Dissertations / Theses on the topic 'Telecommunication - Traffic - Management'

The successful deployment of Universal Mobile Telecommunication System (UMTS) is heavily dependent on Quality of Service (QoS) to be achieved. This thesis addresses a few facets of the QoS issues in UMTS, including traffic shaping and policing, conformance consistency, Call Admission Control (CAC) and resource allocation.

The main focus of this thesis is traffic conditioning related issues for QoS provisioning in UMTS. Assuming an end-to-end QoS scenario supported by IntServ or/and Diffserv architectures, the thesis initially presents an all nodes traffic conditioning-enabled framework in UMTS. Under this framework, the traffic generated at each User Equipment (UE) is regulated by a traffic shaper in the form of a token bucket, and the conformance of the traffic flow is policed at the policing node. The performance of imposing traffic shaping at the UE is studied and compared with the case without shaping. Next, having observed that the performance of the traffic conditioned system is sensitive to the values of the token bucket parameters, the thesis proposes a heuristic approach for searching local and global QoS-aware token bucket parameters. By tuning the system operating at the obtained ’optimal’ shaping parameters, the requirements for all concerned QoS attributes are guaranteed. Furthermore, the thesis studies conformance consistency in a traffic conditioned multi-hop network, by monitoring the conformance status of a traffic flow using an identical token bucket for both traffic shaping and policing. In the presence of variable packet size, the thesis gives a quantitative result, for a simple case, on how much percent of the originally conformant packets may misbehave at further policing node(s). The performance of the aggregated traffic flows and the measures to minimize the effect of conformance deterioration are also studied in the thesis.

Another facet of the QoS issues in UMTS, CAC together with resource allocation, is also studied in the thesis. A priority-oriented framework for QoS management of multimedia services in UMTS is proposed. Based on a traffic class priority definition, the framework is implemented through a priority-oriented CAC, channel congestion control and adaptive bandwidth allocation.

The MultiProtocol Label Switching (MPLS) framework is used in many networks to provide efficient load balancing which distributes the traffic for efficient Quality of Service (QoS) provisioning in the network. If the MPLS framework is combined with Differentiated Services (DiffServ) architecture, together they can provide aggregate-based service differentiation and QoS. The combined use of DiffServ and MPLS in a network is called DiffServ-aware Traffic Engineering (DS-TE). Such DiffServ-based MPLS networks demand development of efficient methods for QoS provisioning. In this thesis, an automated manager for management of these DiffServ-based MPLS networks is proposed. This manager, called Traffic Engineering Automated Manager (TEAM), is a centralized authority for adaptively managing a DiffServ/MPLS domain and it is responsible for dynamic bandwidth and route management. TEAM is designed to provide a novel and unique architecture capable of managing large scale MPLS/DiffServ domains without any human interference. TEAM constantly monitors the network state and reconfigures the network for efficient handling of network events. Under the umbrella of TEAM, new schemes for Label Switched Path (LSP) setup/teardown, traffic routing, and network measurement are proposed and evaluated through simulations. Also, extensions to include Generalized MPLS (GMPLS) networks and inter-domain management are proposed.

Thesis (Ph.D)--Computing, Georgia Institute of Technology, 2008.
Committee Chair: Xu, Jun; Committee Member: Ammar, Mostafa; Committee Member: Feamster, Nick; Committee Member: Ma, Xiaoli; Committee Member: Zegura, Ellen.

The focus of this dissertation is the development of resource management schemes for integrated networks, with the major contributions being: (i) the development of an optimal adaptive buffer management scheme for the packet-switched subsystem, (ii) the integration of a moveable-boundary hybrid switching scheme with the time assigned speech interpolation technique for implementing a congestion control mechanism for the packet-switched subsystem, and (iii) the development of an adaptive hierarchical scheme for implementing the access control and routing functions within the circuit-switched subsystem. The problem of buffer management at an integrated network node is formulated as a nonlinear programming problem with a convex objective function and an interative solution technique with fast convergence is proposed for a real-time implementation of the buffer management scheme in practical environments. In order to exercise an additional degree of control over the packet-blocking probability at each hybrid-switched link within the network, a new multiplexing scheme based on the integration of the moveable-boundary hybrid switching scheme and the time assigned speech interpolation technique is presented in this dissertation. The tradeoff between the corresponding decrease in the packet blocking probability and the increase in the circuit freezeout fraction is demonstrated by a detailed queueing analysis of the multiplexer. Specific algorithms are also presented in this dissertation for the solution of the access control and routing problems within the circuit-switched subsystem. In particular, an access control scheme is developed by solving an integer programming problem formulated using the policy of complete partitioning of the available bandwidth among the competing user classes. As an alternative to the completely partitioned approach, the problem of traffic routing is considered in a network that supports homogeneous traffic classes based on the policy of complete sharing. Finally, for the general case of networks with heterogeneous traffic classes, a hierarchical scheme is developed for the implementation of the access control and the routing functions at two functional levels, where the access control is implemented by the network supervisor who solves an appropriate linear integer programming problem periodically, and the routing function is handled by the individual nodes of the network on a distributed basis. (Abstract shortened with permission of author.)

The Internet is dynamically shared by numerous flows of data traffic. Network congestion occurs when the aggregate flow rate persistently exceeds the network capacity, leading to excessive delivery delay and loss of user data. To control network congestion, a flow needs to adapt the sending rate to its inferred level of congestion, and a packet switch needs to report its local level of congestion. In this framework of Internet congestion control, it is important for flows to react promptly against congestion, and robustly against interfering network events resembling congestion. This is challenging due to the highly dynamic interactions of various network components over a global scale. Prior approaches rely predominantly on empirical observations in experiments for constructing and validating designs. However, without a careful, systematic examination of all viable options, more efficient designs may be overlooked. Moreover, experimental results have limited applicability to scenarios beyond the specific experimental settings. In this thesis, I employ a novel, systematic design approach. I formalize the design process of Internet congestion control from a minimal set of empirical observations. I prove the robustness and optimality of the attained design in general settings, and validate these properties in practical experimental settings. First, I develop a systematic method for enhancing the robustness of flows against interfering events resembling congestion. The class of additive-increase-multiplicative-decrease (AIMD) algorithms in Transmission Control Protocol (TCP) is the set of dominant algorithms governing the flow rate adaptation process. Over the present Internet, packet reordering and non-congestive loss occur frequently and are misinterpreted by TCP AIMD as packet loss due to congestion. This leads to underutilization of network resources. With a complete, formal characterization of the design space of TCP AIMD, I formulate designing wireless TCP AIMD as an optimal control problem over this space. The derived optimal algorithm attains a significant performance improvement over existing enhancements in packet-level simulation. Second, I propose a novel design principle, known as pricing-link-by-time (PLT), that specifies how to set the measure of congestion, or “link price”, at a router to provide prompt feedback to flows. Existing feedback mechanisms require sophisticated parameter tuning, and experience drastic performance degradation with improperly tuned parameters. PLT makes parameter tuning a simple, optional process. It increases the link price as the backlog stays above a threshold value, and resets the price once the backlog goes below the threshold. I prove that such a system exhibits cyclic behavior that is robust against changes in network environment and protocol parameters. Moreover, changing the threshold value can control delay without undermining system performance. I validate these analytical results using packet-level simulation. The incremental deployment of various enhancements have made Internet congestion control highly heterogeneous. The final part of the thesis studies this issue by analyzing the competition among flows with heterogeneous robustness against interfering network events. While rigorous theories have been a major vehicle for understanding system designs, the thesis involves them directly in the design process. This systematic design approach can fully exploit the structural characteristics, and lead to generally applicable, effective solutions.
published_or_final_version
Electrical and Electronic Engineering
Doctoral
Doctor of Philosophy

