NXP Semiconductor à Caen ayant des compétences dans le développement destechnologies System in Package (SiP) et NXP Semiconductor à Eindhoven ayant unespécialité en qualification virtuelle, deux partenariats ont été mis en place pour réaliser uneétude sur la qualification accélérée des composants SiP. Une thèse orientée simulations a étéréalisée à l'université de Delft (Pays-Bas) par Xiaosong Ma et dirigée par Kaspar Jansen, enparallèle une thèse plus expérimentale a été réalisée avec l'université de Bordeaux 1 parCharles Regard, à Caen, et dirigée par Hélène Frémont. Ces deux thèses ont été effectuées enproche collaboration. Dans un premier temps, des véhicules de test ont été définisconjointement. Puis un ensemble de caractérisations des matériaux et de simulations a étémené à Delft, alors que des essais expérimentaux de qualification et des analyses dedéfaillance étaient menés à Caen. Tout au long de ces deux thèses, des échanges constants ontété entretenus afin de corréler les simulations par les expérimentations. Ce besoin industrield'étude sur la qualification des composants SiP vient de la très forte augmentation del'intégration des fonctions au sein des équipements mobiles. En effet la technologie SiPpermet de répondre dans des délais intéressants aux nécessités de miniaturisation imposéespar ces nouveaux développements.L'objectif de ce travail de thèse est donc de mettre en place des méthodes et destechniques pour optimiser la qualification des composants System in Package (SiP). / NXP Semiconductor at Caen, which has System in Package (SiP) technologiesdevelopment competences and NXP Semiconductor at Eindhoven, which has virtualqualification specialization jointed to study fast reliability qualification of SiP products. Athesis focused on simulations started at TU Delft University (Netherlands) with Xiaosong Madirected by Kaspar Jansen and another thesis focused on experimentation started atBordeaux 1 University with Charles Regard directed by Hélène Frémont. These thesis werelead on close collaboration. In a first time, the tests vehicles were defined by both the PhDstudents. Then materials characterizations and simulations were performed at TU Delft, andexperimental qualification tests and physical analyses were performed in Caen. A long ofthese thesis, constant exchanges allowed to correlate simulations by experimentation. Thisindustrial need of SiP product qualification study is due to the strong increase of functionsintegration into mobile equipments. Thus the SiP technology allows to provide in relativeshort time miniaturized products imposed by the new developments.This thesis work's goal is to get methods and techniques to optimize System inPackage (SiP) reliability qualification.

With the emergence of multimedia applications and the upcoming age of Voice over IP (VoIP), Voice setup and resources control protocols such as SIP and H.323 over the Internet are becoming increasingly attractive applications. In the last few years as a real competitor in traditional telephony services (PSTN), SIP has gained much attention when compared with H.323. SIP works at presentation and application layer thus it mainly faces security issue at these layers. The objective of this thesis is to describe the most relevant SIP related security issues and then present security mechanisms that can be deployed to overcome the SIP security related issues. This project work demonstrates the tasks necessary to enhance the SIP security both inside and outside of the network. It is divided into three main parts, where the first part describes the SIP architecture, for example, the SIP rivals, SIP components and how a SIP system works. The second part is about some vulnerability issues of concern to SIP, study of the proposed security mechanism and also analysis on how possible threats to the SIP system such as call hijacking, message tempering and DoS attack, affect the SIP based VoIP system. The third and final part describes different steps that have been taken to avoid SIP attacks, by implementing some of the proposed security mechanisms. In order to test the SIP security, a SIP model is designed, which based on security mechanisms such as firewall, IPSec, DMZ and SIP-TLS. The results are conducted into two different scenarios. In the 1st scenario, the SIP system is tested before implementing the security measurements. In this case, the insecure system was vulnerable to several SIP attacks such as call hijacking, DOS and message tampering. In the 2nd scenario, the system is tested after the implementation of the proposed security mechanisms, where by the system now is only accessible to the authorized users and services. The tested results are also compared and discussed at the end.

