This research looks at the design of an open transport protocol that supports a range of services including multimedia over low data-rate networks. Low data-rate multimedia applications require a system that provides quality of service (QoS) assurance and flexibility. One promising field is the area of content-based coding. Content-based systems use an array of protocols to select the optimum set of coding algorithms. A content-based transport protocol integrates a content-based application to a transmission network. General transport protocols form a bottleneck in low data-rate multimedia communicationbsy limiting throughpuot r by not maintainingt iming requirementsT. his work presents an original model of a transport protocol that eliminates the bottleneck by introducing a flexible yet efficient algorithm that uses an open approach to flexibility and holistic architectureto promoteQ oS.T he flexibility andt ransparenccyo mesi n the form of a fixed syntaxt hat providesa seto f transportp rotocols emanticsT. he mediaQ oSi s maintained by defining a generic descriptor. Overall, the structure of the protocol is based on a single adaptablea lgorithm that supportsa pplication independencen, etwork independencea nd quality of service. The transportp rotocol was evaluatedth rougha set of assessmentos:f f-line; off-line for a specific application; and on-line for a specific application. Application contexts used MPEG-4 test material where the on-line assessmenuts eda modified MPEG-4 pl; yer. The performanceo f the QoSc ontrolledt ransportp rotocoli s often bettert hano thers chemews hen appropriateQ oS controlledm anagemenatl gorithmsa re selectedT. his is shownf irst for an off-line assessmenwt here the performancei s compared between the QoS controlled multiplexer,a n emulatedM PEG-4F lexMux multiplexers chemea, ndt he targetr equirements. The performanceis also shownt o be better in a real environmentw hen the QoS controlled multiplexeri s comparedw ith the real MPEG-4F lexMux scheme.

This research focuses on the investigation of a fast simulation method for broadband telecommunication networks, such as ATM networks and IP networks. As a result of this research, a hybrid simulation model is proposed, which combines the analytical modelling and event-driven simulation modelling to speeding up the overall simulation. The division between foreground and background traffic and the way of dealing with these different types of traffic to achieve improvement in simulation time is the major contribution reported in this thesis. Background traffic is present to ensure that proper buffering behaviour is included during the course of the simulation experiments, but only the foreground traffic of interest is simulated, unlike traditional simulation techniques. Foreground and background traffic are dealt with in a different way. To avoid the need for extra events on the event list, and the processing overhead, associated with the background traffic, the novel technique investigated in this research is to remove the background traffic completely, adjusting the service time of the queues for the background traffic to compensate (in most cases, the service time for the foreground traffic will increase). By removing the background traffic from the event-driven simulator the number of cell processing events dealt with is reduced drastically. Validation of this approach shows that, overall, the method works well, but the simulation using this method does have some differences compared with experimental results on a testbed. The reason for this is mainly because of the assumptions behind the analytical model that make the modelling tractable. Hence, the analytical model needs to be adjusted. This is done by having a neural network trained to learn the relationship between the input traffic parameters and the output difference between the proposed model and the testbed. Following this training, simulations can be run using the output of the neural network to adjust the analytical model for those particular traffic conditions. The approach is applied to cell scale and burst scale queueing to simulate an ATM switch, and it is also used to simulate an IP router. In all the applications, the method ensures a fast simulation as well as an accurate result.

A real dynamic plant is used to compare, test and assess the present theoretical techniques of adaptive, learning or intelligent control under practical criteria. Work of this nature has yet to be carried out if "intelligent control" is to have a place in everyday practice. The project follows a natural pattern of development, the construction of computer programmes being an important part of it. First, a. real plant - a model steam engine - and its electronic interface with a general purpose digital computer are designed and built as part of the project. A rough mathematical model of the plant is then obtained through identification tests. Second, conventional control of the plant is effected using digital techniques and the above mentioned mathematical model, and the results are saved to compare with and evaluate the results of "intelligent control". Third, a few well-known adaptive or learning control algorithms are investigated and implemented. These tests bring out certain practical problems not encountered or not given due consideration in theoretical or simulation studies. Alternatively, these problems materialise because assumptions made on paper are not readily available in practice. The most important of these problematic. assumptions are those relating to computational time and storage, convergence of the adaptive or learning algorithm and the training of the controller. The human operator as a distinct candidate for the trainer is also considered and the problems therein are discussed. Finally, the notion of fuzzy sets and logic is viewed from the control point and a controller using this approach is developed and implemented. The operational advantages and the results obtained, albeit preliminary, demonstrate the potential power of this notion and provide the grounds for further work in this area.

