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

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Elastic Highly Available Cloud Computing

High availability and elasticity are two the cloud computing services technical features. Elasticity is a key feature of cloud computing where provisioning of resources is closely tied to the runtime demand. High availability assure that cloud applications are resilient to failures. Existing cloud solutions focus on providing both features at the level of the virtual resource through virtual machines by managing their restart, addition, and removal as needed. These existing solutions map applications to a specific design, which is not suitable for many applications especially virtualized telecommunication applications that are required to meet carrier grade standards. Carrier grade applications typically rely on the underlying platform to manage their availability by monitoring heartbeats, executing recoveries, and attempting repairs to bring the system back to normal. Migrating such applications to the cloud can be particularly challenging, especially if the elasticity policies target the application only, without considering the underlying platform contributing to its high availability (HA). In this thesis, a Network Function Virtualization (NFV) framework is introduced; the challenges and requirements of its use in mobile networks are discussed. In particular, an architecture for NFV framework entities in the virtual environment is proposed. In order to reduce signaling traffic congestion and achieve better performance, a criterion to bundle multiple functions of virtualized evolved packet-core in a single physical device or a group of adjacent devices is proposed. The analysis shows that the proposed grouping can reduce the network control traffic by 70 percent. Moreover, a comprehensive framework for the elasticity of highly available applications that considers the elastic deployment of the platform and the HA placement of the application’s components is proposed. The approach is applied to an internet protocol multimedia subsystem (IMS) application and demonstrate how, within a matter of seconds, the IMS application can be scaled up while maintaining its HA status

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

Autonomic Overload Management For Large-Scale Virtualized Network Functions

The explosion of data traffic in telecommunication networks has been impressive in the last few years. To keep up with the high demand and staying profitable, Telcos are embracing the Network Function Virtualization (NFV) paradigm by shifting from hardware network appliances to software virtual network functions, which are expected to support extremely large scale architectures, providing both high performance and high reliability. The main objective of this dissertation is to provide frameworks and techniques to enable proper overload detection and mitigation for the emerging virtualized software-based network services. The thesis contribution is threefold. First, it proposes a novel approach to quickly detect performance anomalies in complex and large-scale VNF services. Second, it presents NFV-Throttle, an autonomic overload control framework to protect NFV services from overload within a short period of time, allowing to preserve the QoS of traffic flows admitted by network services in response to both traffic spikes (up to 10x the available capacity) and capacity reduction due to infrastructure problems (such as CPU contention). Third, it proposes DRACO, to manage overload problems arising in novel large-scale multi-tier applications, such as complex stateful network functions in which the state is spread across modern key-value stores to achieve both scalability and performance. DRACO performs a fine-grained admission control, by tuning the amount and type of traffic according to datastore node dependencies among the tiers (which are dynamically discovered at run-time), and to the current capacity of individual nodes, in order to mitigate overloads and preventing hot-spots. This thesis presents the implementation details and an extensive experimental evaluation for all the above overload management solutions, by means of a virtualized IP Multimedia Subsystem (IMS), which provides modern multimedia services for Telco operators, such as Videoconferencing and VoLTE, and which is one of the top use-cases of the NFV technology

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

Aspect oriented service composition for telecommunication applications

This PhD dissertation investigates how to overcome the negative effects of cross cutting concerns in the development of composite service applications. It proposes a combination of dynamic aspect oriented programming with a rules driven service composition mechanism. This combination allows very flexible utilization of aspects based on run-time data. The thesis contributes a join-point model and it integrates techniques for weaving and advice definition into the underlying composition language and execution engine. A particular focus of the thesis is telecommunication applications with their unique model for utilizing heterogeneous constituent services and their severe real-time requirements. Next to its primary use for modular implementation and flexible deployment of concerns in telecommunication applications, the dissertation discusses AOP as a feature for automated management and customization of service applications. The verification of the proposed solution contributes a detailed assessment of run-time performance, including a theoretical model of the AOP implementation. It allows predicting the performance of various alternative solutions. The proposed solution for combined AOP and service composition provides properties, which directly address challenges in pervasive computing and the Internet of things. Thus, this dissertation advances beyond the telecommunication domain with results applicable to various highly relevant technical developments

Core Network Design of Software Defined Radio Testbed

The 4th generation of cellular system (LTE) does not inherit the traditional voice (circuit-switched) capabilities from its predecessors. Instead it relies on its high speed packet-switched core network with IMS (IP Multimedia Subsystem) for voice capabilities. Even though there are temporary solutions available until LTE gets its full deployment and coverage, operators are looking for a long term solution known as VoIMS which uses VoIP with SIP protocol for voice in the LTE network (VoLTE) through the IMS domain. The scope of this thesis work is to design, implement and verify the working of the core network for an LTE type software defined radio (SDR) testbed which is able to initiate, maintain and terminate voice and data connections. First step in this regard is to search and select the tools, programs and technologies that fulfil the network requirement in terms of network performance and user satisfaction. Next is to build, configure and verify the network operations of the designed network. As SDRs are used for testing purposes, the core network is also designed in correspondence to that, i.e., it is a test (lab) core network with configurations that are simple to implement and do not require coding implementation. The core network makes use of the virtualization technology and is realized with the help of open-source solutions, i.e., protocols and technologies that are customizable as required and does not require licensing for their use. These functionalities are implemented with the help of OpenSIPS, an open-source SIP server, DHCP and DNS servers. Demonstration of the core network verifies that successful voice and video call can be made between registered users on two different networks, running VoIP client software on different operating system platforms. The core network provides features such as voice, video, instant messaging, presence, dynamic IP assignment, IP address to name resolution and mobility