Afin d'améliorer la qualité de service offerte aux abonnés, il est nécessaire de faire évoluer les réseaux d'accès GPRS/EDGE. Cette thèse se propose d'étudier différentes problématiques qui visent à améliorer les débits offerts aux utilisateurs et à mettre en oeuvre un mécanisme de handover qui réponde à leurs besoins. La première partie de cette thèse étudie différents mécanismes d'allocation dynamique de ressources sur l'interface Abis. Ces mécanismes visent en particulier à permettre le déploiement de la technologie E-GPRS tout en préservant la structure de l'interface Abis existante, qui s'appuie sur une structure de trame MIC. La seconde partie de cette thèse analyse différentes approches de handover qui peuvent être mises en oeuvre pour assurer la mobilité dans les réseaux E-GPRS. Nous y formulons quelques propositions pour améliorer les performances du basculement et mettre en place un véritable handover. Nous présentons les évolutions récentes de la normalisation, puis exposons les résultats de nos simulations qui permettent de comparer les performances des différents mécanismes. Notre étude montre les gains de performances obtenus par la préservation des états de transmission au niveau RLC. La troisième partie de cette thèse considère le problème des handover inter-systèmes. Nous y avons analysé le basculement entre une station de base E-GPRS et un point d'accès WIFI intégré au réseau d'accès mobile. Nous avons analysé l'impact du passage d'une pile protocolaire à une autre, ainsi que de la rupture de débit. Nos résultats montrent que l'introduction d'une couche de convergence au niveau liaison de données permet de limiter l'impact du handover au niveau réseau d'accès. Nous analysons, par ailleurs, les mécanismes de handover à mettre en place pour le transfert de données en mode Streaming. Nous démontrons l'intérêt de l'activation de la couche de liaison de données dans le cas de transferts avec le protocole UDP. La quatrième partie étudie les mécanismes de compression qui peuvent être mis en oeuvre pour réduire la taille des messages de signalisation SIP. Cette étude vise à réduire le temps de transmission de la signalisation - à travers des bearer bas débit - et à économiser l'utilisation des ressources radio.

At beginning of this paper, it will introduce the SIP protocol major used on the VoIP service at present time. Introduce the development and the application of SIP. And then explain the operation of the SIP protocol by an example. Explain how a SIP client establish the communication with another one by the SIP proxy. After understanding the operation of the SIP, explain why the SIP client behind NAT will meet the fail and introduce the solutions that have developed to conquer the fail. Briefly describe each of these solutions and compare all of them. Next is to describe how the solution of statteful SIP proxy used to avoid the fail of NAT changes the behavior between the SIP clients from end-to-end to master-to-slave. The change of behavior between the SIP clients may affect the quality of the communication in some conditions. To improve this, this paper brings out an idea and implements it. Finally, test the quality of the communication, catch the packet carrying the SIP messages and packet of RTP, and watch the content of SIP messages and the routing of the RTP packet.

Telecommunication industry in national legal norm must be able to provide lawful interception functions of offenders phone. The traditional PSTN and GSM have had such a system that can provide investigating authorities to monitor telephone and mobile phone users. In the meanwhile, IP telephony must provide the same monitor functions. However, the current SIP-based IP telephony is still unable to provide this monitoring function. In my thesis, I designed and implemented a monitoring system structure over SIP. It can efficiently carry out lawful interception without violating SIP communication. Additionally, it will not cause any overload on server, but will be able to monitor immediately. The recorded data can be played back without any delay and distortion. A database is built up first for those who are monitored. When SIP dialog begins, SIP proxy inspects whether a call must be monitored. If it is the case of monitoring, a duplicate packet flow is delivered to the monitor. The monitor can playback. I believe this implementation can become a platform for further work in the lawful interception.

As the network fundamental infrastructures become mature, broadband network turns into the main stream. Sufficient bandwidth makes many applications, for example, voice over IP (VoIP), become possible. Through IP phone, people only need to pay local Internet service fee, which is relatively more inexpensive, to be able to make long-distance call with remote people. After the basic calling facility is ready, additional VoIP services become more and more important. User will demand for more additional service functions. In this thesis, I propose and implement a voice recording facility based on SIP-based VoIP system. Users can record both caller and callee's voice together in digital way. Furthermore, we use this facility to provide a voice message recording service. When callee does not pickup his/her phone, caller's phone will be redirected to voice message recording server. Caller can record his/her voice message into callee's directory on the voice recording server, and callee can listen to his/her own voice message later.

