Dissertationen zum Thema „SPI communication protocol“

The project presented in this report is set out to develop a portable electronic system to be used as a medicine pill container. With the functionality to configure up to twelve daily repeated alarms, the purpose of the medicine dosage system is first and foremost to remind the user when they should take their medicine. Secondly, LED lights and user-recorded voice notifications are to be implemented to further aid the user in taking the right medicine each time. The device is also to have a memory log, recording up to one hundred missed dosages, enabling an authorized medicine professional to verify the medicine adherence of the user.Prior to the start of the project, an outline for the functionality and physical appearance of the device was set by the project owner, Victrix AB. This project covers the hardware and software development, as well as the design choices within. The aim is to follow the proposed functionality specification as close as possible, while making justified hardware and software choices considering simplicity, efficiency, power consumption, and availability. By following the specification, the goal is ultimately to increase the medicine adherence for users of the device developed with this thesis.Using qualitative research methods, a valid background study was created, preceding the development of the medicine dosage system. Hardware for a first prototype of the device was then chosen based on the gathered information about existing technologies and related work. With thorough testing and recurrent information exchange with the client, a prototype of the medicine dosage system, based on an Arduino microcontroller, was constructed. The prototype was evaluated to fulfill 92% of the requirements considered as high priority by Victrix.
Projektet som presenteras i denna rapport är tänkt att utveckla ett portabelt elektroniskt system för användning som en medicinsk pillerbehållare. Med funktionaliteten att konfigurera upp till tolv dagligen upprepande alarm är syftet med medicindoseringssystemet först och främst att påminna användaren när de ska ta sin medicin. Lysdioder och användarens egna inspelade röst som notifikationer ska implementeras för att vidare hjälpa användaren att ta rätt medicin vid varje tillfälle. Enheten ska också ha en minneslogg som sparar upp till etthundra missade doseringar, vilket gör det möjligt för auktoriserad sjukvårdspersonal att verifiera användarens följsamhet till medicineringen.En översiktlig beskrivning av funktionaliteten samt det fysiska utseendet av enheten skrevs av projektägaren Victrix AB innan projektet startades. Det som detta projekt täcker är hårdvaruoch mjukvaruutvecklingen, så väl som där tillhörande designval. Projektet siktar på att följa den föreslagna funktionalitetsspecifikationen så nära som möjligt, och samtidigt göra välgrundade val för hårdoch mjukvara med enkelhet, effektivitet, energiförbrukning och tillgänglighet i åtanke. Genom att följa specifikationen är det slutliga målet att frambringa ökad medicinföljsamhet för användare av den med det här projektet utvecklade enheten.Utvecklingen av medicindoseringssystemet föregicks av en befogad bakgrundsstudie utformad genom användningen av kvalitativa forskningsmetoder. Hårdvara att användas för en första prototyp av enheten valdes sedan baserat på den insamlade informationen om existerande teknologier och relaterat arbete. Genom grundliga tester och regelbundet informationsutbyte med kunden konstruerades en prototyp av medicindoseringssystemet baserat på en Arduinomikrokontroller. Prototypen utvärderades att uppfylla 92% av kraven som Victrix ansåg vara av hög prioritet.

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New versions of SCTP protocol allow the implementation of handover procedures in the transport layer, as well as the supply of a partially reliable communication service. A communication architecture is proposed herein, integrating SCTP with the session initiation protocol, SIP, besides additional protocols. This architecture is intended to handle voice applications over IP networks with mobility requirements. User localization procedures are specified in the application layer as well, using SIP, as an alternative mean to the mechanisms used by traditional protocols, that support mobility in the network layer. The SDL formal specification language is used to specify the operation of a control module, which coordinates the operation of the system component protocols. This formal specification is intended to prevent ambiguities and inconsistencies in the definition of this module, assisting in the correct implementation of the elements of this architecture
Novas vers?es do protocolo SCTP permitem sua utiliza??o para implementa??o de mecanismos de handover em n?vel de transporte, bem como o fornecimento de um servi?o de transmiss?o de dados parcialmente confi?vel. Integrando o SCTP com o protocolo de inicia??o de sess?es, SIP, al?m de utilizar adicionalmente servi?os de outros protocolos auxiliares, uma arquitetura de comunica??o p?de ser proposta, a fim de atender ?s aplica??es de voz sobre IP com requisitos de mobilidade. S?o especificados ainda os procedimentos de localiza??o de usu?rio em n?vel de aplica??o, utilizando o protocolo SIP, como alternativa aos mecanismos empregados por protocolos tradicionais que suportam mobilidade na camada de rede. A linguagem de especifica??o formal SDL ? utilizada para especificar o funcionamento de um M?dulo de Controle, relacionado ? opera??o coordenada dos protocolos que comp?e a arquitetura. Pretende-se assim evitar ambig?idades e inconsist?ncias na defini??o desse m?dulo, o que pode auxiliar em implementa??es corretas de elementos dessa arquitetura

SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.

