L'avènement de la VoIP (Voice over IP) a déclenché une période de profonds changements dans le marché des télécommunications. En particulier, dans le secteur de la téléphonie résidentielle, cette technologie s'est consolidée, rapidement et pour de nombreuses raisons, comme l'évolution de la téléphonie traditionnelle. Dès les tous débuts, et afin d'établir une base de compatibilité permettant l'interconnexion de plusieurs réseaux téléphoniques et la convergence entre les systèmes traditionnels analogiques et leur évolution numérique, l'industrie a demandé l'établissement de cadres normatifs. En réponse à ces besoins, plusieurs standards et protocoles, avec de successives modifications et corrections, ont été publiés dans une période relativement brève. Parmi les plus populaires, SIP (Session Initiation Protocol), un protocole de signalisation, et RTP (Real-Time Transport Protocol), un protocole de transport de flots temps réel, se démarquent et ils sont au coeur de la majorité des applications conçues actuellement. Bien que, aujourd'hui, SIP et RTP sont liés fortement à la téléphonie sur IP, leur portée et leurs possibilités sont beaucoup plus vastes, ce qui déclenche un grand intérêt et justifie l'effort mis dans la conception des implémentations plus performantes et orientées plus spécifiquement à divers serveurs mandataires UA (User Agent). Dans ce contexte, le but du présent projet de maîtrise est de concevoir des piles de protocoles SIP et RTP orientées vers des applications de téléphonie sur IP, dans un environnement embarqué. Des conditions additionnelles sont que les piles doivent être codées en langage C et s'appuyer sur le système d'exploitation en temps réel MicroC/OS-II. Afin de faciliter la portabilité, il doit se prévoir des couches d'abstraction du matériel et du système d'exploitation. Même si les applications ciblées pour le projet sont, principalement, celles de VoIP, la pile SIP doit viser d'autres domaines, notamment des applications de domotique et de contrôle à distance. Cette dernière condition impose, de façon indirecte, d'autres conditions sur la taille du code et la puissance de calcul demandée, car le matériel pour ces types d'applications est d'habitude plus simple et moins puissant que les ordinateurs qui sont souvent utilisés dans les applications professionnelles de communication. Ce mémoire, qui décrit le travail effectué, est organisé en deux parties. La première fait une introduction théorique à la téléphonie sur IP, et sert de fondement à la deuxième partie, où la mise en oeuvre des protocoles SIP et RTP est décrite en détail.L'accent a été mis sur les justifications des décisions prises pendant toute la conception afin d'aider à mieux comprendre la logique appliquée et de permettre sa reconsidération et analyse dans de futures itérations. Comme résultat des contraintes et limitations imposées dans le cadre de ce projet, les piles de protocoles conçues se sont révélées très compactes et performantes, ce que justifie pleinement la continuité du travail dans l'avenir.

Today the telecommunication market is faced with the situation that customers are requesting for new telecommunication services, especially value added services. The concept of Next Generation Networks (NGN) seems to be a solution for this, so this concept finds its way into the telecommunication area. These customer expectations have emerged in the context of NGN and the associated migration of the telecommunication networks from traditional circuit-switched towards packet-switched networks. One fundamental aspect of the NGN concept is to outsource the intelligence of services from the switching plane onto separated Service Delivery Platforms using SIP (Session Initiation Protocol) to provide the required signalling functionality. Caused by this migration process towards NGN SIP has appeared as the major signalling protocol for IP (Internet Protocol) based NGN. This will lead in contrast to ISDN (Integrated Services Digital Network) and IN (Intelligent Network) to significantly lower dependences among the network and services and enables to implement new services much easier and faster. In addition, further concepts from the IT (Information Technology) namely SOA (Service-Oriented Architecture) have largely influenced the telecommunication sector forced by amalgamation of IT and telecommunications. The benefit of applying SOA in telecommunication services is the acceleration of service creation and delivery. Main features of the SOA are that services are reusable, discoverable combinable and independently accessible from any location. Integration of those features offers a broader flexibility and efficiency for varying demands on services. This thesis proposes a novel framework for service provisioning and composition in SIP-based peer-to-peer networks applying the principles of SOA. One key contribution of the framework is the approach to enable the provisioning and composition of services which is performed by applying SIP. Based on this, the framework provides a flexible and fast way to request the creation for composite services. Furthermore the framework enables to request and combine multimodal value-added services, which means that they are no longer limited regarding media types such as audio, video and text. The proposed framework has been validated by a prototype implementation.