With the advent of bandwidth-hungry video and audio applications, demand for bandwidth is expected to exceed supply. Users will require more bandwidth and, as always, there are likely to be more users. As the Internet user base becomes more diverse, there is an increasing perception that Internet Service Providers (ISPs) should be able to differentiate between users, so that the specific needs of different types of users can be met. Differentiated services is seen as a possible solution to the bandwidth problem. Currently, however, the technology used on the Internet differentiates neither between users, nor between applications. The thesis focuses on current and anticipated bandwidth shortages on the Internet, and on the lack of a differentiated service. The aim is to identify methods of managing bandwidth and to investigate how these bandwidth management methods can be used to provide a differentiated service. The scope of the study is limited to networks using both Ethernet technology and the Internet Protocol (IP). Tile study is significant because it addresses current problems confronted by network managers. The key terms, Quality of Service (QoS) and bandwidth management, are defined. “QoS” is equated to a differentiating system. Bandwidth management is defined as any method of controlling and allocating bandwidth. "Installing more capacity" is taken to be a method of bandwidth management. The review of literature concentrates on Ethernet/IP networks. It begins with a detailed examination of definitions and interpretations of the term "Quality of Service" and shows how the meaning changed over the last decade. The review then examines congestion control, including a survey of queuing methods. Priority queuing implemented in hardware is examined in detail, followed by a review of the ReSource reserVation Protocol (RSVP) and a new version of IP (lPv6). Finally, the new standards IEEE 802.1p and IEEE 802.1Q are outlined, and parts of ISO/IEC 15802-3 are analysed. The Integrated Services Architecture (ISA), Differentiated Services (DiffServ) and MultiProtocol Label Switching (MPLS) are seen as providing a theoretical framework for QoS development. The Open Systems Interconnection Reference Model (OSI model) is chosen as the preferred framework for investigating bandwidth management because it is more comprehensive than the alternative US Department of Defence Model (DoD model). A case study of the Edith Cowan University (ECU) data network illustrates current practice in network management. It provides concrete examples of some of the problems, methods and solutions identified in the literary review. Bandwidth management methods are identified and categorised based on the OSI layers in which they operate. Suggestions are given as to how some of these bandwidth management methods are, or can be used within current QoS architectures. The experimental work consists of two series of tests on small, experimental LANs. The tests are aimed at evaluating the effectiveness of IEEE 802.1 p prioritisation. The results suggest that in small Local Area Networks (LANs) prioritisation provides no benefit when Ethernet switches are lightly loaded.

Packet switched wireless networks have become increasingly popular due to improvements in transmission speed, ease of deployment and mobility. Wireless technologies such as the IEEE 802.11 Wireless Local Area Networks provide transmission speeds capable of supporting multimedia applications. However, wireless channels suffer from short term effects such as interference and fading, and long term effects such as signal strength changes that are caused by user mobility. In multi-rate networks, stations can adapt to the channel variations by adjusting their physical transmission rates. This introduces resource management problems as resource usage depends on the application's bit rate as well as the physical transmission rate used. Multimedia applications demand consistent Quality of Service (QoS) performance from the network. This does not fit well with the dynamic nature of wireless networks. In this thesis, we propose a link layer resource manager to maintain application QoS requirements in multi-rate wireless networks. It consists of two components - Resource reservation and Medium Access Control (MAC) parameters selection. The resource reservation algorithm determines the current and future amount of channel resources required by a multimedia application and performs the function of admission control. This prevents any new traffic or physical transmission rate changes from degrading the QoS of the admitted traffic. The design of the MAC parameters selection algorithm is based on the IEEE 802.11e Enhanced Distributed Channel Access (EDCA) scheme. The algorithm enables the provision of QoS to individual multimedia applications with the prioritised service of EDCA. Analytical and simulation studies were performed to demonstrate the efficiency and effectiveness of the algorithms. The results highlighted the ability of the algorithms to mitigate the QoS provision problem in multi-rate wireless networks introduced by channel variations.

Orientadores: Leonardo de Souza Mendes, Mario Lemes Proença Junior
Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação
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Resumo: Estratégias voltadas para a detecção de anomalias em redes de computadores emitem alarmes como forma de notificação ao administrador de rede. Esses alarmes são essenciais para a gerencia de rede, pois são evidencias de uma anormalidade. Entretanto, uma única anomalia pode gerar um numero excessivo de alarmes, tornando a inspeção manual inviável. Nesta dissertação, e apresentado um sistema de correlação de alarmes automatizado, divido em três camadas, que obtém os alarmes primitivos e apresenta ao administrador de rede uma visão global do cenário afetado pela anomalia. A camada de pré-processamento faz a compressão dos alarmes utilizando seus atributos espaciais e temporais, os quais são reduzidos a um único alarme denominado DLA (Alarme em Nível de Equipamento). A camada de correlação busca, através dos DLAs e de informações sobre a topologia da rede, inferir o caminho de propagação da anomalia, sua origem e destino. A camada de apresentação prove a visualização do caminho e elementos de redes afetados pela propagação da anomalia. O sistema apresentado nesta dissertação foi aplicado em diversos cenários que apresentavam anomalias reais detectadas na rede da Universidade Estadual de Londrina. Foi demonstrada sua capacidade de identificar, de forma automatizada, o caminho de propagação do trafego anômalo, proporcionando informações úteis e corretas ao administrador de rede para o diagnostico do problema
Abstract: Anomaly detection systems for computer networks send alarms in order to notify the network administrator. These alarms are essential for network management because they are evidences of an abnormality. However, a single anomaly may generate an excessive volume of alarms, making the manual inspection unfeasible. In this work, it is presented an automated alarm correlation system divided into three layers, which obtains raw alarms and presents to network administrator a global view of the scenario affected by the anomaly. In the preprocessing layer, it is performed the alarm compression using their spatial and temporal attributes, which are reduced to a unique alarm named DLA (Device Level Alarm). The correlation layer aims to infer the anomaly propagation path and its origin and destination using DLAs and network topology information. The presentation layer provides the visualization of the path and network elements affected by the anomaly propagation through the network. The presented system was applied in various scenarios that had real anomalies detected on the State University of Londrina network. It demonstrated its ability to identify in an automated manner the anomalous traffic propagation path, providing useful and accurate information to the network administrator to diagnose the problem
Mestrado
Telecomunicações e Telemática
Mestre em Engenharia Elétrica

Data distribution management (DDM) is a mechanism to interconnect data producers and data consumers in a distributed application. Data producers provide useful data to consumers in the form of messages. For each message produced, DDM determines the set of data consumers interested in receiving the message and delivers it to those consumers. We are particularly interested in DDM techniques for parallel and distributed discrete event simulations. Thus far, researchers have treated synchronization of events (i.e. time management) and DDM independent of each other. This research focuses on how to realize time managed DDM mechanisms. The main reason for time-managed DDM is to ensure that changes in the routing of messages from producers to consumers occur in a correct sequence. Also time managed DDM avoids non-determinism in the federation execution, which may result in non-repeatable executions. An optimistic approach to time managed DDM is proposed where one allows DDM events to be processed out of time stamp order, but a detection and recovery procedure is used to recover from such errors. These mechanisms are tailored to the semantics of the DDM operations to ensure an efficient realization. A correctness proof is presented to verify the algorithm correctly synchronizes DDM events. We have developed a fully distributed implementation of the algorithm within the framework of the Georgia Tech Federated Simulation Development Kit (FDK) software. A performance evaluation of the synchronized DDM mechanism has been completed in a loosely coupled distributed system consisting of a network of workstations connected over a local area network (LAN). We compare time-managed versus unsynchronized DDM for two applications that exercise different mobility patterns: one based on a military simulation and a second utilizing a synthetic workload. The experiments and analysis illustrate that synchronized DDM performance depends on several factors: the simulations model (e.g. lookahead), applications mobility patterns and the network hardware (e.g. size of network buffers). Under certain mobility patterns, time-managed DDM is as efficient as unsynchronized DDM. There are also mobility patterns where time-managed DDM overheads become significant, and we show how they can be reduced.

