The mobile communications industry has undergone rapid developments in the last decade. Consumers and businesses have become increasingly technology savvy and are likely to expect creative wireless access to many services. Consequently, future mobile systems are envisioned to support a wide variety of applications. The support of these diverse services (e.g. data and multimedia) has led to different bandwidth requirements, inevitably rendering many of the traditional radio resource management (RRM) mechanisms deficient. For one, the present radio resource dimensioning strategy that dimensions radio resource based on "busy hour" traffic does not allow operators to efficiently utilised the radio resource, which is scarce and limited. In line with the future demand driven resource allocation, two novel Scalable Resource Allocation (ScRA) techniques are proposed in this thesis to optimise the utilisation of radio resources within multiple services. The proposed novel ScRA techniques performances are evaluated on a typical GSM/GPRS mobile network through our developed dynamic system level simulator. Based on the simulation results, significant radio resource utilization gain, higher data throughput, lesser delay for data connections can be achieved with the use of our ScRA technique over the conventional StRA technique. For example, when the WWW arrival rate reaches approximate 35 Kbits/sec, ScRA technique can attain the throughput difference as high as 27.663%. On the other hand, the system throughput gain can yield as high as 27.44% when the Email arrival rate reached 1769 bits/sec for ScRA technique. Above all, we have also verified that the ScRA techniques perform well within the consideration of desirable voice QoSs at all times. The better performance of this technique is credited to its flexible and adaptive radio channel allocation (or re-allocation) between the multiple services during each sampled time. Key words; Radio Resource Management, Call Admission Control, Scalable Resource Allocation.

Mobile communications technology has increasingly become integrated into modern day life and the rate of this integration is growing rapidly. However, many mobile devices including personal digital assistants (PDA) and mobile phones are predominantly passive, lacking a facility for automated information handling. For example, a mobile phone cannot automatically detect a user's context and react to support that situation. In this thesis, we present a context aware system that automatically detects mobile users' activity context through information gathered from a mobile phone. The thesis contains two main contributions: the development of the context prediction model and the implementation of a context aware system. The development of the context prediction model involves the use of Bayesian networks to solve the uncertainty regarding a user current activity, along with the prediction of possible future activities. A Bayesian probabilistic network is a comprehensive probabilistic computational framework for reasoning under conditions of uncertainty. In this research, we utilise Bayesian networks as a modelling tool to support context reasoning and activity prediction within the context aware domain. A further contribution of this work is the development of a real world context aware system based upon our above theoretical contribution. A Sony Ericsson P910i mobile phone alongside a context aware engine server has been implemented to demonstrate the theoretical principles. The mobile phone is used to capture contextual information from a user and interact with the context engine server. Using a probabilistic approach and uncertainty metric, the Bayesian context prediction network (which is located on the context engine server) predicts a user's likely activity, and contextual information is provided to the user via their mobile phone. A dataset of real-life user activities has been collected to evaluate the accuracy of the predicted activities. The results of the evaluation demonstrated encouraging levels of accuracy for this approach.

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Efficient source coding techniques are necessary to make optimal use of the limited bandwidth available in mobile phone networks. Most current mobile telephone communication systems compress the speech waveform by using speech coders based on the Code Excited Linear Prediction (CELP) model. Such coders give high quality speech at bit rates of 8 kbps and above. Below 8 kbps, the quality of the coded speech degrades rapidly. At rates of 6 kbps and below, parametric speech coders offer better speech quality. These coders reduce the required bit rate by transmitting certain characteristics of the speech waveform to the decoder, rather than attempting to code the waveform itself. The disadvantage of parametric coders is that the maximum achievable quality is limited by assumptions made during the coding of the speech signal. The aim of the research presented is to investigate and eliminate the factors that limit the speech quality of parametric coders. A new pitch synchronous coding model is proposed that operates on individual pitch cycle waveforms of speech rather than longer, fixed length frames as used in classic techniques. In order to implement a pitch synchronous coder, new pitch cycle detection algorithms have been proposed. Pitch synchronous parameter analysis was investigated and several new techniques have been developed. A novel pitch synchronous split-band voicing estimator has been proposed that utilises only the phase of the speech harmonics rather than the periodicity used in traditional techniques. Fixed rate quantisation of pitch synchronous speech parameters has been investigated and a joint quantisation/interpolation scheme has been proposed. This scheme has been applied to the quantisation of the pitch synchronous parameters and has been shown to outperform traditional quantisation techniques. A comparison of a reference parametric coder with its pitch synchronous counterpart has shown that the pitch synchronous paradigm eliminates some of the main factors that limit the speech quality in parametric coders. It is expected that this will lead to the development of speech coders that can produce speech of higher quality than current parametric coders operating at the same bit rate. Key words: Speech Coding, Pitch Synchronous, Sinusoidal Coding, Split-Band LPC Coding.

Numerous examinations of the weaknesses with Authentication of Key Agreement protocol (AKA) of Universal Mobile Telecommunications System (UMTS) have been presented by various researchers. It is plausible to suggest that the majority of the proposed solutions of authentication protocols in the reported investigative works are formulated based on asymmetric algorithms which consume more computational overhead than symmetric algorithms. There is no evidence that the symmetric algorithms could lead to the existing weaknesses.

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Relay based cellular networks (RBCNs) are the technologies that incorporate multi-hop communication into traditional cellular networks. A RBCN can potentially support higher data rates, more stable radio coverage and more dynamic services. In reality, RBCNs still suffer from performance degradation in terms of high user congestion, base station failure and overloading in emergency situations. The focus of this thesis is to explore the potential to improve IEEE802.16j supported RBCN performance in user congestion and emergency situations using adjustments to the RF layer (by antenna adjustments or extensions using multi-hop) and cooperative adjustment algorithms, e.g. based on controlling frequency allocation centrally and using distributed approaches. The first part of this thesis designs and validates network reconfiguration algorithms for RBCN, including a cooperative antenna power control algorithm and a heuristic antenna tilting algorithm. The second part of this thesis investigates centralized and distributed dynamic frequency allocation for higher RBCN frequency efficiency, network resilience, and computation simplicity. It is demonstrated that these benefits mitigate user congestion and base station failure problems significantly. Additionally, interweaving coordinated dynamic frequency allocation and antenna tilting is investigated in order to obtain the benefits of both actions. The third part of this thesis incorporates Delay Tolerate Networking (DTN) technology into RBCN to let users self-organize to connect to functional base station through multi-hops supported by other users. Through the use of DTN, RBCN coverage and performance are improved. This thesis explores the augmentation of DTN routing protocols to let more un-covered users connect to base stations and improve network load balancing

Mobile terminals allow users to access services while on the move. This unique feature has driven the rapid growth in the mobile network industry, changing it from a new technology into a massive industry within less than two decades. Handover is the essential functionality for dealing with the mobility of the mobile users. Compared with the conventional hard handover employed in the GSM mobile networks, the soft handover used in IS-95 and being proposed for 3G has better performance on both link and system level. Previous work on soft handover has led to several algorithms being proposed and extensive research has been conducted on the performance analysis and parameters optimisation of these algorithms. Most of the previous analysis focused on the uplink direction. However, in future mobile networks, the downlink is more likely to be the bottleneck of the system capacity because of the asymmetric nature of new services, such as Internet traffic. In this thesis, an in-depth study of the soft handover effects on the downlink direction of WCDMA networks is carried out, leading to a new method of optimising soft handover for maximising the downlink capacity and a new power control approach.

Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes.

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