Wireless communication equipment is widely available and the number of transmitters and receivers keeps increasing. In addition to communications, wireless nodes can be used for sensing. This project is focuses on human presence detection in indoor environments using measurements such as CSI that can be extracted from radio receivers and labeled using a camera and AI computer vision techniques (YoLo framework). Our goal is to understand if a room is empty or has one or two people by utilizing machine learning algorithms. We have selected SVM (Support Vector Machines) and CNN (Convolutional Neural Networks). These methods will be evaluated in different scenarios such as different locations, bandwidths of 20, 40 and 120MHz, carrier frequencies of 2.4 and 5 GHz, high/low SNR values as well as different antenna configurations (MIMO, SIMO, SISO). Both methods perform very well for classification and specifically in case of CNN it performs better in low SNR compared to SVM. We found that some of the measurements seemed to be outliers and the clustering algorithm DBScan was used in order to identify them. Last but not least, we explore whether the radio can complement computer vision in presence detection since radio waves may propagate through walls and opaque obstacles. / Trådlös kommunikationsutrustning är allmänt tillgänglig och antalet sändare och mottagare fortsätter att öka. Förutom kommunikation kan trådlösa noder användas för avkänning. Detta projekt fokuserar på mänsklig närvarodetektering i inomhusmiljöer med hjälp av mätningar som CSI som kan extraheras från radiomottagare och märkas med hjälp av en kamera och AI datorseende tekniker (YoLo-ramverket). Vårt mål är att förstå om ett rum är tomt eller har en eller två personer genom att använda maskininlärningsalgoritmer. Vi har valt SVM och CNN. Dessa metoder kommer att utvärderas i olika scenarier såsom olika platser, bandbredder på 20, 40 och 120MHz, bärvågsfrekvenser på 2,4 och 5 GHz, höga/låga SNR-värden samt olika antennkonfigurationer (MIMO, SIMO, SISO). Båda metoderna fungerar mycket bra för klassificering och specifikt i fall av CNN presterar den bättre i låg SNR jämfört med SVM. Vi fann att några av mätningarna verkade vara extremvärden och klustringsalgoritmen DBScan användes för att identifiera dem. Sist men inte minst undersöker vi om radion kan komplettera datorseende vid närvarodetektering eftersom radiovågor kan fortplanta sig genom väggar och ogenomskinliga hinder.

The Quality of Service (QoS) routing approach is gaining an increasing interest in the Internet community due to the new emerging Internet applications such as real-time multimedia applications. These applications require better levels of quality of services than those supported by best effort networks. Therefore providing such services is crucial to many real time and multimedia applications which have strict quality of service requirements regarding bandwidth and timeliness of delivery. QoS routing is a major component in any QoS architecture and thus has been studied extensively in the literature. Scalability is considered one of the major issues in designing efficient QoS routing algorithms due to the high cost of QoS routing both in terms of computational effort and communication overhead. Localized quality of service routing is a promising approach to overcome the scalability problem of the conventional quality of service routing approach. The localized quality of service approach eliminates the communication overhead because it does not need the global network state information. The main aim of this thesis is to contribute towards the localised routing area by proposing and developing some new models and algorithms. Toward this goal we make the following major contributions. First, a scalable and efficient QoS routing algorithm based on a localised approach to QoS routing has been developed and evaluated. Second, we have developed a path selection technique that can be used with existing localized QoS routing algorithms to enhance their scalability and performance. Third, a scalable and efficient hierarchical QoS routing algorithm based on a localised approach to QoS routing has been developed and evaluated.

Call quality has been crucial from the inception of telecommunication networks. Operators need to monitor call quality from the end-user¿s perspective, in order to retain subscribers and reduce subscriber ¿churn¿. Operators worry not only about call quality and interconnect revenue loss, but also about network connectivity issues in areas where mobile network gateways are prevalent. Bandwidth quality as experienced by the end-user is equally important in helping operators to reduce churn. The parameters that network operators use to improve call quality are mainly from the end-user¿s perspective. These parameters are usually ASR (answer seizure ratio), PDD (postdial delay), NER (network efficiency ratio), the number of calls for which these parameters have been analyzed and successful calls. Operators use these parameters to evaluate and optimize the network to meet their quality requirements. Analysis of speech quality is a major arena for research. Traditionally, users¿ perception of speech quality has been measured offline using subjective listening tests. Such tests are, however, slow, tedious and costly. An alternative method is therefore needed; one that can be automatically computed on the subscriber¿s handset, be available to the operator as well as to subscribers and, at the same time, provide results that are comparable with conventional subjective scores. QMeter® ¿ a set of tools for signal and bandwidth measurement that have been developed bearing in mind all the parameters that influence call and bandwidth quality experienced by the end-user ¿ addresses these issues and, additionally, facilitates dynamic tariff propositions which enhance the credibility of the operator. This research focuses on call quality parameters from the end-user¿s perspective. The call parameters used in the research are signal strength, successful call rate, normal drop call rate, and hand-over drop rate. Signal strength is measured for every five milliseconds of an active call and average signal strength is calculated for each successful call. The successful call rate, normal drop rate and hand-over drop rate are used to achieve a measurement of the overall call quality. Call quality with respect to bundles of 10 calls is proposed. An attempt is made to visualize these parameters for better understanding of where the quality is bad, good and excellent. This will help operators, as well as user groups, to measure quality and coverage. Operators boast about their bandwidth but in reality, to know the locations where speed has to be improved, they need a tool that can effectively measure speed from the end-user¿s perspective. BM (bandwidth meter), a tool developed as a part of this research, measures the average speed of data sessions and stores the information for analysis at different locations. To address issues of quality in the subscriber segment, this research proposes the varying of tariffs based on call and bandwidth quality. Call charging based on call quality as perceived by the end-user is proposed, both to satisfy subscribers and help operators to improve customer satisfaction and increase average revenue per user. Tariff redemption procedures are put forward for bundles of 10 calls and 10 data sessions. In addition to the varying of tariffs, quality escalation processes are proposed. Deploying such tools on selected or random samples of users will result in substantial improvement in user loyalty which, in turn, will bring operational and economic advantages.

