Tesis: "Control over networks"

1

Ji, Kun. "Real-time control over networks". Texas A&M University, 2003. http://hdl.handle.net/1969.1/5834.

Resumen

A control system in which sensors, actuators, and controllers are interconnected over a communication network is called a networked control system (NCS). Enhanced computational capabilities and bandwidths in the networking technology enabled researchers to develop NCSs to implement distributed control schemes. This dissertation presents a framework for the modeling, design, stability analysis, control, and bandwidth allocation of real-time control over networks. This framework covers key research issues regarding control over networks and can be the guidelines of NCS design. A single actuator ball magnetic-levitation (maglev) system is implemented as a test bed for the real-time control over networks to illustrate and verify the theoretical results of this dissertation. Experimentally verifying the feasibility of Internet-based real-time control is another main objective of this dissertation. First, this dissertation proposes a novel NCS model in which the effects of the networkinduced time delay, data-packet loss, and out-of-order data transmission are all considered. Second, two simple algorithms based on model-estimator and predictor- and timeout-scheme are proposed to compensate for the network-induced time delay and packet loss simultaneously. These algorithms are verified experimentally by the ball maglev test bed. System stability analyses of original and compensated systems are presented. Then, a novel co-design consideration related to real-time control and network communication is also proposed. The working range of the sampling frequency is determined by the analysis of the system stability and network parameters such as time delay, data rate, and data-packet size. The NCS design chart developed in this dissertation can be a useful guideline for choosing the network and control parameters in the design of an NCS. Using a real-time operating system for real-time control over networks is also proposed as one of the main contributions of this dissertation. After a real-time NCS is successfully implemented, advanced control theories such as robust control, optimal control, and adaptive control are applied and formulated to improve the quality of control (QoC) of NCSs. Finally, an optimal dynamic bandwidth management method is proposed to solve the optimal network scheduling and bandwidth allocation problem when NCSs are connected to the same network and are sharing the network resource.

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2

Abebe, Zelalem Teffera. "Process Control over Wireless Sensor Networks". Thesis, KTH, Reglerteknik, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-133584.

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A signicant growth was witnessed in the led of Wireless Sensor Networks (WSNs), the previous decade. Advances in hardware miniaturization coupled with increased processing capabilities and memory capacity have extended the application domains of WSNs. In light of this, standardization organizations led by academia and industries initiated activities for the design of protocols such as IEEE 802.15.4 and IETF RPL (Routing Protocol for Low power and Lossy Networks). IEEE 802.15.4 denes physical and media access layers for WSNs while IETF RPL denes the functionality of the routing layer. This thesis investigates research issues in wireless sensor networks and network controlled systems that control micro-biological processes for water treatment plants. By choosing a process model that can relate to an industrial process, feasibility of control over IEEE 802.15.4 and RPL protocols is evaluated for stability with regards to network delay and packet loss. Settling time and overshoot are measured to indicate control performance. Control messages related to routing and routing table lengths are measured to indicate network stability and scalability. The system model used is a centralized discrete controller controlling a thermal processes running on the sensors. This model is chosen for representing wide industrial networked control systems while adding a WSN dimension based on IEEE 802.15.4 and RPL. The main contribution of this thesis is an experimental study where both the network and controller performance is validated while utilizing commercial o-theshelf sensor platforms. The results from this experimental work include rst the use of established theorems for analyzing control using WSNs. Moreover, the ability of IEEE 802.15.4 and RPL to provide stable communication that is reliable enough for actual industrial control implementation is validated.

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3

Zaidi, Ali Abbas. "Sensing and Control of Dynamical Systems Over Networks". Doctoral thesis, KTH, Kommunikationsteori, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-133337.

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Rapid advances in sensing, computing, and wireless technologies have led to significant interest in the understanding and development of wireless networked control systems. Networked control systems consist of spatially distributed agents such as plants (dynamical systems), sensors, and controllers, that interact to achieve desired objectives. The sensors monitor the plants and communicate measurements to remotely situated controllers. The controllers apply actions to stabilize and control the plants. Such systems have diverse applications in security, surveillance, industrial production, health monitoring, remote surgery, environment management, space missions, and intelligent transport systems. The objective of the thesis is to understand the fundamental limits and principles involved in the design of sensing and control strategies for dynamical systems controlled over communication networks. The thesis has three parts. Part I and Part III consider the design of sensing and control strategies for mean-square stabilization of linear dynamical systems over fundamental communication channels such as point-to-point, relay, multiple-access, broadcast, and interference channels. The sensors and other nodes within the communication network are assumed to have average power transmit constraints. Moreover, the communication links between all agents (plants, sensors, controllers) are modeled as Gaussian channels. Necessary as well as sufficient conditions for mean-square stabilization over various network topologies are derived. The necessary conditions are arrived at using information theoretic arguments such as properties of mutual information, directed information, and differential entropy. The sufficient conditions are obtained using delay-free sensing and control policies. These conditions quantify the effect of communication network parameters such as transmit powers, channel noise, and channel interference on the stability of the plant(s). Different settings where linear policies are optimal, asymptotically optimal (in certain parameters of the system) and suboptimal have also been identified. Part II considers the design of real-time sensing and control strategies for minimization of a quadratic cost function of the state process of a system over Gaussian networks. Two fundamental Gaussian networks are considered: i) cascade network and ii) parallel network. For each network, non-linear sensing and control schemes are proposed and sub-optimality of linear strategies is discussed. The results reveal fundamental limits on the performance of linear systems controlled over Gaussian networks. The methods used to derive these results reveal a close interplay between information theory and control theory.

QC 20131031

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4

梁鉅輝 y Kui-fai Leung. "Enhancing transmission control protocol performance over wireless networks". Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2002. http://hub.hku.hk/bib/B31227570.

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5

Ekman, Rasmus. "Automobile Control Systems : Transition from Controller Area Networks to Ethernets". Thesis, KTH, Radio Systems Laboratory (RS Lab), 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-148052.

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Due to concerns about the negative impacts of powering vehicles using fossil fuel and the future availability of fossil fuel, there has been an increased focus on electric vehicles. However, current electric vehicle energy efficiency is a key problem as these vehicles are not as efficient as fossil-fueled vehicles. One way of decreasing a vehicle’s energy consumption is to reduce the weight of the vehicle, while still ensuring the safety and reliability of the vehicle. Controller Area Network (CAN) systems have been used in vehicles to realize real-time applications, however the low peak data rates of CAN have begun to limit the applications that can be realized. This bachelor’s thesis project focuses on secure communication within a vehicle using Ethernet. Additionally, the use of Power over Ethernet can be used for powering some of the network attached devices within the vehicle. The goal is to reduce the number of components and the weight of the vehicle while continuing to ensure the security and reliability of the communication – even when the network grows in size (either in physical size or in number of connected devices). This thesis shows that an Ethernet based system can serve as a possible replacement candidate for the CAN system due to its low latencies and high bandwidth. Ethernet is also a very scalable system with none of the limitations that a CAN system have.
Den negativa påverkan av fossila bränslen har de senaste årtionden haft en negativ på planeten, mängden fossila bränslen över världen konsumeras även i en högre takt än vad som produceras. Därför har fokusen för att finna förnybara energi källor som både är effektiva och inte påverkar miljön på ett negativt sätt ökat. Därför är elbilar en viktig del i konverteringen av enheter som drivs av fossila bränslen till förnybara energikällor. Ett av problemen i en elbil är att energi konsumptionen är inte lika effektiv som fossila bränslen inom bil industrin. Ett sätt att sänka energi konsumptionen är att minska mängden komponenter inom en bil för att minska på vikten, utan att påverka säkerheten och tillförlitligheten. Tidigare har man använt sig av ett CAN system för att försäkra sig om systemet fungerar felfritt i realtid, problematiken med detta system är att när nätverket ökar i storlek så sätter de fysiska begränsningarna av detta system stop för den garanterade säkerheten. Detta kandidatexamensarbete kommer att fokusera på den interna kommunikationen i en elbil med hjälp av ett ethernet baserat kommunkations system över CAN systemet. Power over Ethernet tekinken kommer att tillämpas för de systemen som kan drivas av detta system. Målet är att reducera antalet komponenter som behövs och att garantera säkerheten och tillförlitligheten av den interna kommunikationen när nätverket av komponenter ökar i storlek. Det här kandidatarbetet visar att Ethernet kan ersätta det nuvarande CAN systemet ef-tersom att Ethernet erbjuder låga latenser och hög bandbredd. Detta arbete visar även att Et-hernet är väldigt skalbart och har inte begränsingarna som ett CAN system har.

