The work discusses about the PBX Yate, describes its configuration from the initial installation, through the basic setup and description of the way through the logs of user registration, the syntax for defining the calling plan, including examples, and additional services, including demonstration of practical verification of the functionality in the computer lab. Are also created two samples job for more familiar with the principles of operation and the Yate PBX settings.

The master’s thesis presents routing protocols, SIP handshake, describes modern VoIP networks and its features. Thesis is primary focused on networks with integrated MPLS. The most known advantages and disadvantages were described of modern network based on such technology. Further, networks and a router with installed Cisco Call Manager Express 7.1 were configured, which is provided by CISCO. For recreating of fully working telephone network IP PBX10 has been used from SMC Networks. For connection between different phones within these integrated networks, SIP trunk was configured on SMC PBX10 as well as on router, which is connecting three configured Cisco IP phones. Each phone has its own number according to the dial plan. For calls from one network to another the special pattern has been established, so the number of each caller can be easily identified by added digit. On one of the CISCO routers was shown a configuration of MPLS and OSPF protocols. With an analyzer VePAL TX300e, the lab network has been measured to analyze QoS parameters of the network according to standard RFC 2544. Thesis also gives references to lab devices, which have been used accordingly to the work.

Die vorliegende Arbeit beschreibt SIP Clients für Linux und Windows. Die Programme funktionieren in einem leistungsstarken Netzwerk, wie z.B. dem Intranet der Technischen Universität Chemnitz, problemlos. Alle Funktionen wurden in einer VoIP-Umgebung getestet. Die Sprachübertragungqualität über das Internet mittels dieser Clients ist jedoch nicht Gegenstand der Arbeit und somit auch nicht erprobt worden. In vielen Unternehmensbereichen fällt häufig der Begriff Echtzeitkommunikation in Verbindung mit einer geeigneten Infrastruktur, ausgelöst durch den Zuwachs der verfügbaren Netzwerkbandbreite und somit immer realistischer werdender Sprach- und Videoübertragungen. Ein Ansatz für die Anwendungen ist das Protokoll SIP (Session Initiation Protocol), welches für die Signalisierung der Video- und Sprachübertragung verwendet wird.

Context awareness is an important aspect of pervasive and ubiquitous computing. By utilizing contextual information gathered from the environment, applications can adapt to the user’s specific situation. In this thesis, user context is used to automatically discover multimedia devices and services that can be used by a mobile Session Initiation Protocol (SIP) user agent. The location of the user is captured using various sensing technologies to allow users of our SIP user agent to interact with network attached projectors, speakers, and cameras in a home or office environment. In order to determine the location of the user, we have developed and evaluated a context aggregation framework that gathers and analyzes contextual information from various sources such as passive infrared sensors, infrared beacons, light intensity, and temperature sensors. Once the location of the user is determined, the Service Location Protocol (SLP) is used to search for services. For this purpose, we have implemented a mobile SLP user agent and integrated it with an existing SIP user agent. The resulting mobile SIP user agent is able to dynamically utilize multimedia devices around it without requiring the user to do any manual configuration. This thesis also addressed the challenge of building trust relationship between the user agent and the multimedia services. We propose a mechanism which enables the user agent authenticate service advertisements before starting to redirect media streams. The measurements we have performed indicate that the proposed context aggregation framework provides more accurate location determination when additional sensors are incorporated. Furthermore, the performance measurements indicate that the delay incurred by introducing context awareness to the SIP user agent is acceptable for a small deployment such as home and office environment. In order to realize large scale deployments, future investigations are recommended to further improve the performance of the framework. / Att vara medveten om kontexten är en viktig synpunkt av präglande och allestädes närvarande uppskattning av omgivningen. Genom att utnyttja den kontextuella informationen som samlats in från omgivningen, kan applikationen anpassas till användarens specifika situation. I denna avhandling använder man användarens sammanhang för att automatiskt upptäcka multimedia utrustning och tjänster som kan användas av en mobil Session Initiation Protokoll (SIP) användaragent. Placeringen av användaren mäter man med hjälp av olika sensorer för att låta användare av vår SIP användaragent att interagera med nätverk tillkopplat projektorer, högtalare och kameror i hem eller kontorsmiljöer. För att avgöra var användaren befinner har vi utvecklat och utvärderat en sammanhangsstruktur som samlar in och analyserar innehållsbaserad information från olika källor; passiva infraröda sensorer, infraröd beacons, ljusstyrkan och temperaturgivare. Efter bestämmaning användarens placering den så kallade Service Location Protocol (SLP) användas för att söka efter tjänster. För detta ändamål har vi genomfört en mobil SLP användaragent och integrerat denna med ett befintligt SIP användaragent. Den resulterande i mobil SIP användaragent som dynamiskt kan utnyttja multimedia utrustning runt omkring utan att kräva att användaren skall kunna göra någon manuell konfiguration. Avhandlingen tar även upp den utmaningen som krävs för att bygga förtroende mellan användaragenten och multimedia tjänster. Vi föreslår en mekanism som gör det möjligt för användaragenten att verifiera tjänstannonsering innan man börjar dirigera medieströmmar. Dessutom så indikerar mätningarna av prestanda att fördröjningen som man utsätter den för genom att introducera ”medvetenhet om sammanhanget” till SIP användaragenten är acceptabel på hemma eller i en företagsmiljö. För att stora spridningar skall bli verklighet så rekommenderas det att göra mer forskning för att förbättra prestanda.

