessive message overhead during QoS connection establishment. In this thesis, we present a new QoS routing algorithm that is a combination of the above-mentioned two schemes (i.e., global state and flooding based). The algorithm aims at minimizing the message overhead associated with these two schemes and still maintaining the positive aspects of both of them. The basic idea of the algorithm is: to reach to a destination, the path(s) will always pass through a specific set of intermediate nodes. The algorithm discovers such intermediate nodes (limited by a hop count threshold value needed to reach there). When a QoS connection request arrives at a node, it selects the feasible path leading to the intermediate node for the requested destination. The QoS connection establishment message (or routing message) is forwarded along this path. When the message arrives at the intermediate node, the further path is decided through same logic. To decide the path that leads to the intermediate node, the algorithm maintains the link state related to these intermediate nodes, and link state updates are restricted only with regard to these intermediate nodes. Because of this restriction in link state updation, one has less message overhead, compared to the global state based routing scheme. Further, the algorithm tries to group these intermediate nodes in such a way that the routing message need be sent to only one of the grouped intermediate nodes, and still makes sure that all the possible paths are covered. Therefore, one has a reduced message overhead because of grouping.

Currently, several mature and commercial component models (for e.g. EJB, .NET, COM+) exist on the market. These technologies were designed largely for applications with business-oriented non-functional requirements such as data persistence, confidentiality, and transactional support. They provide only limited support for the development of components and applications with non-functional properties (NFPs) like QoS (e.g. throughput, response time). The integration of QoS into component infrastructure requires among other things the support of components’ QoS contract specification, negotiation, adaptation, etc. This thesis focuses on contract negotiation. For applications in which the consideration of non-functional properties (NFPs) is essential (e.g. Video-on-Demand, eCommerce), a component-based solution demands the appropriate composition of the QoS contracts specified at the different ports of the collaborating components. The ports must be properly connected so that the QoS level required by one is matched by the QoS level provided by the other. Generally, QoS contracts of components depend on run-time resources (e.g. network bandwidth, CPU time) or quality attributes to be established dynamically and are usually specified in multiple QoS-Profiles. QoS contract negotiation enables the selection of appropriate concrete QoS contracts between collaborating components. In our approach, the component containers perform the contract negotiation at run-time. This thesis addresses the QoS contract negotiation problem by first modelling it as a constraint satisfaction optimization problem (CSOP). As a basis for this modelling, the provided and required QoS as well as resource demand are specified at the component level. The notion of utility is applied to select a good solution according to some negotiation goal (e.g. user’s satisfaction). We argue that performing QoS contract negotiation in multiple phases simplifies the negotiation process and makes it more efficient. Based on such classification, the thesis presents heuristic algorithms that comprise coarse-grained and fine-grained negotiations for collaborating components deployed in distributed nodes in the following scenarios: (i) single-client - single-server, (ii) multiple-clients, and (iii) multi-tier scenarios. To motivate the problem as well as to validate the proposed approach, we have examined three componentized distributed applications. These are: (i) video streaming, (ii) stock quote, and (iii) billing (to evaluate certain security properties). An experiment has been conducted to specify the QoS contracts of the collaborating components in one of the applications we studied. In a run-time system that implements our algorithm, we simulated different behaviors concerning: (i) user’s QoS requirements and preferences, (ii) resource availability conditions concerning the client, server, and network bandwidth, and (iii) the specified QoS-Profiles of the collaborating components. Under various conditions, the outcome of the negotiation confirms the claim we made with regard to obtaining a good solution.

Progressive download video services, such as YouTube and podcasts, are responsible for a major part of the transmitted data volume in the Internet and it is expected, that they will also strongly affect mobile networks. Streaming video quality mainly depends on the sustainable throughput achieved during transmission. To ensure acceptable video quality in mobile networks (with limited capacity resources) the perceived quality by the customer (QoE) needs to be monitored by estimation. For that, the streaming video quality needs to be measured and monitored permanently. For TCP based progressive download we propose to extract the the video timestamps which are encoded within the payload of the TCP segments by decoding the video within the payload. The actual estimation is then done by play out buffer fill level calculations based on the TCP segment timestamp and their internal play out timestamp. The perceived quality for the user is derived from the number and duration of video stalls. Algorithms for decoding Flash Video, MP4 and WebM Video have already been implemented. After deriving the play out time it is compared to the timestamp of the respective TCP segment. The result of this comparison is an estimate of the fill level of the play out buffer in terms of play out time within the client. This estimation is done without access to the end device. The same measurement procedure can be applied for any TCP based progressive download Internet service. Video was simply taken as an example because of its current large share in traffic volume in operator networks.