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6

Li, Yuliang. "Congestion control for scalable video transmission over IP networks". Thesis, University of Bristol, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.441312.

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7

Chotikapong, Yotsapak. "TCP/IP and ATM over LEO satellite networks". Thesis, University of Surrey, 2000. http://epubs.surrey.ac.uk/638/.

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8

Shi, Xiaohan. "A reliable real-time transport protocol for networked control systems over wireless networks". Thesis, Queensland University of Technology, 2012. https://eprints.qut.edu.au/60160/1/Xiaohan_Shi_Thesis.pdf.

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Deploying wireless networks in networked control systems (NCSs) has become more and more popular during the last few years. As a typical type of real-time control systems, an NCS is sensitive to long and nondeterministic time delay and packet losses. However, the nature of the wireless channel has the potential to degrade the performance of NCS networks in many aspects, particularly in time delay and packet losses. Transport layer protocols could play an important role in providing both reliable and fast transmission service to fulfill NCS’s real-time transmission requirements. Unfortunately, none of the existing transport protocols, including the Transport Control Protocol (TCP) and the User Datagram Protocol (UDP), was designed for real-time control applications. Moreover, periodic data and sporadic data are two types of real-time data traffic with different priorities in an NCS. Due to the lack of support for prioritized transmission service, the real-time performance for periodic and sporadic data in an NCS network is often degraded significantly, particularly under congested network conditions. To address these problems, a new transport layer protocol called Reliable Real-Time Transport Protocol (RRTTP) is proposed in this thesis. As a UDP-based protocol, RRTTP inherits UDP’s simplicity and fast transmission features. To improve the reliability, a retransmission and an acknowledgement mechanism are designed in RRTTP to compensate for packet losses. They are able to avoid unnecessary retransmission of the out-of-date packets in NCSs, and collisions are unlikely to happen, and small transmission delay can be achieved. Moreover, a prioritized transmission mechanism is also designed in RRTTP to improve the real-time performance of NCS networks under congested traffic conditions. Furthermore, the proposed RRTTP is implemented in the Network Simulator 2 for comprehensive simulations. The simulation results demonstrate that RRTTP outperforms TCP and UDP in terms of real-time transmissions in an NCS over wireless networks.

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9

Alnuem, M. A. "Improving TCP performance over heterogeneous networks : the investigation and design of End to End techniques for improving TCP performance for transmission errors over heterogeneous data networks". Thesis, University of Bradford, 2009. http://hdl.handle.net/10454/3347.

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Transmission Control Protocol (TCP) is considered one of the most important protocols in the Internet. An important mechanism in TCP is the congestion control mechanism which controls TCP sending rate and makes TCP react to congestion signals. Nowadays in heterogeneous networks, TCP may work in networks with some links that have lossy nature (wireless networks for example). TCP treats all packet loss as if they were due to congestion. Consequently, when used in networks that have lossy links, TCP reduces sending rate aggressively when there are transmission (non-congestion) errors in an uncongested network. One solution to the problem is to discriminate between errors; to deal with congestion errors by reducing TCP sending rate and use other actions for transmission errors. In this work we investigate the problem and propose a solution using an end-to-end error discriminator. The error discriminator will improve the current congestion window mechanism in TCP and decide when to cut and how much to cut the congestion window. We have identified three areas where TCP interacts with drops: congestion window update mechanism, retransmission mechanism and timeout mechanism. All of these mechanisms are part of the TCP congestion control mechanism. We propose changes to each of these mechanisms in order to allow TCP to cope with transmission errors. We propose a new TCP congestion window action (CWA) for transmission errors by delaying the window cut decision until TCP receives all duplicate acknowledgments for a given window of data (packets in flight). This will give TCP a clear image about the number of drops from this window. The congestion window size is then reduced only by number of dropped packets. Also, we propose a safety mechanism to prevent this algorithm from causing congestion to the network by using an extra congestion window threshold (tthresh) in order to save the safe area where there are no drops of any kind. The second algorithm is a new retransmission action to deal with multiple drops from the same window. This multiple drops action (MDA) will prevent TCP from falling into consecutive timeout events by resending all dropped packets from the same window. A third algorithm is used to calculate a new back-off policy for TCP retransmission timeout based on the network's available bandwidth. This new retransmission timeout action (RTA) helps relating the length of the timeout event with current network conditions, especially with heavy transmission error rates. The three algorithms have been combined and incorporated into a delay based error discriminator. The improvement of the new algorithm is measured along with the impact on the network in terms of congestion drop rate, end-to-end delay, average queue size and fairness of sharing the bottleneck bandwidth. The results show that the proposed error discriminator along with the new actions toward transmission errors has increased the performance of TCP. At the same time it has reduced the load on the network compared to existing error discriminators. Also, the proposed error discriminator has managed to deliver excellent fairness values for sharing the bottleneck bandwidth. Finally improvements to the basic error discriminator have been proposed by using the multiple drops action (MDA) for both transmission and congestion errors. The results showed improvements in the performance as well as decreases in the congestion loss rates when compared to a similar error discriminator.

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10

Pesonen, Joonas. "Stochastic Estimation and Control over WirelessHART Networks: Theory and Implementation". Thesis, KTH, Reglerteknik, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-105172.

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There is currently a high interest of replacing traditional wired networks with wireless technology. Wireless communications can provide several advantages for process industries with aspect to exibility, maintenance and installation. The WirelessHART protocol provides a standardized wireless technology for large automation networks that explore wireless communication. However, wireless networks introduce time delays and losses in the communication system, which denes requirements for designing estimators and controllers that can tolerate and compensate for the losses and delays. This thesis consists of several contributions. First, we develop tools for analyzing the delay and loss probabilities in WirelessHART networks with unreliable transmission links. For given network topology, routing and transmission schedule the developed tools can be used to determine the latency distributions of individual packets and quantify that a packet will arrive within a prescribed deadline. Secondly, we consider estimation and control when sensor and control messages are sent over WirelessHART networks. The network losses and latencies are modelled and compensated for by timevarying Kalman lters and LQG controllers. Both optimal controllers, of high implementation complexity, and simple suboptimal schemes are considered. The control strategies are evaluated on a simulation model of a flotation process in a Boliden mine where the wired sensors of the existent solution are replaced by a WirelessHART network scheduled for time-optimal data collection. Finally, we implement a WirelessHART-compliant sensor on a Tmote sky device and perform real experiments of wireless control on a water tank process.

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11

Li, Ming. "Resource discovery and fair intelligent admission control over scalable Internet /". Electronic version, 2004. http://adt.lib.uts.edu.au/public/adt-NTSM20050314.180037/index.html.

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12

Kang, Intae. "Topology control for broadcasting over energy constrained wireless ad hoc networks /". Thesis, Connect to this title online; UW restricted, 2005. http://hdl.handle.net/1773/6061.

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13

Alheraish, Abdulmohsen Abdulrahman. "Reliable VBR video communications using error control techniques over ATM networks". Thesis, University of Strathclyde, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.248517.

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14

Perera, G. M. Ryan. "Error control for compressed video transmission over next generation mobile networks". Thesis, University of Surrey, 2018. http://epubs.surrey.ac.uk/844961/.