Voice over Internet Protocol (VoIP) is a service that has recently gained a lot of attention from the telecommunications (telecom) world since both Internet service providers (ISPs) and telecommunications operators have realized the important advantages that it can offer. Although traditional telephony is well established both in the telecom world and in our daily lives, VoIP is now competing with it by offering cost savings, simplicity, and introducing new ways of communicating. Internet service providers have already started deploying efficient VoIP services for their customers and carriers are transforming their network infrastructures in order to be able to accommodate the requirements of VoIP traffic. There are a lot of essential factors that both providers and carriers have to take into consideration in order to efficiently build and operate VoIP technologies. Proper service planning and well-established monitoring and troubleshooting procedures are vital for successful VoIP service. This thesis focuses on commercial VoIP implementation at the carrier’s side and investigates how a carrier can efficiently maintain and troubleshoot their VoIP infrastructure so as to comply with the Service Level Agreements (SLAs) they have signed with their customers (ISP providers), as well as analyses proactive actions that can betaken for minimizing the resources required for customer support. As an outcome, this thesis presents efficient ways of network planning and monitoring, as well as it provides conclusions regarding what are the efficient methods for troubleshooting the carrier’s VoIP products inboth technical and organizational level. / Röst över Internet Protokoll (VoIP) är en tjänst som nyligen har fått ökad uppmärksamhet inom telekommunikations (telecom) branschen eftersom att både Internetleverantörer (ISPs) och telecom operatörer har insett vilka fördelar som tjänsten erbjuder. Även om traditionell telefoni är väl etablerad i både telecombranschen och vår vardag, så kan VoIP konkurrera genom att erbjuda kostnadsbesparingar, förenkling, och introducera nya sätt att kommunicera på. IP leverantörer har redan påbörjat lansering av effektiva VoIP tjänster till sina kunder och telecom carriers bygger om sin nätverksstruktur för att möta kraven av VoIP traffik. Det finns många faktorer att bejaka för både IP leverantörer och telecom carriers för att effektivt bygga och driva VoIP nätverk. Noggrann produktplanering och väletablerad övervakning samt felsökningsprocedurer är en vital del i en framgångsrik VoIP tjänst. Denna avhandling fokuserar på VoIP implementering hos en telecom carrier och hur en telecom carrier effektivt kan underhålla och felsöka VoIP infrastruktur för att möta de servicenivåavtal de har skrivit med sina kunder (IP leverantörer), samt analysera det förebyggande åtgärder som kan tas för att minimera de resurser som behövs till kundtjänst. Denna avhandling presenteras effektiva tillvägagångssätt för planering och övervakning samt erbjuder effektiva,teknisk och organisationella metoder för felsökning av en telecom carriers VoIP produkter.