Le secteur mondial de télécommunications a connu des bouleversements sans précédent pendant ces dernières années avec l’avènement de nouveaux services et de nouvelles technologies de communication. Les opérateurs de télécommunications subissent en effet une décroissance continue de leur revenu par utilisateur en raison d’une compétition toujours accrue et, dans une certaine mesure, de la saturation du marché pour les services les plus traditionnels. Afin d’attirer de nouveaux usagers, de retenir ceux qui existent déjà et d’augmenter le revenu par usager, les services de communication proposés par les opérateurs doivent prendre en compte le contexte de l’utilisateur. L’interfonctionnement entre les solutions proposées pour la prise en compte du contexte et les réseaux de communication actuels et de nouvelle génération représentent un grand défi tant pour les opérateurs que les fournisseurs de services de communications. Ce travail de thèse traite des questions relatives au développement des systèmes de communications adaptés au contexte en proposant un agent coté réseau nommé INCA (Intelligent Network-based Communication Assistant). L’INCA permet une gestion avancée des communications SIP (Session Initiation Protocol) adaptée aux informations de contexte qui sont recueillies à travers un cadre dédié à la publication et la notification de contexte. Son architecture multicouche est basée sur un modèle de couche générique et elle implémente une approche orientée plan pour la gestion des sessions SIP. Elle repose aussi sur un nouveau modèle de communication avec prise en compte du contexte pour permettre une adaptation des communications basée sur les préférences utilisateur
In recent years, the world telecommunications sector has undergone unprecedented changes driven mainly by the deployment of new communication technologies and services. Telecom operators are suffering from a steady decline in their revenues per user due to fierce competition and market saturation for traditional services. In order to attract new customers and retain existing ones, communication services proposed by these operators need to be aware of the user’s context, which includes information related to the user himself as well as his environment (e. G. His location, current activities and available devices). Unfortunately, interworking proposed context-aware solutions with current and next-generation networks still represents a big challenge for communication service providers as well as operators. This thesis addresses issues related to the development of context-aware communication systems by proposing a network-based agent called INCA (Intelligent Network-based Communication Assistant). INCA provides advanced management of SIP communications based on context information that is retrieved through a dedicated framework for context publication and notification. Its multilayered architecture is based on a generic layer model and implements a plan-centric approach for SIP session management. It also relies on a new context-aware communication model for providing communication adaptation based on user preferences