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Video data claims a significant portion of global mobile data communications, currently standing at 55%. This demand outburst has been due to exceptional display technologies, on-demand video, gaming and live video streaming, to name a few. Despite the massive data rates supported by modern mobile communication technologies, video data is starting to overload mobile networks. This is particularly true in links with low connectivity, where repeat requests flood the system. As solutions for this inevitable demand growth, in addition to efficient video compression methods, more video data error resilience must be sought. One reason video traffic is vulnerable to channel errors is the method it is treated at transmission; treatment as any other generic data type. Video is a unique data type because its ultimate user is not a machine but a human, and the contents within the data are interdependent on each other. Based on its properties, video compression, transmission methodology, and the decoding function must be adapted. By considering video communication as a collaborative effort of these three functions, error resilience can be effectively implemented. Analysis of radio resources available for data transmission in a multipath fading channel reveals that some resources are more robust than others. In the first contribution of this thesis, this characteristic is utilised to impose more resilience to more sensitive data within the video. Reliable means of forecasting the relative robustness of each radio resource are designed. Criteria for identifying the sensitivity of different video data segments are formulated. Finally, a technology is presented to map data to radio resources such that maximum received video quality is achieved. While the focus of the first contribution was on harmonising the transmission methods with the features of the compressed video payload, the second contribution takes an alternative route to error resilience by focusing on the decoder. The compressed video payload entails some identifiable syntax elements, some of which follow a predictable pattern. This feature is exploited to improve error recovery at an iterative turbo decoder. An algorithm to identify the video frame boundaries in corrupted compressed sequences is formulated, along with algorithms to deduce the correct values for selected fields in the compressed stream. Modifying the turbo extrinsic information using these corrections act as reinforcements in the turbo decoding iterative process. Most communication protocols transmit data as blocks in an ordered sequence and await the acknowledgement of the receiver to determine the next block to be transmitted. This gives rise to latency issues and the overloading of the network when the link connectivity is poor. A solution is presented for video data in the final contribution, where the concept of a digital fountain is hired. A two-dimensional forward error correction strategy is introduced for a digital fountain, where first, the video payload is LDPC encoded and then turbo encoded. A joint decoding strategy is designed between the turbo decoder and the LDPC decoder to recover the video data in an iterative manner. Taken together, these contributions are solutions for the video data burden on mobile networks; solutions which reduce the necessity for re-transmissions. The presented error resilience techniques are updates to the existing transmission methodology and the decoding function. They explore a new paradigm of improving coverage and channel throughput.

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15

Abdallah, AbouSheaisha Abdallah Sabry. "Cross-layer Control for Adaptive Video Streaming over Wireless Access Networks". Diss., Virginia Tech, 2016. http://hdl.handle.net/10919/78844.

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Over the last decade, the wide deployment of wireless access technologies (e.g. WiFi, 3G, and LTE) and the remarkable growth in the volume of streaming video content have significantly altered the telecommunications field. These developments introduce new challenges to the research community including the need to develop new solutions (e.g. traffic models and transport protocols) to address changing traffic patterns and the characteristics of wireless links and the need for new evaluation methods that generate higher fidelity results under more realistic scenarios. Unfortunately, for the last two decades, simulation studies have been the main tool for researchers in wireless networks. In spite of the advantages of simulation studies, overall they have had a negative influence on the credibility of published results. In partial response to this simulation crisis, the research community has adopted testing and evaluation using implementation-based experiments. Implementation-based experiments include field experiments, prototypes, emulations, and testbeds. An example of an implementation-based experiment is the MANIAC Challenge, a wireless networking competition that we designed and hosted, which included creation and operation of ad hoc networks using commodity hardware. However, the lack of software tools to facilitate these sorts of experiments has created new challenges. Currently, researchers must practice kernel programming in order to implement networking experiments, and there is an urgent need to lower the barriers of entry to wireless network experimentation. With respect to the growth in video traffic over wireless networks, the main challenge is a mismatch between the design concepts of current internet protocols (e.g. the Transport Control Protocol (TCP)) and the reality of modern wireless networks and streaming video techniques. Internet protocols were designed to be deployed over wired networks and often perform poorly over wireless links; video encoding is highly loss tolerant and delay-constrained and yet, for reasons of expedience is carried using protocols that emphasize reliable delivery at the cost of potentially high delay. This dissertation addresses the lack of software tools to support implementation-based networking experiments and the need to improve the performance of video streaming over wireless access networks. We propose a new software tool that allows researchers to implement experiments without a need to become kernel programmers. The new tool, called the Flexible Internetwork Stack (FINS) Framework, is available under an open source license. With our tool, researchers can implement new network layers, protocols, and algorithms, and redesign the interconnections between the protocols. It offers logging and monitoring capabilities as well as dynamic reconfigurability of the modules' attributes and interconnections during runtime. We present details regarding the architecture, design, and implementation of the FINS Framework and provide an assessment of the framework including both qualitative and quantitative comparison with significant previous tools. We also address the problem of HTTP-based adaptive video streaming (HAVS) over WiFi access networks. We focus on the negative influence of wireless last-hop connections on network utilization and the end-user quality of experience (QoE). We use a cross-layer approach to design three controllers. The first and second controllers adopt a heuristic cross-layer design, while the third controller formulates the HAVS problem as a Markov decision process (MDP). By solving the model using reinforcement learning, we achieved 20% performance improvement (after enough training) with respect to the performance of the best heuristic controller under unstable channel conditions. Our simulation results are backed by a system prototype using the FINS Framework. Although it may seem predictable to achieve more gain in performance and in QoE by using cross-layer design, this dissertation not only presents a new technique that improves performance, but also suggests that it is time to move cross-layer and machine-learning-based approaches from the research field to actual deployment. It is time to move cognitive network techniques from the simulation environment to real world implementations.
Ph. D.

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16

Sandberg, Henrik, Maben Rabi, Mikael Skoglund y Karl Henrik Johansson. "Estimation over heterogeneous sensor networks". KTH, ACCESS Linnaeus Centre, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-30325.

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Design trade-offs between estimation performance, processing delay and communication cost for a sensor scheduling problem is discussed. We consider a heterogeneous sensor network with two types of sensors: the first type has low-quality measurements, small processing delay and a light communication cost, while the second type is of high quality, but imposes a large processing delay and a high communication cost. Such a heterogeneous sensor network is common in applications, where for instance in a localization system the poor sensor can be an ultrasound sensor while the more powerful sensor can be a camera. Using a time-periodic Kalman filter, we show how one can find an optimal schedule of the sensor communication. One can significantly improve estimation quality by only using the expensive sensor rarely. We also demonstrate how simple sensor switching rules based on the Riccati equation drives the filter into a stable time-periodic Kalman filter. ï¿œ 2008 IEEE.

QC 20110224

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17

Bao, Lei. "Control over Low-Rate Noisy Channels". Doctoral thesis, KTH, Kommunikationsteori, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-10641.

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Networked embedded control systems are present almost everywhere. A recent trendis to introduce radio communication in these systems to increase mobility and flex-ibility. Network nodes, such as the sensors, are often simple devices with limitedcomputing and transmission power and low storage capacity, so an important prob-lem concerns how to optimize the use of resources to provide sustained overall sys-tem performance. The approach to this problem taken in the thesis is to analyzeand design the communication and control application layers in an integrated man-ner. We focus in particular on cross-layer design techniques for closed-loop controlover non-ideal communication channels, motivated by future control systems withvery low-rate and highly quantized sensor communication over noisy links. Severalfundamental problems in the design of source–channel coding and optimal controlfor these systems are discussed.The thesis consists of three parts. The first and main part is devoted to the jointdesign of the coding and control for linear plants, whose state feedback is trans-mitted over a finite-rate noisy channel. The system performance is measured by afinite-horizon linear quadratic cost. We discuss equivalence and separation proper-ties of the system, and conclude that although certainty equivalence does not holdin general it can still be utilized, under certain conditions, to simplify the overalldesign by separating the estimation and the control problems. An iterative opti-mization algorithm for training the encoder–controller pairs, taking channel errorsinto account in the quantizer design, is proposed. Monte Carlo simulations demon-strate promising improvements in performance compared to traditional approaches.In the second part of the thesis, we study the rate allocation problem for statefeedback control of a linear plant over a noisy channel. Optimizing a time-varyingcommunication rate, subject to a maximum average-rate constraint, can be viewedas a method to overcome the limited bandwidth and energy resources and to achievebetter overall performance. The basic idea is to allow the sensor and the controllerto communicate with a higher data rate when it is required. One general obstacle ofoptimal rate allocation is that it often leads to a non-convex and non-linear problem.We deal with this challenge by using high-rate theory and Lagrange duality. It isshown that the proposed method gives a good performance compared to some otherrate allocation schemes.In the third part, encoder–controller design for Gaussian channels is addressed.Optimizing for the Gaussian channel increases the controller complexity substan-tially because the channel output alphabet is now infinite. We show that an efficientcontroller can be implemented using Hadamard techniques. Thereafter, we proposea practical controller that makes use of both soft and hard channel outputs.
QC 20100623

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18

Ors, Tolga. "Traffic and congestion control for ATM over satellite to provide QoS". Thesis, University of Surrey, 1998. http://epubs.surrey.ac.uk/843458/.