Video conversation on mobile devices is popularizing with the advent of 3G. The enhanced network capacity thus available enables transmission of video data over the internet. It has been forecasted by several VOIP service organizations that the present IVR systems will evolve into Voice and Video Response (IVVR) Systems. However, this evolution has many technical challenges on the way. Architectures to implement queuing systems for video data and standards for inter conversion of video data between the formats supported by calling parties are two of these challenges. This thesis is an analysis of queues and media transcoding for IVVRs. A major effort in this work involves constructing a prototype IVVR queuing system. The system is constructed by using an open source server named Asterisk and MySql database. Asterisk is a SIP based Public Exchange Server (PBX) and also a development environment for VOIP based IVRs. Functional scenarios for SIP session establishment and the corresponding session setup times for this queueing model are measured. The results indicate that the prototype serves as a sufficient model for a queue, although a significant delay is introduced for session establishment.  The work also includes analysis of integrating DiaStar™, is a SIP based media transcoding engine to this queue. However, this system is not complete to function with DiaStar for media translation. The study concludes with a mention of the areas for future work on this particular system and the general state of IVVR queuing systems in the industry. / Videosamtal på mobila enheter är popularisera med tillkomsten av 3G. Den förbättrade nätkapacitet så tillgänglig möjliggör överföring av videodata över Internet. Det har prognos av flera VOIP serviceorganisationer att de nuvarande IVR-system kommer att utvecklas till röst och video Response (IVVR) System. Dock har denna utveckling många tekniska utmaningar på vägen. Arkitekturer för att genomföra kösystem för videodata och standarder för bland konvertering av videodata mellan format som stöds för uppringande är två av dessa utmaningar. Denna avhandling är en analys av köer och media kodkonvertering för IVVRs. En stor insats i detta arbete innebär att bygga en prototyp IVVR kösystem. Systemet är konstruerat med hjälp av en öppen källkod-server som heter Asterisk och MySQL-databas. Asterisk är en SIP-baserad Public Exchange Server (PBX) och även en utvecklingsmiljö för VOIP-baserade IVRs. Funktionella scenarier för SIP session etablering och motsvarande sessionen inställningar för den föreslagna kö modell mäts. Resultaten indikerar att prototypen tjänar som en tillräcklig modell för en kö, även om en betydande fördröjning införs för sessionsupprättandebegäran. Arbetet omfattar även analys av integrering DiaStar™ är en SIP-baserad media kodkonvertering motor till denna kö. Emellertid är detta system inte helt att fungera med DiaStar för media translation. The studie avslutas med ett omnämnande av de områden för framtida arbete med detta system och det allmänna tillståndet i IVVR kö-system i branschen.

A arquitetura IP Multimedia Subsystem, proposta pelo consórcio 3rd Generation Partnership Project como base para o suporte à convergência entre telefonia móvel e a Internet, define uma série de elementos arquiteturais, entre os quais, o componente Call Session Control Function e o protocolo Session Initiation Protocol. Session Initiation Protocol é um protocolo da camada de aplicação utilizado para estabelecer, modificar e terminar sessões multimídia entre dispositivos. Em redes baseadas na arquitetura IP Multimedia Subsystem, o Session Initiation Protocol é o responsável pela comunicação entre dispositivos e a rede, e entre os componentes responsáveis pelo gerenciamento de sessão. Nos últimos anos, estudos detectaram degradação de desempenho em redes baseadas na arquitetura IP Multimedia Subsystem em função das características centralizadas do Session Initiation Protocol e dos componentes de gerenciamento de sessão. Este trabalho apresenta uma arquitetura distribuída para redes baseadas em IP Multimedia Subsystem, tendo como fundamento o paradigma de computação paralela baseado em espaço de tuplas onde os servidores são organizados em uma rede P2P, com objetivo de prover uma infraestrutura escalável e tolerante a falhas. A validação da arquitetura em termos de desempenho e escalabilidade se deu através de modelagem formal e simulação com Redes de Petri Coloridas. / The IP Multimedia Subsystem architecture, proposed by the 3rd Generation Partnership Project consortium as basis to support the convergence between mobile networks and the Internet, defines a set of architectural elements, among them, the Call Session Control Function and the Session Initiation Protocol. The Session Initiation Protocol is an application layer protocol used to establish, modify and terminate sessions between devices. On the IP multimedia subsystem based network, the Session Initiation Protocol play a key role on the communication between devices and the network, and between session management components. In the last years, studies have detected a performance bottleneck on IP multimedia subsystem networks due to centralized characteristic of the Session Initiation Protocol and in Session Control components. This work shows a distributed architecture for IP Multimedia Subsystem networks based on the tuple space paradigm, and the servers structured in a P2P network, aiming to achieve a scalable and fault-tolerant infrastructure. The validation of the architecture on the performance and scalability took place through the Coloured Petri Net formal modeling and simulation.

