Designing and prototyping WebRTC and IMS integration using open source tools

WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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A Cross Domain Next Generation Network IPTV Client for Media Center environments

Functions, which can be summarized to the keyword Internet Protocol Television (IPTV) describe the transmission of video services to users via Internet Protocol (IP). Accompanying to this new television transmission path Home Theatre PCs (HTPC) running a so called Media Center platform are more and more entering the living rooms as a companion for the popular LCD and Plasma displays. Perfect ease of use and the visual integration on the screen and also into the living room is raising their acceptance. These HTPCs are a central node for multimedia services such as TV, radio and email within the networked household. Thus, there are good preconditions for the use of a HTPC as end device for Telco operator driven IPTV and telecommunication services. In the context of this diploma thesis possibilities for the provisioning of IPTV and Next Generation Network (NGN) services on a converged multimedia home entertainment platform for the living room will be investigated, especially Vista Media Center platforms. For this reason, standardization activities will be investigated, which deal with the integration of IPTV and telecommunication services into NGN. The validation of the results will be achieved by the design and implementation of a Vista Media Center Add-In, which can be integrated as an IP Multimedia Subsystem (IMS) based User Agent (UA) in ETSI TISPAN Release 2 IPTV infrastructures. Additionally, a Cross Domain messaging service for IMS based UA is created, which enables a cross-network communication between users

Contributions to presence-based systems for deploying ubiquitous communication services