Les travaux de recherche de la thèse de Rami Troudi répondent à trois problématiques :-la première est de concevoir un onduleur triphasé multiniveau pour des applications à base d’énergies renouvelables connectées au réseau de distribution, ou pour la motorisation des véhicules électriques. Dans la première application, l’utilisation de bras multiniveau limite fortement les inductances de couplage avec le réseau tandis que dans la deuxième application, le couple délivré par le moteur est de meilleure qualité.-la deuxième consiste à concevoir une structure de convertisseur DC-DC permettant de n’avoir qu’une seule source continue pour alimenter cet onduleur multiniveau. -la troisième est la conception d’une architecture de commande temps réel à base de microcontrôleurs permettant d’avoir une grande capacité d’évolution et de calcul et une facilité d’industrialisation.Le manuscrit de la thèse est organisé en quatre chapitres. Le premier chapitre présente un état de l’art des structures d’onduleur multiniveau. Cette technologie est devenue aujourd’hui un thème de recherche important. Ce chapitre donne les avantages et les inconvénients de chaque topologie d’onduleur multiniveau conventionnel ainsi que les nouvelles topologies permettant une réduction du nombre de composants. Ce chapitre fait aussi un état de l’art des structures de hacheurs avec une ou plusieurs entrées-sorties (MISO, MIMO et SIMO). Ce chapitre présente aussi les avantages et les inconvénients de chaque famille de structure avec leur commande. La fin du chapitre présente les nouvelles topologies retenues pour l’onduleur multiniveau et le hacheur SIQO (une entrée-quatre sorties).Le deuxième chapitre est consacré à la présentation de la structure de l’onduleur multiniveau proposé, ainsi qu’à l’étude de son mode de fonctionnement, de sa commande rapprochée et de son utilisation dans une application en boucle fermée. Ce chapitre montre que cette structure a l’avantage de minimiser les pertes dans les composants de puissance en ayant, à chaque instant, peu de composants qui conduisent le courant de chaque bras, ce qui permet d’augmenter son rendement. En plus, ce chapitre montre la simplicité de la commande rapprochée de l’onduleur en utilisant un algorithme très simple. Des essais expérimentaux sont donnés à la fin du chapitre après le descriptif de la maquette d’essai.Le troisième chapitre traite en détail la structure du convertisseur DC-DC SIQO utilisé pour alimenter l’onduleur multiniveau, son mode de fonctionnement, sa modélisation et le développement d’une commande multi-entrée multi-sortie (MIMO). Cette structure est conçue à partir d’un couplage de la structure SEPIC avec la structure à accumulation magnétique et du dédoublement de chaque sortie par un système d’aiguillage qui permet ainsi d’obtenir quatre sorties à partir d’une seule entrée DC. Chaque structure (SEPIC et à accumulation) gère deux sorties avec le calcul de deux rapports cycliques. Pour cela, une synthèse d’asservissement basée sur une méthode H_∞ est présentée pour être robuste aux variations des courants et aux changements de consigne. Les résultats des essais expérimentaux sont donnés à la fin du chapitre après le descriptif de la maquette d’essai.Le chapitre quatre aborde le développement de l’architecture de commande à base de microcontrôleurs. Cette structure est appliquée au contrôle de l’onduleur triphasé. Ce chapitre décrit toutes les fonctions qui composent cette architecture au niveau matériel et logiciel. Le fait de répartir les besoins matériels et algorithmiques sur plusieurs microcontrôleurs permet de faciliter l’évolution des demandes de fonctions supplémentaires à savoir le diagnostic et la reconfiguration d’un bras, ainsi que l’ajout de la fonction de filtrage actif. Cette architecture repose sur une communication par bus SPI (Serial Peripheral Interface) qui permet des échanges rapides entre les microcontrôleurs et aussi vers un système IHM (Interface Homme Machine)
Rami Troudi's thesis research work addresses three problematics:- the first is to design a three-phase multilevel inverter for applications based on renewable energies connected to the power grid, or for electric motor drive of electric vehicles. In the first application, the use of multilevel arms greatly limit the coupling inductances with the power grid, while in the second application, the torque delivered by the motor is of better quality.- the second is to design a DC-DC converter structure having only one DC source destinated to supply this multilevel inverter.- the third is the design of a real-time control architecture based on microcontrollers leading to a large capacity of evolution and calculation and an ease industrialization.The thesis manuscript is organized into four chapters.The first chapter presents a state of art of multilevel inverter structures. This technology is becoming an important research topic today. This chapter gives the advantages and disadvantages of each conventional multilevel inverter topology as well as the new topologies with a reduction in the number of components. This chapter also give a reviews of the state of art of chopper structures with one or multiple inputs-outputs (MISO, MIMO and SIMO). This chapter also presents the advantages and the disadvantages of each family of structure with their regulation. The end of the chapter presents the new topologies retained for the multilevel inverter and the SIQO chopper (one input-four outputs).The second chapter is devoted to the presentation of the structure of the proposed multilevel inverter, as well as to the study of its mode of operation, its close control device and its use in a closed loop application. This chapter shows that this structure has the advantage of minimizing losses in power components by having, at any time, few components that conduct the current of each arm, which allows to increase its efficiency. In addition, this chapter shows the simplicity of the used close control of the inverter employing a very simple algorithm. Experimental tests are given at the end of the chapter after the description of the test bed.The third chapter discusses in detail the structure of the SIQO DC-DC converter used to supply the multilevel inverter, its operation mode, its modeling and the development of a multi-input multi-output (MIMO) control. This structure is designed from a coupling of the SEPIC structure with the buck-boost structure and the doubling of each output by a switching system which leads to obtain four outputs from a single DC input. Each structure (SEPIC and buck-boost) manages two outputs with the calculation of two duty cycles. For this, a control synthesis based on an Hinfini method is presented to be robust to the variations of the currents and the changes of the setpoint. The results of the experimental tests are given at the end of the chapter after the description of the test bed.Chapter four discusses the development of one architecture based on multi-microcontroller system. This structure is applied to the control of the three-phase multilevel inverter. This chapter describes all the functions that compose this architecture at the hardware and software level. The distribution of the hardware and algorithmic needs several microcontrollers makes it easier to evolve the demands for additional functions, namely the diagnosis and reconfiguration of an arm, as well as the addition of the active filtering function. This architecture is based on SPI (Serial Peripheral Interface) bus communication which allows rapid exchanges between the microcontrollers and also towards an HMI (human-machine interfaces) system

