Les réseaux sans fil multi-saut (WMN : Wireless multi-hop Networks) sont passés du stade de simple curiosité pour revêtir aujourd'hui un intérêt certain aussi bien du point de vue de la communauté de recherche que des opérateurs de réseaux et services. En analysant les services et applications fournis au sein des réseaux WMNs, nous pouvons constater que certaines applications telles que la visioconférence, la VoIP, etc sont sensibles au délai et nécessitent une certaine qualité de service (QoS). D'autres applications telles que le transfert de fichier, le streaming vidéo, etc. sont gourmands en terme d'utilisation de bande passante. Par conséquent, les architectures de communication des réseaux WMNs doivent intégrer des mécanismes de routage efficaces et adaptés pour répondre aux besoins des services et applications envisagés. Dans cette thèse, Nous nous intéressons à la problématique du routage dans les réseaux WMNs. Notre objectif est de proposer une nouvelle approche de routage qui prend en compte différents métriques de coûts. Tout d'abord, nous avons montré que le routage sous contraintes multiples est un problème NP complet et que trois étapes sont nécessaires à la conception d'une nouvelle solution de routage: (i) modélisation de l'interférence, (ii) l'estimation de la de la bande passante restante, (iii) l'estimation du délai à un saut. Suivant cette vision, nous avons proposé deux variantes du protocole de routage OLSR (SP-OLSR, S2P-OLSR) se basant sur la métrique SINR. Les résultats des simulations ont montré l'intérêt de la proposition dans un contexte de communication vocale (VoIP). Ensuite, nous avons proposé un algorithme d'estimation d'interférence à 2 sauts (2-HEAR) afin d'estimer la bande passante disponible. Puis, et sur la base de cet algorithme, nous avons proposé une nouvelle métrique de routage pour les WMNs: Estimated Balanced Capacity (EBC) en vue de parvenir à l'équilibrage de charge entre des différents flux. La dernière question abordée dans cette thèse est celle de l'estimation du délai à un saut. La solution proposée donne une borne du délai en se basant sur un modèle de file d'attente de type G/G/1. Enfin, nous avons englobé toutes les précédentes contributions pour mettre en place une nouvelle approche de routage hybride sous contraintes multiples. Ce protocole comporte une partie proactive utilisant la nouvelle métrique de routage (EBC) et une partie réactive qui permet de prendre en compte le délai relative à une connexion donné. / There is a growing interest in wireless multi-hop networks (WMNs) since there are promising in opening new business opportunity for network operators and service providers. This research field aims at providing wireless communication means to carry different types of applications (FTP, Web browsing, video streaming, in addition to VoIP). Such applications have different constraints and their specific requirements in terms of Quality of Service (QoS) or performance metrics (delay jitter, end-to-end delay). We examine, in this thesis, the problem of routing in WMNs. Our main goal is to propose a new multi-metrics routing capable to fit these particular needs. In this thesis, we make several contributions toward WMN multi-constrained routing. First, we show that the multi-constrained path finding problem is NP-Complete and inherently a cross-layer issue, and that three steps are necessary to design the multi-metric routing protocol: (i) modeling of the inferring signal, (ii) estimation of the remaining bandwidth, (iii) estimation of the one-hop delay. Second, moving in such direction, we propose two enhanced versions of the OLSR routing protocol. The suggested protocols consider the SINR as a routing metric to build a reliable topology graph. Performance evaluation shows that utilizing such routing metric helps to improve significantly the VoIP application quality in the context of ad hoc network while maintaining a reasonable overhead cost. Third, we have proposed a 2-Hop interference Estimation Algorithm (2-HEAR) in order to estimate the available bandwidth. Then, and based on such algorithm, we have proposed a novel routing metric for WMNs: Estimated Balanced Capacity (EBC) in order to achieve load-balancing among the different flows. The next issue tackled in this thesis is the one-hop delay estimation, the one-hop delay is estimated by means of an analytical model based on G/G/1 queue. Finally, we have encompassed all the previous contributions to address our main goal, i.e. the design of a multi-constrained routing protocol for WMNs. A hybrid routing protocol is then proposed. This protocol is a junction of two parts : a proactive part that makes use of the previously estimated constraint, and a reactive part, which is triggered ”on demand” when news applications are expressed.

Ce manuscrit synthétise les activités de recherche que nous avons menées, au cours des ces dernières années dans le domaine de la gestion de réseaux, et contient une présentation des enjeux pour la décénie à venir. Il contient également un projet de recherche ambitieux visant á répondre aux défis de la sécurité de l'Internet du futur. Nous présentons nos travaux sur l'audit de sécurité en abordant le problème du "fingerprinting", c'est à dire la détection par prise d'empreintes d'une couche protocolaire et/ou d'un équipement. La prise d'empreintes d'un système est une action essentielle dans l'audit de sécurité d'un réseau. L'objectif de ce processus consiste dans la détection d'une version spécifique d'un service/équipement par l'analyse du trafic véhiculé sur le réseau. Nous avons elaboré deux approches de "fingerprinting". La première s'appuie sur l'analyse des arbres d'analyse syntactique pour les messages d'un protocole. Nous abordons ensuite la problématique du "fingerprinting" pour le cas des protocoles dont nous ne disposons pas de spécifications. L'approche que nous présentons est capable d'inférer les divers types de messages et de construire une (ou plusieurs) machine d'états. Ces machines d'états sont les fondements pour définir le comportement. Nous proposons à la suite une approche de fingerprinting qui permet la prise en compte du comportement d'une souche protocolaire. Nous développons ensuite la problématique liée à la surveillance de sécurité d'un réseau. Celle-ci comprend deux parties principales: le monitoring pour la détection d'intrusions et un pot de miel pour la VoIP. La dernière contribution est une approche pro-active élevée pour la détection de failles de sécurité par un processus de type "fuzzing".

