The dissertation addresses the communication issues of interconnecting the different government sectors LANs, and access to the global Internet. Network capacities are being purposely overengineered in today's commercial Internet. Any network provider, be it a commercial Internet Service Provider (ISP) or Information Technology Service department at government sector, company or university site, will design network bandwidth resources in such a way that there will be virtually no data loss, even during the worst possible network utilization scenario. Thus, the service delivered by today's end-to-end wide area Internet would be perfect if it wasn't for the inter-domain connections, such as Internet access link to the ISP, or peering points between ISPs. The thesis studies the performance of the network in Eritrea, displaying the problems of Local Area Networks (LANs) and Wide Area Networks (WAN) and suggesting initial solutions and investigating the performance of (WAN) through the measured traffic analysis between Asmara LAN and Massawa LAN, using queuing theory system (M/M/1 and M/M/2) solution. The dissertation also uses OPNET IT Guru simulation software program ·to study the performance of LAN and WLAN in Eritrea. The items studied include traffic, collision, packet loss, and queue delay. Finally in order to follow the current trends, we study the performance ofVOIP links in Eritrean WANs environment, with a focus on five different link capacities: 28 kbps, 33 kbps, 64 kbps, and 128 kbps for voice and 256/512 kbps for voice and data. Using the R value as a measure of mean opinion score (MOS), we determine that the 33 kbps link would be adequate for Eritrean WANs. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2006.

The Internet has experienced a tremendous growth over the past two decades and with that growth have come severe congestion problems. Research efforts to alleviate the congestion problem can broadly be classified into three groups: Cl) Router based congestion detection; (2) Generation and transmission of congestion notification signal to the traffic sources; (3) End-to-end algorithms which control the flow of traffic between the end hosts. This dissertation has largely addressed the first two groups which are basically router initiated. Router based congestion detection mechanisms, commonly known as Active Queue Management (AQM), can be classified into two groups: conventional mathematical analytical techniques and fuzzy logic based techniques. Research has shown that fuzzy logic techniques are more effective and robust compared to the conventional techniques because they do not rely on the availability of a precise mathematical model of Internet. They use linguistic knowledge and are, therefore, better placed to handle the complexities associated with the non-linearity and dynamics of the Internet. In spite of all these developments, there still exists ample room for improvement because, practically, there has been a slow deployment of AQM mechanisms. In the first part of this dissertation, we study the major AQM schemes in both the conventional and the fuzzy logic domain in order to uncover the problems that have hampered their deployment in practical implementations. Based on the findings from this study, we model the Internet congestion problem as a multi-objective problem. We propose a Fuzzy Logic Congestion Detection (FLCD) which synergistically combines the good characteristics of the fuzzy approaches with those of the conventional approaches. We design the membership functions (MFs) of the FLCD algorithm automatically by using Multi-objective Particle Swarm Optimization (MOPSO), a population based stochastic optimization algorithm. This enables the FLCD algorithm to achieve optimal performance on all the major objectives of Internet congestion control. The FLCD algorithm is compared with the basic Fuzzy Logic AQM and the Random Explicit Marking (REM) algorithms on a best effort network. Simulation results show that the FLCD algorithm provides high link utilization whilst maintaining lower jitter and packet loss. It also exhibits higher fairness and stability compared to its basic variant and REM. We extend this concept to Proportional Differentiated Services network environment where the FLCD algorithm outperforms the traditional Weighted RED algorithm. We also propose self learning and organization structures which enable the FLCD algorithm to achieve a more stable queue, lower packet losses and UDP traffic delay in dynamic traffic environments on both wired and wireless networks. In the second part of this dissertation, we present the congestion notification mechanisms which have been proposed for wired and satellite networks. We propose an FLCD based dual explicit congestion notification algorithm which combines the merits of the Explicit Congestion Notification (ECN) and the Backward Explicit Congestion Notification (BECN) mechanisms. In this proposal, the ECN mechanism is invoked based on the packet marking probability while the BECN mechanism is invoked based on the BECN parameter which helps to ensure that BECN is invoked only when congestion is severe. Motivated by the fact that TCP reacts to tbe congestion notification signal only once during a round trip time (RTT), we propose an RTT based BECN decay function. This reduces the invocation of the BECN mechanism and resultantly the generation of reverse traffic during an RTT. Compared to the traditional explicit notification mechanisms, simulation results show that the new approach exhibits lower packet loss rates and higher queue stability on wired networks. It also exhibits lower packet loss rates, higher good-put and link utilization on satellite networks. We also observe that the BECN decay function reduces reverse traffic significantly on both wired and satellite networks while ensuring that performance remains virtually the same as in the algorithm without BECN traffic reduction. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, 2006.