L'application MédiclP est un logiciel de télémédecine permettant à des spécialistes de la santé d'entrer en communication lors de situations d'urgence. Ce prototype, développé depuis 2005, permet des communications audio ainsi que le transfert d'électrocardiogrammes en temps réel. Le scénario typique visé par ce projet était de permettre à une équipe ambulancière qui récupère un blessé sur la route, ou quelqu'un victime d'un malaise cardiaque, d'entrer en contact avec les hôpitaux les plus près afin de déterminer lequel est le plus apte à recevoir adéquatement ce patient. Cette approche permettrait d'améliorer la préparation, la qualité et la rapidité des opérations médicales à l'hôpital lorsque le patient se présente. Le présent projet, SécureMédic, se veut un moyen d'aborder le problème de la sécurité entourant ce prototype, étant donné son utilisation dans un contexte médical. Une analyse de MédicIP a permis d'identifier quatre failles de sécurité critiques reliées aux authentifications usagers, aux établissements des conférences, aux transferts des données audio ainsi qu'aux transferts des électrocardiogrammes. La contribution majeure de ce projet a été la création d'une infrastructure dédiée au processus d'authentification des usagers. Le système développé permet deux types d'authentification, fournissant ainsi un excellent niveau de robustesse. De plus, le serveur principal développé dans cette infrastructure intègre des mesures de protection permettant de minimiser les impacts de certains types d'attaque. Le projet SécureMédic a permis de démontrer la faisabilité de la sécurisation d'une application de télémédecine en utilisant les techniques de protection et les standards actuels de l'industrie. Les résultats de tests comparatifs ont cependant permis de constater que des impacts reliés à la performance ont été engendrés par l'ajout des mesures de sécurité, dus principalement aux ressources requises pour le chiffrement et le déchiffrement des données dans un environnement multimédia temps réel. Bien qu'il soit encore trop tôt pour envisager le déploiement du système actuel dans le milieu de la santé, les projets MédicIP et SécureMédic sont un pas dans la bonne direction. Le prototype actuel répond aux exigences techniques voulues, répond à un besoin bien réel et est parmi les premières solutions concrètes à démontrer la faisabilité d'un tel système. D'ici quelques années, l'apparition de solutions similaires est assurée.

Ce mémoire traite la problématique de la mobilité des sessions pour le transfert des communications multimédias basées sur le protocole SIP. Plusieurs aspects sont étudiés et des mécanismes proposés afin de permettre la mobilité des sessions avant, durant et après leur établissement. En plus d'une communication impliquant deux intervenants, Il a été possible d'étendre l'utilisation de la mobilité des sessions pour l'appliquer aux scénarios de communications en mode conférence regroupant plusieurs intervenants. Les mécanismes de mobilité de session développés au cours de cette thèse sont par la suite déployés pour (1) permettre des transformations entre différentes topologies de conférences et (2) construire un modèle de conférence adapté pour l'échange de la voix au sein des groupes de communication à large échelle.

Pas de résumé en français
Session Initiation Protocol (SIP) is the widely used signaling protocol for voiceand video communication as well as other multimedia applications. Despiteof its flexibility and a common standard that can be leveraged to efficientlycombine a wide array of communication systems and technologies, it is exposedto a number of problems, including the vulnerability to several types of attacksdue to its open nature, in particular, and lack of a clear defense line. Likewise,flooding attack is one of the most destructive attacks targeting both User AgentServer (UAS) and User Agent Client (UAC), leading to a Denial of Service (DoS)in VoIP applications. In particular, INVITE message is considered as one of themajor root causes of flooding attacks in SIP. This is due to the fact that an attackermay send numerous INVITE requests without waiting for responses from theUAS or proxy in order to exhaust their resources. Moreover, SPIT problem inSIP is also a challenging issue which needs proper attention and appropriatesolutions.Most of the solutions proposed to overcome the flooding attacks are eitherdifficult to deploy in practice or require significant changes in the SIP servers.Additionally, the diverse nature of flooding attacks offers a huge challenge toenvisage appropriate prevention mechanisms. In this survey, we present acomprehensive study on flooding attacks against SIP by addressing its differentvariants and analyzing its consequences. We also classify the existing solutionscorresponding to different flooding behaviors, types and targets, and then weperform an extensive investigation of their main weaknesses and strengths.Additionally, we also take into account the underlying assumptions of eachsolution for a better understanding of its limitations. Specifically, we havethoroughly analyzed SPIT problems and few of the existing solutions proposedfor their prevention.The theoretical framework derived from our extensive literature survey led us topropose a solution for handling specific number of SIP requests in a particulartime window. Our proposed "Light Weight Scheme" is implemented in a SERSIP server. The evaluation results presented in this thesis depict the satisfactoryperformance of this approach. In order to cope with SIP flooding attacks, wepropose another solution based on "Strategy Based Proxy". This solution isdesigned for a SIP proxy that calculates the probability of a call being maliciouson the basis of its current experience. The obtained experience is also utilized tocalculate the probabilities of a successful call setup. This approach is useful forboth state-ful and state-less proxy servers.For dealing with SPIT, we have designed a 2-step solution. In first step, weextract the useful information from the VoIP traffic. In second step, we apply aNaive Bayes classifier on the date extracted from first step to determine whetherthe nature of an incoming SIP call is malicious or it is a harmless routine call.With this mechanism, we can detect the SPIT calls from a group of incomingSIP calls. Finally, we presents a detailed discussion and conclusions derivedfrom our case study carried out in this thesis along with future directions andpotential research areas related to VoIP security threats

