The evolution of the intemet protocol (IP) to offer quality of service (QoS) makes it a suitable core network protocol for next generation networks (NGN). The QoS features incorporated to IP will enable future lP-based wireless networks to meet QoS requirements of various multimedia traffic. The Differentiated Service (Diffserv) Architecture is a promising QoS technology due to its scalability which arises from traffic flow aggregates. For this reason, in this dissertation a network infrastructure based on DiffServ is assumed. This architecture provides assured service (AS) and premium service (PrS) classes in addition to best-effort service (BE). The medium access control (MAC) protocol is one of the important design issues in wireless networks. In a wireless network carrying multimedia traffic, the MAC protocol is required to provide simultaneous support for a wide variety of traffic types, support traffic with delay and jitter bounds, and assign bandwidth in an efficient and fair manner among traffic classes. Several MAC protocols capable of supporting multimedia services have been proposed in the literature, the majority of which were designed for wireless A1M (Asynchronous Transfer Mode). The focus of this dissertation is on time division multiple access and code division multiple access (TDMAlCDMA) based MAC protocols that support QoS in lP-based wireless networks. This dissertation begins by giving a survey of wireless MAC protocols. The survey considers MAC protocols for centralised wireless networks and classifies them according to their multiple access technology and as well as their method of resource sharing. A novel TDMAlCDMA based MAC protocol incorporating techniques from existing protocols is then proposed. To provide the above-mentioned services, the bandwidth is partitioned amongst AS and PrS classes. The BE class utilizes the remaining bandwidth from the two classes because it does not have QoS requirements. The protocol employs a demand assignment (DA) scheme to support traffic from PrS and AS classes. BE traffic is supported by a random reservation access scheme with dual multiple access interference (MAl) admission thresholds. The performance of the protocol, i.e. the AS or PrS call blocking probability, and BE throughput are evaluated through Markov analytical models and Monte-Carlo simulations. Furthermore, the protocol is modified and incorporated into IEEE 802.16 broadband wireless access (BWA) network. / Thesis (M.Sc.)-University of KwaZulu-Natal, Durban, 2005.

The rapid growth of wireless subscribers and services as well as the increased use of internet services, suggest that wireless internet access will increase rapidly over the next few years. This will require the provision of high data rate wireless communication services. However the problem of a limited and expensive radio spectrum coupled with the problem of the wireless fading channel makes it difficult to provide these services. For these reasons, the research area of high data rate, bandwidth efficient and reliable wireless communications is currently receiving much attention. Concatenated codes are a class of forward error correction codes which consist of two or more constituent codes. These codes achieve reliable communications very close to the Shannon limit provided that sufficient diversity, such as temporal or spatial diversity, is available. Space-time trellis codes (STTCs) merge channel coding and transmit antenna diversity to improve system capacity and performance. The main focus of this dissertation is on STTCs and concatenated STTCs in quasi-static and rapid Rayleigh fading channels. Analytical bounds are useful in determining the behaviour of a code at high SNRs where it becomes difficult to generate simulation results. A novel method is proposed to analyse the performance of STTCs and the accuracy of this analysis is compared to simulation results where it is shown to closely approximate system performance. The field of concatenated STTCs has already received much attention and has shown improved performance over conventional STTCs. It was recently shown that double concatenated convolutional codes in AWGN channels outperform simple concatenated codes. Motivated by this, two double concatenated STTC structures are proposed and their performance is compared to that of a simple concatenated STTCs. It is shown that double concatenated STTCs outperform simple concatenated STTCs in rapid Rayleigh fading channels. An analytical model for this system in rapid fading is developed which combines the proposed analytical method for STTCs with existing analytical techniques for concatenated convolutional codes. The final part of this dissertation considers a direct-sequencejslow-frequency-hopped (DSj SFH) code division multiple access (CDMA) system with turbo coding and multiple transmit antennas. The system model is modified to include a more realistic, time correlated Rayleigh fading channel and the use of side information is incorporated to improve the performance of the turbo decoder. Simulation results are presented for this system and it is shown that the use of transmit antenna diversity and side information can be used to improve system performance. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2002.