Active Queue Management (AQM) schemes are used for ensuring the Quality of Service (QoS) in telecommunication networks. However, they are sensitive to parameter settings and have weaknesses in detecting and controlling congestion under dynamically changing network situations. Another drawback for the AQM algorithms is that they have been applied only on the Markovian models which are considered as Short Range Dependent (SRD) traffic models. However, traffic measurements from communication networks have shown that network traffic can exhibit self-similar as well as Long Range Dependent (LRD) properties. Therefore, it is important to design new algorithms not only to control congestion but also to have the ability to predict the onset of congestion within a network. An aim of this research is to devise some new congestion control methods for communication networks that make use of various traffic characteristics, such as LRD, which has not previously been employed in congestion control methods currently used in the Internet. A queueing model with a number of ON/OFF sources has been used and this incorporates a novel congestion prediction algorithm for AQM. The simulation results have shown that applying the algorithm can provide better performance than an equivalent system without the prediction. Modifying the algorithm by the inclusion of a sliding window mechanism has been shown to further improve the performance in terms of controlling the total number of packets within the system and improving the throughput. Also considered is the important problem of maintaining QoS constraints, such as mean delay, which is crucially important in providing satisfactory transmission of real-time services over multi-service networks like the Internet and which were not originally designed for this purpose. An algorithm has been developed to provide a control strategy that operates on a buffer which incorporates a moveable threshold. The algorithm has been developed to control the mean delay by dynamically adjusting the threshold, which, in turn, controls the effective arrival rate by randomly dropping packets. This work has been carried out using a mixture of computer simulation and analytical modelling. The performance of the new methods that have
Ministry of Higher Education in Egypt and the Egyptian Cultural Centre and Educational Bureau in London

Active Queue Management (AQM) schemes are used for ensuring the Quality of Service (QoS) in telecommunication networks. However, they are sensitive to parameter settings and have weaknesses in detecting and controlling congestion under dynamically changing network situations. Another drawback for the AQM algorithms is that they have been applied only on the Markovian models which are considered as Short Range Dependent (SRD) traffic models. However, traffic measurements from communication networks have shown that network traffic can exhibit self-similar as well as Long Range Dependent (LRD) properties. Therefore, it is important to design new algorithms not only to control congestion but also to have the ability to predict the onset of congestion within a network. An aim of this research is to devise some new congestion control methods for communication networks that make use of various traffic characteristics, such as LRD, which has not previously been employed in congestion control methods currently used in the Internet. A queueing model with a number of ON/OFF sources has been used and this incorporates a novel congestion prediction algorithm for AQM. The simulation results have shown that applying the algorithm can provide better performance than an equivalent system without the prediction. Modifying the algorithm by the inclusion of a sliding window mechanism has been shown to further improve the performance in terms of controlling the total number of packets within the system and improving the throughput. Also considered is the important problem of maintaining QoS constraints, such as mean delay, which is crucially important in providing satisfactory transmission of real-time services over multi-service networks like the Internet and which were not originally designed for this purpose. An algorithm has been developed to provide a control strategy that operates on a buffer which incorporates a moveable threshold. The algorithm has been developed to control the mean delay by dynamically adjusting the threshold, which, in turn, controls the effective arrival rate by randomly dropping packets. This work has been carried out using a mixture of computer simulation and analytical modelling. The performance of the new methods that have.

Orientadores: Leonardo de Souza Mendes, Mario Lemes Proença Junior
Tese (doutorado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação
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Resumo: Anomalias em redes de computadores são desvios súbitos e acentuados que ocorrem no tráfego em consequência de diversas situações como defeitos em softwares, uso abusivo de recursos da rede, falhas em equipamentos, erros em configurações e ataques. Nesta tese, é proposto um sistema de detecção de anomalias em redes de computadores baseado em três níveis de análise. O primeiro nível de análise é responsável por comparar os dados coletados em um objeto SNMP (Simple Network Management Protocol) com o perfil de operações normais da rede. O segundo nível de análise correlaciona os alarmes gerados no primeiro nível de análise utilizando um grafo de dependências que representa as relações entre os objetos SNMP monitorados. O terceiro nível de análise reúne os alarmes de segundo nível utilizando informações sobre a topologia de rede e gera um alarme de terceiro nível que reporta a propagação da anomalia pela rede. Os testes foram realizados na rede da Universidade Estadual de Londrina, utilizando situações reais. Os resultados mostraram que a proposta apresentou baixas taxas de falsos positivos combinadas a altas taxas de detecção. Além disso, o sistema foi capaz de correlacionar alarmes gerados para diferentes objetos SNMP em toda a rede, produzindo conjuntos menores de alarmes que ofereceram ao administrador de redes uma visão panorâmica do problema
Abstract: Anomalies in computer networks are unexpected and significant deviations that occur in network traffic due to different situations such as software bugs, unfair resource usage, failures, misconfiguration and attacks. In this work, it is proposed an anomaly detection system based on three levels of analysis. The first level of analysis is responsible for comparing the data collected from SNMP (Simple Network Management Protocol) objects with the profile of network normal behavior. The second level of analysis correlates the alarms generated by the first level of analysis by using a dependency graph, which represents the relationships between the SNMP objects. The third level of analysis correlates the second level alarms by using network topology information. The third level generates a third level alarm that presents the anomaly propagation path through the network. Tests were performed in the State University of Londrina network, exploring real situations. Results showed that the proposal presents low false positive rates and high detection rates. Moreover, the proposed system is able to correlate alarms that were generated for SNMP objects at different places of the network, producing smaller sets of alarms that offer a wide-view of the problem to the network administrator
Doutorado
Telecomunicações e Telemática
Doutor em Engenharia Elétrica

Thesis (MSc (Mathematical Sciences. Computer Science))--University of Stellenbosch, 2008.
Consider a path-oriented telecommunications network where calls arrive to each route in a Poisson process. Each call brings on average a fixed number of packets that are offered to route. The packet inter-arrival times and the packet lengths are exponentially distributed. Each route can queue a finite number of packets while one packet is being transmitted. Each accepted packet/call generates an amount of revenue for the route manager. At specified time instants a route manager can acquire additional capacity (“interface capacity”) in order to carry more calls and/or the manager can acquire additional buffer space in order to carry more packets, in which cases the manager earns more revenue; alternatively a route manager can earn additional revenue by selling surplus interface capacity and/or by selling surplus buffer space to other route managers that (possibly temporarily) value it more highly. We present a method for efficiently computing the buying and the selling prices of buffer space. Moreover, we propose a bandwidth reallocation scheme capable of improving the network overall rate of earning revenue at both the call level and the packet level. Our reallocation scheme combines the Erlang price [4] and our proposed buffer space price (M/M/1/K prices) to reallocate interface capacity and buffer space among routes. The proposed scheme uses local rules and decides whether or not to adjust the interface capacity and/or the buffer space. Simulation results show that the reallocation scheme achieves good performance when applied to a fictitious network of 30-nodes and 46-links based on the geography of Europe.

An energy-aware on-demand Bluetooth scatternet formation and routing protocol taking into account network architecture and traffic pattern is proposed. The scatternet formation protocol is able to cope with multiple sources initiating traffic simultaneously as well as prolong network lifetime. A modified Inquiry scheme using extended ID packet is introduced for fast device discovery and power efficient propagation of route request messages with low delay. A mechanism employing POLL packets in Page processes is proposed to transfer scatternet formation and route reply information without extra overhead. In addition, the energy aware forwarding nodes selection scheme is based on local information and results in more uniform network resource utilization and improved network lifetime. Simulation results show that this protocol can provide scatternet formation with reasonable delay and with good load balance which results in prolonged network lifetime for Bluetooth-based wireless sensor networks. In this research, a metric-based scatternet formation algorithm for the Bluetooth-based sensor motes is presented. It optimizes the Bluetooth network formation from the hop distance and link quality perspectives. In addition, a smart repair mechanism is proposed to deal with link/node failure and recover the network connectivity promptly with low overhead. The experiments with the Intel Mote platform demonstrate the effectiveness of the optimizations. This research also investigates the scalability of ad hoc routing protocols in very large-scale wireless ad hoc networks. A comprehensive simulation study is conducted of the performance of an on-demand routing protocol on a very large-scale, with as many as 50,000 nodes in the network. The scalability analysis is addressed based on various network sizes, node density, traffic load, and mobility. The reasons for packet loss are analyzed and categorized at each network layer. Based on the observations, we observe the effect of the parameter selection and try to exhaust the scalability boundary of the on-demand routing protocol for wireless ad hoc networks.