Now a day, one of the most attractive research topics in the area of Intelligent Traffic Control is

Inter-vehicle communication (V2V communication). In V2V communication, a vehicle can

communicate to its neighbouring vehicles even in the absence of a central Base Station. The

concept of this direct communication is to send vehicle safety messages one-to-one or one-to-

many vehicles via wireless connection. Such messages are usually short in length and have very

short lifetime in which they must reach the destination. The Inter-vehicle communication system

is an ad-hoc network with high mobility and changing number of nodes, where mobile nodes

dynamically create temporary sensor networks and transferring messages from one network to

others by using multiple hops due to limitation of short range.

The goal of the project is to investigate some basic research questions in order to organize such

sensor networks and at the same time highlight the appropriate routing protocol that support

mobile ad hoc networks in an efficient and reliable manner.

In our investigation, we have answered the technical issues in order to construct a V2V

communication system. We have also studied some mobile ad hoc network routing protocols in

detail and then selected the DSR (Dynamic Source Routing) for our V2V communication and

then simulated it according to our system requirements. We are quite satisfied by the result of

DSR, but at the same time much more work is required to come up with an absolute application

for the end user.

Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323.

Le routage dans les réseaux de capteurs, est un service essentiel qui transmet les lectures des capteurs à certains points de collecte de données dans le réseau sur la base des relais multi-saut. Cette tâche est particulièrement difficile car elle doit être réalisé d'une manière efficace au niveau de consommation de ressources et avec une quantité limitée d'informations disponible. La facilité de mise à l'échelle et l'utilisation d'information local pour fonctionner ont permis au routage géographique être considéré comme une approche prometteuse. Cependant, lors de son implémentation, certains problèmes subsistent en raison des difficultés pratiques. Dans ce travail de recherche, deux problématiques inhérentes aux protocoles de routages géographique ont été étudiés: i) Le coût associé: aux évanouissements liés aux obstacles et aux multi-trajets suivis par un signal transmis sur un canal radio, aux changements rapides des conditions physiques du canal de transmission and ii) l'administration de resources affectés à chaque noeud appartenant au réseau. Afin de résoudre ce problème, deux protocoles ont été présentés: un protocole de routage géographique avec communications coopératives, Beaconless Cooperative Geographic cross-layer protocol for ad hoc and sensor networks (CoopGeo) et un protocole de routage basé sur le principe d'extension de couverture: Relay-Aware Cooperative Routing (RACR).

