The Virtual Device: Expanding Wireless Communication Services Through Service Discovery and Session Mobility

We present a location-based, ubiquitous service architecture, based on the Session Initiation Protocol (SIP) and a service discovery protocol that enables users to enhance the multimedia communications services available on their mobile devices by discovering other local devices, and including them in their active sessions, creating a 'virtual device.' We have implemented our concept based on Columbia University's multimedia environment and we show its feasibility by a performance analysis

A prototype and demonstrator of Akogrimo’s architecture: An approach of merging grids, SOA, and the mobile Internet

The trend of merging telecommunication infrastructures with traditional Information Technology (IT) infrastructures is ongoing and important for commercial service providers. The driver behind this development is, on one hand, the strong need for enhanced services and on the other hand, the need of telecommunication operators aiming at value-added service provisioning to a wide variety of customers. In the telecommunications sector, the IP Multimedia Subsystem (IMS) is a promising service platform, which may become a ''standard'' for supporting added-value services on top of the next generation network infrastructure. However, since its range of applicability is bound to SIP- enabled services, IMS extensions are being proposed by ''SIPifying'' applications. In parallel to these developments within the traditional IT sector, the notion of Virtual Organizations (VO) enabling collaborative businesses across organizational boundaries is addressed in the framework of Web Services (WS) standards implementing a Service-oriented Architecture (SOA). Here, concepts for controlled resource and service sharing based on WS and Semantic Technologies have been consolidated. Since the telecommunications sector has become, in the meantime ''mobile'', all concepts brought into this infrastructure must cope with the dynamics mobility brings in. Therefore, within the Akogrimo project the VO concept has been extended towards a Mobile Dynamic Virtual Organization (MDVO) concept, additionally considering key requirements of mobile users and resources. Especial attention is given to ensure the duality of the merge of both, SOA and IMS approaches to holistically support SOA-enabled mobile added-value services and their users. This work describes major results of the Akogrimo project, paying special attention to the overall Akogrimo architecture, the prototype implemented, and the key scenario in which the instantiated Akogrimo architecture shows a very clear picture of applicability, use, and an additional functional evaluation

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

On the Design of a SIP-Based Binding Middleware for Next Generation Home Network Services

Abstract. This paper proposes a two-layer component-based middleware frame-work that copes with the complexity of managing and constructing efficient and useful SIP-based home services. In the first layer, the device integration frame-work overcomes the heterogeneity of media home devices by providing protocol-independent components that reify the underlying devices. At the second layer, the binding framework allows constructing open mobile media bindings between SIP and non SIP communication protocol endpoints including media home de-vices. The openness of our framework is motivated by the need of constructing highly flexible home services such as context aware adaptation, session mobility, media session enrichment and QoS. Our framework is implemented as part of a context-aware adaptive middleware on top of the OSGi platform and an illustra-tive use case is shown.

A Unified Mobility Management Architecture for Interworked Heterogeneous Mobile Networks