Un des aspects les plus intéressants des identités numériques du type "Me-Sphere" est sa variabilité : les usagers d’Internet utilisent les technologies de l’écriture et de l’image, les espaces privés/publiques des RS et les dispositifs de connectivité à Internet afin de transformer leurs identités numériques. Nous explorons ce développement hypermédiatisé de la Digital Identity. Notre hypothèse de recherche est : l’hypermédiatisation reconfigure les manières de produire et de percevoir des informations et des connaissances. Cela signifie que la production de contenus d'information a des implications logiques, esthétiques et éthiques liées à la pratique social de l'identité numérique. Mots-clés : Identité numérique (Digital ID), Réseaux socionumériques (RS), Interaction, Hypermédiatisation, Me-Sphere / One of the most interesting aspects of "Me-Sphere" digital identities is their variability: (a) Internet users employ the technologies of writing and images, (b) private/public areas of social network, and (c) connectivity devices of Internet, to transform their digital identities. We explore this hyper-mediatized development of Personal Identity. Our research hypothesis is: the hyper-mediatization reconfigures the production and information of knowledge. This means that the production of information contents has logical, aesthetic and ethical implications on the social practice of digital identity. Keywords Digital identity (digital ID), Social Networks (RS), Interaction, Hyper-mediatization, Me-Sphere

A arquitetura IP Multimedia Subsystem, proposta pelo consórcio 3rd Generation Partnership Project como base para o suporte à convergência entre telefonia móvel e a Internet, define uma série de elementos arquiteturais, entre os quais, o componente Call Session Control Function e o protocolo Session Initiation Protocol. Session Initiation Protocol é um protocolo da camada de aplicação utilizado para estabelecer, modificar e terminar sessões multimídia entre dispositivos. Em redes baseadas na arquitetura IP Multimedia Subsystem, o Session Initiation Protocol é o responsável pela comunicação entre dispositivos e a rede, e entre os componentes responsáveis pelo gerenciamento de sessão. Nos últimos anos, estudos detectaram degradação de desempenho em redes baseadas na arquitetura IP Multimedia Subsystem em função das características centralizadas do Session Initiation Protocol e dos componentes de gerenciamento de sessão. Este trabalho apresenta uma arquitetura distribuída para redes baseadas em IP Multimedia Subsystem, tendo como fundamento o paradigma de computação paralela baseado em espaço de tuplas onde os servidores são organizados em uma rede P2P, com objetivo de prover uma infraestrutura escalável e tolerante a falhas. A validação da arquitetura em termos de desempenho e escalabilidade se deu através de modelagem formal e simulação com Redes de Petri Coloridas. / The IP Multimedia Subsystem architecture, proposed by the 3rd Generation Partnership Project consortium as basis to support the convergence between mobile networks and the Internet, defines a set of architectural elements, among them, the Call Session Control Function and the Session Initiation Protocol. The Session Initiation Protocol is an application layer protocol used to establish, modify and terminate sessions between devices. On the IP multimedia subsystem based network, the Session Initiation Protocol play a key role on the communication between devices and the network, and between session management components. In the last years, studies have detected a performance bottleneck on IP multimedia subsystem networks due to centralized characteristic of the Session Initiation Protocol and in Session Control components. This work shows a distributed architecture for IP Multimedia Subsystem networks based on the tuple space paradigm, and the servers structured in a P2P network, aiming to achieve a scalable and fault-tolerant infrastructure. The validation of the architecture on the performance and scalability took place through the Coloured Petri Net formal modeling and simulation.