Next-Generation Networks (NGNs) will converge the existing fixed and wireless networks. These networks rely on the IMS (IP Multimedia Subsystem), introduced by the 3GPP. The presence service came into being in instant messaging applications. A user¿s presence information consists in any context that is necessary for applications to handle and adapt the user's communications. The presence service is crucial in the IMS to deploy ubiquitous services. SIMPLE is the standard protocol for handling presence and instant messages. This protocol disseminates users' presence information through subscriptions, notifications and publications. SIMPLE generates much signaling traffic for constantly disseminating presence information and maintaining subscriptions, which may overload network servers. This issue is even more harmful to the IMS due to its centralized servers. A key factor in the success of NGNs is to provide users with always-on services that are seamlessly part of their daily life. Personalizing these services according to the users' needs is necessary for the success of these services. To this end, presence information is considered as a crucial tool for user-based personalization. This thesis can be briefly summarized through the following contributions: We propose filtering and controlling the rate of presence publications so as to reduce the information sent over access links. We probabilistically model presence information through Markov chains, and analyzed the efficiency of controlling the rate of publications that are modeled by a particular Markov chain. The reported results show that this technique certainly reduces presence overload. We mathematically study the amount of presence traffic exchanged between domains, and analyze the efficiency of several strategies for reducing this traffic. We propose an strategy, which we call Common Subscribe (CS), for reducing the presence traffic exchanged between federated domains. We compare this strategy traffic with that generated by other optimizations. The reported results show that CS is the most efficient at reducing presence traffic. We analyze the load in the number of messages that several inter-domain traffic optimizations cause to the IMS centralized servers. Our proposed strategy, CS, combined with an RLS (i.e., a SIMPLE optimization) is the only optimization that reduces the IMS load; the others increase this load. We estimate the efficiency of the RLS, thereby concluding that the RLS is not efficient under certain circumstances, and hence this optimization is discouraged. We propose a queuing system for optimizing presence traffic on both the network core and access link, which is capable to adapt the publication and notification rate based on some quality conditions (e.g, maximum delay). We probabilistically model this system, and validate it in different scenarios. We propose, and implement a prototype of, a fully-distributed platform for handling user presence information. This approach allows integrating Internet Services, such as HTTP or VoIP, and optimizing these services in an easy, user-personalized way. We have developed SECE (Sense Everything, Control Everything), a platform for users to create rules that handle their communications and Internet Services proactively. SECE interacts with multiple third-party services for obtaining as much user context as possible. We have developed a natural-English-like formal language for SECE rules. We have enhanced SECE for discovering web services automatically through the Web Ontology Language (OWL). SECE allows composing web services automatically based on real-world events, which is a significant contribution to the Semantic Web. The research presented in this thesis has been published through 3 book chapters, 4 international journals (3 of them are indexed in JCR), 10 international conference papers, 1 demonstration at an international conference, and 1 national conferenceNext-Generation Networks (NGNs) son las redes de próxima generación que soportaran la convergencia de redes de telecomunicación inalámbricas y fijas. La base de NGNs es el IMS (IP Multimedia Subsystem), introducido por el 3GPP. El servicio de presencia nació de aplicaciones de mesajería instantánea. La información de presencia de un usuario consiste en cualquier tipo de información que es de utilidad para manejar las comunicaciones con el usuario. El servicio de presencia es una parte esencial del IMS para el despliegue de servicios ubicuos. SIMPLE es el protocolo estándar para manejar presencia y mensajes instantáneos en el IMS. Este protocolo distribuye la información de presencia de los usuarios a través de suscripciones, notificaciones y publicaciones. SIMPLE genera mucho tráfico por la diseminación constante de información de presencia y el mantenimiento de las suscripciones, lo cual puede saturar los servidores de red. Este problema es todavía más perjudicial en el IMS, debido al carácter centralizado de sus servidores. Un factor clave en el éxito de NGNs es proporcionar a los usuarios servicios ubicuos que esten integrados en su vida diaria y asi interactúen con los usuarios constantemente. La personalización de estos servicios basado en los usuarios es imprescindible para el éxito de los mismos. Para este fin, la información de presencia es considerada como una herramienta base. La tesis realizada se puede resumir brevemente en los siguientes contribuciones: Proponemos filtrar y controlar el ratio de las publicaciones de presencia para reducir la cantidad de información enviada en la red de acceso. Modelamos la información de presencia probabilísticamente mediante cadenas de Markov, y analizamos la eficiencia de controlar el ratio de publicaciones con una cadena de Markov. Los resultados muestran que este mecanismo puede efectivamente reducir el tráfico de presencia. Estudiamos matemáticamente la cantidad de tráfico de presencia generada entre dominios y analizamos el rendimiento de tres estrategias para reducir este tráfico. Proponemos una estrategia, la cual llamamos Common Subscribe (CS), para reducir el tráfico de presencia entre dominios federados. Comparamos el tráfico generado por CS frente a otras estrategias de optimización. Los resultados de este análisis muestran que CS es la estrategia más efectiva. Analizamos la carga en numero de mensajes introducida por diferentes optimizaciones de tráfico de presencia en los servidores centralizados del IMS. Nuestra propuesta, CS, combinada con un RLS (i.e, una optimización de SIMPLE), es la unica optimización que reduce la carga en el IMS. Estimamos la eficiencia del RLS, deduciendo que un RLS no es eficiente en ciertas circunstancias, en las que es preferible no usar esta optimización. Proponemos un sistema de colas para optimizar el tráfico de presencia tanto en el núcleo de red como en la red de acceso, y que puede adaptar el ratio de publicación y notificación en base a varios parametros de calidad (e.g., maximo retraso). Modelamos y analizamos este sistema de colas probabilísticamente en diferentes escenarios. Proponemos una arquitectura totalmente distribuida para manejar las información de presencia del usuario, de la cual hemos implementado un prototipo. Esta propuesta permite la integracion sencilla y personalizada al usuario de servicios de Internet, como HTTP o VoIP, asi como la optimizacón de estos servicios. Hemos desarrollado SECE (Sense Everything, Control Everything), una plataforma donde los usuarios pueden crear reglas para manejar todas sus comunicaciones y servicios de Internet de forma proactiva. SECE interactúa con una multitud de servicios para conseguir todo el contexto possible del usuario. Hemos desarollado un lenguaje formal que parace como Ingles natural para que los usuarios puedan crear sus reglas. Hemos mejorado SECE para descubrir servicios web automaticamente a través del lenguaje OWL (Web Ontology Language)

A criação de redes de próxima geração IMS usando produtos Open Source

Estágio realizado na PT InovaçãoTese de mestrado integrado. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200

A Decentralized Session Management Framework for Heterogeneous Ad-Hoc and Fixed Networks