Progressive download video services, such as YouTube, are responsible for a major part of the transmitted data volume in the Internet and it is expected, that they also will strongly affect mobile networks. Streaming video quality mainly depends on the sustainable throughput achieved during transmission. In order to achieve an acceptable video quality in mobile networks (with limited capacity resources), traffic engineering mechanisms have to be applied. For that, the streaming video quality needs to be measured and monitored permanently. Therefore, the video timestamps which are encoded within the payload of the TCP segments have to be extracted. For that it is necessary to decode the video within the transported payload. Algorithms for decoding Flash Video, MP4 and WebM Video have already been implemented as a demonstration implementation in support of the network based measurement contribution to SG12 by Chemnitz University for TCP encoded progressive download Internet services. In the demonstration, the derived play out buffering from the monitored traffic is being output internally. A second application is then used to graphically display the estimation result. The measurement and estimation is solely done within a measurement point of an operator network without access to the client’s end device.

La tendance actuelle des réseaux de capteurs sans fil est d’avoir un seul réseau supportant plusieurs applications et fournissant plusieurs QoS. Dans cette thèse, nous étudions les techniques de cross-layering afin d’améliorer les performances et de fournir de la QoS. Tout d’abord, nous généralisons le concept de la méthode d’accès MaCARI en proposant une architecture multi-couches où plusieurs combinaisons de protocoles MAC-routage sont utilisées. Une file d’attente est associée à chaque combinaison, et chaque combinaison est activée pour une période précise. Le but est de profiter de ces combinaisons pour offrir différentes QoS. Cependant, cette architecture cause un problème de dimensionnement des périodes, ce qui a un impact sur les performances du réseau. Nous proposons, ensuite, des techniques de cross-layering en échangeant les paquets entre les différentes files d’attente afin de résoudre le problème de dimensionnement. Durant sa période, chaque combinaison traite tous les paquets de sa file d’attente ainsi que les paquets des files d’attente d’autres périodes. Nous montrons par simulation que notre approche améliore les performances du réseau. / The current trend in wireless sensor networks is to have a single network supporting serveral applications and providing several QoS. In this thesis, we study the cross-layering techniques in order to improve the network performance and provide several QoS. Fistly, we generalize the concept of the access method MaCARI by proposing a multi-stack architecture in which several MAC-routing combination protocols are used. A queue is associated to each combination, and each combination is active for a specified period. The purpose consists in using these combinations in order to provide different QoS. However, this architecture yields to a dimensioning problem for the periods reducing the network performance. Secondly, we propose cross-layering techniques by exchanging packets between different queues to solve the dimensioning problem. During its period, each combination treats all the packets of its queue and the packets related to queues associated to other periods. We show by simulations that our approach improves the network performance.

Submitted by Fernando Souza (fernandoafsou@gmail.com) on 2017-08-16T11:08:57Z No. of bitstreams: 1 arquivototal.pdf: 14592184 bytes, checksum: e1af48f8997b6af87f0fc91dd6d4a697 (MD5) / Made available in DSpace on 2017-08-16T11:08:57Z (GMT). No. of bitstreams: 1 arquivototal.pdf: 14592184 bytes, checksum: e1af48f8997b6af87f0fc91dd6d4a697 (MD5) Previous issue date: 2016-08-31 / One of the most important aspects concerning the implementation of a Wireless Local Area Network (WLAN) is that in an area covered by several Access Points (APs), there must exist mechanisms to keep the user connected at any time, regardless of his/her physical location in the network and, primarily, ensure that the applications do not suffer from loss of connectivity. In this context, handoff performance, which consists of changing the AP as the station moves within the same area, is a preponderant factor when it comes to applications that are delay-sensitive. The process is usually started by the stations that once disconnected from an AP, start scanning the existing channels searching for other available APs in a smooth fashion, but in the order of seconds, which makes it difficult to use real-time applications. The challenge is to create