Les systèmes de 3G offrent de nouveaux services support (bearer services) à plus hauts débits pour les modes de transmission "paquet". Ces services vont coexister avec la voix (ou d'autres services temps réels), des scénarios de trafic mixte, voix et données, doivent être envisagés. La norme UMTS permet effectivement aux utilisateurs d'avoir plus d'un service activé simultanément. Les différentes classes de trafic augmentent la complexité de la gestion des ressources radios. Dans ce contexte, deux types de fonctions sont étudiés: l'allocation de TFCI et l'ordonnancement de paquets. Leur impact sur la qualité de service (QoS) ainsi que sur la capacité du système est évalué. On propose des améliorations de ces mécanismes dans le but d'augmenter la capacité du système et par conséquent d'améliorer la QoS des utilisateurs. Les études se restreignent au sens montant, c'est à dire aux transmissions du mobile (User Equipment ou UE) vers le réseau. Un premier mécanisme pour lequel un effort d'amélioration est fait, est l'adaptation du lien radio par variation du débit instantané transmis. On simule le cas d'une transmission multiservice (voix et données). L'UE doit partager un débit global qui lui est alloué entre les différents services activés. Ces derniers sont véhiculés dans des radio bearers (tuyaux supports). À chaque intervalle élémentaire de transmission (Transmission Time Interval, TTI), l'UE sélectionne un sous-débit pour chaque bearer; ceci se fait par la sélection d'un "format de transport" à appliquer pendant la durée TTI. Cette procédure s'effectue dans la couche MAC (Medium Access Control); le résultat de la sélection est une combinaison de formats de transport (Transport Format Combination, TFC) que la couche physique doit utiliser. La procédure, nommée sélection de TFC, permet d'adapter la transmission des différents services aux conditions variables de la propagation radio: elle détermine notablement la performance de transmission. L'algorithme de sélection de TFC est tracé dans ses grandes lignes dans la norme. Un de ses principes est de favoriser le trafic temps réel au détriment des services de données par paquet. Cependant, le trafic temps réel peut être perturbé par le trafic de données sous certaines conditions, en particulier pour les mobiles éloignés de la station de base (Node B). On propose un algorithme de sélection de TFC qui limite ces perturbations et qui offre une plus large zone de couverture aux services temps réels. En plus, il améliore la QoS du service de données et le débit effectif de l'UE sans augmenter sa puissance de transmission. Un autre type d'études concerne l'ordonnancement de paquets entre les différents utilisateurs ou UEs. C'est une procédure qui est contrôlée par la partie fixe du réseau. Nous l'étudions principalement par simulation en considérant divers mécanismes ou variations. Un premier mécanisme est nommé fast Variable Spreading Factor (fast VSF): les UEs distants changent rapidement leur facteur d'étalement (SF) afin de conserver une puissance de transmission constante, ce qui vise à stabiliser l'interférence inter-cellulaire. Un deuxième mécanisme étudié est un accès paquet décentralisé (decentralized mode) utilisant une information en retour sur le niveau global d'interférence dans la cellule. Un troisième mécanisme nommé "fast scheduling" (ordonnancement rapide) raccourcit le cycle d'ordonnancement. Les résultats ont montré que dans le cas de faible ou moyenne charge dans la cellule, le mode décentralisé réduit le délai par paquet jusqu'à 25 %. L'ordonnancement rapide augmente la capacité du système jusqu' à 10%. En plus, il améliore la QoS perçue par les utilisateurs en terme de débit par utilisateur et délai par paquet transmis.

Our study on interdomain traffic engineering focuses on AS Path Prepending (ASPP), a popular way for inbound traffic engineering. In order to improve the current situation that ISPs often practise this approach in a trial-and-error basis, we propose a greedy algorithm to help ISPs perform this approach systematically and efficiently. Then we demonstrate two fundamental issues of decentralized selfish traffic engineering, routing instability and global network performance degradation, based on an abstract model where ISPs perform traffic engineering for their individual load balance. We also present a real-world pathologic case of prepending instability from our measurement study. Some simple guidelines are given for ISPs to avoid such routing instability.
Our work on peering strategy is to help ISPs understand the economic implications of various traffic patterns and make proper decisions to optimize their business. We first conduct an economic analysis for an overlay streaming network to gain some insights on the free ride phenomenon. We further improve the analysis by taking the response of subscribers into consideration and formulate the dynamic market as a multi-leader-follower game to capture the Nash Equilibrium of the routing tussle among the major players of the Internet marketplace. Based on this framework together with a gravity traffic model, we present some important observations on the implications of overlays on ISPs' peering strategy.
Over the past several years, numerous types of "overlay" networks change the interdomain traffic pattern and ISPs lose the routing control of some interdomain traffic flows due to the application layer routing. As a result, some ISPs may provide unintended transit service for other local ISPs. It upsets the traditional business model and makes ISPs' peering strategies more complicated.
The Internet has quickly evolved into a vast global network owned and operated by thousands of interconnected Internet Service Providers. Each of these ISPs, as one autonomous system, has its individual economic interests. ISPs can achieve their objectives through peering strategy and interdomain traffic engineering. These two issues are important for ISPs' business and have significant implications on the Internet architecture.
Wang, Hui.
"September 2007."
Adviser: Dah Ming Chiu.
Source: Dissertation Abstracts International, Volume: 69-08, Section: B, page: 4865.
Thesis (Ph.D.)--Chinese University of Hong Kong, 2007.
Includes bibliographical references (p. 163-170).
Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Electronic reproduction. [Ann Arbor, MI] : ProQuest Information and Learning, [200-] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Abstracts in English and Chinese.
School code: 1307.

by Kit-man Chan.
Thesis (M.Phil.)--Chinese University of Hong Kong, 1994.
Includes bibliographical references (leaves 82-86).
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Overview of Integrated Service Digital Networks --- p.1
Chapter 1.2 --- Multirate Loss Networks --- p.5
Chapter 1.3 --- Previous Work --- p.7
Chapter 1.4 --- Organization --- p.11
Chapter 1.5 --- Publications --- p.12
Chapter 2 --- Call Admission in Multirate Loss Networks --- p.13
Chapter 2.1 --- Introduction --- p.13
Chapter 2.2 --- Two Adaptive Routing Rules --- p.15
Chapter 2.3 --- Call Admission Policies --- p.17
Chapter 2.4 --- Analysis of Call Admission Policies --- p.25
Chapter 2.4.1 --- "The CS, LO, GB and the EB Policies" --- p.25
Chapter 2.4.2 --- The DP Policy --- p.29
Chapter 2.5 --- Performance Comparisons --- p.32
Chapter 2.6 --- Concluding Remarks --- p.35
Chapter 3 --- Least Congestion Routing in Multirate Loss Networks --- p.41
Chapter 3.1 --- Introduction --- p.41
Chapter 3.2 --- The M2 and MTB Routings --- p.42
Chapter 3.2.1 --- M2 Routing --- p.43
Chapter 3.2.2 --- MTB Routing --- p.43
Chapter 3.3 --- Bandwidth Sharing Policies and State Aggregation --- p.45
Chapter 3.4 --- Analysis of M2 Routing --- p.47
Chapter 3.5 --- Analysis of MTB Routing --- p.50
Chapter 3.6 --- Numerical Results and Discussions --- p.53
Chapter 3.7 --- Concluding Remarks --- p.56
Chapter 4 --- The Least Congestion Routing in WDM Lightwave Networks --- p.60
Chapter 4.1 --- Introduction --- p.60
Chapter 4.2 --- Architecture and Some Design Issues --- p.62
Chapter 4.3 --- The Routing Rule --- p.66
Chapter 4.4 --- Analysis of the LC Routing Rule --- p.67
Chapter 4.4.1 --- Fixed Point Model --- p.67
Chapter 4.4.2 --- Without Direct-link Priority --- p.68
Chapter 4.4.3 --- With Direct-link Priority --- p.72
Chapter 4.5 --- Performance Comparisons --- p.73
Chapter 4.6 --- Concluding Remarks --- p.75
Chapter 5 --- Conclusions and Future Work --- p.79
Chapter 5.1 --- Future Work --- p.80