Resource Management is a crucial aspect in the next generation cellular networks since the use of W-CDMA technology gives an inherent flexibility in managing the system capacity. The concept of a “Service Level Agreement” (SLA) also plays a very important role as it is the means to guarantee the quality of service provided to the customers in response to the level of service to which they have subscribed. Hence there is a need to introduce effective SLA-based policies as part of the radio resource management. This work proposes the application of intelligent agents in SLA-based control in resource management, especially when congestion occurs. The work demonstrates the ability of intelligent agents in improving and maintaining the quality of service to meet the required SLA as the congestion occurs. A particularly novel aspect of this work is the use of learning (here Case Based Reasoning) to predict the control strategies to be imposed. As the system environment changes, the most suitable policy will be implemented. When congestion occurs, the system either proposes the solution by recalling from experience (if the event is similar to what has been previously solved) or recalculates the solution from its knowledge (if the event is new). With this approach, the system performance will be monitored at all times and a suitable policy can be immediately applied as the system environment changes, resulting in maintaining the system quality of service.

The increasing use of interactive multimedia applications over the Internet has created a problem of congestion. This is because a majority of these applications do not respond to congestion indicators. This leads to resource starvation for responsive flows, and ultimately excessive delay and losses for all flows therefore loss of quality. This results in unfair sharing of network resources and increasing the risk of network ‘congestion collapse’. Current Congestion Control Mechanisms such as ‘TCP-Friendly Rate Control’ (TFRC) have been able to achieve ‘fair-share’ of network resource when competing with responsive flows such as TCP, but TFRC’s method of congestion response (i.e. to reduce Packet Rate) is not ideally matched for interactive multimedia applications which maintain a fixed Frame Rate. This mismatch of the two rates (Packet Rate and Frame Rate) leads to buffering of frames at the Sender Buffer resulting in delay and loss, and an unacceptable reduction of quality or complete loss of service for the end-user. To address this issue, this thesis proposes a novel Congestion Control Mechanism which is referred to as ‘TCP-friendly rate control – Fine Grain Scalable’ (TFGS) for interactive multimedia applications. This new approach allows multimedia frames (data) to be sent as soon as they are generated, so that the multimedia frames can reach the destination as quickly as possible, in order to provide an isochronous interactive service. This is done by maintaining the Packet Rate of the Congestion Control Mechanism (CCM) at a level equivalent to the Frame Rate of the Multimedia Encoder.The response to congestion is to truncate the Packet Size, hence reducing the overall bitrate of the multimedia stream. This functionality of the Congestion Control Mechanism is referred to as Packet Size Truncation (PST), and takes advantage of adaptive multimedia encoding, such as Fine Grain Scalable (FGS), where the multimedia frame is encoded in order of significance, Most to Least Significant Bits. The Multimedia Adaptation Manager (MAM) truncates the multimedia frame to the size indicated by the Packet Size Truncation function of the CCM, accurately mapping user demand to available network resource. Additionally Fine Grain Scalable encoding can offer scalability at byte level granularity, providing a true match to available network resources. This approach has the benefits of achieving a ‘fair-share’ of network resource when competing with responsive flows (as similar to TFRC CCM), but it also provides an isochronous service which is of crucial benefit to real-time interactive services. Furthermore, results illustrate that an increased number of interactive multimedia flows (such as voice) can be carried over congested networks whilst maintaining a quality level equivalent to that of a standard landline telephone. This is because the loss and delay arising from the buffering of frames at the Sender Buffer is completely removed. Packets sent maintain a fixed inter-packet-gap-spacing (IPGS). This results in a majority of packets arriving at the receiving end at tight time intervals. Hence, this avoids the need of using large Playout (de-jitter) Buffer sizes and adaptive Playout Buffer configurations. As a result this reduces delay, improves interactivity and Quality of Experience (QoE) of the multimedia application.