a more efficient mechanism to manage this process, providing the required Quality of Service (QoS) for the applications. This work presents the use of mechanisms to improve the performance of the handoff process by employing an approach based on the a Software DefinedWireless Network (SDWN), which was implemented to validate the proposal. The tests performed showed a decrease in the time spent in the handoff, contributing to guarantee the QoS of the applications. / Um dos aspectos mais importantes na implantação de uma rede local sem fio Wireless Local Area Network (WLAN) é que em uma área coberta por vários Access Points (APs), devem existir mecanismos que sejam suficientes para que o usuário possa se manter conectado a qualquer hora independentemente de sua localização física dentro da rede e, principalmente, garantir que as aplicações não sofram com falhas ou perdas de conexão. Neste contexto, o desempenho do handoff, que consiste na troca de AP à medida que uma estação se move dentro de uma mesma área, é fator determinante para o uso de aplicações sensíveis ao atraso. Normalmente o processo é iniciado pelas estações, que ao se desconectarem de um AP iniciam uma varredura nos canais existentes buscando outros APs disponíveis de forma suave, porém na ordem de segundos, o que dificulta o uso de aplicações em tempo real. O desafio é criar um mecanismo mais eficiente para gerenciar este processo provendo a Qualidade de Serviço (QoS) necessária para as aplicações. Este trabalho apresenta uma solução denominada Detection and dIScovery Control in Handoff (DISpatCH) que utiliza mecanismos para melhorar o desempenho do processo de handoff utilizando uma abordagem baseada na arquitetura Software Defined Wireless Network (SDWN) os quais foram implementados para validar a proposta. Testes realizados demonstraram uma diminuição no tempo gasto no handoff contribuindo para garantir o QoS das aplicações.

Made available in DSpace on 2014-12-17T14:55:03Z (GMT). No. of bitstreams: 1 EdsonMSN.pdf: 1297152 bytes, checksum: ddeef5119a3b863368e35112b7fbbd68 (MD5) Previous issue date: 2006-08-02 / New multimedia applications that use the Internet as a communication media are pressing for the development of new technologies, such as: MPLS (Multiprotocol Label Switching) and DiffServ. These technologies introduce new and powerful features to the Internet backbone, as the provision of QoS (Quality of Service) capabilities. However, to obtain a true end-to-end QoS, it is not enough to implement such technologies in the network core, it becomes indispensable to extend such improvements to the access networks, what is the aim of the several works presently under development. To contribute to this process, this Thesis presents the RSVP-SVC (Resource Reservation Protocol Switched Virtual Connection) that consists in an extension of RSVP-TE. The RSVP-SVC is presented herein as a mean to support a true end-to-end QoS, through the extension of MPLS scope. Thus, it is specified a Switched Virtual Connection (SVC) service to be used in the context of a MPLS User-to-Network Interface (MPLS UNI), that is able to efficiently establish and activate Label Switched Paths (LSP), starting from the access routers that satisfy the QoS requirements demanded by the applications. The RSVP-SVC was specified in Estelle, a Formal Description Technique (FDT) standardized by ISO. The edition, compilation, verification and simulation of RSVP-SVC were made by the EDT (Estelle Development Toolset) software. The benefits and most important issues to be considered when using the proposed protocol are also included / O protocolo proposto nesta Tese, denominado Resource Reservation Protocol Switched Virtual Connection (RSVP-SVC), que consiste numa extens?o do RSVP-TE, vai de encontro ao surgimento de novas aplica??es multim?dia, que usam a Internet como meio de interconex?o. Tais aplica??es pressionam pelo desenvolvimento de novas tecnologias, tais como: MPLS, DiffServ e RSVP-TE, que introduzem novas e eficientes caracter?sticas ao backbone Internet, proporcionando uma significativa diferen?a no que tange ? provis?o de QoS (Quality of Service). O presente trabalho leva em conta o fato de que para se conseguir uma QoS fim-a-fim verdadeira, n?o basta implementar tais tecnologias no n?cleo da rede, ? imprescind?vel, estender tais melhorias ?s redes de acesso e qui?? ?s redes locais. Nesse sentido, muitos trabalhos est?o atualmente em desenvolvimento. ? no intuito de contribuir com este processo que este trabalho apresenta a defini??o de uma UNI MPLS SVC atrav?s do RSVP-SVC. Essa extens?o d? ao RSVP-TE a capacidade de estabelecer t?neis LSP (Label Switched Path) a partir de conex?es discadas, ampliando portanto o escopo de utiliza??o do MPLS, levando-o at? ?s redes locais atrav?s das redes de acesso, e provendo tamb?m suporte a uma QoS fim-a-fim verdadeira. O RSVP-SVC foi especificado em Estelle, que ? uma linguagem de especifica??o formal padronizada pela ISO. A edi??o, compila??o, verifica??o e simula??o do RSVP-SVC foi feita atrav?s do programa EDT (Estelle Development Toolset). Ademais, tanto os benef?cios quanto ?s quest?es mais importantes a serem consideradas quando do uso deste protocolo s?o apresentados