The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments.
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Voice over IP (VoIP) is a relatively new technology that enables voice calls over data networks.With VoIP it is possible to lower expenses, and increase functionality and flexibility. FromSwedish Armed Forces point of view, the security issue is of great importance, why the focus inthis report is on the security aspect of the two most common open-source VoIP-protocols H.323and SIP, some of the most common attacks, and counter-measures for those attacks.Because of the level of complexity with a network running H.323 or SIP, and the fact that it hasyet to stand the same level of trial as of traditional telephony, a VoIP-system includes manyknown security-issues, and probably at present many unknown security flaws. The conclusion is that it takes great knowledge and insight about a VoIP-network based onH.323 or SIP to make the network satisfyingly safe as it is today, and is therefore perhaps not asuitable solution for the Swedish Armed Forces today for their more sensitive communications.
Voice over IP (VoIP) är en datakommunkationsteknik som möjliggör röstsamtal överdatanätverk. Med VoIP är det möjligt att sänka kostnader, utöka funktionalitet och flexibilitet.Från Försvarsmaktens perspektiv är säkerhetsfrågan med VoIP av stor vikt, därför läggs speciellfokus för denna rapport på säkerhetsaspekten av de två största öppna VoIP-protokollen H.323och SIP, några av de vanligaste attackerna, och åtgärder mot dessa attacker. Eftersom uppbyggnaden av ett H.323- eller SIP-baserat nätverk är komplext och inte allsbeprövat i samma utsträckning som traditionell telefoni, innehåller det många kända säkerhetshåloch förmodligen för närvarande många okända säkerhetsbrister. Slutsatsen är att det krävs mycket stor kunskap och insikt hur ett VoIP-nätverk baserat på H.323eller SIP fungerar för att göra nätverket tillräckligt säkert i nuläget, vilket gör det till en tveksamttillfredställande lösning för Försvarsmakten idag för deras kommunikation av känsligare slag.

Dans un futur proche, les technologies d’accès sans fil telles que le WiFi, leWiMAX et l’UMTS coexisteront. Néanmoins, cette coexistence quotidienne ne signifie pas qu’ils seront en mesure d’être pleinement inter-opérationnels. Nous pensons alors que le roaming concernera différents réseaux d’accès gérés par différentes entités : des opérateurs de réseaux cellulaires, des fournisseurs d’accès Internet (FAI) et certaines organisations ou individus. Notre travail de recherche s’articule ainsi autour d’un scénario de roaming hétérogène, concernant les réseaux d’accès sans fil gérés par différents opérateurs. Notre objectif principal est de fournir une architecture de roaming qui permet l’interopérabilité des réseaux hétérogènes dans le cadre d’un environnement multi-opérateurs, et ce sans grands changements dans les architectures sans fil actuelles. L’un des objectifs de la nouvelle génération des réseaux sans fil NGWNs (Next GenerationWireless Networks) est la possibilité de transférer de manière transparente les services entre les réseaux sans fil de différentes technologies d’accès. L’objectif est de tirer partie de la popularité et des hauts débits des UWNs (UnlicensedWireless Networks) afin d’améliorer les services mobiles. Bien qu’il existe déjà des solutions qui tentent de répondre à la problématique de la mobilité de service, la plupart d’entre elles sont basées sur l’hypothèse que les opérateurs mobiles sont également propriétaires des réseaux WiFi / WiMAX. La réalité est que, en dépit du fait que les opérateurs mobiles investissent énormément dans le déploiement de leurs propres UWNs, de nombreux UWNs indépendants peuvent être utilisés comme des extensions de ces réseaux mobiles
In a future, existing wireless access technologies such as WiFi (Wireless Fidelity), WiMAX (Worldwide Interoperability for Microwave Access) and UMTS (Universal Mobile Telecommunication System) will coexist on daily basis. Nevertheless, this daily coexistence does not imply that they will be able to fully interoperate. Thus, my research work focuses on a specific roaming scenario which is formed by heterogeneous wireless access networks managed by different operators. Our main objective is to provide a seamless (for the mobile user) roaming architecture to enable network interoperability under a heterogeneous multi-operator wireless environment, all this without major changes in current wireless architectures. In this regard, we also aim at reducing the roaming signaling exchange caused by the authentication and authorization mechanisms in the network registration and session initiation process while providing robust network security. The results obtained through computer simulation indicated that the use of our approach reduces significantly the network registration and session initiation delay, hence outperforming the traditional Roaming-SIP method. We also confirmed that the wireless delay introduced by the VN increases considerably when increasing the traffic congestion level in the Visiting Networks. Thus, from the simulations results we can state that by reducing the signaling message exchange, and maintaining acceptable congestion levels in the wireless network, hence reducing the wireless transmission delay, we can improved the overall de-lay in both network registration and session initiaition process