The continuous miniaturisation of sensors, as well as the progression in wearable electronics, embedded software, digital signal processing and biomedical technologies, have led to new usercentric networks, where devices can be carried in the user’s pockets, attached to the user’s body. Body-centric wireless communications (BCWCs) is a central point in the development of fourth generation mobile communications. Body-centric wireless networks take their place within the personal area networks, body area networks and sensor networks which are all emerging technologies that have a wide range of applications (such as, healthcare, entertainment, surveillance, emergency, sports and military). The major difference between BCWC and conventional wireless systems is the radio channels over which the communication takes place. The human body is a hostile environment from a radio propagation perspective and it is therefore important to understand and characterise the effects of the human body on the antenna elements, the radio channel parameters and, hence, system performance. This thesis focuses on the study of body-worn antennas and on-body radio propagation channels. The performance parameters of five different narrowband (2.45 GHz) and four UWB (3.1- 10.6 GHz) body-worn antennas in the presence of human body are investigated and compared. This was performed through a combination of numerical simulations and measurement campaigns. Parametric studies and statistical analysis, addressing the human body effects on the performance parameters of different types of narrowband and UWB antennas have been presented. The aim of this study is to understand the human body effects on the antenna parameters and specify the suitable antenna in BCWCs at both 2.45 GHz and UWB frequencies. Extensive experimental investigations are carried out to study the effects of various antenna types on the on-body radio propagation channels as well. Results and analysis emphasize the best body-worn antenna for reliable and power-efficient on-body communications. Based on the results and analysis, a novel dual-band and dual-mode antenna is proposed for power-efficient and reliable on-body and off-body communications. The on-body performance of the DBDM antenna at 2.45 GHz is compared with other five narrowband antennas. Based on the results and analysis of six narrowband and four UWB antennas, antenna specifications and design guidelines are provided that will help in selecting the best body-worn antenna for both narrowband and UWB systems to be applied in body-centric wireless networks (BCWNs). A comparison between IV the narrowband and UWB antenna parameters are also provided. At the end of the thesis, the subject-specificity of the on-body radio propagation channel at 2.45 GHz and 3-10 GHz was experimentally investigated by considering eight real human test subjects of different shapes, heights and sizes. The subject-specificity of the on-body radio propagation channels was compared between the narrowband and UWB systems as well.

Optimizing radio resource within a network and across cooperating heterogeneous networks is the focus of this thesis. Cooperation in a multi-network environment is tackled by investigating network selection mechanisms. These play an important role in ensuring quality of service for users in a multi-network environment. Churning of mobile users from one service provider to another is already common when people change contracts and in a heterogeneous communication environment, where mobile users have freedom to choose the best wireless service-real time selection is expected to become common feature. This real time selection impacts both the technical and the economic aspects of wireless network operations. Next generation wireless networks will enable a dynamic environment whereby the nodes of the same or even different network operator can interact and cooperate to improve their performance. Cooperation has emerged as a novel communication paradigm that can yield tremendous performance gains from the physical layer all the way up to the application layer. Game theory and in particular coalitional game theory is a highly suited mathematical tool for modelling cooperation between wireless networks and is investigated in this thesis. In this thesis, the churning behaviour of wireless service users is modelled by using evolutionary game theory in the context of WLAN access points and WiMAX networks. This approach illustrates how to improve the user perceived QoS in heterogeneous networks using a two-layered optimization. The top layer views the problem of prediction of the network that would be chosen by a user where the criteria are offered bit rate, price, mobility support and reputation. At the second level, conditional on the strategies chosen by the users, the network provider hypothetically, reconfigures the network, subject to the network constraints of bandwidth and acceptable SNR and optimizes the network coverage to support users who would otherwise not be serviced adequately. This forms an iterative cycle until a solution that optimizes the user satisfaction subject to the adjustments that the network provider can make to mitigate the binding constraints, is found and applied to the real network. The evolutionary equilibrium, which is used to 3 compute the average number of users choosing each wireless service, is taken as the solution. This thesis also proposes a fair and practical cooperation framework in which the base stations belonging to the same network provider cooperate, to serve each other‘s customers. How this cooperation can potentially increase their aggregate payoffs through efficient utilization of resources is shown for the case of dynamic frequency allocation. This cooperation framework needs to intelligently determine the cooperating partner and provide a rational basis for sharing aggregate payoff between the cooperative partners for the stability of the coalition. The optimum cooperation strategy, which involves the allocations of the channels to mobile customers, can be obtained as solutions of linear programming optimizations.