This dissertation exammes two of the rlsing technologies in the world of wireless, cellular communications - CDMA and the software radio. This thesis covers many of the issues related to these two emerging field s of wireless communications, establish ing a theoretical framework for the broader issues of implementation. To this end, the thesis covers many of the basic issues of spread spectrum communications, in addition to establishing the need for, and defining the role of, the software radio. Amalgamation of these two key areas of interest is embellished in a presentation of many of the concerns of implementing a specific CDMA system on a particular type of software radio - the Alcatel Altech Telecomms Flexible Radio Platform. Of primary concern in the research methodology embraced in this thesis is the mastering of a variety of analysis and implementation tools. Once the theoretical background has been substantiated by current expositions, the thesis launches along a highly deterministic route. First, the research issues are tested in a mathematical environment for suitability to the given task. Second, an analysis of the appropriateness of the technique for the software radio environment is undertaken, culminating in the attempted deployment within the hardware environmenl. Rigorous testing of the input/output mapping characteristics of the hardware instantiations created in this manner complements the research methodology with a viability study. This procedure is repeated with many elements of the CDMA system design as they are examined, simu lated, deployed and tested. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2000.

The Transmission Control Protocol (TCP) is the most widely used transport protocol in the Internet. TCP is a reliable transport protocol that is tuned to perform well in wired networks where packet losses are mainly due to congestion. Wireless channels are characterized by losses due to transmission errors and handoffs. TCP interprets these losses as congestion and invokes congestion control mechanisms resulting in degradation of performance. TCP is usually layered over the Internet protocol (lP) at the network layer. JP is not reliable and does not provide for any Quality of Service (QoS). The Internet Engineering Task Force (IETF) has provided two techniques for providing QoS in the Internet. These include Integrated Services (lntServ) and Differentiated Services (DiffServ). IntServ provides flow based quality of service and thus it is not scalable on connections with large flows. DiffServ has grown in popularity since it is scalable. A packet in a DiffServ domain is classified into a class of service according to its contract profile and treated differently by its class. To provide end-to-end QoS there is a strong interaction between the transport protocol and the network protocol. In this dissertation we consider the performance of the TCP over a wireless channel. We study whether the current TCP protocols can deliver the desired quality of service faced with the challenges they have on wireless channel. The dissertation discusses the methods of providing for QoS in the Internet. We derive an analytical model for TCP protocol. It is extended to cater for the wireless channel and then further differentiated services. The model is shown to be accurate when compared to simulation. We then conclude by deducing to what degree you can provide the desired QoS with TCP on a wireless channel. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2001.

Future wireless networks will have to handle varying combinations of multimedia traffic that present the network with numerous quality of service (QoS) requirements. The continuously growing demand for mobile phones has resulted in radio spectrum becoming a precious resource that cannot be wasted. The current second-generation mobile networks are designed for voice communication and, even with the enhancements being implemented to accommodate data, they cannot efficiently handle the multimedia traffic demands that will be introduced in the near future. This thesis begins with a survey of existing wireless ATM (WATM) protocols, followed by an examination of some medium access control (MAC) protocols, supporting multimedia traffic, and based on code division multiple access (CDMA) physical layers. A WATM link layer protocol based on a CDMA physical layer, and incorporating techniques from some of the surveyed protocols, is then proposed. The MAC protocol supports a wide range of service requirements by utilising a flexible scheduling algorithm that takes advantage of the graceful degradation of CDMA with increasing user interference to schedule cells for transmission according to their maximum bit error rate (BER) requirements. The data link control (DLC) accommodates the various traffic types by allowing virtual channels (VCs) to make use of forward error correction (FEc) or retransmission techniques. The proposed link layer protocol has been implemented on a Blue Wave Systems DSP board that forms part of Alcatel Altech Telecoms' software radio platform. The details and practicality of the implementation are presented. A simulation model for the protocol has been developed using MIL3 's Opnet Modeler. Hence, both simulated and measured performance results are presented before the thesis concludes with suggestions for improvements and future work. / Thesis (M.Sc.)-University of Natal, Durban, 2000.