Wireless technologies are continuously evolving. Second generation cellular networks have gained worldwide acceptance. Wireless LANs are commonly deployed in corporations or university campuses, and their diffusion in public hotspots is growing. Third generation cellular systems are yet to affirm everywhere; still, there is an impressive amount of research ongoing for deploying beyond 3G systems. These new wireless technologies combine the characteristics of WLAN based and cellular networks to provide increased bandwidth. The common direction where all the efforts in wireless technologies are headed is towards an IP-based communication. Telephony services have been the killer application for cellular systems; their evolution to packet-switched networks is a natural path. Effective IP telephony signaling protocols, such as the Session Initiation Protocol (SIP) and the H 323 protocol are needed to establish IP-based telephony sessions. However, IP telephony is just one service example of IP-based communication. IP-based multimedia sessions are expected to become popular and offer a wider range of communication capabilities than pure telephony. In order to conjoin the advances of the future wireless technologies with the potential of IP-based multimedia communication, the next step would be to obtain ubiquitous communication capabilities. According to this vision, people must be able to communicate also when no support from an infrastructured network is available, needed or desired. In order to achieve ubiquitous communication, end devices must integrate all the capabilities necessary for IP-based distributed and decentralized communication. Such capabilities are currently missing. For example, it is not possible to utilize native IP telephony signaling protocols in a totally decentralized way. This dissertation presents a solution for deploying the SIP protocol in a decentralized fashion without support of infrastructure servers. The proposed solution is mainly designed to fit the needs of decentralized mobile environments, and can be applied to small scale ad-hoc networks or also bigger networks with hundreds of nodes. A framework allowing discovery of SIP users in ad-hoc networks and the establishment of SIP sessions among them, in a fully distributed and secure way, is described and evaluated. Security support allows ad-hoc users to authenticate the sender of a message, and to verify the integrity of a received message. The distributed session management framework has been extended in order to achieve interoperability with the Internet, and the native Internet applications. With limited extensions to the SIP protocol, we have designed and experimentally validated a SIP gateway allowing SIP signaling between ad-hoc networks with private addressing space and native SIP applications in the Internet. The design is completed by an application level relay that permits instant messaging sessions to be established in heterogeneous environments. The resulting framework constitutes a flexible and effective approach for the pervasive deployment of real time applications.The invention of the phone has radically changed the way people communicate, as it allowed persons to get in contact instantly no matter of their location. However, phone communication has been confined for decades to a fixed location, be it one's own house or a phone boot. The widespread affirmation of cellular technologies has had for fixed telephony a similar impact that the invention of the phone has had on communications years before. With mobile phones, people are enabled to talk with each other anytime and anywhere. Internet has also revolutionized the way people communicate. E-mails have soon become one of the Internet killer applications. Later on, instant messaging, popularly known as chatting, has gained huge consensus among net surfers. Only recently, the use of the Internet for voice communication is becoming mainstream, and the so called Voice over IP (VoIP) applications (Skype is probably the most famous for the masses) are becoming common use. Despite its popularity, Internet still suffers from the inherent limitations that affected early telephony: it is fixed. The usage of Internet on the move still does not constitute the easiest and most satisfactory user experience, due to capabilities and limitations of the access technology, terminals, services and applications. Efforts for mobilizing the Internet are ongoing both in the industrial and in the academic worlds, but several bricks are needed to build the wall of mobile Internet. This dissertation provides one of these bricks, describing a solution that allows the deployment of multimedia applications (chat, VoIP, gaming) in mobile environments. In other words, this dissertation gives solutions for facilitating ubiquitous Internet-based communication, anytime and anywhere. The vision that we want to become true is that Internet must become mobile in the same way as fixed telephony has become mobile thanks to the cellular technology. More than this, we do not want that users are limited by the presence of an infrastructure to communicate with each other. In order to achieve this, we present solutions to deploy Internet-based services and applications in environments where no support from servers is available. In other words, we enable direct device-to-device, user-to-user Internet communication. Our contribution is mainly focused on the steps needed to establish the communication, the so called session establishment or signaling phase. We have validated our signaling framework by building a chat application that utilizes its features and works in server-less environments. The custom server-less solution does not prohibit to connect at the same time with the Internet, so that one can engage in a chess game using direct communication with a person in the proximity while having a chat in progress with a friend using standard Internet services. The challenge that we had to face is that Internet services and applications are usually built implying support from a centralized server. In order to deploy direct user-to-user Internet services, while maintaining interoperability with mainstream services, we had to enhance native Internet services to work without infrastructure support, without sacrificing interoperability with standard Internet applications. To conclude, we have placed our brick on the still yet to be completed wall of mobile Internet. Our hope is that one day, thanks also to this brick, everybody will be able to enjoy Internet-based applications as easily as now it is possible to use mobile telephony services

Integrating Context-Awareness in the IP Multimedia Subsystem for Enhanced Session Control and Service Provisioning Capabilities