Esta dissertação de mestrado aborda a Composição de Web services baseada em atributos de Qualidade de Serviço (CWSbQ). Foi selecionado o domínio de sistemas de e-commerce, por se tratar de uma área em franca expansão, tanto no cenário nacional como internacional. Além disso, os sistemas de e-commerce podem se beneficiar da CWSbQ devido à sua interoperabilidade e atendimento aos requisitos de qualidade de serviço de seus clientes. Foram implementados dez algoritmos para resolver esse problema, usando várias técnicas diferentes, sendo elas: busca aleatória, busca exaustiva, busca heurística, busca meta-heurística e um algoritmo híbrido. Foram realizados experimentos de avaliação de desempenho, considerando tamanhos de espaço de busca diferentes e dois deadlines estabelecidos. As principais contribuições deste trabalho foram o desenvolvimento de três novos algoritmos: Heurística Gulosa (HG), Heurística Gulosa 2 (HG2) e Algoritmo Genético Duplamente Híbrido (AGDH); o uso de planejamento de experimentos e avaliação de desempenho para aperfeiçoar Algoritmos Genéticos (AGs) usados no problema de CWSbQ; e o estudo de adequação dos algoritmos de CWSbQ para diferentes situações / This MSc dissertation addresses the QoS-aware Web services composition (QWSC) problem. The field of e-commerce systems was selected because it comprises an area in wide expansion, both in national and international scenarios. Furthermore, e-commerce systems can benefit from QWSC due to its interoperability and compliance to the requirements of quality of service of the customers. Ten algorithms were implemented to solve this problem by using several different techniques, namely: random search, exhaustive search, heuristic search, meta-heuristic and a hybrid algorithm. Performance evaluation was made considering different search space sizes for two different established deadlines. The main contributions of this work were the development of three new algorithms: Greedy Heuristic (GH), Greedy Heuristic 2 (GH2) and Doubly Hybrid Genetic Algorithm (DHGA); the use of planning of experiments and performance evaluation to improve the genetic algorithms (GAs) used in QWSC problem; and a suitability study of QWSC algorithms for different situations

Submitted by Luciana Ferreira (lucgeral@gmail.com) on 2015-10-22T10:13:08Z No. of bitstreams: 2 Dissertação - Diego Cruz Abrahão - 2015.pdf: 9765017 bytes, checksum: 6f29592215e8451edda4aa5ad24238f4 (MD5) license_rdf: 23148 bytes, checksum: 9da0b6dfac957114c6a7714714b86306 (MD5) / Approved for entry into archive by Luciana Ferreira (lucgeral@gmail.com) on 2015-10-22T10:15:40Z (GMT) No. of bitstreams: 2 Dissertação - Diego Cruz Abrahão - 2015.pdf: 9765017 bytes, checksum: 6f29592215e8451edda4aa5ad24238f4 (MD5) license_rdf: 23148 bytes, checksum: 9da0b6dfac957114c6a7714714b86306 (MD5) / Made available in DSpace on 2015-10-22T10:15:40Z (GMT). No. of bitstreams: 2 Dissertação - Diego Cruz Abrahão - 2015.pdf: 9765017 bytes, checksum: 6f29592215e8451edda4aa5ad24238f4 (MD5) license_rdf: 23148 bytes, checksum: 9da0b6dfac957114c6a7714714b86306 (MD5) Previous issue date: 2015-03-19 / In this paper we propose two schemes for allocating resource blocks for transmission LTE downlink to maximize the throughput of the system, to guarantee QoS (Quality of Service) parameters for the users and reduce the data loss rate of network. The first proposed scheme uses the Max-min criterion and the second employs a fuzzy inference system to calculate the priorities of users and make scheduling decisions. Both schemes use an estimated effective bandwidth of traffic flows of users. The effective bandwidth of a traffic flow is the rate required to meet a criterion of probability of data loss rate. In this work, the effective bandwidth is estimated adaptively by the parameters of multifractal modeling βMWM (β-Multifractal Wavelet Mode). Are made simulations of the algorithms proposed, considering different propagation models with multipath fading and with different numbers of users in the network. The simulation results are compared with other algorithms presented in the literature, using parameters such as: throughput of the system, data loss rate and fairness index. It is also proposed to use predict of the SNR (Signal-to-Noise Ratio) of users, in the scheduling