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In broadband multimedia satellite networks it is necessary to multiplex bursty streams of traffic with differing Quality of Service (QoS) requirements to maximise the utilisation of the satellite link bandwidth. Providing the desired QoS of each service, in a multi-service environment is a major challenge for satellite networks. Asynchronous Transfer Mode (ATM) which provides hard QoS guarantees is suitable for a multi-service satellite environment. ATM has been developed as a vehicle for multimedia communications and is widely regarded as one of the most important and fastest-growing communications technology of this decade. The design of suitable traffic and congestion control algorithms is one of the most important challenge for the success of an ATM-based satellite network. This thesis develops and optimises a traffic and congestion control mechanism which can provide users the required QoS for ATM over satellite networks. In order to provide QoS differentiation for end-to-end communication it is proposed to use both loss and delay priorities, which are determined form the required Cell Loss Rate (CLR) and Cell Transfer Delay (CTD) parameters, for each service class. A multiple shared buffer scheduling (MSBS) policy considering both delay and loss priorities, is proposed and evaluated for scheduling and discarding of ATM cells. It is shown that both the CTD and CLR requirements of ATM services can be met by the MSBS scheme. A combined preventive/reactive control scheme incorporating an adaptive Leaky Bucket (LB) is investigated for the satellite environment. It has been found that reactive control improves the cell loss due to congestion for time scales larger than the propagation delay. As the satellite air interface bandwidth is currently one of the most expensive commodities in the service provision, an adaptive MAC protocol that can support the ATM service classes whilst maximising the bandwidth utilisation, is proposed and evaluated. The mapping of ATM service classes to MAC classes and the use of a prioritised request queue provides the QoS differentiation required by ATM networks. It is shown that a pure reservation system performs poorly for very bursty user traffic. The user population which can be supported using Random Access (RA) for very bursty users with short burst duration is higher. The system throughput can be maximised, by making this protocol adaptive to changing traffic characteristics. It is shown that the utilisation of the frame capacity and the total number of users served can be improved by using this protocol.

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19

Hossain, Mohammad Istiak. "Designing Efficient Access Control to Comply Massive-Multiservice IoT over Cellular Networks". Licentiate thesis, KTH, Radio Systems Laboratory (RS Lab), 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-214974.

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Internet of Things (IoT) has come in reality to improve our living quality. Automation is embraced in all the possible business verticals that have diverse communication needs ranged from static devices’ sporadic transmission to mobile devices’ every minute transmission. Despite, there are many technologies available today to support IoT services; cellular systems can play a vital role for IoT services, like wearables, vehicular, and industrial IoT, rollout which have either mobility or security concern. IoT services generated traffic are foreseen as a sporadic-bursty traffic. As the cellular networks are designed to serve continuous data traffic, the existing system’s access control mechanism cannot efficiently conform to the burstiness of traffic. This limits the scope of the network scalability in terms of simultaneous serving devices’ capacity. Also, this bursty pattern can extensively increase the rate of network’s congestion incident. In this thesis, we focus on these underlying challenges to support a large number of heterogeneous IoT services with existing services over the same radio network. An important question for supporting IoT services over cellular networks is how detrimental are the effects of IoT services on other services of cellular networks. This dissertation seeks to answer this with quantitative results to indicate the real constraints of existing networks. An important conclusion is that existing cellular system is incompetent to support the bursty arrival of massive IoT devices in terms of radio networks’ access control plane’s scalability. Therefore, this dissertation presents solutions to overcome the identified limitations of access control planes. To improve the performance of the access control plane, we incorporate a vertical core network controlled group management scheme that can assure the operator’s granular control over capillary gateways. Besides, this introduces a unique handover opportunity between cellular and capillary network vertices. Then, we present a simple but efficient initial access mechanism to overcome the initial access collision at the very early stage. Finally, we show the impact of access collision and retransmission on the initial access resource dimensioning.We present a practical traffic model that is realistic for the traffic scenario for mixed-traffic. Our presented results and analysis depict the trade-offs between access rate, retransmission and resource allocation over time and frequency.Our results reveal that with proposed schemes of the cellular system’s access control plane can be scalable and resilient to accommodate a large number of IoT devices without incurring extra delay or need of resources to the system.

QC 20170928

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20

Xu, Jianxuan. "Performance evaluation of TCP over optical channels and heterogeneous networks". [Tampa, Fla.] : University of South Florida, 2004. http://purl.fcla.edu/fcla/etd/SFE0000314.

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21

Henriksson, Erik. "Compensating for Unreliable Communication Links in Networked Control Systems". Licentiate thesis, Stockholm : Skolan för elektro- och systemteknik, Kungliga Tekniska högskolan, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-11059.

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22

Jia, Guihua. "Performance evaluation of congestion control protocols and loss differentiation algorithms over wireless networks". Thesis, University of Ottawa (Canada), 2007. http://hdl.handle.net/10393/27857.

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Congestion control protocols for wireless networks must be efficient, TCP-friendly, and robust to random wireless loss. Based on these criteria we have evaluated several different congestion control protocols, such as TCP Westwood, TFRC, MULTFRC, RAP, and IFTP. In wireless network environments, most of these protocols do not work well since wireless losses are counted as congestion losses. Therefore, it is necessary to extend these congestion control protocols with end-to-end Loss Differentiation Algorithms (LDA). We evaluate several existing LDA schemes, including Biaz, mBiaz, Spike, ZigZag, ZBS, PLC, SPLD, and TD, with simulation results showing different drawbacks for each scheme. We thus propose a new LDA scheme: the mSpike scheme. The mSpike scheme classifies the loss type according to the mean and deviation of the relative one-way trip time. The simulation results show that the mSpike scheme has better performance and fewer problems in most of the situations evaluated. We also test the combination of MULTFRC and mSpike under different wireless lossy environments. The simulation results show that they have a high utilization of the available bandwidth. Therefore, this combination would be a good choice for applications with high bandwidth requirements.

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23

Kasiolas, Anastasios. "Intelligent control in multimedia traffic policing, shaping and congestion avoidance over broadband networks". Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp03/MQ39837.pdf.

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Buse, D. P. "Information management, condition monitoring and control of power systems over internet protocol networks". Thesis, University of Liverpool, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.402263.

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25

Kesavareddigari, Himaja. "Analysis and Control of the Propagation of Failures and Misinformation over Complex Networks". The Ohio State University, 2020. http://rave.ohiolink.edu/etdc/view?acc_num=osu1588320404278331.

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26

Javadtalab, Abbas. "An End-to-End Solution for High Definition Video Conferencing over Best-Effort Networks". Thesis, Université d'Ottawa / University of Ottawa, 2015. http://hdl.handle.net/10393/31954.