The 3GPP-defined IP Multimedia Subsystem (IMS) is becoming the de-facto standard for IP-based multimedia communication services. It consists of an overlay control and service layer that is deployed on top of IP-based mobile and fixed networks. This layer encompasses a set of common functions (e.g. session control functions allowing the initiation/modification/termination of sessions) and service logics that are needed for the seamless provisioning of IP multimedia services to users, via different access technologies. As it continues to evolve, the IMS still faces several challenges including: the enabling of innovative and personalized services that would appeal to users and increase network operators' revenues; its interaction with other types of networks (e.g. wireless sensor networks) as means to enhance its capabilities; and the support of advanced QoS schemes that would manage the network resources in an efficient and adaptive manner. The context-awareness concept, which comes from the pervasive computing field, signifies the ability to use situational information (or context) in support to operations and decision making and for the provision of relevant services to the user. Context-awareness is considered to enhance users' experience and is seen as an enabler to adaptability and service personalization - two capabilities that could play important roles in telecommunication environments. This thesis focuses on the introduction of the context-awareness technology in the IMS, as means to enhance its session control and service provisioning capabilities. It starts by presenting the necessary background information, followed by a derivation of requirements and a review of the related work. To ensure the availability of contextual information within the network, we then propose an architecture for context information acquisition and management in the IMS. This architecture leverages and extends the 3GPP presence framework. Building on the capabilities of this architecture, we demonstrate how the managed information could be integrated in IMS operations, at the control and service levels. Showcasing control level integration, we propose a novel context-aware call differentiation framework as means to offer enhanced QoS support (for sessions/calls) in IMS-based networks. This framework enables the differentiation between different categories of calls at the IMS session control level, via dynamic and adaptive resource allocation, in addition to supporting a specialized charging model. Furthermore, we also propose a framework for enhanced IMS emergency communication services. This framework addresses the limitations of existing IP-based emergency solutions, by offering three main improvements: a QoS-enhanced emergency service; a context-aware personalized emergency service; and a conferencing-enhanced emergency service. We demonstrate the use of context awareness at the IMS service level using two new context-aware IMS applications. Finally, to validate our solutions and evaluate their performance, we build various proof-of-concept prototypes and OPNET simulation model

Convergence du web et des services de communication

Les services de communication, du courrier postal à la téléphonie, en passant par la voix et la vidéo sur IP (Internet Protocol), la messagerie électronique, les salons de discussion sur Internet, les visioconférences ou les télécommunications immersives ont évolué au fil du temps. Un système de communication voix-vidéo sur IP est réalisé grâce à deux couches architecturales fondamentales : la couche de signalisation et la couche média. Le protocole de signalisation est utilisé pour créer, modifier et terminer des sessions multimédias entre des participants. La couche de signalisation est divisée en deux sous-couches - la couche de service et celle de contrôle - selon la spécification de l IP Multimedia Subsystem (IMS). Deux systèmes de communication largement utilisés sont l IMS et SIP Pair-à- Pair (P2P SIP). Les fournisseurs de services, qui se comportent en tant qu intermédiaires entre appelants et appelés, implémentent les systèmes de communication, contrôlant strictement la couche signalisation. Or ces fournisseurs de services ne prennent pas en compte la diversité des utilisateurs. Cette thèse identifie trois barrières technologiques dans les systèmes de communication actuels et plus précisément concernant la couche de signalisation. I. Un manque d ouverture et de flexibilité dans la couche de signalisation pour les utilisateurs. II. Un développement difficile des services basés sur le réseau et les sessions. III. Une complexification du la couche de signalisation lors d un très grand nombre d appels. Ces barrières technologiques gênent l innovation des utilisateurs avec ces services de communication. Basé sur les barrières technologiques listées cidessus, le but initial de cette thèse est de définir un concept et une architecture de système de communication dans lequel chaque individu devient un fournisseur de service. Le concept, "My Own Communication Service Provider" (MOCSP) et le système MOCSP sont proposés, accompagné d un diagramme de séquence. Ensuite, la thèse fournit une analyse qui compare le système MOCSP avec les systèmes de communication existants en termes d ouverture et de flexibilité. La seconde partie de la thèse présente des solutions pour les services basés sur le réseau ou les sessions, mettant en avant le système MOCSP proposé. Deux services innovants, user mobility et partial session transfer/retrieval (PSTR) sont pris comme exemples de services basés sur le réseau ou les sessions. Les services basés sur un réseau ou des sessions interagissent avec une session ou sont exécutés dans une session. Dans les deux cas, une seule entité fonctionnelle entre l appelant et l appelé déclenche le flux multimédia pendant l initialisation de l appel et/ou en cours de communication. De plus, la coopération entre le contrôle d appel réseau et les différents pairs est facilement réalisé. La dernière partie de la thèse est dédiée à l extension de MOCSP en cas de forte densité d appels, elle inclut une analyse comparative. Cette analyse dépend de quatre facteurs - limite de passage à l échelle, niveau de complexité, ressources de calcul requises et délais d établissement de session - qui sont considérés pour évaluer le passage à l échelle de la couche de signalisation. L analyse comparative montre clairement que la solution basée sur MOCSP est simple et améliore l usage effectif des ressources de calcul par rapport aux systèmes de communication traditionnelsDifferent 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 systemsEVRY-INT (912282302) / SudocSudocFranceF