algorithms, using linear prediction of multi-step filter, in view of the delay in receipt of the channel quality information of the users in the base station and the variation of the signal propagation conditions. The multi-step prediction filter is used with the algorithms of allocation of resource blocks proposed in this work to estimate the signal-to-noise-ratio of users and set well, modulation schemes and code to be used in the LTE network. / Neste trabalho são propostos dois esquemas de alocação de blocos de recurso para a transmissão de downlink LTE, visando maximizar a vazão de dados total do sistema, garantir parâmetros de QoS (Quality of Service) dos usuários e diminuir a taxa de perda de dados da rede. O primeiro esquema proposto utiliza o critério Max-min e o segundo emprega um sistema de inferência fuzzy para calcular as prioridades dos usuários e tomar decisões de escalonamento. Ambos os esquemas utilizam estimações de banda efetiva dos fluxos de tráfego dos usuários. A banda efetiva de um fluxo de tráfego é a taxa necessária para atender a um critério de probabilidade de perda de dados. Neste trabalho, a banda efetiva é estimada de forma adaptativa, através dos parâmetros da modelagem multifractal βMWM (β-Multifractal Wavelet Mode). São realizadas simulações dos algoritmos propostos de alocação de blocos de recurso, considerando diferentes modelos de propagação com desvanecimento de multipercursos e com diferentes quantidades de usuários na rede. Os resultados das simulações são comparados com outros algoritmos apresentados na literatura, através de parâmetros, como: vazão de dados do sistema, taxa de perda de dados e índice de justiça. É proposta ainda, a utilização da predição da relação sinalruído (SNR - Signal-to-Noise Ratio) dos usuários nos algoritmos de escalonamento, utilizando filtro de predição linear de multi-passos, tendo em vista o atraso no recebimento da informação da qualidade do canal dos usuários na estação base e a variação das condições de propagação do sinal. O filtro de predição de multi-passos é utilizado junto com os algoritmos de alocação de blocos de recurso propostos neste trabalho, para estimar a relação sinal-ruído dos usuários e definir, assim, os esquemas de modulação e código a serem utilizados na rede LTE.

Les réseaux sans fil sont en plein développement du fait de la flexibilité de leur interface, qui permet aux utilisateurs de se communiquer directement entre eux ou de se connecter facilement à Internet en onde radio sans mettre en place préalablement d'infrastructures lourdes, telles que des câbles filaires. Parmi les différentes technologies de réseaux sans fil, l'IEEE 802.11/Wi-Fi est devenu une technologie plus connue et plus utilisée pour construire des réseaux sans fil à haut débit dans une zone à forte concentration d'utilisateurs, telle que les aéroports, les campus ou les sites industriels. L'engouement pour les réseaux sans fil et notamment pour les réseaux Wi-Fi a fait émerger de nouvelles nécessités, tel que se déplace dans les réseaux sans fil tout en restant connecté à Internet. Dans les réseaux sans fil, le déplacement d’un utilisateur implique parfois un changement de Point d’accès (AP) au réseau. On désigne généralement ce fait un handover de niveau 2, du fait que le changement d'AP n’implique que les deux premières couches du modèle OSI. Si les deux APs se situent dans des réseaux différents, le changement d'AP implique aussi le changement de réseau pour cet utilisateur. On dénomme généralement cette situation un handover de niveau 3, par le fait que cet utilisateur devrait changer son réseau d’attachement et son adresse IP pour maintenir la connexion à Internet et que ce changement intervient sur la couche réseau du model OSI. La procédure du handover de niveau 2 dans les réseaux Wi-Fi est gérée par la norme IEEE 802.11 et celle de niveau 3 est gérée par le protocole IP Mobile. Le protocole IP Mobile est un protocole standardisé par l'IETF qui permet à l'utilisateur de maintenir ses communications en cours et de rester connecté à Internet tout en masquant d'une manière transparente le changement de réseau. Ainsi, l'utilisateur peut se déplace dans les réseaux Wi-Fi tout en maintenant les communications en cours et restant connecté à Internet grâce à la norme IEEE 802.11 et au protocole IP Mobile. Cependant, le délai introduit par ces deux procédures du handover est trop long, les communications en cours sont interrompus pendant ces procédures, naturellement, cela ne peut pas répondre aux exigences qualitatives des applications temps réel comme la vidéo conférence ou la voix sur IP. Diverses propositions qui ont été faites pour réduire le délai de ces procédures du handover et améliorer leur performance. Cependant, ces