The second problem that is addressed in the thesis is about traffic analysis and classification. Accurate identification of network applications is important to many network activities. Traditional port-based technique has become much less effective since many new applications no longer use well-known fixed port numbers. In this thesis, we propose a novel profile-based approach to identify traffic flows belonging to the target application. In contrast to classifying traffic based on statistics of individual flows in previous studies, we build behavioral profiles of the target application, which describe dominant communication patterns of the application. Based on the behavior profiles, a two-level matching is used in identifying new traffic. We demonstrate the effectiveness of our method on campus traffic traces. Our results show that one can identify the popular P2P applications with very high accuracy.
This thesis represents new intelligent methods for monitoring and classifying network traffic. Internet traffic flow measurement is vitally important for network management, accounting and performance studies. Cisco's NetFlow is a widely deployed flow measurement solution that uses a configurable static sampling rate to control processor and memory usage on the router and the amount of reporting flow records generated. But during flooding attacks the memory and network bandwidth consumed by flow records can increase beyond what is available. Currently available countermeasures have their own problems In this thesis, we propose an entropy based adaptive flow aggregation algorithm. Relying on information-theoretic techniques, the algorithm efficiently identifies the clusters of attack flows in real time and aggregates those large number of short attack flows into a few metaflows. Compared to currently available solutions, our solution not only alleviates the problem in memory and export bandwidth, but also significantly improves the accuracy of legitimate flows. We evaluate our system using both synthetic trace file and real trace files from the Internet.
Hu, Yan.
Adviser: Dah-Mino Chen.
Source: Dissertation Abstracts International, Volume: 70-06, Section: B, page: 3600.
Thesis (Ph.D.)--Chinese University of Hong Kong, 2008.
Includes bibliographical references (leaves 128-135).
Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Electronic reproduction. [Ann Arbor, MI] : ProQuest Information and Learning, [200-] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Abstracts in English and Chinese.
School code: 1307.

Lau Tsz-ming.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2000.
Includes bibliographical references (leaves 57-[61]).
Abstracts in English and Chinese.
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Integrated Services Architecture --- p.2
Chapter 1.2 --- Cross-path Switch --- p.4
Chapter 1.2.1 --- Path Switching --- p.5
Chapter 1.3 --- Organization of Thesis --- p.9
Chapter 2 --- Module Architecture --- p.10
Chapter 2.1 --- Introduction --- p.10
Chapter 2.2 --- Notable Features --- p.11
Chapter 3 --- Connection Admission Control and Resource Allocation --- p.14
Chapter 3.1 --- Introduction --- p.14
Chapter 3.2 --- Connection Admission Control --- p.15
Chapter 3.2.1 --- Guaranteed Service --- p.15
Chapter 3.2.2 --- Controlled-Load Service --- p.18
Chapter 3.3 --- Resource Allocation --- p.27
Chapter 4 --- Resource Management --- p.31
Chapter 4.1 --- Introduction --- p.31
Chapter 4.2 --- Scheduling Algorithm --- p.32
Chapter 4.2.1 --- Input and Output Module --- p.32
Chapter 4.2.2 --- Central Module --- p.34
Chapter 4.3 --- Buffer Management --- p.39
Chapter 4.3.1 --- Buffer Partitioning --- p.40
Chapter 4.3.2 --- Dicard Policy --- p.40
Chapter 5 --- Design Issue of Cross-path Switch --- p.43
Chapter 5.1 --- Introduction --- p.43
Chapter 5.2 --- Stability Condition --- p.44
Chapter 5.3 --- Supplementary Admission Control Scheme --- p.46
Chapter 5.4 --- Simulation --- p.50
Chapter 6 --- Conclusion --- p.55
Bibliography --- p.57

Leung Ka Hing.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.
Includes bibliographical references (leaves 94-96).
Abstracts in English and Chinese.
Chapter 1 --- Introduction --- p.1
Chapter 2 --- Characterization and Performance Analysis --- p.8
Chapter 2.1 --- Two-class Proportional Differentiated Service --- p.11
Chapter 2.2 --- N-class Proportional DS --- p.13
Chapter 3 --- Experiments of Proportional Delay Differentiation using the It- erative Algorithm --- p.20
Chapter 3.1 --- Experiments using Poisson arrivals --- p.21
Chapter 3.2 --- Experiments using other arrivals distributions --- p.33
Chapter 4 --- Dynamic Adjustment --- p.37
Chapter 4.1 --- Adjustment algorithms --- p.37
Chapter 5 --- Experiments of Proportional Delay Differentiation using Dy- namic Adjustment --- p.41
Chapter 5.1 --- Illustration of dynamic adjustment --- p.43
Chapter 5.2 --- Poisson --- p.45
Chapter 5.3 --- Pareto --- p.48
Chapter 5.4 --- MMPP --- p.54
Chapter 5.5 --- Heterogeneous traffic classes --- p.60
Chapter 5.6 --- Experiments for short time-scale analysis --- p.62
Chapter 6 --- Multiple nodes --- p.69
Chapter 7 --- Summary of the Experiments Results --- p.75
Chapter 8 --- Improvement of WTP --- p.78
Chapter 8.1 --- Algorithm --- p.78
Chapter 8.2 --- Experiments --- p.80
Chapter 9 --- Possible Extensions --- p.85
Chapter 9.1 --- Application Extension --- p.85
Chapter 9.2 --- Performance Quantification --- p.87
Chapter 10 --- Conclusion --- p.90

by Wing Ho Andy Yuen.
Thesis (M.Phil.)--Chinese University of Hong Kong, 1997.
Includes bibliographical references (leaves 85-[90]).
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Design Issues in Wireless Cellular Systems --- p.1
Chapter 1.1.1 --- Channel Assignment --- p.2
Chapter 1.1.2 --- Handoff --- p.3
Chapter 1.1.3 --- Multiple Accesses --- p.3
Chapter 1.1.4 --- Mobility Management --- p.4
Chapter 1.2 --- Motivation of the thesis --- p.5
Chapter 1.3 --- Outline of the thesis --- p.6
Chapter 2 --- Background Studies --- p.9
Chapter 2.1 --- Current Standards --- p.9
Chapter 2.2 --- Mobility Models --- p.11
Chapter 2.2.1 --- Fluid Flow Model --- p.12
Chapter 2.2.2 --- Markovian model --- p.13
Chapter 2.3 --- Dynamic versus Static Location Strategy --- p.14
Chapter 2.4 --- Location Registration Strategies --- p.14
Chapter 2.4.1 --- Time Based Strategy --- p.15
Chapter 2.4.2 --- Geographic Based Strategy --- p.17
Chapter 2.4.3 --- Distance Based Strategy --- p.18
Chapter 2.4.4 --- Miscellaneous Strategies --- p.19
Chapter 2.5 --- Summary --- p.20
Chapter 3 --- Hybrid Bloom Filter Location Update Algorithm --- p.21
Chapter 3.1 --- Introduction --- p.21
Chapter 3.2 --- System Model --- p.22
Chapter 3.3 --- Hybrid Bloom Filter Algorithm --- p.23
Chapter 3.3.1 --- Location Update Protocol --- p.23
Chapter 3.3.2 --- Paging Protocol --- p.25
Chapter 3.4 --- Performance Evaluation --- p.27
Chapter 3.4.1 --- Comparison of the hybrid and time based Bloom filter algorithms --- p.32
Chapter 3.5 --- Numerical Studies --- p.33
Chapter 3.5.1 --- Cost versus mobility --- p.34
Chapter 3.5.2 --- Cost versus call arrival rate --- p.37
Chapter 3.6 --- Summary --- p.39
Chapter 4 --- A Dynamic Location Area Assignment Algorithm --- p.40
Chapter 4.1 --- Geographic versus Distance Based Strategies --- p.41
Chapter 4.2 --- System Model --- p.42
Chapter 4.2.1 --- cell layout --- p.42
Chapter 4.2.2 --- mobility model --- p.43
Chapter 4.2.3 --- cost function --- p.44
Chapter 4.3 --- Dynamic Location Area Assignment Algorithm --- p.45
Chapter 4.3.1 --- Measurement --- p.46
Chapter 4.3.2 --- "Computation of (iopt, jopt)" --- p.46
Chapter 4.3.3 --- Computation of location area size k --- p.51
Chapter 4.4 --- Numerical Studies --- p.52
Chapter 4.5 --- Summary --- p.57
Chapter 5 --- Paging Cost Reduction Using Bloom Filtering As Auxilliary Strategy --- p.64
Chapter 5.1 --- Introduction --- p.64
Chapter 5.2 --- A Case Study - Joint DLA-HBF Algorithm --- p.65
Chapter 5.2.1 --- The Algorithm --- p.65
Chapter 5.2.2 --- Performance Evaluation --- p.67
Chapter 5.3 --- Summary --- p.68
Chapter 6 --- Conclusion --- p.70
Chapter 6.1 --- Summary of Results --- p.70
Chapter 6.2 --- Future Research Directions --- p.71
Appendix --- p.73
Chapter A --- Optimality of the Hybrid Bloom Filter Algorithm --- p.73
Chapter B --- Derivation of the Expected First Passage Time ExTi and EyTj --- p.77
Chapter C --- Optimality of the Dynamic Location Area Algorithm --- p.81
Bibliography --- p.85

by Guo, Xiao-Lei.
Thesis (Ph.D.)--Chinese University of Hong Kong, 1996.
Includes bibliographical references (p. 138-[147]).
Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Mode of access: World Wide Web.