Different communication services from delivery of written letters to telephones, voice/video over Internet Protocol(IP), email, Internet chat rooms, and video/audio conferences, immersive communications have evolved over time. A communication system of voice/video over IP is the realization of a two fundamental layered architecture, signaling layer and media layer. The signaling protocol is used to create, modify, and terminate media sessions between participants. The signaling layer is further divided into two layers, service layer and service control layer, in the IP Multimedia Subsystem (IMS) specification. Two widely used communication systems are IMS, and Peer-to-Peer Session Initiation Protocol (P2P SIP). Service providers, who behave as brokers between callers and callees, implement communication systems, heavily controlling the signaling layer. These providers do not take the diversity aspect of end users into account. This dissertation identifies three technical barriers in the current communication systems especially in the signaling layer. Those are: I. lack of openness and flexibility in the signaling layer for end users. II. difficulty of development of network-based, session-based services. III. the signaling layer becomes complex during the high call rate. These technical barriers hinder the end-user innovation with communication services. Based on the above listed technical barriers, the first part of this thesis defines a concept and architecture for a communication system in which an individual user becomes the service provider. The concept, My Own Communication Service Provider (MOCSP) and MOCSP system is proposed and followed by a call flow. Later, this thesis provides an analysis that compares the MOCSP system with existing communication systems in terms of openness and flexibility. The second part of this thesis presents solutions for network-based, session based services, leveraging the proposed MOCSP system. Two innovative services, user mobility and partial session transfer/retrieval are considered as examples for network-based, session-based services. The network-based, sessionbased services interwork with a session or are executed within a session. In both cases, a single functional entity between caller and callee consistently enables the media flow during the call initiation and/or mid-call. In addition, the cooperation of network call control and end-points is easily achieved. The last part of the thesis is devoted to extending the MOCSP for a high call rate and includes a preliminary comparative analysis. This analysis depends on four factors - scalability limit, complexity level, needed computing resources and session setup latency - that are considered to specify the scalability of the signaling layer. The preliminary analysis clearly shows that the MOCSP based solution is simple and has potential for improving the effective usage of computing resources over the traditional communication systems

Les systèmes de production manufacturiers sont une classe des systèmes à événements discrets. Leur taille nécessite de distribuer le logiciel de contrôle sur une architecture industrielle de plusieurs ordinateurs reliés en réseau. Dans ce contexte, il devient essentiel d'être capable d'évaluer l'impact d'une architecture réseau spécifique sur les services des systèmes manufacturiers en termes de la performance et la qualité. Les performances du réseau sous-jacent peuvent notamment nuire à la productivité du système. Dans la méthodologie traditionnelle proposée dans la littérature, cet aspect n'est pas pris en compte au niveau conception. Cependant, la modélisation de tels systèmes est importante pour vérifier certaines propriétés. Dans cette thèse, nous proposons une approche de modélisation par composants à l'aide des réseaux de Petri haut niveau pour la modélisation de certains protocoles de réseaux afin d'évaluer les systèmes manufacturiers comme étant des systèmes distribués. La sélection des réseaux de Petri est justifiée par leur pouvoir d'expression en ce qui concerne la modélisation des systèmes distribués et concurrents. L'approche par composants permet de diminuer la complexité de la modélisation et encourage la généricité, la modularité et la réutilisabilité des composants prêt-à-utiliser. Cela permet de construire facilement de nouveaux modèles et de réduire les coûts de développement de systèmes. En outre, cela peut aider à une meilleure gestion des services et des protocoles et à changer facilement/modifier un élément du système. Notre modélisation permet enfin d'évaluer ces systèmes par le biais de simulations centralisées

The diploma thesis deals with the design and implementation of the system that will control the water supply for the family house and its garden. The system aslo allows you to store a history that can be viewed via the web interface. The web interface also allows you to set system parameters and monitor the current status.