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.

This thesis investigates reinforcement learning algorithms suitable for learning in large state space problems and coevolution. In order to learn in large state spaces, the state space must be collapsed to a computationally feasible size and then generalised about. This thesis presents two new implementations of the classic temporal difference (TD) reinforcement learning algorithm Sarsa that utilise fuzzy logic principles for approximation, FQ Sarsa and Fuzzy Sarsa. The effectiveness of these two fuzzy reinforcement learning algorithms is investigated in the context of an agent marketplace. It presents a practical investigation into the design of fuzzy membership functions and tile coding schemas. A critical analysis of the fuzzy algorithms to a related technique in function approximation, a coarse coding approach called tile coding is given in the context of three different simulation environments; the mountain-car problem, a predator/prey gridworld and an agent marketplace. A further comparison between Fuzzy Sarsa and tile coding in the context of the nonstationary environments of the agent marketplace and predator/prey gridworld is presented. This thesis shows that the Fuzzy Sarsa algorithm achieves a significant reduction of state space over traditional Sarsa, without loss of the finer detail that the FQ Sarsa algorithm experiences. It also shows that Fuzzy Sarsa and gradient descent Sarsa(λ) with tile coding learn similar levels of distinction against a stationary strategy. Finally, this thesis demonstrates that Fuzzy Sarsa performs better in a competitive multiagent domain than the tile coding solution.

A Mobile Ad Hoc Networks (MANET) is a collection of mobile nodes that can communicate with each other using multihop wireless links without utilizing any fixed based-station infrastructure and centralized management. Each mobile node in the network acts as both a host generating flows or being destination of flows and a router forwarding flows directed to other nodes. Future applications of MANETs are expected to be based on all-IP architecture and be capable of carrying multitude real-time multimedia applications such as voice and video as well as data. It is very necessary for MANETs to have an efficient routing and quality of service (QoS) mechanism to support diverse applications. This thesis proposes an on-demand Node-Disjoint Multipath Routing protocol (NDMR) with low broadcast redundancy. Multipath routing allows the establishment of multiple paths between a single source and single destination node. It is also beneficial to avoid traffic congestion and frequent link breaks in communication because of the mobility of nodes. The important components of the protocol, such as path accumulation, decreasing routing overhead and selecting node-disjoint paths, are explained. Because the new protocol significantly reduces the total number of Route Request packets, this results in an increased delivery ratio, smaller end-to-end delays for data packets, lower control overhead and fewer collisions of packets. Although NDMR provides node-disjoint multipath routing with low route overhead in MANETs, it is only a best-effort routing approach, which is not enough to support QoS. DiffServ is a standard approach for a more scalable way to achieve QoS in any IP network and could potentially be used to provide QoS in MANETs because it minimises the need for signalling. However, one of the biggest drawbacks of DiffServ is that the QoS provisioning is separate from the routing process. This thesis presents a Multipath QoS Routing protocol for 111 supporting DiffServ (MQRD), which combines the advantages of NDMR and DiffServ. The protocol can classify network traffic into different priority levels and apply priority scheduling and queuing management mechanisms to obtain QoS guarantees.