Ng Ping-chung.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2005.
Includes bibliographical references (leaves 71-72).
Abstracts in English and Chinese.
Chapter Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Overview and Motivation --- p.1
Chapter 1.2 --- Background of Offered Load Control --- p.2
Chapter 1.3 --- Background of Stability Control --- p.3
Chapter 1.4 --- Organization of the Thesis --- p.4
Chapter Chapter 2 --- Performance Problems and Solutions --- p.6
Chapter 2.1 --- Simulation Set-up --- p.6
Chapter 2.2 --- High Packet-Drop Rate --- p.7
Chapter 2.3 --- Re-routing Instability --- p.8
Chapter 2.3.1 --- Hidden-Node Problem --- p.8
Chapter 2.3.2 --- Ineffectiveness of Solving Hidden-Node Problem with RTS/CTS …… --- p.9
Chapter 2.4 --- Solutions to High-Packet Loss Rate and Re-routing Instability --- p.10
Chapter 2.4.1 --- Link-Failure Re-routing --- p.11
Chapter 2.4.2 --- Controlling Offered Load --- p.13
Chapter 2.5 --- Verification of Simulation Results with Real-life Experimental Measurements --- p.14
Chapter Chapter 3 --- Offered Load Control --- p.16
Chapter 3.1 --- Capacity Limited by the Hidden-node and Exposed-node Problems --- p.16
Chapter 3.1.1 --- Signal Capture --- p.18
Chapter 3.1.2 --- Analysis of Vulnerable Period induced by Hidden Nodes --- p.20
Chapter 3.1.3 --- Analysis of Vulnerable Period induced by Exposed Nodes --- p.21
Chapter 3.1.4 --- Sustainable Throughput --- p.22
Chapter 3.2 --- Capacity Limited by Carrier Sensing Property --- p.23
Chapter 3.3 --- Numerical Results --- p.26
Chapter 3.4 --- General Throughput Analysis of a Single Multi-hop Traffic Flow --- p.29
Chapter 3.5 --- Throughput Analysis on Topologies with Variable Distances between Successive Nodes --- p.31
Chapter Chapter 4 --- Discussions of Other Special Cases --- p.33
Chapter 4.1 --- A Carrier-sensing Limited Example --- p.33
Chapter 4.2 --- A Practical Solution to Improve Throughput --- p.34
Chapter Chapter 5 --- Achieving Fairness in Other Network Topologies --- p.36
Chapter 5.1 --- Lattice Topology --- p.36
Chapter Chapter 6 --- Stability Control --- p.39
Chapter 6.1 --- Ad-hoc routing protocols --- p.39
Chapter 6.2 --- Proposed scheme --- p.40
Chapter 6.2.1 --- Original AODV --- p.41
Chapter 6.2.2 --- AODV with Proposed Scheme --- p.42
Chapter 6.2.2.1 --- A Single Flow in a Single Chain of Nodes --- p.43
Chapter 6.2.2.2 --- Real-break Case --- p.44
Chapter 6.3 --- Improvements --- p.45
Chapter Chapter 7 --- Impacts of Data Transmission Rate and Payload Size --- p.48
Chapter 7.1 --- Signal Capture --- p.48
Chapter 7.2 --- Vulnerable region --- p.50
Chapter Chapter 8 --- Performance Enhancements in Multiple Flows --- p.53
Chapter 8.1 --- Impacts of Re-routing Instability in Two Flow Topology --- p.53
Chapter 8.2 --- Impacts of Vulnerable Periods in Multiple Flow Topologies --- p.55
Chapter 8.2.1 --- The Vulnerable Period induced by Individual Hidden-terminal Flow --- p.57
Chapter 8.2.2 --- The Number of Hidden-terminal Flows --- p.58
Chapter 8.2.3 --- Correlation between Hidden-terminal Flows --- p.60
Chapter Chapter 9 --- Conclusion --- p.63
Chapter Appendix A: --- General Throughput Analysis of a Single Multi-hop Traffic Flow --- p.67
Chapter A.l --- Capacity Limited by Hidden-node and Exposed-Node --- p.67
Chapter A.1.1 --- Sustainable Throughput --- p.68
Chapter A.2 --- Capacity Limited by Carrier Sensing Property --- p.68
Bibliography --- p.71

Chen, Liang.
Thesis (Ph.D.)--Chinese University of Hong Kong, 2013.
Includes bibliographical references (leaves 164-175).
Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Abstracts also in Chinese.

Lau Kwok Hung.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2006.
Includes bibliographical references (leaves 66-67).
Abstracts in English and Chinese.
Acknowledgement --- p.1
Abstract --- p.2
摘要 --- p.3
Chapter Chapter 1 --- Introduction --- p.6
Chapter Chapter 2 --- Background and Related Work --- p.9
Chapter 2.1 --- Application-Layer Bandwidth Differentiation --- p.9
Chapter 2.2 --- Related Work --- p.14
Chapter 2.2.1 --- Bandwidth Differentation --- p.14
Chapter 2.2.2 --- Shared Congestion Management --- p.15
Chapter 2.2.3 --- Flow Partition --- p.16
Chapter Chapter 3 --- VPS Protocol Architecture --- p.17
Chapter 3.1 --- Virtual and Actual Flows --- p.18
Chapter 3.2 --- VPS Controller --- p.21
Chapter Chapter 4 --- ACK Translation --- p.25
Chapter 4.1 --- Fast Retransmit and Fast Recovery --- p.27
Chapter 4.2 --- Timeout --- p.30
Chapter 4.3 --- Packet and ACK Reordering --- p.33
Chapter 4.4 --- False Duplicate ACK Suppression --- p.35
Chapter 4.5 --- Maxburst --- p.37
Chapter 4.6 --- Memory Overhead and Computation Complexity --- p.38
Chapter Chapter 5 --- Bandwidth Differentiation --- p.41
Chapter 5.1 --- Distribution of Virtual Packets --- p.41
Chapter 5.2 --- Temporary Suspension of Actual Flows --- p.43
Chapter 5.3 --- Receive Window Limit --- p.44
Chapter 5.4 --- Limited Data Transmission --- p.44
Chapter Chapter 6 --- Performance Evaluatoin --- p.45
Chapter 6.1 --- Performance Metric --- p.45
Chapter 6.2 --- Simulation Setup --- p.46
Chapter 6.3 --- Performance over Different Time Scales --- p.47
Chapter 6.4 --- Performance over Different Bottleneck Bandwidth --- p.53
Chapter 6.5 --- Performance over Different Application-specified Ratios --- p.54
Chapter 6.6 --- Performance over Different Number of Flows --- p.57
Chapter 6.7 --- Heterogeneous Receivers --- p.60
Chapter Chapter 7 --- Conclusions and Future Work --- p.65
Bibliography --- p.66

Yang Yang.
"July 2002."
Thesis (Ph.D.)--Chinese University of Hong Kong, 2002.
Includes bibliographical references (2nd gr., p. 1-9).
Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Mode of access: World Wide Web.
Abstracts in English and Chinese.

D. Tech. Electrical Engineering.
Objectives of the study is to propose a suitable feature extraction and classication approach that is capable of adapting to the non-linear nature and the noise contained in the time series data. The end goal of subscriber classication in this study is to utilise the subscriber information extracted for a new radio resource optimisation model that focuses on the Channel Allocation Problem CAP. Although there they have been various models proposed in literature for solving of the CAP problem, the utilisation of subscriber related information in the CAP has not been directly considered.