<p>The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.</p>

<p>Session Initiation Protocol (SIP) arises as a new standard of establishing and releasing connections for vast variety of multimedia applications. The protocol may be used for voice calls, video calls, video conferencing, gaming and many more.</p><p>The 3GPP (3^rd Generation Partnership Project) suggests SIP as the signalling solution for 3^rd generation telephony. Thereby, this purely IP-centric protocol appears as a promising alternative to older signalling systems such as H.323, SS7 or analog signals in PSTN. In contrast to them, SIP does not focus on communication with PSTN network. It is more similar to HTTP than to any of the mentioned protocols. </p><p>The main standardisation body behind Session Initiation Protocol is The Internet Engineering Task Force (IETF). The most recent paper published on SIP is RFC 3261 [5]. Moreover, there are working groups within IETF that publish suggestions and extensions to the main standard. One of those extensions is “Caller Preferences for the Session Initiation Protocol (SIP)” [1]. </p><p>This document describes a set of new rules that allow a caller to express preferences about request handling in servers. They give ability to select which Uniform Resource Identifiers (URI) a request gets routed to, and to specify certain request handling directives in proxies and redirect servers. It does so by defining three new request header fields, Accept-Contact, Reject-Contact, and Request-Disposition, which specify the caller preferences. [1]. </p><p>The aim of this project is to extend the existing software with caller preferences and evaluate the new functionality.</p>

<p>Traditional public switched telephone networks (PSTN) are replaced more and more by VoIP services these days.  Although it is good for saving costs, the disadvantage of this development is that VoIP networks are less secure than the traditional  way of transmitting voice. Because VoIP networks are being deployed in open environments and rely on other network  services, the VoIP service itself becomes vulnerable to potential attacks against its infrastructure or other services  it relies on.</p><p>This thesis will present a discussion of security issues of the Session Initiation Protocol (SIP), the signalling protocol for  VoIP services. The main focus is on active attacks against the protocol that aim to reduce the service's availability -- so called  Denial of Service (DoS) attacks.</p><p>Existing countermeasures and detection schemes do not adequately differentiate between DoS attacks. However, the differentiation  is important with respect to performance loss, as various protection schemes involve more computationally intensive processes.</p><p>Based on that discussion, this thesis attempts to provide an ontological approach to describing, and eventually preventing attacks from  having their intended effects.</p>

The success of the Internet has brought significant changes to the telecommunication industry. One of the remarkable outcomes of this evolution is Voice over IP (VoIP), which enables realtime voice communications over packet switched networks for a lower cost than traditional public switched telephone networks (PSTN). Nevertheless, security and privacy vulnerabilities pose a significant challenge to hindering VoIP from being widely deployed. The main object of this thesis is to define and elaborate unexplored security and privacy risks on standardized VoIP protocols and their implementations as well as to develop suitable countermeasures. Three research questions are addressed to achieve this objective: Question 1:  What are potential unexplored threats in a SIP VoIP network with regard to availability, confidentiality and privacy by means of unwanted traffic and information disclosure? Question 2:  How far are existing security and privacy mechanisms sufficient to counteract these threats and what are their shortcomings? Question 3:  How can new countermeasures be designed for minimizing or preventing the consequences caused by these threats efficiently in practice? Part I of the thesis concentrates on the threats caused by "unwanted traffic", which includes Denial of Service (DoS) attacks and voice spam. They generate unwanted traffic to consume the resources and annoy users. Part II of this thesis explores unauthorized information disclosure in VoIP traffic. Confidential user data such as calling records, identity information, PIN code and data revealing a user's social networks might be disclosed or partially disclosed from VoIP traffic. We studied both threats and countermeasures by conducting experiments or using theoretical assessment. Part II also presents a survey research related to threats and countermeasures for anonymous VoIP communication.

The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.

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Das Ziel dieser Arbeit ist es, ein Handover-Verfahren für ein Siemens Handset zu entwickeln. Die Entwicklungsumgebung beruht auf den Wireless-LAN Standards 802.11 der IEEE (Institute of Electrical and Electronics Engineers). Dabei liegen die Schwerpunkte auf den Standardisierungen 802.11f und 802.11i, wobei sich eine neue Arbeitsgruppe (IEEE 802.11r) direkt mit dem Thema "Handover" beschäftigen wird. Das Handset soll selbständig die Verwaltung und Einleitung des Handovers übernehmen und lediglich insofern vom Access Point unterstützt werden, dass dieser als Informationssammler dient und somit Entscheidungshilfen geben kann.

Eine kurze Einführung zu "Voice over IP" (dem Telefonieren über Datennetze). Es wird ein Überblick über technische Anforderungen und Lösungen geben. Behandelte Gebiete sind Audio-Codecs, das Transportprotokoll RTP sowie die Signalisierungsdienste SIP und H.323.