Modelling, Dimensioning and Optimization of 5G Communication Networks, Resources and Services

This reprint aims to collect state-of-the-art research contributions that address challenges in the emerging 5G networks design, dimensioning and optimization. Designing, dimensioning and optimization of communication networks resources and services have been an inseparable part of telecom network development. The latter must convey a large volume of traffic, providing service to traffic streams with highly differentiated requirements in terms of bit-rate and service time, required quality of service and quality of experience parameters. Such a communication infrastructure presents many important challenges, such as the study of necessary multi-layer cooperation, new protocols, performance evaluation of different network parts, low layer network design, network management and security issues, and new technologies in general, which will be discussed in this book

Hybrid switching : converging packet and TDM flows in a single platform

Optical fibers have brought fast and reliable data transmission to today’s network. The immense fiber build-out over the last few years has generated a wide array of new access technologies, transport and network protocols, and next-generation services in the Local Area Network (LAN), Metropolitan Area Network (MAN), and Wide Area Network (WAN). All these different technologies, protocols, and services were introduced to address particular telecommunication needs. To remain competitive in the market, the service providers must offer most of these services, while maintaining their own profitability. However, offering a large variety of equipment, protocols, and services posses a big challenge for service carriers because it requires a huge investment in different technology platforms, lots of training of staff, and the management of all these networks. In today’s network, service providers use SONET (Synchronous Optical NETwork) as a basic TDM (Time Division Multiplexing) transport network. SONET was primarily designed to carry voice traffic from telephone networks. However, with the explosion of traffic in the Internet, the same SONET based TDM network is optimized to support increasing demand for packet based Internet network services (data, voice, video, teleconference etc.) at access networks and LANs. Therefore the service providers need to support their Internet Protocol (IP) infrastructure as well as in the legacy telephony infrastructure. Supporting both TDM and packet services in the present condition needs multilayer operations which is complex, expensive, and difficult to manage. A hybrid switch is a novel architecture that combines packets (IP) and TDM switching in a unified access platform and provides seamless integration of access networks and LANs with MAN/WAN networks. The ability to fully integrate these two capabilities in a single chassis will allow service providers to deploy a more cost effective and flexible architecture that can support a variety of different services. This thesis develops a hybrid switch which is capable of offering bundled services for TDM switching and packet routing. This is done by dividing the switch’s bandwidth into VT1.5 (Virtual Tributary -1.5) channels and providing SONET based signaling for routing the data and controlling the switch’s resources. The switch is a TDM based architecture which allows each switch’s port to be independently configured for any mixture of packet and TDM traffic, including 100% packet and 100% TDM. This switch allows service providers to simplify their edge networks by consolidating the number of separate boxes needed to provide fast and reliable access. This switch also reduces the number of network management systems needed, and decreases the resources needed to install, provision and maintain the network because of its ability to “collapse” two network layers into one platform. The scope of this thesis includes system architecture, logic implementation, and verification testing, and performance evaluation of the hybrid switch. The architecture consists of ingress/egress ports, an arbiter and a crossbar. Data from ingress ports is carried to the egress ports via VT1.5 channels which are switched at the cross point of the crossbar. The crossbar setup and channel assignments at ingress port are done by the arbiter. The design was tested by simulation and the hardware cost was estimated. The performance results showed that the switch is non-blocking, provide differentiated service, and has an overall effective throughput of 80%. This result is a significant step towards the goal of building a switch that can support multiprotocol and provide different network capabilities into one platform. The long-term goal of this project is to develop a prototype of the hybrid switch with broadband capability

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Quality-of-service management in IP networks

Quality of Service (QoS) in Internet Protocol (IF) Networks has been the subject of active research over the past two decades. Integrated Services (IntServ) and Differentiated Services (DiffServ) QoS architectures have emerged as proposed standards for resource allocation in IF Networks. These two QoS architectures support the need for multiple traffic queuing systems to allow for resource partitioning for heterogeneous applications making use of the networks. There have been a number of specifications or proposals for the number of traffic queuing classes (Class of Service (CoS)) that will support integrated services in IF Networks, but none has provided verification in the form of analytical or empirical investigation to prove that its specification or proposal will be optimum. Despite the existence of the two standard QoS architectures and the large volume of research work that has been carried out on IF QoS, its deployment still remains elusive in the Internet. This is not unconnected with the complexities associated with some aspects of the standard QoS architectures. [Continues.

Quality of service technologies for multimedia applications in next generation networks

Next Generation Networks are constantly evolving towards solutions that allow the operator to provide advanced multimedia applications with QoS guarantees in heterogeneous, multi-domain and multi-services networks. Other than the unquestionable advantages inherent the ability to simultaneously handle traffic flows at different QoS levels, these architectures require management systems to efficiently perform quality guarantees and network resource utilization. These issues have been addressed in this thesis. DiffServ-aware Traffic Engineering (DS-TE) has been considered as reference architecture for the deployment of the quality management systems. It represents the most advanced technology to accomplish either network scalability and service granularity goals. On the basis of DS-TE features, a methodology for traffic and network resource management has been defined. It provides some rules for QoS service characterization and allows to implement Traffic Engineering policies with a class-based approach. A set of basic parameters for quality evaluation has been defined, that are the Key Performance Indicators; some mathematical model to derive the statistical nature of traffic have been analyzed and an algorithm to improve the fulfillment of quality of service targets and to optimize network resource utilization. It is aimed at reducing the complexity inherent the setting of some of the key parameters in the NGN architectures. Multidomain scenarios with technologies different from DS-TE have been also evaluated, defining some methodologies for network interoperability. Simulations with Opnet Modeler confirmed the efficacy of the proposed system in computing network configurations with QoS targets. With regard to QoS performance at the application level, video streaming applications in wireless domains have been particularly addressed. A rate control algorithm to adjust the rate on a per-window basis has been defined, making use of a short-term prediction of the network delay to keep the probability of playback buffer starvation lower than a desired threshold during each window. Finally, a framework for mutual authentication in web applications has been proposed and evaluated. It integrates an IBA password technique with a challenge-response scheme based on a shared secret key for image scrambling. The wireless environment is mainly addressed by the proposed system, which tries to overcome the severe constraints on security, data transmission capability and user friendliness imposed by such environment

Journal of Telecommunications and Information Technology, 2009, nr 3

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