Ho Chi-Fong.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2000.
Includes bibliographical references (leaves 63-[64]).
Abstracts in English and Chinese.
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Introduction --- p.1
Chapter 1.2 --- The 2.5G Systems --- p.3
Chapter 1.2.1 --- HSCSD --- p.3
Chapter 1.2.2 --- GPRS --- p.3
Chapter 1.2.3 --- EDGE --- p.4
Chapter 1.2.4 --- IS-136 --- p.4
Chapter 1.3 --- The Evolution from 2G/2.5G to 3G --- p.4
Chapter 1.3.1 --- GSM Data Evolution --- p.4
Chapter 1.3.2 --- TDMA Data Evolution --- p.5
Chapter 1.3.3 --- CDMA Data Evolution --- p.6
Chapter 1.4 --- UTRA --- p.7
Chapter 1.4.1 --- UTRA FDD --- p.8
Chapter 1.4.2 --- UTRA TDD --- p.18
Chapter 1.4.3 --- Transport Channels --- p.25
Chapter 2 --- Spreading Factor Optimization for FDD Downlink --- p.27
Chapter 2.1 --- The Optimal Channel Splitting Problem --- p.28
Chapter 2.2 --- Spreading Factor Optimization for FDD Downlink Dedicated Chan- nel --- p.30
Chapter 3 --- Random Access Channel Stability Control --- p.33
Chapter 3.1 --- Random Access Slotted Aloha --- p.33
Chapter 3.1.1 --- System model --- p.33
Chapter 3.1.2 --- Probability of Code-Collision --- p.34
Chapter 3.1.3 --- Throughput Analysis of Random Access in TD/CDMA System --- p.37
Chapter 3.1.4 --- Retransmission --- p.42
Chapter 3.1.5 --- System Delay --- p.42
Chapter 3.2 --- Random Access Channel Stability Control --- p.43
Chapter 3.2.1 --- System Model --- p.43
Chapter 3.2.2 --- Random Access Procedure --- p.44
Chapter 3.3 --- Random Access Channel Stability Control Alogrithm --- p.47
Chapter 3.3.1 --- Simulation --- p.49
Chapter 3.4 --- Multi-class Model --- p.55
Chapter 4 --- Conclusions and Topics for Future Study --- p.60
Chapter 4.1 --- Thesis Conclusions --- p.60
Chapter 4.2 --- Future Work --- p.61
Chapter 4.2.1 --- Downlink and Uplink resource allocation in TDD --- p.61
Chapter 4.2.2 --- Resource Unit Packing in TDD --- p.62
Chapter 4.2.3 --- Other Topics --- p.62
Bibliography --- p.63