The buzzword of this decade has been convergence: the convergence of telecommunications, Internet, entertainment, and information technologies for the seamless provisioning of multimedia services across different network types. Thus the future Next Generation Mobile Network (NGMN) can be envisioned as a group of co-existing heterogeneous mobile data networking technologies sharing a common Internet Protocol (IP) based backbone. In such all-IP based heterogeneous networking environments, ongoing sessions from roaming users are subjected to frequent vertical handoffs across network boundaries. Therefore, ensuring uninterrupted service continuity during session handoffs requires successful mobility and session management mechanisms to be implemented in these participating access networks. Therefore, it is essential for a common interworking framework to be in place for ensuring seamless service continuity over dissimilar networks to enable a potential user to freely roam from one network to another. For the best of our knowledge, the need for a suitable unified mobility and session management framework for the NGMN has not been successfully addressed as yet. This can be seen as the primary motivation of this research. Therefore, the key objectives of this thesis can be stated as: To propose a mobility-aware novel architecture for interworking between heterogeneous mobile data networks To propose a framework for facilitating unified real-time session management (inclusive of session establishment and seamless session handoff) across these different networks. In order to achieve the above goals, an interworking architecture is designed by incorporating the IP Multimedia Subsystem (IMS) as the coupling mediator between dissipate mobile data networking technologies. Subsequently, two different mobility management frameworks are proposed and implemented over the initial interworking architectural design. The first mobility management framework is fully handled by the IMS at the Application Layer. This framework is primarily dependant on the IMS’s default session management protocol, which is the Session Initiation Protocol (SIP). The second framework is a combined method based on SIP and the Mobile IP (MIP) protocols, which is essentially operated at the Network Layer. An analytical model is derived for evaluating the proposed scheme for analyzing the network Quality of Service (QoS) metrics and measures involved in session mobility management for the proposed mobility management frameworks. More precisely, these analyzed QoS metrics include vertical handoff delay, transient packet loss, jitter, and signaling overhead/cost. The results of the QoS analysis indicates that a MIP-SIP based mobility management framework performs better than its predecessor, the Pure-SIP based mobility management method. Also, the analysis results indicate that the QoS performances for the investigated parameters are within acceptable levels for real-time VoIP conversations. An OPNET based simulation platform is also used for modeling the proposed mobility management frameworks. All simulated scenarios prove to be capable of performing successful VoIP session handoffs between dissimilar networks whilst maintaining acceptable QoS levels. Lastly, based on the findings, the contributions made by this thesis can be summarized as: The development of a novel framework for interworked heterogeneous mobile data networks in a NGMN environment. The final design conveniently enables 3G cellular technologies (such as the Universal Mobile Telecommunications Systems (UMTS) or Code Division Multiple Access 2000 (CDMA2000) type systems), Wireless Local Area Networking (WLAN) technologies, and Wireless Metropolitan Area Networking (WMAN) technologies (e.g., Broadband Wireless Access (BWA) systems such as WiMAX) to interwork under a common signaling platform. The introduction of a novel unified/centralized mobility and session management platform by exploiting the IMS as a universal coupling mediator for real-time session negotiation and management. This enables a roaming user to seamlessly handoff sessions between different heterogeneous networks. As secondary outcomes of this thesis, an analytical framework and an OPNET simulation framework are developed for analyzing vertical handoff performance. This OPNET simulation platform is suitable for commercial use

A structural and functional specification of a SCIM for service interaction management and personalisation in the IMS

The Internet Protocol Multimedia Subsystem (IMS) is a component of the 3G mobile network that has been specified by standards development organisations such as the 3GPP (3rd Generation Partnership Project) and ETSI (European Telecommunication Standards Institute). IMS seeks to guarantee that the telecommunication network of the future provides subscribers with seamless access to services across disparate networks. In order to achieve this, it defines a service architecture that hosts application servers that provide subscribers with value added services. Typically, an application server bundles all the functionality it needs to execute the services it delivers, however this view is currently being challenged. It is now thought that services should be synthesised from simple building blocks called service capabilities. This decomposition would facilitate the re-use of service capabilities across multiple services and would support the creation of new services that could not have originally been conceived. The shift from monolithic services to those built from service capabilities poses a challenge to the current service model in IMS. To accommodate this, the 3GPP has defined an entity known as a service capability interaction manager (SCIM) that would be responsible for managing the interactions between service capabilities in order to realise complex services. Some of these interactions could potentially lead to undesirable results, which the SCIM must work to avoid. As an added requirement, it is believed that the network should allow policies to be applied to network services which the SCIM should be responsible for enforcing. At the time of writing, the functional and structural architecture of the SCIM has not yet been standardised. This thesis explores the current serv ice architecture of the IMS in detail. Proposals that address the structure and functions of the SCIM are carefully compared and contrasted. This investigation leads to the presentation of key aspects of the SCIM, and provides solutions that explain how it should interact with service capabilities, manage undesirable interactions and factor user and network operator policies into its execution model. A modified design of the IMS service layer that embeds the SCIM is subsequently presented and described. The design uses existing IMS protocols and requires no change in the behaviour of the standard IMS entities. In order to develop a testbed for experimental verification of the design, the identification of suitable software platforms was required. This thesis presents some of the most popular platforms currently used by developers such as the Open IMS Core and OpenSER, as well as an open source, Java-based, multimedia communication platform called Mobicents. As a precursor to the development of the SCIM, a converged multimedia service is presented that describes how a video streaming application that is leveraged by a web portal was implemented for an IMS testbed using Mobicents components. The Mobicents SIP Servlets container was subsequently used to model an initial prototype of the SCIM, using a mUlti-component telephony service to illustrate the proposed service execution model. The design focuses on SIP-based services only, but should also work for other types of IMS application servers as well