Within recent years, the concept of Next Generation Networks (NGN) has become widely accepted within the telecommunication area, in parallel with the migration of telecommunication networks from traditional circuit-switched technologies such as ISDN (Integrated Services Digital Network) towards packet-switched NGN. In this context, SIP (Session Initiation Protocol), originally developed for Internet use only, has emerged as the major signalling protocol for multimedia sessions in IP (Internet Protocol) based NGN. One of the traditional limitations of IP when faced with the challenges of real-time communications is the lack of quality support at the network layer. In line with NGN specification work, international standardisation bodies have defined a sophisticated QoS (Quality of Service) architecture for NGN, controlling IP transport resources and conventional IP QoS mechanisms through centralised higher layer network elements via cross-layer signalling. Being able to centrally control QoS conditions for any media session in NGN without the imperative of a cross-layer approach would result in a feasible and less complex NGN architecture. Especially the demand for additional network elements would be decreased, resulting in the reduction of system and operational costs in both, service and transport infrastructure. This thesis proposes a novel framework for QoS optimisation for media sessions in SIP-based NGN without the need for cross-layer signalling. One key contribution of the framework is the approach to identify and logically group media sessions that encounter similar QoS conditions, which is performed by applying pattern recognition and clustering techniques. Based on this novel methodology, the framework provides functions and mechanisms for comprehensive resource-saving QoS estimation, adaptation of QoS conditions, and support of Call Admission Control. The framework can be integrated with any arbitrary SIP-IP-based real-time communication infrastructure, since it does not require access to any particular QoS control or monitoring functionalities provided within the IP transport network. The proposed framework concept has been deployed and validated in a prototypical simulation environment. Simulation results show MOS (Mean Opinion Score) improvement rates between 53 and 66 percent without any active control of transport network resources. Overall, the proposed framework comes as an effective concept for central controlled QoS optimisation in NGN without the need for cross-layer signalling. As such, by either being run stand-alone or combined with conventional QoS control mechanisms, the framework provides a comprehensive basis for both the reduction of complexity and mitigation of issues coming along with QoS provision in NGN.

VoIP(Voice over IP) is one of the most important applications on the Internet. As the voice coding enhancing, VoIP provides a good voice quality with low bandwidth. Therefore, the IP telephony services have a price advantage with PSTN. In the near future, VoIp may replace with the PSTN to provide a better telephony service. In this paper, we discuss how to design and implement an embedded VoIp user agent system. With the help of DSP(Digital Signal Processor), we speed up the voice data processing. We use Linux as the embedded operating system and other open source library to implement a VoIP user agent base on SIP(Session Initiation Protocol) standard. With the hardware and software co-design, we build up an embedded VoIP Integrated Access Device.

The Session Initiation Protocol (SIP) was designed for wire line networks. It was developed to initiate, modify and terminate sessions between two hosts on a network. When the Internet expanded to include wireless hosts, SIP did not scale well for these wireless hosts because of the nature of the wireless channel. Also, there were issues with mobility and real time communication. This thesis proposes improvements to some of the extensions to SIP, for better performance over wireless channels. We investigate the call setup time for various transport mechanisms viz. TCP and UDP, and study the performance of a dynamic Session Timers compared to the current standard of a periodic refresh mechanism, where the frequency of UPDATEs vary with the condition of the wireless channel. We also propose a handoff algorithm that reduces the handover time with decreased packet losses.