propositions sont soit imparfaites, soit non-implémentables à cause de leur complexité. En partant du principe que les réseaux Wi-Fi et les routeurs d'accès sont déjà massivement implantés dans le monde universitaire et dans les entreprises, nous proposons d'ajouter une nouvelle fonctionnalité, appelé E-HCF (Extended Handover Control Function) dans un routeur sans modifier les autres équipements du réseau. Le routeur pourvu de cette fonctionnalité est dénommé le routeur E-HCF. Pour réduire le délai des procédures du handover, la fonctionnalité E-HCF permet au routeur de générer une topologie des APs en utilisant la théorie des graphes de voisinage et de maintenir un pool d'adresses IP disponibles dans sa base de données. Quand le Nœud mobile (MN) a besoin de changer son AP, le routeur E-HCF peut proposer au MN une liste des APs potentiellement utilisables qui sont choisis et classés par un algorithme de sélection et de classement que nous avons élaboré dans la thèse. Si le changement d'AP implique un changement de réseau, le MN doit changer d'adresse IP. Dans ce cas, le routeur E-HCF peut attribuer une adresse IP unique à ce MN. Le MN peut donc utiliser cette adresse sans exécuter la phase d'Auto-configuration d'adresses ni exécuter la procédure de Détection d'adresse dupliquée. Avec cette nouvelle fonctionnalité E-HCF, nous pouvons réduire le délai des procédures du handover de quelques secondes à une centaine de millisecondes. / Wireless networks are in full development because of the flexibility of their interfaces, which allow users to be easily connected to the Internet. Among various technologies of wireless networks, IEEE 802.11/Wi-Fi technology is becoming better known and more used to construct high speed wireless networks in areas with high concentration of users, such as airports, campuses or industrial sites. The passion for wireless networks and in particular for Wi-Fi networks has given rise to new uses of the Internet, such as moving in wireless networks while still being connected. In Wi-Fi networks, the user's movement may sometimes lead to a change of Access Points (APs) to the network. This fact is generally named the handover of layer 2 because this change involves only the first two layers of the OSI model. If the two APs are located in different networks, the change of AP would entail a change of network for the user. This situation is generally termed, the handover of layer 3 because the user should change his network and his IP address to maintain connection to the Internet. Therefore, this change intervenes on the network layer of the OSI model. The process of the handover of layer 2 is handled by the IEEE 802.11 standard and that of layer 3 is controlled by the Mobile IP protocol. The Mobile IP protocol is a protocol standardized by IETF, which allows users to change network, while maintaining their actual connection to the Internet. Consequently, users can connect to the Internet, while keep moving in Wi-Fi networks in control of the IEEE 802.11 standard and the Mobile IP protocol. However, the delay induced by these procedures of handover is too long. As such, this generally leads to the cut-off of current communications, hence impacting adversely on the qualitative requirements of real-time applications, such as video conferencing or voice over IP. Various proposals have been made to reduce the delay of handover procedures and to improve their performances. However, these proposals are either imperfect, or non-implementable because of their complexity. Based on the premise that Wi-Fi networks and access routers are already massively implanted in academia and in industry, we propose to add a new functionality, called E-HCF (Extended Handover Control Function) in routers, without modifying other network equipments. A router equipped with this functionality is called an E-HCF router. To reduce the delay of handover procedures, the E-HCF functionality allows a router to generate a topology of APs by using the neighbourhood graph theory and to maintain a pool of available IP addresses in its database. When a Mobile Node (MN) needs to change its AP, the E-HCF router may propose to the latter a list of potentially usable APs, which are selected and classified by an algorithm of selection and classification that we developed in the thesis. If the change of APs involves a change of network, the MN must change its IP address. In this case, the E-HCF router can assign a unique IP address to this MN. The MN can thus use this address without engaging in the process of Stateless Address Autoconfiguration or the procedure of Duplicate Address Detection. With this new E-HCF functionality, we can reduce the delay of handover procedures from a few seconds to one hundred milliseconds.