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Video streaming applications over best-effort networks, such as the Internet, have become very popular among Internet users. Watching live sports and news, renting movies, watching clips online, making video calls, and participating in videoconferences are typical video applications that millions of people use daily. One of the most challenging aspects of video communication is the proper transmission of video in various network bandwidth conditions. Currently, various devices with different processing powers and various connection speeds (2G, 3G, Wi-Fi, and LTE) are used to access video over the Internet, which offers best-effort services only. Skype, ooVoo, Yahoo Messenger, and Zoom are some well-known applications employed on a daily basis by people throughout the world; however, best-effort networks are characterized by dynamic and unpredictable changes in the available bandwidth, which adversely affect the quality of the video. For the average consumer, there is no guarantee of receiving an exact amount of bandwidth for sending or receiving video data. Therefore, the video delivery system must use a bandwidth adaptation mechanism to deliver video content properly. Otherwise, bandwidth variations will lead to degradation in video quality or, in the worst case, disrupt the entire service. This is especially problematic for videoconferencing (VC) because of the bulkiness of the video, the stringent bandwidth demands, and the delay constraints. Furthermore, for business grade VC, which uses high definition videoconferencing (HDVC), user expectations regarding video quality are much higher than they are for ordinary VC. To manage network fluctuations and handle the video traffic, two major components in the system should be improved: the video encoder and the congestion control. The video encoder is responsible for compressing raw video captured by a camera and generating a bitstream. In addition to the efficiency of the encoder and compression speed, its output flow is also important. Though the nature of video content may make it impossible to generate a constant bitstream for a long period of time, the encoder must generate a flow around the given bitrate. While the encoder generates the video traffic around the given bitrate, congestion management plays a key role in determining the current available bandwidth. This can be done by analyzing the statistics of the sent/received packets, applying mathematical models, updating parameters, and informing the encoder. The performance of the whole system is related to the in-line collaboration of the encoder and the congestion management, in which the congestion control system detects and calculates the available bandwidth for a specific period of time, preferably per incoming packet, and informs rate control (RC) to adapt its bitrate in a reasonable time frame, so that the network oscillations do not affect the perceived quality on the decoder side and do not impose adverse effects on the video session. To address these problems, this thesis proposes a collaborative management architecture that monitors the network situation and manages the encoded video rate. The goal of this architecture is twofold: First, it aims to monitor the available network bandwidth, to predict network behavior and to pass that information to the encoder. So encoder can encode a suitable video bitrate. Second, by using a smart rate controller, it aims for an optimal adaptation of the encoder output bitrate to the bitrate determined by congestion control. Merging RC operations and network congestion management, to provide a reliable infrastructure for HDVC over the Internet, represents a unique approach. The primary motivation behind this project is that by applying videoconference features, which are explained in the rate controller and congestion management chapter, the HDVC application becomes feasible and reliable for the business grade application even in the best-effort networks such as the Internet.

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27

Ibrahim, Abdulkareem B. "Error resilience and concealment in MVC video over wireless networks". Thesis, Brunel University, 2015. http://bura.brunel.ac.uk/handle/2438/10484.

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Multi-view video is capable of presenting a full and accurate depth perception of a scene. The concept of multi-view video is becoming more useful especially in 3D display systems by enhancing the viewing of high resolution stereoscopic images from arbitrary viewpoints without the use of any special glasses. Like monoscopic video, the multi-view video is faced with different challenges such as: reliable compression, storage and bandwidth due to the increased number of views as well as the high sensitivity to transmission errors. All these may lead to a detrimental effect on the reconstructed views. The work in this thesis investigates the problems and challenges of transmission losses in a multi-view video bitstream over error prone wireless networks. Based on the network simulation results, the proposed technique is capable of addressing the problem of transmission losses. In practical wireless networks, transmission errors are inevitable and pose a serious challenge to the coded video data. The aim of this research effort is to examine the effect of these errors in a multi-view video bitstream when transmitted over a lossy channel. Moreover, this research work aims to develop a novel scheme that can make the multi-view coded videos more robust to transmission errors by minimizing the error effects and improving the perceptual quality. Multi-layer data partitioning as an error resilient technique is developed in JMVC 8.5 reference software in order to make the multi-view video bitstream more robust during transmission. In addition to that, we propose a simple decoding scheme that can support the decoding of the multi-layer data partitioning bitstream over channels with high error rate. The proposed technique is benchmarked with the already existing H.264/AVC data partitioning technique. The work in this thesis also employs the use of group of pictures as a coding parameter to investigate and reduce the effects of transmission errors in multi-view video transmitted over a very high error rate channel. The experiments are carried out with different error loss rates in order to evaluate the performance of these techniques in terms of perceptual quality when transmitted over a simulated erroneous channel. Errors are introduced using the Sirannon network simulator. The error performance of each technique is evaluated and analysed both objectively and subjectively after reconstruction. The results of the research investigation and simulation are presented and analysed in chapter six of the thesis.

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28

Begen, Ali C. "Enhancing the Multimedia Experience in Emerging Networks". Diss., Georgia Institute of Technology, 2006. http://hdl.handle.net/1853/14010.

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As multimedia processing and networking technologies, products and services evolve, the number of users communicating, collaborating and entertaining over the IP networks is growing rapidly. With the emergence of pervasive and ubiquitous multimedia services, this proliferation creates an abundant increase in the amount of the Internet backbone traffic. This brings the problem of efficient transmission of real-time and time-sensitive media content to the fore. Effective multimedia services demand appropriate application-specific and media-aware solutions, without which the full benefits of such services will not be realized. Poor approaches often lead to system performance degradations such as unacceptable presentation quality perceived by the users, possible network collapses due to the high-bandwidth nature of the multimedia applications, and poor performance observed by other data-oriented applications due to the unresponsiveness of multimedia flows. From a networking perspective, traditional approaches consider the application data as "sacred" and do not differentiate any part of it from the rest. While this keeps the data-delivery mechanisms, namely, the transport-layer protocols, as plain as possible, it also precludes these mechanisms from interpreting the media content and tailoring their actions according to the importance of the content. Given that this naive approach cannot satisfy the specific needs of each and every one of the today's emerging applications ranging from videotelephony to video-on-demand, from distance education to telemedicine, from remote surveillance to online video gaming, the study of Multimedia Transport Protocols (MMTP) is overdue. An MMTP solution basically integrates the multimedia content information into the responsible data-delivery mechanisms along with the requirements of the invoking application and network characteristics to deliver the highest level of service quality. In other words, an MMTP solution offers a unified environment where all cooperating protocol components interact with each other and make the best use of this collaboration to fulfill their respective duties. The focus of this thesis is on the design and evaluation of a set of end-to-end and system-level MMTP solutions for scalable, reliable, and high quality multimedia services in ever-changing, complex and heterogeneous computing and communication environments.

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29

Abuhaija, Belal Saleh. "A simulation framework for service continuity over multi access wireless networks". Thesis, University of South Wales, 2010. https://pure.southwales.ac.uk/en/studentthesis/a-simulation-framework-for-service-continuity-over-multi-access-wireless-networks(5b370ae0-47b2-43b7-b0f2-91b1fd5fd3b8).html.

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Mobile communication systems have continued to evolve by the release of new standards for HSPA and the release of new standards for LTE in release 8 and release 9. The new releases aim at providing higher data rates to accommodate the envisioned services of 3GPP in voice, data, image transmission, video, multimedia service and broadband services. Catering for a wide variety of services to satisfy the demands imposed on mobile networks by the user diversity and demanding applications, the air interface has been identified as the major bottleneck in the mobile networks. Network planning engineers and operators are deploying the new air interfaces in the same cell sites, which gives rise to several internetworking issues among the different air interfaces from radio resources management to service call continuity issues due to user's mobility and changing point of attachment. Deployment of different air interfaces coupled with traffic diversity requests further complicates the managements of the mobile networks. One of The main objectives of this research is to propose and evaluate solutions that address the internetworking of the different radio air interfaces in proposing a call admission control algorithm that can utilize different air interfaces capabilities and to determine which traffic types are better suited to be serviced by an interface. The proposed algorithm will consider the availability of the interface, the load of the network and the user equipment capabilities. Another main objective of this research is to propose and evaluate solutions that address changing the point of attachment of the users due to mobility in the form of handover algorithm. The proposed algorithm will consider the coverage of the node, direction and speed of the user, the network load, the air interface availability and the user equipment capabilities. Another main objective of this research is to design and implement a simulation system which includes all 3GPP standardized technologies. The simulation tool was designed as a discrete event simulation (DBS) system which includes all the standardized air interface technologies and services. The simulation tool was designed using Visual C# to take advantage of the object oriented capabilities of the Windows environment and libraries. The simulation tool was essential in evaluating the proposed algorithms in the first two objectives.