En raison de la complexité croissante des systèmes réactifs, le test est devenu un des éléments essentiels dans le processus de leur développement. Les tests de conformité avec des méthodes formelles concernent la correction du contrôle fonctionnel, par le biais des tests d'un système en boîte noire avec une spécification formelle du système. Les techniques passives de test sont utilisées lorsque l’exécution des systèmes testés ne peut pas être perturbée ou l'interface du système n'est pas fournie. Les techniques passives de test sont fondées sur l'observation et la vérification des propriétés du comportement d'un système sans interférer avec son fonctionnement normal. Les tests contribuent également à établir les comportements anormaux pendant l’exécution sur la base de l'observation de toute déviation d'un comportement prédéterminé. L'objectif principal de cette thèse est de présenter une nouvelle approche pour la mise en place des tests passifs fondés sur l'analyse des parties contrôle et données du système sous test. Au cours des dernières décennies, de nombreuses théories et outils ont été développés pour effectuer les tests de conformité. De fait, les spécifications ou les propriétés des systèmes réactifs sont souvent modélisés par différentes variantes de Labeled Transition Systems (LTS). Toutefois, ces méthodes ne prennent pas explicitement en compte les parties données du système, étant donné que le modèle sous-jacent de LTS n’est pas en mesure de le faire. Par conséquent, avec ces approches il est nécessaire d'énumérer les valeurs des données avant la modélisation du système. Cela conduit souvent au problème de l'explosion combinatoire de l'état-espace. Pour palier à cette limitation, nous avons étudié un modèle appelé Input-Output Symbolic Transition Systems (IOSTS) qui inclut explicitement toutes les données d'un système réactif. De nombreuses techniques de tests passives prennent uniquement en considération la partie du contrôle du système en négligeant les données, ou elles sont confrontées à une quantité énorme de données du processus. Dans notre approche, nous prenons en compte la partie contrôle et données en intégrant les concepts d'exécution symbolique et nous améliorons l'analyse de traces en introduisant des techniques de slicing des traces d’exécution. Les propriétés sont décrites à l'aide d'IOSTS et nous illustrons dans notre approche comment elles peuvent être testées sur l'exécution réelle des traces en optimisant l'analyse. Ces propriétés peuvent être conçues pour tester la conformité fonctionnelle d'un protocole ainsi que des propriétés de sécurité. Au-delà de l'approche théorique, nous avons développé un outil logiciel qui implémente les algorithmes présentés dans nos travaux. Enfin, comme preuve de concept de notre approche et de l'outil logiciel, nous avons appliqué les techniques à deux études de cas réels : le protocole SIP et le protocole Bluetooth / Due to the increasing complexity of reactive systems, testing has become an important part in the process of the development of such systems. Conformance testing with formal methods refers to checking functional correctness, by means of testing, of a black-box system under test with respect to a formal system specification, i.e., a specification given in a language with a formal semantics. In this aspect, passive testing techniques are used when the implementation under test cannot be disturbed or the system interface is not provided. Passive testing techniques are based on the observation and verification of properties on the behavior of a system without interfering with its normal operation, it also helps to observe abnormal behavior in the implementation under test on the basis of observing any deviation from the predefined behavior. The main objective of this thesis is to present a new approach to perform passive testing based on the analysis of the control and data part of the system under test. During the last decades, many theories and tools have been developed to perform conformance testing. However, in these theories, the specifications or properties of reactive systems are often modeled by different variants of Labeled Transition Systems (LTS). However, these methodologies do not explicitly take into account the system's data, since the underlying model of LTS are not able to do that. Hence, it is mandatory to enumerate the values of the data before modeling the system. This often results in the state-space explosion problem. To overcome this limitation, we have studied a model called Input-Output Symbolic Transition Systems (IOSTS) which explicitly includes all the data of a reactive system. Many passive testing techniques consider only the control part of the system and neglect data, or are confronted with an overwhelming amount of data values to process. In our approach, we consider control and data parts by integrating the concepts of symbolic execution and we improve trace analysis by introducing trace slicing techniques. Properties are described using Input Output Symbolic Transition Systems (IOSTSs) and we illustrate in our approach how they can be tested on real execution traces optimizing the trace analysis. These properties can be designed to test the functional conformance of a protocol as well as security properties. In addition to the theoretical approach, we have developed a software tool that implements the algorithms presented in this paper. Finally, as a proof of concept of our approach and tool we have applied the techniques to two real-life case studies: the SIP and Bluetooth protocol