Ring networks are appropriate for the full range of network levels, including multiprocessor systems, local area computer networks and high speed backbones. The most well known and widely implemented examples are the IBM token ring and FDDI networks. Ring networks have the advantages of high channel utilisation and bounded delay if an n-limited service policy is used. The packet transfer delay, defined as the average time a packet spends in the network from the time it is generated until the time it is received at its destination node, improves with the number of rings on which a node is connected. However, many ring connections are not economically feasible since the cost of the ring interface increases with the number of rings. There has been an abundance of previous work on single token ring networks. A number of papers on slotted rings, register insertion rings and more complex ring architectures have also been published. However, there is very little existing literature on multiple ring networks as well as ring networks in clustered traffic environments, i.e. where nodes from the same cluster tend to communicate more with each other than with other nodes in the network. This thesis focuses on two network topologies that make use of multiple rings and are well suited to clustered traffic environments: the two-connected multiple ring (2-MR) and the destination removal double ring (DRDR). For the 2-MR network, three different practical token-based protocols are investigated in an attempt to optimise performance. It is further shown that significant performance improvements can be achieved by employing a slotted ring protocol rather than the token ring protocol. The DRDR network is also examined and its performance compared to the aforementioned architectures. For each of the six cases, both random and clustered traffic patterns are considered and compared. Analytical results are derived which are verified by results obtained from computer simulations. Furthermore, we look at exact methods of analysing ring networks. A mean value analysis of a single token ring network with a I-limited service discipline is performed, which clearly shows the complexity exact methods introduce. Finally, although it has been stated in the literature that an exact analysis of a multiple symmetrical token ring network is intractable, we present a novel Markov chain approach that gives exact results for near zero loads. / Thesis (M.Sc.-Electronic Engineering)-University of Natal, 1998.

The document describes the development of a micro-processor based terminal multiplexer to connect four terminals to a standard Hewlett Packard series 1000 mini-computer. The project was required to fulfill the dual roll of both increasing the number of terminals that the HPI000 could support and of reducing the peripheral load on the host CPU. The final product occupied a standard 200mm square HP size interface card and used an 8085 micro-processor and several 8085 family peripheral chips to provide four full duplex serial channels and a high speed data link with the host. A multi-tasking executive was written to control the multiplexer software which was finally implemented as 15 independent tasks occupying 8 kilo-bytes of eprom. The software was written to perform all terminal interaction and editing in order to reduce the host CPU involvement to a single interrupt per record. The resultant interface proved capable of handling an aggregate throughput in excess of 4000 characters per second which was sufficient to cope with all four terminals running at 9600 bits per second, even when all four were transferring in burst mode. The interface also proved to be between five and eighteen times less demanding on the host than the two standard Hewlett Packard interfaces then available. When compared to the low cost HP12531 interface, the multiplexer increased the 9600b/s terminal handling capability of the host from 3 terminals to 52. / Thesis (M.Sc.-Electronic Engineering)-University of Natal, 1987.

Natural phenomena such as rainfall are responsible for communication service disruption, leading to severe outages and bandwidth inefficiency in both terrestrial and satellite systems, especially above 10 GHz. Rainfall attenuation is a source of concern to radio engineers in link budgeting and is primarily related to the rainfall mechanism of absorption and scattering of millimetric signal energy. Therefore, the study of rainfall microstructure can serve as a veritable means of optimizing network parameters for the design and deployment of millimetric and microwave links. Rainfall rate and rainfall drop-size are two microstructural parameters essential for the appropriate estimation of local rainfall attenuation. There are several existing analytical and empirical models for the prediction of rainfall attenuation and their performances largely depend on regional and climatic characteristics of interest. In this study, the thrust is to establish the most appropriate models in South African areas for rainfall rate and rainfall drop-size. Statistical analysis is derived from disdrometer measurements sampled at one-minute interval over a period of two years in Durban, a subtropical site in South Africa. The measurements are further categorized according to temporal rainfall regimes: drizzle, widespread, shower and thunderstorm. The analysis is modified to develop statistical and empirical models for rainfall rate using gamma, lognormal, Moupfouma and other ITU-R compliant models for the control site. Additionally, rain drop-size distribution (DSD) parameters are developed from the modified gamma, lognormal, negative exponential and Weibull models. The spherical droplet assumption is used to estimate the scattering parameters for frequencies between 2 GHz and 1000 GHz using the disdrometer diameter ranges. The resulting proposed DSD models are used, alongside the scattering parameters, for the prediction and estimation of rainfall attenuation. Finally, the study employs correlation and regression techniques to extend the results to other locations in South Africa. The cumulative density function analysis of rainfall parameters is applied for the selected locations to obtain their equivalent models for rainfall rate and rainfall DSD required for the estimation of rainfall attenuation. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2011.