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30

Mushtaq, Mubashar. "QoS provisioning for adaptive video streaming over P2P networks". Thesis, Bordeaux 1, 2008. http://www.theses.fr/2008BOR13744.

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Actuellement, nous constatons une augmentation de demande de services vidéo sur les réseaux P2P. Ces réseaux jouent un rôle primordial dans la transmission de contenus multimédia à grande échelle pour des clients hétérogènes. Cependant, le déploiement de services vidéo temps réel sur les réseaux P2P a suscité un grand nombre de défis dû à l’hétérogénéité des terminaux et des réseaux d’accès, aux caractéristiques dynamiques des pairs, et aux autres problèmes hérités des protocoles TCP/IP, à savoir les pertes de paquets, les délais de transfert et la variation de la bande passante de bout-en-bout. Dans ce contexte, l’objectif de cette thèse est d’analyser les différents problèmes et de proposer un mécanisme de transport vidéo temps réel avec le provisionnement de la qualité de Service (QoS). Ainsi, nous proposons trois contributions majeures. Premièrement, il s’agit d’un mécanisme de streaming vidéo adaptatif permettant de sélectionner les meilleurs pair émetteurs. Ce mécanisme permet de structurer les pairs dans des réseaux overlay hybrides avec une prise en charge des caractéristiques sémantiques du contenu et des caractéristiques physiques du lien de transport. Nous présentons ensuite un mécanisme d’ordonnancement de paquets vidéo combiné à une commutation de pairs et/ou de flux pour assurer un transport lisse. Finalement, une architecture de collaboration entre les applications P2P et les fournisseurs de services / réseaux est proposée pour supporter un contrôle d’admission de flux
There is an increasing demand for scalable deployment of real-time multimedia streaming applications over Internet. In this context, Peer-to-Peer (P2P) networks are playing an important role for supporting robust and large-scale transmission of multimedia content to heterogeneous clients. However, the deployment of real-time video streaming applications over P2P networks arises lot of challenges due to heterogeneity of terminals and access networks, dynamicity of peers, and other problems inherited from IP network. Real-time streaming applications are very sensitive to packet loss, jitter / transmission delay, and available end-to-end bandwidth. These elements have key importance in QoS provisioning and need extra consideration for smooth delivery of video streaming applications over P2P networks. Beside the abovementioned issues, P2P applications lack of awareness in constructing their overlay topologies and do not have any explicit interaction with service and network providers. This situation leads to inefficient utilization of network resources and may cause potential violation of peering agreements between providers. The aim of this thesis is to analyze these issues and to propose an adaptive real-time transport mechanism for QoS provisioning of Scalable Video Coding (SVC) applications over P2P networks. Our contributions in this dissertation are threefold. First, we propose a hybrid overlay organization mechanism allowing intelligent organization of sender peers based on network-awareness, media- awareness, and quality-awareness. This overlay organization is further used for an appropriate selection of best sender peers, and the efficient switching of peers to ensure a smooth video delivery when any of the sender peers is no more reliable. Second, we propose a packet video scheduling mechanism to assign different parts of the video content to specific peers. Third, we present a service provider driven P2P network framework that enables effective interaction between service / network providers and P2P applications to perform QoS provisioning mechanism for the video streaming

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31

Almadi, Soloman Moses. "Converged IP-over-standard ethernet progress control networks for hydrocarbon process automation applications controllers". Thesis, Brunel University, 2011. http://bura.brunel.ac.uk/handle/2438/8365.

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The maturity level of Internet Protocol (IP) and the emergence of standard Ethernet interfaces of Hydrocarbon Process Automation Application (HPAA) present a real opportunity to combine independent industrial applications onto an integrated IP based network platform. Quality of Service (QoS) for IP over Ethernet has the strength to regulate traffic mix and support timely delivery. The combinations of these technologies lend themselves to provide a platform to support HPAA applications across Local Area Network (LAN) and Wide Area Network (WAN) networks. HPAA systems are composed of sensors, actuators, and logic solvers networked together to form independent control system network platforms. They support hydrocarbon plants operating under critical conditions that — if not controlled — could become dangerous to people, assets and the environment. This demands high speed networking which is triggered by the need to capture data with higher frequency rate at a finer granularity. Nevertheless, existing HPAA network infrastructure is based on unique autonomous systems, which has resulted in multiple, parallel and separate networks with limited interconnectivity supporting different functions. This created increased complexity in integrating various applications and resulted higher costs in the technology life cycle total ownership. To date, the concept of consolidating HPAA into a converged IP network over standard Ethernet has not yet been explored. This research aims to explore and develop the HPAA Process Control Systems (PCS) in a Converged Internet Protocol (CIP) using experimental and simulated networks case studies. Results from experimental and simulation work showed encouraging outcomes and provided a good argument for supporting the co-existence of HPAA and non-HPAA applications taking into consideration timeliness and reliability requirements. This was achieved by invoking priority based scheduling with the highest priority being awarded to PCS among other supported services such as voice, multimedia streams and other applications. HPAA can benefit from utilizing CIP over Ethernet by reducing the number of interdependent HPAA PCS networks to a single uniform and standard network. In addition, this integrated infrastructure offers a platform for additional support services such as multimedia streaming, voice, and data. This network‐based model manifests itself to be integrated with remote control system platform capabilities at the end user's desktop independent of space and time resulting in the concept of plant virtualization.

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32

Gao, Chengwei. "Priority statistical multiplexing and two-level congestion control for video transmission over ATM networks /". Thesis, Connect to this title online; UW restricted, 1997. http://hdl.handle.net/1773/5889.

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Cen, Song. "Congestion control and error concealment for real-time video transmission over Lossy Packet Networks /". Diss., Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 2002. http://wwwlib.umi.com/cr/ucsd/fullcit?p3061641.

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34

Hsiao, Hsu-Feng. "Multimedia streaming congestion control over heterogeneous networks : from distributed computation and end-to-end perspectives /". Thesis, Connect to this title online; UW restricted, 2005. http://hdl.handle.net/1773/5946.

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35

Fredholm, Kenth y Kristian Nilsson. "Implementing an application for communication and quality measurements over UMTS networks". Thesis, Linköping University, Department of Electrical Engineering, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-1666.

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The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.

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36

ITO, Maria Silvia. "A control – theoretic approach for adaptive streaming over HTTP". Universidade Federal de Pernambuco, 2015. https://repositorio.ufpe.br/handle/123456789/16342.

