Protocols, performance assessment and consolidation on interfaces for standardization – D3.3

The following document presents a detailed description of the protocol for the “ Control Channels for the Cooperation of the Cognitive Management System ” (C4MS) which provides the necessary means to enable proper management of Opportunistic Networks. Additionally, the document defines the methodology that was applied for the purpose of signalling evaluation. The protocol overview presented in section 2 of the main document, provides the C4MS principles. The section includes, among others, the description of the protocol identifiers, procedures, protocol state machines and message format as well as the security asp ects. Section 3 provides a high-level description of the data structures defined within the scope of OneFIT project. The data structures are classified into five categories, i.e.: Profiles, Context, Decisions,Knowledge and Policies. The high level description is complemented by some detailed data structures in the Appendix to D3.3 Section 3[10]. Section 4 provides details on the evaluation methodology applied for the purpose of C4MS performance assessment. The section presents the evaluation plan along with a description of metrics that are to be exploited in the scope of WP3. Section 5 and Section 6 are composed of the signalling evaluation results. Section 5 focuses on the estimation of the signalling load imposed by ON management in different ON phases. Additionally some results for the initialization phase (not explicitly mentioned in the previous phases of the project)and security related aspects are also depicted. Section 6 on the other hand is focused on the evaluation of the signalling traffic generated by different ON related algorithms. Conclusions to the document are drawn in section 7. Detailed description of the C4MS procedures, implementation options based on IEEE 802.21, DIAMTER and 3GPP are depicted in the appendix to the D3.3[10] . Additionally, the appendix incorporates the detailed definition of the information data structures and final set of Message Sequence Charts (MSCs) provided for the OneFIT project.Peer ReviewedPreprin

Separation Framework: An Enabler for Cooperative and D2D Communication for Future 5G Networks

Soaring capacity and coverage demands dictate that future cellular networks need to soon migrate towards ultra-dense networks. However, network densification comes with a host of challenges that include compromised energy efficiency, complex interference management, cumbersome mobility management, burdensome signaling overheads and higher backhaul costs. Interestingly, most of the problems, that beleaguer network densification, stem from legacy networks' one common feature i.e., tight coupling between the control and data planes regardless of their degree of heterogeneity and cell density. Consequently, in wake of 5G, control and data planes separation architecture (SARC) has recently been conceived as a promising paradigm that has potential to address most of aforementioned challenges. In this article, we review various proposals that have been presented in literature so far to enable SARC. More specifically, we analyze how and to what degree various SARC proposals address the four main challenges in network densification namely: energy efficiency, system level capacity maximization, interference management and mobility management. We then focus on two salient features of future cellular networks that have not yet been adapted in legacy networks at wide scale and thus remain a hallmark of 5G, i.e., coordinated multipoint (CoMP), and device-to-device (D2D) communications. After providing necessary background on CoMP and D2D, we analyze how SARC can particularly act as a major enabler for CoMP and D2D in context of 5G. This article thus serves as both a tutorial as well as an up to date survey on SARC, CoMP and D2D. Most importantly, the article provides an extensive outlook of challenges and opportunities that lie at the crossroads of these three mutually entangled emerging technologies.Comment: 28 pages, 11 figures, IEEE Communications Surveys & Tutorials 201

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

Smart PIN: performance and cost-oriented context-aware personal information network

The next generation of networks will involve interconnection of heterogeneous individual networks such as WPAN, WLAN, WMAN and Cellular network, adopting the IP as common infrastructural protocol and providing virtually always-connected network. Furthermore, there are many devices which enable easy acquisition and storage of information as pictures, movies, emails, etc. Therefore, the information overload and divergent content’s characteristics make it difficult for users to handle their data in manual way. Consequently, there is a need for personalised automatic services which would enable data exchange across heterogeneous network and devices. To support these personalised services, user centric approaches for data delivery across the heterogeneous network are also required. In this context, this thesis proposes Smart PIN - a novel performance and cost-oriented context-aware Personal Information Network. Smart PIN's architecture is detailed including its network, service and management components. Within the service component, two novel schemes for efficient delivery of context and content data are proposed: Multimedia Data Replication Scheme (MDRS) and Quality-oriented Algorithm for Multiple-source Multimedia Delivery (QAMMD). MDRS supports efficient data accessibility among distributed devices using data replication which is based on a utility function and a minimum data set. QAMMD employs a buffer underflow avoidance scheme for streaming, which achieves high multimedia quality without content adaptation to network conditions. Simulation models for MDRS and QAMMD were built which are based on various heterogeneous network scenarios. Additionally a multiple-source streaming based on QAMMS was implemented as a prototype and tested in an emulated network environment. Comparative tests show that MDRS and QAMMD perform significantly better than other approaches

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

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

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

Xcast Based Routing Protocol For Push To Talk Application In Mobile Ad Hoc Networks

Mobile ad-hoc networks comprise a type of wireless network that can be easily created without the need for network infrastructure or administration. These networks are organized and administered into temporary and dynamic network topologies. Unfortunately, mobile ad-hoc networks suffer from some limitations related to insufficient bandwidth. The proliferation of new IP Multimedia subsystem services (IMs), such as Push-to-talk (PTT) applications consume large amounts of bandwidth, resulting in degraded QoS performance of mobile ad-hoc networks. In this thesis, a Priority XCAST based routing protocol (P-XCAST) is proposed for mobile ad-hoc networks to minimize bandwidth consumption. P-XCAST is based on demand route requests and route reply mechanisms for every destination in the PXCAST layer. To build the network topology and fill up the route table for nodes, the information in the route table is used to classify the XCAST list of destinations according to similarities on their next hop. Furthermore, P-XCAST is merged with a proposed Group Management algorithm to handle node mobility by classifying nodes into two types: group head and member. The proposed protocol was tested using the GloMoSim network simulator under different network scenarios to investigate Quality of Service (QoS) performance network metrics. P-XCAST performance was better by about 20% than those of other tested routing protocols by supporting of group size up to twenty receivers with an acceptable QoS. Therefore, it can be applied under different network scenarios (static or dynamic). In addition Link throughput and average delay was calculated using queuing network model; as this model is suitable for evaluating the IEEE 802.11 MAC that is used for push to talk applications. The analytical results for link throughput and average delay were used to validate the simulated results