Wireless radio access technologies have been progressively evolving to meet the high data rate demands of consumers. The deployment and success of voice-based second generation networks were enabled through the use of the Global System for Mobile Communications (GSM) and the Interim Standard Code Division Multiple Access (lS-95 CDMA) networks. The rise of the high data rate third generation communication systems is realised by two potential wireless radio access networks, the Wideband Code Division Multiple Access (WCDMA) and the CDMA2000. These networks are based on the use of various types of codes to initiate, sustain and terminate the communication links. Moreover, different codes are used to separate the transmitting base stations. This dissertation focuses on base station identification aspects of the Frequency Division Duplex (FDD) WCDMA networks. Notwithstanding the ease of deployment of these networks, their asynchronous nature presents serious challenges to the designer of the receiver. One of the challenges is the identification of the base station identity by the receiver, a process called Cell Search. The receiver algorithms must therefore be robust to the hostile radio channel conditions, Doppler frequency shifts and the detrimental effects of carrier frequency offsets. The dissertation begins by discussing the structure and the generation of WCDMA base station data along with an examination of the effects of the carrier frequency offset. The various cell searching algorithms proposed in the literature are then discussed and a new algorithm that exploits the correlation length structure is proposed and the simulation results are presented. Another design challenge presented by WCDMA networks is the estimation of carrier frequency offset at the receiver. Carrier frequency offsets arise due to crystal oscillator inaccuracies at the receiver and their effect is realised when the voltage controlled oscillator at the receiver is not oscillating at the same carrier frequency as that of the transmitter. This leads to a decrease in the receiver acquisition performance. The carrier frequency offset has to be estimated and corrected before the decoding process can commence. There are different approaches in the literature to estimate and correct these offsets. The final part of the dissertation investigates the FFT based carrier frequency estimation techniques and presents a new method that reduces the estimation error. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2006.

The use of error-correction coding techniques in communication systems has become extremely imperative. Due to the heavy constraints faced by systems engineers more attention has been given to developing codes that converge closer to the Shannon theoretical limit. Turbo codes exhibit a performance a few tenths of a decibel from the theoretical limit and has motivated a lot of good research in the channel coding area in recent years. In the under-mentioned dissertation, motivated by turbo codes, we study the use of three new error-correction coding schemes: Repeat-Punctured Superorthogonal Convolutional Turbo Codes, Dual-Repeat-Punctured Turbo Codes and Dual-Repeat-Punctured Superorthogonal Convolutional Turbo Codes, applied to the additive white Gaussian noise channel and the frequency non-selective or flat Rayleigh fading channel. The performance of turbo codes has been shown to be near the theoretical limit in the AWGN channel. By using orthogonal signaling, which allows for bandwidth expansion, the performance of the turbo coding scheme can be improved even further. Since the resultant is a low-rate code, the code is mainly suitable for spread-spectrum modulation applications. In conventional turbo codes the frame length is set equal to the interleaver size; however, the codeword distance spectrum of turbo codes improves with an increasing interleaver size. It has been reported that the performance of turbo codes can be improved by using repetition and puncturing. Repeat-punctured turbo codes have shown a significant increase in performance at moderate to high signal-to-noise ratios. In this thesis, we study the use of orthogonal signaling and parallel concatenation together with repetition (dual and single) and puncturing, to improve the performance of the superorthogonal convolutional turbo code and the conventional turbo code for reliable and effective communications. During this research, three new coding schemes were adapted from the conventional turbo code; a method to evaluate the union bounds for the AWGN channel and flat Rayleigh fading channel was also established together with a technique for the weight-spectrum evaluation. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2007.