WCDMA promises to achieve high-speed internet, high quality image transmission and high-speed data services with larger system capacity. However, Multiple Access Interference is one of the major causes of transmission impairment, which reduces the link capacity in WCDMA systems. The Adaptive Antenna Array (AAA) technique reduces multiple access interference by directing antenna beam nulls towards the interfering signals by weighting the received signals from all antennas before combining the signals. With the very rapid advancement of wireless personal communications services, a new challenge to the cellular industry is the integration of multiple systems and applications on a single device. A software radio technique offers a possible solution to achieve this goal including international roaming and multiple standard operations within the same geographical area. The main attraction of a software radio is it's flexibility, in that it can be programmed for emerging cellular standards allowing it to be updated with new software without any changes in the hardware infrastructure. A software radio incorporating adaptive array beamforming at the receiver can increase the total carried traffic in a system and transmit power while the probability of call blocking and forced termination can also be decreased. This dissertation examines WCDMA, AAA, power control and software radio techniques in the world of wireless communication systems. Once the theoretical background of CDMA and AAA has been substantiated, the thesis establishes the need for power control in mobile systems by examining simulation results. An AAA receiver with six antenna elements is proposed and evaluated in different environments as a precursor to implementation. It can be inferred that when the link is interference limited, the link capacity can be increased and it has been shown that the AAA receiver with six antenna elements increases the link capacity to about 2.9 times that of the single antenna RAKE receiver. This thesis also examines the basic concepts of VHDL and considers this as the principle means to program reconfigurable core FPGA's in the software radio. A three-layered (PC/DSP/FPGA) software radio test bed is used to implement an AAA receiver. The architecture of the test bed is designed in such a way that it can be used to evaluate the performance of various FPGA based transceivers and coding schemes etc. Many of the desirable features and flexibilities inherent in the software radio concept are available on this test bed and the system has proved to be capable of high speed digital processing and is ideally suited to the development of time critical system components. The bit error rate achieved using the implemented receiver is assessed and compared to simulation results in an environment incorporating Rayleigh fading and AWGN. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2001.

An ad-hoc network is a collection of mobile nodes (wireless communication devices) that transmit data over systems that do not require any centralized control, such as that found in cellular networks. This makes ad-hoc networks suitable for military type applications, since there is no need for an established backbone infrastructure and hence no single-point-of-failure. However, other uses of ad-hoc systems include search and rescue missions, law enforcement operations, commercial and educational communication of laptop (and other handheld device) data, as well as in the transmission of environmental sensor information. The mobile ad-hoc concept brings many design challenges. The dynamic freedom of movement from mobile nodes causes random, sometimes rapidly time changing topologies, which are inappropriate for use through traditional wired protocols. In addition, wireless networks generally contain greater bandwidth, processing and power constraints than their wired counterparts, since they are implemented on embedded mobile, handheld devices. Thus, a different approach is needed in the wireless network domain. This has resulted in wireless routing protocols employing adaptive, multi-hop, distributed methodologies in which each node additionally acts as a router for each of its neighbouring nodes, in order to achieve a large degree of network connectivity. However, due to the broadcast nature of wireless transmissions, ad-hoc systems contain a point-to- multipoint communication architecture, making it well suited to multi-path traffic. One such application is in multicasting, which sends data from one source to two (or more) destinations. But, due to the shared characteristics of the communication channel, such traffic may cause multiple contentions and collisions to occur, which will degrade the efficiency and performance of a protocol. This dissertation examines these different design tradeoffs through the use of a freely available simulation package, known as NS-2 (Network Simulator - version 2). In addition, a novel routing protocol, known as LAMP (Location Aided Multicasting Protocol), is developed to handle time-bounded audio information, which is employed in a network that consists of sixteen commercial handheld devices. LAMP utilizes a destination-sequenced, next-hop routing table to forward multicast data. Since mobility causes neighbouring nodes to continually change, next-hop links need to be periodically updated. But, between each update period, a next-hop link may become broken. Thus, if a packet is required to be routed, for which its' next-hop link is unknown, LAMP reverts to a localized location aided flood to find a path to that destination. However, since flooding causes network congestion, it is only employed when its' table forwarding scheme fails. Results have shown that LAMP improves packet delivery ratios by up to 5% over exisiting flood-limiting schemes: Furthermore, LAMP has been shown to be comparable to leading schemes, even when employed to route data to a single source-destination pair. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2003.