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Submitted by Fabio Sobreira Campos da Costa (fabio.sobreira@ufpe.br) on 2016-04-06T13:08:48Z No. of bitstreams: 2 license_rdf: 1232 bytes, checksum: 66e71c371cc565284e70f40736c94386 (MD5) A CONTROL-THEORETIC APPROACH FOR ADAPTIVE STREAMING OVER HTT.pdf: 5118913 bytes, checksum: fc08706cd245c8c0de45e4db894cb0a5 (MD5)
Made available in DSpace on 2016-04-06T13:08:48Z (GMT). No. of bitstreams: 2 license_rdf: 1232 bytes, checksum: 66e71c371cc565284e70f40736c94386 (MD5) A CONTROL-THEORETIC APPROACH FOR ADAPTIVE STREAMING OVER HTT.pdf: 5118913 bytes, checksum: fc08706cd245c8c0de45e4db894cb0a5 (MD5) Previous issue date: 2015-08-24
FACEPE
Video applications account for a large percentage of Internet traffic, and their utilization tends to grow. In this scenario, multimedia providers are required to deliver video and/or audio content in a seamless manner, regardless of network conditions and client device capabilities. However, current commercial services are not efficient enough, and the academic proposals do not take full advantage of their potential. To this end, we propose a control-theoretic adaptive streaming over HTTP system, which adapts the video quality to the network conditions in a video session. With this system, we aim at delivering video with the highest Quality of Experience possible, while leveraging the available network resources. Our system is located at the client side, and comprises a control system and a state machine. Together, they select the video level that is requested to the server, a standard HTTP server, which sends video chunks as the client requests them. To select the control system and state machine that best fit our requirements, we perform several experiments with different parameter values combinations. Our parameter selection is accomplished in two steps, the last one resulting in the combination that best fits our needs. Then, we compare our system performance to an existent system in the academia. Our experimental results show the benefits of providing more aggressive decisions on the video quality selection. However, they also show that such aggressiveness may, sometimes, compromise the system performance. Overall, though, our system outperforms the baseline of our experiments.
A computação em nuvem é um paradigma computacional que vem sendo amplamente utilizado ao longo dos últimos anos devido as suas características de provisionamento de recursos de forma escalável, onde o usuário paga apenas por aquilo que consumir. Esse modelo computacional possibilita que diversos serviços sejam ofertados a partir da sua Infraestrutura como Serviço (IaaS - Infrastructure as a Service). Porém, a falha de componentes dos recursos da nuvem é algo bastante comum e que afeta diretamente a disponibilidade dos serviços que os utilizam. Garantir alta disponibilidade em serviços na nuvem é um desafio para provedores, que podem utilizar de técnicas como análise de sensibilidade para identificar gargalos de disponibilidade ou ainda fazer uso de mecanismos de tolerância à falhas para atingir melhores resultados de confiabilidade. Desse modo, esse trabalho tem como proposta realizar a avaliação de disponibilidade em arquiteturas para o provimento do serviço de VoD streaming na nuvem, baseado na plataforma Eucalyptus. O trabalho está dividido em dois estudos de caso. Primeiro, apresentamos uma arquitetura básica, sem mecanismos de redundância. Através de modelagem hierárquica formada por diagramas de blocos de confiabilidade (RBD - Reliability Block Diagram) e cadeias de Markov de tempo contínuo (CTMC - Continuous Time Markov Chain), realizamos a avaliação da disponibilidade para essa arquitetura, e em seguida alcançamos a validação do modelo que representa o funcionamento do serviço. Por fim, aplicamos a técnica de análise de sensibilidade paramétrica para identificar gargalos de disponibilidade na arquitetura básica. As análises seguintes que foram esse estudo de caso observam a disponibilidade do serviço em arquiteturas com redundância no modo warm-stand guiadas por índices de sensibilidade, apresentando ao final uma comparação dos valores de disponibilidade para cada arquitetura obtida a partir de componentes redundantes. O segundo estudo de caso analisa o comportamento da disponibilidade total do serviço a partir da variação no tempo de ativação dos mecanismos redundantes. Nossos resultados expressaram que a adoção de componentes redundantes no modo warm-stand na infraestrutura da nuvem proporciona ganhos de disponibilidade no serviço analisado. Também observamos que a variação do tempo de ativação do componente de maior confiabilidade tem maior impacto na disponibilidade total do serviço de VoD streaming.

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37

Hussain, Khadim y Choudhry Humayun Saleem. "Using Peer to Peer Over Wireless Ad Hoc Networks as an Emergency Command and Control System". Thesis, Halmstad University, School of Information Science, Computer and Electrical Engineering (IDE), 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-2400.

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Peer-to-Peer networking technology is evolving rapidly. P2P networks overcome the limitations of client/server networks where each computer shares resources of other computer. There are different types of P2P networks depending upon their functionalities. Peer to peer networks provide long list of features like: selection of nearby peers, redundant storage, effi-cient search/location of data items, data performance or guarantees etc. It is important to see that how P2P can work with wireless ad hoc networks and why it is important. What are im-portant issues which come across by using P2P and wireless ad hoc networks together. How TCP/IP stack can be affected. How the whole system looks like in which P2P techniques are used with wireless ad hoc routing protocols to perform different tasks and services.

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38

Dolz, Algaba Daniel. "Estimation and fault diagnosis strategies for networked control systems". Doctoral thesis, Universitat Jaume I, 2014. http://hdl.handle.net/10803/285103.

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Communication networks increase flexibility of industrial monitoring, supervisory and control systems. However, they introduce delays or even dropouts on the transmitted information that affect the performance and robustness on the decision and control mechanisms in the system. This thesis contributes theoretically to the state estimation and fault diagnosis problem over networks.

First, we study the state estimation problem. Motivated by reducing the implementation computational load of Luenberger-type estimators, we focus on predefined gain approaches for different network transmission conditions.

In general, we propose jump estimators whose gains are related to the different network-induced data reception scenarios. We define the estimator complexity in terms of the number of different stored gains. Considering constant successful transmission probabilities, our main contribution here is the design of jump linear estimators to attain favorable trade-offs between estimation performance and estimator complexity. We show that one can reduce the estimator complexity while guaranteeing a similar performance than the optimal Kalman Filter. When dropouts are governed by a non-stationary stochastic process, the successful transmission probability is time-varying and may be unknown. For this case, we propose an estimator whose gains are scheduled in real-time with rational functions of the estimated packet arrival rate. We turn the design procedure into an optimization problem over polynomials that is numerically solved employing sum-of-squares (SOS) decomposition techniques.

Second, motivated by reducing the network resource consumption without considerably degrading the estimation performance, we study the jointly design of jump linear estimators and predefined network operation conditions (co-design) to guarantee a favorable trade-off. Focusing on wireless networks with self-powered nodes, where transmitting is the most energy consuming task, we analyze two approaches for the network operation: event-based transmissions and power control.

For the event-based approach, we use a Send-on-Delta protocol which reduces the number of transmissions with respect to transmitting at each sampling instant. However, it leads to an unknown successful transmission probability. For this framework, we contribute by characterizing this uncertainty and including it on the stochastic behavior of the estimator by means of a SOS-based design. Power control strategies are developed over a multi-hop wireless network with fading channels. Instead of reducing the number of transmission, power control acts directly on the transmission power. Higher transmission powers imply higher successful transmission probability values.

Finally, motivated by the need of assuring a reliable operation of the networked system, we study the fault diagnosis problem. We explore and point out the trade-offs between fast fault detection and fault tracking conditions. We design jump estimatorbased fault diagnosers in which we can specify the minimum detectable faults, false alarm rate and response time to faults. Another contribution is a tightened version of existing false alarm rate bounds.

Moreover, we also address the case when the control input is transmitted through a network without delivery acknowledgement. In this case, we improve fault diagnosisaccuracy by scheduling in real time the estimator jumping gains with rational functions that depend on a statistic of the control input error (difference between the control command being applied in the process and the one being used in the estimator). Again, we use a SOS-based solution approach to make the design problem tractable.

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39

Sangal, Rahul. "Performance analysis of the transmission control protocol over low earth orbit satellite communication systems". Ohio University / OhioLINK, 1999. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1175629275.

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40

Jayaram, Ranjith S. "Performance Evaluation of TEAR, a TCP-friendly Flow Control Protocol, Over the Internet and Wireless Networks". NCSU, 2001. http://www.lib.ncsu.edu/theses/available/etd-20010717-083717.

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TCP Emulation at Receivers (TEAR) is a TCP-friendly protocol that has been proposed for real-time multimedia flow control. Most best-effort traffic on the Internet is well-served by TCP, the dominant transport protocol. However, many applications with real-time constraints, such as multimedia streaming find TCP's response to congestion quite severe and too drastic to deliver acceptable end-user quality. TEAR was designed in order to provide smoothly varying rate changes for such applications while being friendly to competing TCP flows. In this thesis, weevaluate and verify TEAR's performance over the Internet. We verify TEAR's fairness to TCP, the smoothness of its rate fluctuations and its stability in the presence of network perturbations. We then adapt TEAR to run over wireless networks and consider using round-trip delay instead of packet loss as a congestion indication for wireless networks. We present the results of our experiments with TEAR over commercially deployed wireless networks in South Korea. We recount our experiences with the loss-delay characteristics of these networks. We analyze how TEAR competes with TCP, which is known to suffer from severe degradations in environments where the underlying network is unreliable. We then study TEAR's rate variations and the increased longer-term predictability it provides over wireless networks. Finally, we compare the performance of a reliable protocol we built over TEAR with TCP.