Kubernetes is a container orchestration system that has been rising in popularity in recent years. The modular nature of Kubernetes allows the usage of different storage solutions, and for cloud environments, cloud-native distributed storage solutions maybe attractive due to their redundant nature. There are many tools for cloud-native distributed storage available on the market today with differing features and performance. Choosing the correct one for an organisation can be difficult. Organisations utilising Kubernetes in cloud environments would like to be as performance efficient as possible to save on costs and resources. This study aims to offer a benchmark and analysis for some of the most popular tools, to help organisations choose the ‘best’ solution for their operational needs, from a performance perspective. The benchmarks compare three cloud-native distributed storage solutions, OpenEBS, Portworx, and Rook-Ceph on both Amazon Elastic Kubernetes Service (EKS) and Azure Kubernetes Service (AKS). For a baseline comparison, the study will also benchmark the cloud providers own solutions; Azure Disk Storage, and Amazon Elastic Block Storage. The study compares these solutions from three key metrics; bandwidth, latency, and IOPS, in both read and write performance. / <p>Det finns övrigt digitalt material (t.ex. film-, bild- eller ljudfiler) eller modeller/artefakter tillhörande examensarbetet som ska skickas till arkivet.</p><p>There are other digital material (eg film, image or audio files) or models/artifacts that belongs to the thesis and need to be archived.</p>

The angle of arrival estimation of multiple sources plays a vital role in the field of array signal processing as MIMO systems can be employed at both the transmitter and the receiver end and the system capacity, reliability and throughput can be significantly increased by using array signal processing. Almost all applications require accurate direction of arrival (DOA) estimation to localize the sources of the signals. Another important parameter of localization systems is the array geometry and sensor design which can be application specific and is used to estimate the DOA. In this work, various array geometries and arrival estimation algorithms are studied and then a new scheme for multiple source estimation is proposed and evaluated based on the performance of subspace and non-subspace decomposition methods. The proposed scheme has shown to outperform the conventional Multiple Signal Classification (MUSIC) estimation and Bartlett estimation techniques. The new scheme has a better performance advantage at low and high signal to noise ratio values (SNRs). The research work also studies different array geometries for both single and multiple incident sources and proposes a geometry which is cost effective and efficient for 3, 4, and 5 antenna array elements. This research also considers the shape of the ground plane and its effects on the angle of arrival estimation and in addition it shows how the mutual couplings between the elements effect the overall estimation and how this error can be minimised by using a decoupling matrix. At the end, a novel miniaturised multi element reconfigurable antenna to represent the receiver base station is designed and tested. The antenna radiation patterns in the azimuth angle are almost omni-directional with linear polarisation. The antenna geometry is uniplanar printed logspiral with striplines feeding network and biased components to improve the impedance bandwidth. The antenna provides the benefit of small size, and re-configurability and is very well suited for the asset tracking applications.

No / Fairness in bandwidth resource allocation is highly significance to the advancement of the future generation mobile and wireless technologies. It is likely that restriction of bandwidth due to the employment of some scheduling scheme would not be an appropriate option for the future development of communication systems. However, there is need to consider an implementation that would lead to good network performance and avoid unguaranteed bandwidth delivery. This paper focusses on evaluating the performance of Bandwidth Allocation using Dynamic Label Switching Paths (LSPs) Tunnelling and Label Distribution Protocol (LDP) signalling in Multi-Protocol Label Switching (MPLS) network. This will make provision for bandwidth allocation and reservation possible. An appropriate bandwidth allocation would have a positive impact on throughput as well as the delay. The results of an IP (Internet Protocol) Network without MPLS enabled is compared with MPLS model network. Furthermore, implementation of dynamic and static LSPs models are presented with about 75% decrease in packet delay variation for dynamic LSP when compared from static LSP. In addition, the models of bandwidth estimation, bandwidth allocation, delay and jitter are provided. Performance metrics used in this respect for multimedia services (Voice and Video conferencing) confirm that the modified models are improved in comparison with the baseline, having highest throughput of about 51% increment, and packet delay variation decreases drastically.

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