WiFi can be used to provide cost-effective last-mile IP connectivity to rural users. In initial rollout, hotspots or hotzones can be positioned at community centres such as schools, clinics, hospitals or call-centres. The research will investigate maximizing coverage using physical and higher layer techniques. The study will consider a typical South African rural region, with telecommunications services traffic estimates. The study will compare several IEEE 802.11 deployment options based on the requirements of the South African case in order to recommend options that improve performance. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2007.

The central focus of this dissertation is mobile ad hoc networks (MANETs) and their security. MANETs are autonomous networks of wireless nodes connected in an ad hoc manner, and have unique characteristics that make them difficult to secure. The principal aims of this investigation are to discuss the research and evaluation of existing mechanisms to secure MANETs and to design the implementation of a unique security mechanism. Key management is a major challenge in these networks due to the lack of fixed network infrastructure. In presenting a survey of the existing key management solutions for MANETs, the findings indicate that most security attacks target the network layer and more specifically the routing protocol. Consequently, the provision of secure routes is a vital element for trust establishment, and accordingly a survey is provided of the existing secure ad hoc routing protocols. The observation is made that most secure ad hoc routing protocols assume the existence of a key management system to certify, authenticate, and distribute keying information. Mobile ad hoc networks cannot assume the existence of a centralized authority member to perform key management tasks, and the problem of key management must be addressed. A novel key management solution called Direct Indirect Trust Distribution (DITD) is proposed for an on-demand ad hoc routing protocol. The solution includes a trust evaluation mechanism and a key distribution scheme to distribute keying information in the form of certificates. The key distribution scheme performs localized certificate exchanges following the routing procedure. A security evaluation metric is proposed that aggregates trust along a path based on a security metric and the path distance. The proposed solution is implemented on a modified AODV routing protocol, and simulated on the ns2 Network Simulator. Simulations are conducted in order to compare the performance of the AODV and DITD protocols. The simulation results show that the DITD model provides key distribution and trust path selection with minimal effect on the routing agent. The findings of the investigation confirm that DITD can be used as a basis for the operation of existing security protocols requiring a secure key distribution mechanism. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2010.

Increase in technological advancements in fields of telecommunications, computers and television have prompted the need to exchange video, image and audio files between people. Transmission of such files finds numerous multimedia applications such as, internet multimedia, video conferencing, videophone, etc. However, the transmission and rece-ption of these files are limited by the available bandwidth as well as storage capacities of systems. Thus there is a need to develop compression systems, such that required multimedia applications can operate within these limited capacities. This dissertation presents two well established coding approaches that are used in modern' image and video compression systems. These are the wavelet and fractal methods. The wavelet based coder, which adopts the transform coding paradigm, performs the discrete wavelet transform on an image before any compression algorithms are implemented. The wavelet transform provides good energy compaction and decorrelating properties that make it suited for compression. Fractal compression systems on the other hand differ from the traditional transform coders. These algorithms are based on the theory of iterated function systems and take advantage of local self-similarities present in images. In this dissertation, we first review the theoretical foundations of both wavelet and fractal coders. Thereafter we evaluate different wavelet and fractal based compression algorithms, and assess the strengths and weakness in each case. Due to the short-comings of fractal based compression schemes, such as the tiling effect appearing in reconstructed images, a wavelet based analysis of fractal image compression is presented. This is the link that produces fractal coding in the wavelet domain, and presents a hybrid coding scheme called fractal-wavelet coders. We show that by using smooth wavelet basis in computing the wavelet transform, the tiling effect of fractal systems can be removed. The few wavelet-fractal coders that have been proposed in literature are discussed, showing advantages over the traditional fractal coders. This dissertation will present a new low-bit rate video compression system that is based on fractal coding in the wavelet domain. This coder makes use of the advantages of both the wavelet and fractal coders discussed in their review. The self-similarity property of fractal coders exploits the high spatial and temporal correlation between video frames. Thus the fractal coding step gives an approximate representation of the coded frame, while the wavelet technique adds detail to the frame. In this proposed scheme, each frame is decomposed using the pyramidal multi-resolution wavelet transform. Thereafter a motion detection operation is used in which the subtrees are partitioned into motion and non-motion subtrees. The nonmotion subtrees are easily coded by a binary decision, whereas the moving ones are coded using the combination of the wavelet SPIHT and fractal variable subtree size coding scheme. All intra-frame compression is performed using the SPIHT compression algorithm and inter-frame using the fractal-wavelet method described above. The proposed coder is then compared to current low bit-rate video coding standards such as the H.263+ and MPEG-4 coders through analysis and simulations. Results show that the proposed coder is competitive with the current standards, with a performance improvement been shown in video sequences that do not posses large global motion. Finally, a real-time implementation of the proposed algorithm is performed on a digital signal processor. This illustrates the suitability of the proposed coder being applied to numerous multimedia applications. / Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2004.