The objective of this project was to design and test a secure IP-based architecture suitable for short duration transactions. This included the development of a prototype test-bed in which various operating scenarios (such as cryptographic options, various IP-based architectures and fault tolerance) were demonstrated. A solution based on SIP secured with TLS was tested on two IP based architectures. Total time, CPU time and heap usage was measured for each architecture and encryption scheme to examine the viability of such a solution. The results showed that the proposed solution stack was able to complete transactions in reasonable time and was able to recover from transaction processor failure. This research has demonstrated a possible architecture and protocol stack suitable for IP-based transaction networks. The benefits of an IP-based transaction network include reduced operating costs for network providers and clients, as shared IP infrastructure is used, instead of maintaining a separate IP and X.25 network.

This thesis deals with an introduction of new functionalities in communication via IP Multimedia Subsystem (IMS) at the beginning. There are described functions of each part and introduced protocols which are used by this system. Thesis presents the method of positioning with Global Positioning System (GPS) together with methods for the distance measuring of two points on the Earth. Services provided by IMS are developed in Service Development Studio which is shortly mentioned too. The developed service is divided into two parts: a client side part and a server side part, which is formed like a servlet. This program component, which is running on server is mentioned at the end of teoretical part. The procedure of service creating, setup and functions are in description of the created service, which provides informations based on geographical coordinates to user, is described after the teoretical part.

This master's thesis describes suggestion of an application designed for marketing research utilizing development environment of Ericsson company using services IMS and SIP. The essay examines the options offered by programming languages Java, PHP, of the database system MySQL, language XML and SIP messages. A creation of a server - client pair of applications used for marketing research was the main focus of this paper was. These applications can be used for distribution of questionnaires and their subsequent collection by the server, which allows for an output of statistics based on the acquired data. It is not necessary to work with all the applications, it is possible to filter based on one or more categories and only have the selected questionnaires sent. The questionnaire results will be shown in a form of straightforward table in the server’s console window. The XML language has been selected for the storing the questionnaires. All technologies and methods used are described within the essay. An integral part of the module is also the creation of a database of questions, which has been created in the PHP programming language. MySQL has been used as a database server. The application output is an XML file, which can be used as an input for an IMS server. The results of this thesis are demonstrated using screenshots taken during the application testing.

>Magister Scientiae - MSc

Deep Hypersaline Anoxic Basin (DHAB) are extreme hypersaline environments located on the seafloor which, despite conditions that are hardly favourable to life, can support thriving ecosystems and demonstrate unique microbial communities. Information on the diversity and functioning of microbial communities living in these harsh environments remains limited, partly because of access to samples and the difficulty of maintaining the sampled microorganisms once back in the laboratory. The brine seawater interface (BSI) is the most active part of DHAB ecosystems and exhibits different successive microbial communities. This study focuses on the Afifi DHAB, a relatively shallow site described in 2020 and located near the eastern coast of the Red Sea, and uses the DNA-Stable Isotope Probing (DNA-SIP) method to link microorganisms’ identity and their associated functions. Seawater layers of different salinities (50 to 250 PSU) from the Afifi BSI were sampled and incubated under anaerobic conditions with different carbon sources: glucose, pyruvate and acetate. Through this approach, the study successfully identified carbon consumers in different layers of the Afifi DHAB and constructed a putative carbon flow among the microorganisms present. Specifically, Vibrionaceae and Halanaerobiaceae are the predominant glucose utilizers, while Thalassospiraceae, Desulfovibrionaceae and Desulfobacteraceae are the main pyruvate and acetate utilizers. Several sulfate-reducing bacteria groups are also identified as potential secondary glucose and pyruvate utilizers, i.e., they utilize the metabolites produced by glucose and pyruvate first consumers. Layers specificities are also revealed, e.g., glucose is consumed by Deferribacteraceae, Rhodobacteraceae and Pseudoalteromonadaceae in the upper, less salty layer, and by Acetothermiia, Melioribacteraceae and Spirochaetacea in the more saline underlying layer. This study provides an initial functional characterization of the microbial community within the Afifi DHAB, highlighting insights into the ecological dynamics and metabolic activities of microorganisms in this unique ecosystem.