Η ασύρματη επικοινωνία αποκτά ιδιαίτερη αξία σε μια χώρα όπως η Ελλάδα, που η μορφολογία του εδάφους της δεν επιτρέπει πολλές φορές τη χρήση εναλλακτικών μέσων μετάδοσης όπως για παράδειγμα οι οπτικές ίνες. Ειδικότερα ο τομέας της κινητής τηλεφωνίας είναι ένας ταχύτατα εξελισσόμενος τομέας ο οποίος στις μέρες μας βρίσκεται σε ένα στάδιο μετεξέλιξής του καθώς το πέρασμα από τη δεύτερη στην τρίτη γενιά είναι πλέον γεγονός. Στη μεγάλη εξέλιξη του τομέα αυτού συμβάλουν τα μέγιστα και οι απαιτήσεις των σύγχρονων καιρών για ένα ενοποιημένο και λειτουργικό σύστημα κινητής τηλεφωνίας παρέχοντας πληθώρα υπηρεσιών στους πελάτες – χρήστες του. Είναι γεγονός ότι, τα τελευταία χρόνια, η χρήση των κινητών δικτύων τρίτης γενιάς – UMTS (Universal Mobile Telecommunications System) έχει αρχίσει να επεκτείνεται. Τα νέα αυτά κινητά δίκτυα αντικαθιστούν τα υπάρχοντα κινητά δίκτυα δεύτερης γενιάς και επιπλέον προσφέρουν προηγμένες υπηρεσίες στους κινητούς χρήστες. Στην πραγματικότητα είμαστε περισσότερο κοντά παρά ποτέ στο όραμα της ενοποίησης των δικτύων παγκοσμίως καθώς επίσης και στο όραμα του “Mobile Broadband”. Είναι εύλογο λοιπόν, οι χρήστες των κινητών δικτύων τρίτης γενιάς να έχουν πλέον την απαίτηση να εκτελούν εφαρμογές και να προσπελαύνουν υπηρεσίες οι οποίες μέχρι σήμερα μπορούσαν να διατεθούν αποκλειστικά από τα συμβατικά ενσύρματα δίκτυα. Έτσι λοιπόν στις μέρες μας ακούμε για υπηρεσίες πραγματικού χρόνου όπως mobile internet, mobile TV, mobile gaming, mobile streaming κ.α. Στόχος της παρούσας διδακτορικής διατριβής είναι η μελέτη και η ανάλυση των μηχανισμών που κρύβονται πίσω από τις παραπάνω εφαρμογές πραγματικού χρόνου. Πρόκειται για μια προσπάθεια να αναλυθούν όλοι οι υπάρχοντες μηχανισμοί μετάδοσης δεδομένων σε πραγματικό χρόνο πάνω από κινητά δίκτυα επικοινωνιών αλλά επίσης και να προταθούν νέοι μηχανισμοί για την όσο το δυνατόν βέλτιστη (από άποψη απόδοσης και ικανοποίησης του τελικού χρήστη) μετάδοση των δεδομένων. Γενικότερα οι μηχανισμοί μετάδοσης δεδομένων διακρίνονται σε δύο βασικές κατηγορίες: • Μηχανισμοί για μετάδοση δεδομένων σημείου προς σημείο (point-to-point data transmission - Unicast). • Μηχανισμοί για μετάδοση δεδομένων από ένα σημείο προς πολλά σημεία (point-to-multipoint data transmission). Όσον αφορά την πρώτη κατηγορία μηχανισμών στην παρούσα διδακτορική διατριβή αναλύονται και προτείνονται μηχανισμοί οι οποίοι προσαρμόζουν το ρυθμό μετάδοσης των δεδομένων ανάλογα με τις συνθήκες φόρτου που επικρατούν στο δίκτυο. Οι μηχανισμοί αυτοί που ονομάζονται μηχανισμοί προσαρμογής του ρυθμού μετάδοσης των δεδομένων είναι κυρίως μηχανισμοί οι οποίοι χρησιμοποιούνται για μετάδοση εφαρμογών πραγματικού χρόνου όπως είναι για παράδειγμα η μετάδοση video σε πραγματικό χρόνο ή μια βιντεοκλήση μεταξύ δύο συνδρομητών ενός δικτύου κινητής τηλεφωνίας. Οι μηχανισμοί προσαρμογής της μετάδοσης πολυμέσων είναι μηχανισμοί μετάδοσης πολυμεσικών (adaptive streaming multimedia) δεδομένων πάνω από δίκτυα, οι οποίοι έχουν τη δυνατότητα να προσαρμόζουν τη μετάδοση των πολυμεσικών δεδομένων στην τρέχουσα κατάσταση του δικτύου. Για την υλοποίηση μηχανισμών προσαρμογής της μετάδοσης απαιτείται ανάπτυξη μηχανισμών τόσο για την παρακολούθηση της κατάστασης του δικτύου όσο και για την προσαρμογή των πολυμεσικών δεδομένων στις εκάστοτε δικτυακές συνθήκες. Ο κύριος στόχος αυτών των μηχανισμών είναι η προσαρμογή του ρυθμού μετάδοσης δεδομένων στο δίκτυο κάθε φορά που οι δικτυακές συνθήκες μεταβάλλονται. Ένα από τα σημαντικότερα χαρακτηριστικά ενός δικτύου κινητών επικοινωνιών τρίτης γενιάς είναι η εισαγωγή της υπηρεσίας Multimedia Broadcast / Multicast Service (MBMS). To MBMS έχει σαν κύριο σκοπό την υποστήριξη IP εφαρμογών πανεκπομπής (broadcact) και πολυεκπομπής (multicast) επιτρέποντας με αυτό τον τρόπο την παροχή υπηρεσιών υψηλού ρυθμού μετάδοσης σε πολλαπλούς χρήστες με οικονομικό τρόπο. Έτσι λοιπόν, όσον αφορά τη δεύτερη κατηγορία μηχανισμών, η multicast μετάδοση δεδομένων σε κινητά δίκτυα επικοινωνιών είναι μια νέα λειτουργικότητα η οποία βρίσκεται ακόμη στο στάδιο των δοκιμών και της προτυποποίησης της. Ένας multicast μηχανισμός μεταδίδει τα δεδομένα μόνο μία φορά πάνω από κάθε σύνδεσμο που αποτελεί τμήμα των μονοπατιών προς τους προορισμούς. Είναι προφανής η αύξηση της απόδοσης που προσφέρει το multicasting λόγω του γεγονότος ότι εκμεταλλεύεται την κατανομή των χρηστών μέσα στο δίκτυο προς όφελος της οικονομίας στην αποστολή πακέτων. Στην παρούσα διδακτορική διατριβή προτείνεται ένας multicast μηχανισμός ο οποίος προσφέρει αξιόπιστη μετάδοση δεδομένων από έναν κεντρικό εξυπηρετητή προς μια ομάδα κινητών χρηστών ενός δικτύου UMTS. Ο συγκεκριμένος μηχανισμός εκτός από τη λειτουργικότητα της multicast δρομολόγησης των πακέτων στους κόμβους του δικτύου, υποστηρίζεται επίσης από ένα σχήμα διαχείρισης της ομάδας των multicast χρηστών. Επιπλέον, στο μηχανισμό έχει ενσωματωθεί επιπλέον λειτουργικότητα η οποία εξασφαλίζει την αδιάλειπτη μετάδοση των δεδομένων στην ομάδα των multicast χρηστών ακόμα και όταν αυτοί βρίσκονται σε διαρκή κίνηση (Handover functionality). Επιπλέον, στην παρούσα διδακτορική διατριβή γίνεται μια προσπάθεια να αξιολογηθούν όλοι οι υπάρχοντες μηχανισμοί που μπορούν να χρησιμοποιηθούν στο UMTS για μετάδοση δεδομένων από έναν αποστολέα προς μια ομάδα παραληπτών. Συγκεκριμένα, οι τρεις μηχανισμοί που αναλύονται είναι ο Broadcast μηχανισμός, o Multiple Unicast μηχανισμός και ο Multicast μηχανισμός. Η αξιολόγηση των μηχανισμών γίνεται με χρήση ενός αναλυτικού μοντέλου το οποίο μετρά το τηλεπικοινωνιακό κόστος μετάδοσης των δεδομένων από τον έναν κόμβο του δικτύου στον άλλον. Το συγκεκριμένο μοντέλο αναπτύχθηκε στα πλαίσια της παρούσας διδακτορικής διατριβής. Οι μηχανισμοί αξιολογούνται για διάφορες τοπολογίες του δικτύου και διαφορετικές κατανομές των χρηστών στο δίκτυο. Τέλος, αντικείμενο της παρούσας διδακτορικής διατριβής αποτελεί η αξιολόγηση των καναλιών μεταφοράς του UMTS και τα οποία μπορούν να χρησιμοποιηθούν για τη multicast μετάδοση των δεδομένων. Ειδικότερα, η επιλογή του κατάλληλου καναλιού μεταφοράς των δεδομένων στο ασύρματο μέσο είναι μια δύσκολη διαδικασία καθώς μια λανθασμένη επιλογή καναλιού μπορεί να οδηγήσει στην αστοχία ενός ολόκληρου κελιού. Τα κανάλια τα οποία αξιολογούνται είναι τα: Forward Access Channel, High Speed – Downlink Shared Channel και Dedicated Channel. Τα παραπάνω κανάλια μεταφοράς αξιολογούνται με βάση το ρυθμό μετάδοσης τους, την απαιτούμενη ισχύ που πρέπει να ανατεθεί από το σταθμό βάσης για καθένα από αυτά, τον αριθμό των χρηστών που μπορούν να εξυπηρετήσουν και τέλος την ποιότητα υπηρεσιών για κάθε χρήστη.
As communications technology is being developed, users’ demand for multimedia services raises. Meanwhile, the Internet has enjoyed tremendous growth in recent years. Consequently, there is a great interest in using the IP-based networks to provide multimedia services. One of the most important areas in which the issues are being debated, is the development of standards for the Universal Mobile Telecommunications System (UMTS). UMTS constitutes the third generation (3G) of cellular wireless networks which aims to provide high-speed data access along with real time voice calls. Wireless data is one of the major boosters of wireless communications and one of the main motivations of the next generation standards. The recent years, the usage of third generation cellular networks has begun to rise all over the world. These new infrastructures substitute the existed second generation cellular networks and offer broadband services to mobile users. Through the 3G mobile networks, the mobile users have the opportunity to run applications and realize services that offered until today only by wired networks. Such broadband services are mobile Internet, mobile TV, mobile gaming, mobile streaming, video calls etc. The main target of this dissertation is the study and the analysis of the mechanisms that are operated behind the above mentioned services and applications. More specifically, we analyze already existed mechanisms used for the transmission of real time services over 3G networks and furthermore we propose new mechanisms for the effective data transmission in 3G networks in terms of network performance and satisfaction of the mobile user. Generally, we consider two categories of mechanisms for the data transmission: • Mechanisms for point to point data transmission. • Mechanisms for point to multipoint data transmission. Regarding the first category of the mechanisms, in this dissertation, we analyze and propose mechanisms for real time data transmission in 3G networks. We focus firstly on schemes that reliable transmit the real time data to mobile users and secondly on mechanisms for adaptive multimedia transmission in UMTS. Bandwidth is a valuable and limited resource for UMTS and every wireless network, in general. Therefore, it is of extreme importance to exploit this resource in the most efficient way. It is essential for a wireless network to have an efficient bandwidth allocation scheme in order the mobile user to experience both real time applications and Internet applications such as HTTP or SMTP. Consequently, when a user experiences a real time application, there should be enough bandwidth available at any time for any other application that the mobile user might realize. In addition, when two different applications run together, the network should guarantee that there is no possibility for any of the above-mentioned applications to prevail against the other by taking all the available channel bandwidth. Taking into consideration the fact that Internet applications adopt mainly TCP as the transport protocol, while real time applications mainly use RTP, the network should guarantee that RTP does not prevail against the TCP traffic. Consequently, this means that there should be enough bandwidth available in the wireless channel for the Internet applications to run properly. To this direction, rate control of real time applications is an important issue in mobile networks. With the aid of rate control schemes the network could adapt the packet transmission rate of real time applications according to the current network conditions giving the opportunity to the mobile users to experience both real time and non real time applications at the same time in their mobile devices. Regarding the second category of the mechanisms, although UMTS networks offer high capacity, the expected demand will certainly overcome the available resources. Thus, the multicast transmission over the UMTS networks constitutes a challenge and an area of research. To this direction the third Generation Partnership Project (3GPP) is currently standardizing the Multimedia Broadcast/Multicast Service (MBMS) framework of UMTS. In this dissertation, we present a new mechanism for the efficient multicast data routing in UMTS. The proposed mechanism is enhanced with multicast group management functionality as well as with functionality related to the user mobility (handover and relocation). Furthermore, it is known that multicasting is more efficient method of supporting group communication than unicasting or broadcasting, as it allows transmission and routing of packets to multiple destinations using fewer network resources. In this dissertation, the three above mentioned methods of supporting group communication in UMTS are analyzed in terms of their performance. The critical parameters of primary interest for the evaluation of any method are the packet delivery cost and the scalability of the method. Finally, this dissertation analyses the role of power control in the multicast transmission in UMTS. It is proposed a power control scheme for the efficient radio bearer selection in MBMS. The choice of the most efficient transport channel in terms of power consumption is a key point for the MBMS since a wrong transport channel selection for the transmission of the MBMS data could result to a significant decrease in the total capacity of the system. Various UMTS transport channels are examined for the transmission of the multicast data and a new algorithm is proposed for the more efficient usage of power resources in the base station.