Transmission of digital information through a wireless channel with resolvable multipaths or a bandwidth limited channel results in intersymbol interference (1SI) among a number of adjacent symbols. The design of an equalizer is thus important to combat the ISI problem for these types of channels and hence provides reliable communication. Channel coding is used to provide reliable data transmission by adding controlled redundancy to the data. Turbo equalization (TE) is the joint design of channel coding and equalization to approach the achievable uniform input information rate of an ISI channel. The main focus of this dissertation is to investigate the different TE techniques used for a static frequency selective additive white Gaussian noise (AWGN) channel. The extrinsic information transfer (EXIT) chart is used to analyse the iterative equalization/decoding process and to determine the minimum signal to noise ratio (SNR) in order to achieve convergence. The use of the Minimum Mean Square Error (MMSE) Linear Equalizer (LE) using a priori information has been shown to achieve the same performance compared with the optimal trellis based Maximum A Posterior (MAP) equalizer for long block lengths. Motivated by improving the performance of the MMSE LE, two equalization schemes are initially proposed: the MMSE Linear Equalizer with Extrinsic information Feedback (LE-EF (1) and (U)). A general structure for the MMSE LE, MMSE Decision Feedback Equalizer (DFE) and two MMSE LE-EF receivers, using a priori information is also presented. The EXIT chart is used to analyse the two proposed equalizers and their characteristics are compared to the existing MAP equalizer, MMSE LE and MMSE DFE. It is shown that the proposed MMSE LE-EF (1) does have an improved performance compared with the existing MMSE LE and approaches the MMSE Linear Equalizer with Perfect Extrinsic information Feedback (LE-PEF) only after a large number of iterations. For this reason the MMSE LE-EF is shown to suffer from the error propagation problem during the early iterations. A novel way to reduce the error propagation problem is proposed to further improve the performance of the MMSE LE-EF (I). The MAP equalizer was shown to offer a much improved performance over the MMSE equalizers, especially during the initial iterations. Motivated by using the good quality of the MAP equalizer during the early iterations and the hybrid MAP/MMSE LE-EF (l) is proposed in order to suppress the error propagation problem inherent in the MMSE LE-EF (I). The EXIT chart analysis reveals that the hybrid MAP/MMSE LE-EF (l) requires fewer iterations in order to achieve convergence relative to the MMSE LE-EF (l). Simulation results demonstrate that the hybrid MAP/MMSE LE-EF (I) has a superior performance compared to the MMSE LE-EF (I) as well as approaches the performance of both the MAP equalizer and MMSE LE-PEF at high SNRs, at the cost of increased complexity relative to the MMSE LEEF (I) receiver. The final part of this dissertation considers the use of precoders in a TE system. It was shown in the literature that a precoder drastically improves the system performance. Motivated by this, the EXIT chart is used to analyse the characteristics of four different precoders for long block lengths. It was shown that using a precoder results in a loss in mutual information during the initial equalization stage. However" we show by analysis and simulations that this phenomenon is not observed in the equalization of all precoded channels. The slope of the transfer function, relating to the MAP equalization of a precoded ISI channel (MEP), during the high input mutual information values is shown to play an important role in determining the convergence of precoded TE systems. Simulation results are presented to show how the precoders' weight affects the convergence of TE systems. The design of the hybrid MAP/MEP equalizer is also proposed. We also show that the EXIT chart can be used to compute the trellis code capacity of a precoded ISI channel. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2003.