Les protocoles plug-and-play couplés avec les architectures logicielles rendent nos maisons ubiquitaires. Les équipements domestiques qui supportent ces protocoles peuvent être détectés automatiquement, configurés et invoqués pour une tâche donnée. Actuellement, plusieurs protocoles coexistent dans la maison, mais les interactions entre les dispositifs ne peuvent pas être mises en action à moins que les appareils supportent le même protocole. En plus, les applications qui orchestrent ces dispositifs doivent connaître à l'avance les noms des services et dispositifs. Or, chaque protocole définit un profil standard par type d'appareil. Par conséquent, deux appareils ayant le même type et les mêmes fonctions mais qui supportent un protocole différent publient des interfaces qui sont souvent sémantiquement équivalentes mais syntaxiquement différentes. Ceci limite alors les applications à interagir avec un service similaire. Dans ce travail, nous présentons une méthode qui se base sur l'alignement d'ontologie et la génération automatique de mandataire pour parvenir à une adaptation dynamique de services
Ubiquitous systems imagined by Mark Weiser are emerging thanks to the development of embedded systems and plug-n-play protocols like the Universal Plug aNd Play (UPnP), the Intelligent Grouping and Resource Sharing (IGRS), the Device Pro le for Web Services (DPWS) and Apple Bonjour. Such protocols follow the service oriented architecture (SOA) paradigm and allow an automatic device and service discovery in a home network. Once devices are connected to the local network, applications deployed for example on a smart phone, a PC or a home gateway, discover the plug-n-play devices and act as control points. The aim of such applications is to orchestrate the interactions between the devices such as lights, TVs and printers, and their corresponding hosted services to accomplish a specific human daily task like printing a document or dimming a light. Devices supporting a plug-n-play protocol announce their hosted services each in its own description format and data content. Even similar devices supporting the same services represent their capabilities in a different representation format and content. Such heterogeneity along with the protocols layers diversity, prevent applications to use any available equivalent device on the network to accomplish a specific task. For instance, a UPnP printing application cannot interacts with an available DPWS printer on the network to print a document. Designing applications to support multiple protocols is time consuming since developers must implement the interaction with each device pro le and its own data description. Additionally, the deployed application must use multiple protocols stacks to interact with the device. More over, application vendors and telecoms operators need to orchestrate devices through a common application layer, independently from the protocol layers and the device description. To accomplish interoperability between plug-n-play devices and applications, we propose a generic approach which consists in automatically generating proxies based on an ontology alignment. The alignment contains the correspondences between two equivalent devices descriptions. Such correspondences actually represent the proxy behaviour which is used to provide interoperability between an application and a plug and play device. For instance, the generated proxy will announce itself on the network as a UPnP standard printer and will control the DPWS printer. Consequently, the UPnP printing application will interact transparently with the generated proxy which adapts and transfers the invocations to the real DPWS printer. We implemented a prototype as a proof of concept that we evaluated on several real UPnP and DPWS equivalent devices

碩士
國立中正大學
資訊工程研究所
100
Network security has become an important issue since various attacks are appearing in the IP network, and the key exchange protocol is one of the most important cryptography mechanisms to protect end-to-end communication by the secret session key encryption. In this paper, we focus on the four-party key exchange protocol in an environment in which two clients are registered under two distinct servers, and both registered parties have to communicate through their corresponding server respectively. We integrate SIP with the three-way key exchange and agreement protocol (TW-KEAP), denoted by SIP+TW-KEAP. The SIP+TW-KEAP uses an attribute (a=crypto) of SDP in the SIP message body to carry key parameters without exchanging any extra message according to RFC 4568 SDES (Session description protocol security descriptions for media streams). It makes two communication parties have a secret session key to protect their subsequent communications on an efficient way, and intends servers to involve with the key exchange procedure to derive the session key for the lawful interception support. In addition, we utilize GNU oSIP library and crypto++ library to implement the proposed scheme. The SIP+TW-KEAP has not only the shortest total service time based on the experimental results but also the shortest queuing delay, wireless transmission delay, and energy consumption from numerical analysis, as compared with the current alternative.

M. Tech. Electrical Engineering.
Widespread deployment of wireless local area networks (WLANs) and a gradual increase in streaming applications have brought about a demand for improved Quality of Service (QoS) in wireless networks. The IEEE 802.11e standard was proposed to provide QoS mechanisms for assigning high priority to delay-sensitive applications. However, Internet traffic is still dominated by TCP based applications, and the negative effects of the IEEE 802.11e service differentiation scheme on TCP performance in the presence of high priority traffic are becoming a challenging issue. TCP has been found to perform poorly in wireless networks, including IEEE 802.11e; more applications with higher QoS demands use UDP in the transport layer than TCP. Therefore, the QoS of low priority traffic in 802.11e is not guaranteed in networks highly loaded with high priority traffic. This is aggravated by the class differentiation introduced in current QoS protocols, which results in TCP applications being starved during high traffic load. The motivation of this work is to enhance the interaction between the TCP and MAC protocols in order to improve TCP performance in WLANs.

M. Tech. Electrical Engineering.
Presents an IPv6-based mobility support scheme for vehicular scenarios. The focus of this study is on modelling of an efficient network layer handover scheme suitable for vehicular network environments. In order to achieve this, a Vehicular Fast handovers for Mobile IPv6 (VFMIPv6) network layer protocol has been proposed.

M. Tech. Electrical engineering.
Discusses the interference and bandwidth adjusted ETX routing metric uses a logical interference model that refers to the interference arising from the Carrier Sensing Multiple Access with Collision Avoidance (CSMA-CA) based Medium Access Control. This approach of capturing inter-flow is complex and more restrictive. In this dissertation, a more realistic and less restrictive approach based on the information available at the physical layer (signal strength) is used to capture the interference.In contrast to the logical interference model, this has the excellent advantage of measuring the parameters using online data traffic. The question is whether actual capacity improvements can be achieved by considering the physical interference model.