Media Processing in Video Conferences for Cooperating Over the Top and Operator Based Networks

Telecom operators have dominated the communication industry for a long time by providing services with guaranteed quality of service. Such services are provided by the operator at the cost of maintaining a high grade network. With the introduction of broadband and internet, many over the top (OTT) services have emerged. These services use the underlying operator networks as a mere bit pipe while all service intelligence resides in the application running on the client device. Introduction of OTT services has seen a good response from general users who are no longer bound to services provided by the network operator. This in turn has caused operators and telecom companies to loose the ownership of their customers. This thesis takes media processing in video conferencing as a case study to compare the two competing domains of operator networks and OTT networks. Both domains offer video conferencing to end users, but they follow different architectures. The study shows that OTT services can perform much better if they utilize support of the underlying network. This will also bring the user base back to the network operator. The proposal is to turn the competition into cooperation between both parties. Assessments are done from both technical as well as business perspectives to assert that such cooperative agreements are possible and should be experimented in real life

Evolução da telefonia na web

Mestrado em Sistemas de InformaçãoCom a ameaça imposta às operadoras por aplicações OTT como WhatsApp ou Skype, diversas iniciativas coordenadas pela GSMA foram criadas para tentar responder a este fenómeno. Paralelamente, com a evolução de tecnologias como HTML5 e WebRTC, novos serviços como o Twilio têm surgido, oferecendo APIs para o desenvolvimento de novas aplicações Web. No entanto, a integração destas tecnologias em tradicionais redes de telecomunicações não faz parte das actuais especificações. Sendo assim, o objectivo desta dissertação consiste na especificação e implementação de um protótipo baseado nestas tecnologias emergentes, integrado com uma rede IMS. Primeiramente, foi feito um estudo do estado de arte, definindo requisitos e casos de uso a serem explorados. De seguida, o desenho da solução foi feito e implementado, tendo sido criado uma plataforma que alia WebRTC e a OneAPI da GSMA (que define funcionalidades básicas para operadores), oferecendo interoperabilidade entre ambos os mundos. A solução é composta por um servidor aplicacional que expõe a API e gateway WebRTC, tendo sido testada e considerada adaptada às necessidades estabelecidas.With the threat to operators by OTT applications such as Skype or WhatsApp, several initiatives coordinated by GSMA were created in an effort to respond to this phenomenon. In parallel, with the evolution of technologies such as HTML5 and WebRTC, new services such as Twilio are now available, offering APIs for web application development. However, the integration of these technologies and traditional telecommunication networks is not a part of the current standards. As such, the objective of this dissertation is the specification and implementation of a prototype based on these emerging technologies, integrated in an IMS network. First, a state-of-the-art analysis was made, defining requirements and use-cases to be explored. Secondly, the design and implementation of the solution was done, creating a platform that unites WebRTC and GSMA’s OneAPI (which exposes basic operator features), offering interoperability between both worlds. The solution is composed by an application server that exposes the API and a WebRTC gateway, having been successfully tested and adapted to the established needs

Quality of Service optimisation framework for Next Generation Networks

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