It is very convenient for human to use Internet for communications. The VoIP service is a good example. Because voice transmission through Internet becomes mature, people can remotely talk with each other by IP phones inexpensively. Thus, VoIP system can be integrated with Public Switched Telephone Network (PSTN) in some groups, organizations, or companies to reduce the cost for communication. Additionally, it also can be another kind of free consultation channel for customers and users. To take advantage of VoIP system, the people who may be inside a company, be the Internet visitors, or be provisional guests can receive services and question answers immediately and freely. However, it needs a lot of procedures to connect to the VoIP system, e.g., to install a softphone which must be booted to register an VoIP server, to even configure IP address, port number, protocol, encryption, and outbound proxy, etc. This configuration sometimes is difficult and it may need to setup every time when customers want use VoIP phone. It is not an easy job for a user, for example, to download the software from Internet, and to read some documents to setup system parameters and then to operate it, especially for those people who are not familiar with computers. Thus, this inconvenience may cause VoIP service not to be easily promoted. Therefore, the purpose of this paper is to solve this problem and make users have a web interface and without worrying about how to setup the system. I combine Web interface with VoIP phone to become a ¡§webphone¡¨. To take advantage of the characteristics of generality and facility of Web, guests can click the button on the Web pages to trigger the VoIP component inside the Web page to connect to the VoIP system, then make communication with other people. At the same time, it can avoid exposing the information of server address, account, password to public and ward off dangerous attacks from Internet.

With the development and improvement of network bandwidth, Voice over Internet Protocol (VoIP) technology begins to bloom. VoIP gradually replaces voice over telecom networks because of its low cost. For instance, VoIP technology divides voice data into different packets and transmits over Internet Protocol, so that it does not require setting up a particular path as the telecom networks does. Although there is much VoIP software available in the market, most of them need pre-installation before using. Our web-based VoIP technology allows users to take advantage of the service with either home computers or public computers. Users only need to log on to our website, then our system will provide users the client program via Java Applet with automatic installation. Users only need to close the browser to terminate the service; the program data will not be left on the computer. In order to provide users to use our service in anytime and anywhere, we need to recognize Network Address Translation (NAT) as one of our biggest barriers. Including Session Initiation Protocol (SIP), instant message and voice data packet could be blocked by NAT. So, we propose a complete solution of NAT traversal. Our system, SIP Communicator, provides a communication platform to users by supporting instant message, VoIP and session transferring. Users can log on to our system via any computers that are JRE supported to communicate with other clients.

Because of Skype, Voice over IP(VoIP) becomes much more hot and popular. It has always been considered to be a killer application of new generation of the Internet. With its distinct economic advantage and good voice quality, it has the tendency to replace the traditional Public Service Telephone Network(PSTN). Moreover, Session Initiation Protocol(SIP) is proposed by Next Generation Network(NGN) to be the first choice of voice and multimedia network control protocol. Client-server is the key architecture of SIP. Although this architecture is simple and easy to maintain, and even it has faster response time than P2P, the server may cause problems while the client is increasing. Moreover, it is possible that the large load may cause the service stop anytime. In this paper, we discuss how to design and implement an embedded VoIP user agent based on SIP standard. It can improve efficiency of SIP signaling delivery through the user characteristic of making calls with limited resources and reduce the number of services that servers to provide while calling. We also design a graphical user interface. The interface lets users feel friendly and make the method of the system more efficient.

Open source PBX Kamailio and OpenSIPS diploma thesis covers familiarization with appointed SIP exchanges and with their power comparing. A detailed installation instructions on the operating system Ubuntu is the aim of this work too. The work includes the historical development of telephone exchanges with a focus on the latest generation. The following is SIP protocol basic description and components that can be composed SIP exchanges. Another part is devoted to the development of exchanges Kamailio and OpenSIPS. The thesis contain the archutecture and configuration file description. The practical part of the thesis deals with high-capacity switches, and comparing it in terms of memory and computational demands. Selected measurements are compared with the Asterisk PBX.