Third generation cellular communication systems based on CDMA techniques have shown great scope for improvement in system capacity. Over the last decade, there has been significant interest in DS-CDMA detectors. The conventional detector, the optimal detector and a number of sub-optimal multi-user detectors (MUD) have been extensively analyzed in the literature. Recently, the reduction of power consumption in DS-CDMA systems has also become another important consideration in both system design and in implementation. In order to support wireless multimedia services, all CDMA-based systems for third generation systems have a large bandwidth and a high data rate, therefore the power consumed by the digital signal processor (DSP) is high. This thesis focuses on power consumption in the adaptive Minimum Mean Square Error (MMSE) detector which is based on the Least Mean Square (LMS) algorithm. This thesis presents a literature survey on MUD and adaptive filter algorithms. A system model of the quantized LMS-MMSE MUD is proposed and its performance is analyzed. The quantization effects in the finite precision LMS-MMSE adaptive MUD including the steady-state weight covariance, mean square error (MSE) and bit error rate (BER) versus wordlength of data and coefficient are investigated when both the data and filter coefficients are quantized. The effects of wordlength size on power consumption are investigated and the tradeoff between the power consumption and performance degradation and the optimal allocation of bits to data and to LMS coefficients under power constraint is presented. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2003.

Third generation wireless systems have adopted Direct-Sequence/Code-Division Multiple-Access (DS/CDMA) as the multiple access scheme of communication. This system would typically operate in a multipath fading channel. This dissertation only deals with the task of channel estimation at the base station where the multipath delays and attenuations for each user are estimated. This information is used to aid the recovery of data that was transmitted by each user. Subspace-based algorithms are popularly used to perform the task of channel estimation because they have the desirable property of perfectly estimating the channel in a noise-free environment. In this dissertation a new subspace-based channel estimation algorithm for DS/CDMA systems is presented. The proposed algorithm is based on the Parametric Subspace algorithm by Perros-Meilhac et al. for single-user systems. The main focus of this dissertation is to convert the Parametric Subspace algorithm from a single-user system to a multi-user DS/CDMA system. It has been shown in the literature that by using information of the pulse-shaping filter in the Channel Subspace algorithm, the variance of the channel estimates is decreased. However, this has only been applied to a single-user system. There are several subspace algorithms that have been proposed for DS/CDMA systems. Most of these algorithms sample the received signal at the chip rate, making it impossible to exploit knowledge of the pulse-shaping filter in the channel estimation algorithm. In this dissertation a new subspace-based channel estimation algorithm is derived for a DS/CDMA system with multiple receive antennas, where the output is oversampled with respect to the chip rate. By oversampling the received signal, knowledge of the pulse-shaping filter is used in the channel estimation algorithm. It is shown that the variance of the channel estimate for the proposed subspace algorithm is less than the Torlak/Xu subspace algorithm that does not exploit information of the pulse-shaping filter. A mathematical expression of the mean square error of estimation for the new algorithm is also derived. It was shown that the analytic expression provides a good approximation of the actual MSE for high SNR. The Parametric Subspace Delay Estimation (PSDE) algorithm was developed by Perros-Meilhac et al. to estimate the multipath delays introduced by the communications channel. The limitation of the PSDE algorithm is that the performance of the algorithm deteriorates as the power of the multipath signals decrease with increasing delay time. This dissertation proposes a modified version of the PSDE algorithm, called the Modified Parametric Subspace Delay Estimation (MPSDE) algorithm, which performs better than the PSDE algorithm in an environment where the power of the multipath signals varies. The final part of this dissertation discusses the Torlak/Xu channel estimation algorithm and the Bensley/Aazbang delay estimation algorithm. In order to compare the performance of these two subspace algorithms, the Torlak/Xu algorithm is converted to a delay estimation algorithm that is called the Parametric TX algorithm. The performance of the Bensley/Aazbang delay estimation algorithm and the proposed Parametric TX algorithm are compared and it is shown that the Parametric TX algorithm offers the better performance. / Thesis (M.Sc.Eng.)-University of Natal, Durban, 2003.