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41

Li, Lian. "The design and implementation of a real-time multimedia synchronization control system over high-speed communications networks". Thesis, University of Ottawa (Canada), 1994. http://hdl.handle.net/10393/6738.

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Sychronization is considered as a key issue in distributed multimedia systems. In a real-time multimedia presentation, data objects of different media types or coding formats are delivered from distributed media-storing servers to the remote client simultaneously over high-speed networks. The multiple streams need to be synchronized so that the multimedia document can be presented in the way specified by its creator. The synchronization research involves issues such as temporal relationship modeling, extending network protocols and supporting the implementation of applications where the synchronization control mechanisms integrate with other system functionality, such as the ATM network transmissions. The video coding/decoding and the distributed database management. In this thesis, we investigate a software synchronization control system for a target presentational application, i.e., a Multimedia News-on-demand service. Relying on the Quality of Services (QoS) supported by the ATM-based virtual connections, the system prevents major multi-stream mismatches through a delivery scheduling operation. Moreover, the synchronization errors brought by the inevitable network delay variations are recovered through a Stream Synchronization Protocol (SSP) in order to preserve the presentation quality. We apply the Time Flow Graph (TFG) to model the temporal relationships among the media components so that the scheduling and recovering operations can be efficient. Synchronization QoS parameters are employed in the SSP control. In addition, the differences between the characterization of coded and uncoded data streams are taken into account. We present a priority-based synchronization control for coded data, e.g., the MPEG-2 video stream. For the implementation of such a control system, we elaborate a set of data structure specifications and algorithms. As well, we develop the software modules to implement the synchronization control prototype.

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42

Li, Zhi. "Fuzzy logic based robust control of queue management and optimal treatment of traffic over TCP/IP networks". University of Southern Queensland, Faculty of Sciences, 2005. http://eprints.usq.edu.au/archive/00001461/.

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Improving network performance in terms of efficiency, fairness in the bandwidth, and system stability has been a research issue for decades. Current Internet traffic control maintains sophistication in end TCPs but simplicity in routers. In each router, incoming packets queue up in a buffer for transmission until the buffer is full, and then the packets are dropped. This router queue management strategy is referred to as Drop Tail. End TCPs eventually detect packet losses and slow down their sending rates to ease congestion in the network. This way, the aggregate sending rate converges to the network capacity. In the past, Drop Tail has been adopted in most routers in the Internet due to its simplicity of implementation and practicability with light traffic loads. However Drop Tail, with heavy-loaded traffic, causes not only high loss rate and low network throughput, but also long packet delay and lengthy congestion conditions. To address these problems, active queue management (AQM) has been proposed with the idea of proactively and selectively dropping packets before an output buffer is full. The essence of AQM is to drop packets in such a way that the congestion avoidance strategy of TCP works most effectively. Significant efforts in developing AQM have been made since random early detection (RED), the first prominent AQM other than Drop Tail, was introduced in 1993. Although various AQMs also tend to improve fairness in bandwidth among flows, the vulnerability of short-lived flows persists due to the conservative nature of TCP. It has been revealed that short-lived flows take up traffic with a relatively small percentage of bytes but in a large number of flows. From the user’s point of view, there is an expectation of timely delivery of short-lived flows. Our approach is to apply artificial intelligence technologies, particularly fuzzy logic (FL), to address these two issues: an effective AQM scheme, and preferential treatment for short-lived flows. Inspired by the success of FL in the robust control of nonlinear complex systems, our hypothesis is that the Internet is one of the most complex systems and FL can be applied to it. First of all, state of the art AQM schemes outperform Drop Tail, but their performance is not consistent under different network scenarios. Research reveals that this inconsistency is due to the selection of congestion indicators. Most existing AQM schemes are reliant on queue length, input rate, and extreme events occurring in the routers, such as a full queue and an empty queue. This drawback might be overcome by introducing an indicator which takes account of not only input traffic but also queue occupancy for early congestion notification. The congestion indicator chosen in this research is traffic load factor. Traffic load factor is in fact dimensionless and thus independent of link capacity, and also it is easy to use in more complex networks where different traffic classes coexist. The traffic load indicator is a descriptive measure of the complex communication network, and is well suited for use in FL control theory. Based on the traffic load indicator, AQM using FL – or FLAQM – is explored and two FLAQM algorithms are proposed. Secondly, a mice and elephants (ME) strategy is proposed for addressing the problem of the vulnerability of short-lived flows. The idea behind ME is to treat short-lived flows preferably over bulk flows. ME’s operational location is chosen at user premise gateways, where surplus processing resources are available compared to other places. By giving absolute priority to short-lived flows, both short and long-lived flows can benefit. One problem with ME is starvation of elephants or long-lived flows. This issue is addressed by dynamically adjusting the threshold distinguishing between mice and elephants with the guarantee that minimum capacity is maintained for elephants. The method used to dynamically adjust the threshold is to apply FL. FLAQM is deployed to control the elephant queue with consideration of capacity usage of mice packets. In addition, flow states in a ME router are periodically updated to maintain the data storage. The application of the traffic load factor for early congestion notification and the ME strategy have been evaluated via extensive experimental simulations with a range of traffic load conditions. The results show that the proposed two FLAQM algorithms outperform some well-known AQM schemes in all the investigated network circumstances in terms of both user-centric measures and network-centric measures. The ME strategy, with the use of FLAQM to control long-lived flow queues, improves not only the performance of short-lived flows but also the overall performance of the network without disadvantaging long-lived flows.

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43

Canovas, Sergio Roberto de Mello. "Integração entre redes LonWorks e redes IP: aplicações, requisitos e soluções". Universidade de São Paulo, 2006. http://www.teses.usp.br/teses/disponiveis/3/3141/tde-22042007-142526/.

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As redes de controle tendem a ser cada vez mais utilizadas nos sistemas de automação. Sua interconexão com a Internet também apresenta uma interessante alternativa, gerando uma demanda para novos estudos. Este trabalho considera a interconexão entre redes de controle LonWorks e redes baseadas no Internet Protocol (IP) sob o ponto de vista de quatro categorias de aplicações: monitoração remota de redes LonWorks via rede IP; gerenciamento remoto de redes LonWorks via rede IP; utilização de backbone IP para interligar diferentes redes LonWorks; e utilização de backbone IP para interligar diferentes redes LonWorks com aplicações de requisitos de tempo real. Quatro questões delimitam o escopo do estudo: que tipos de soluções existem para a interconexão entre redes LonWorks e IP; que tipos de parâmetros de rede devem ser considerados para uma solução baseada em redes LonWorks/IP e como eles podem ser utilizados para estabelecer requisitos de uma certa aplicação; a quais requisitos cada uma das quatro categorias de aplicações referidas está associada; quais tipos de soluções, dentre as apresentadas, melhor atendem a cada uma das quatro categorias de aplicações. As respostas a essas questões foram embasadas tanto em estudos teóricos quanto em experimentos. Através do levantamento do estado da arte dessa área, reunindo o conhecimento atual, procurou-se uniformizar a nomenclatura associada e apresentar uma série de elementos e conceitos que servem como base teórica para avaliação e projeto de soluções de interconexão entre redes LonWorks e IP. Introduziu-se também novas explicações, exemplos e considerações sobre o assunto. Desenvolveu-se um aplicativo que permite emular uma situação de laço de controle sobre redes LonWorks e IP, obtendo-se uma ferramenta prática para a realização de testes ou mesmo para auxiliar na elaboração de projetos de novas redes LonWorks interligadas a redes IP. Como contribuições do trabalho, destacam-se: levantamento de parâmetros de rede quantitativos para estabelecimento de requisitos de aplicações LonWorks e IP; desenvolvimento de um aplicativo para experimento de controle sobre LonWorks/IP; análise do conhecimento atual sobre a tecnologia LonWorks, harmonizando as diversas visões; elucidação dos relacionamentos entre as soluções tecnológicas para interconexão LonWorks/IP com as categorias de aplicações da tecnologia.
Control networks utilization on automation systems has been increasing. Their interconnection with the Internet is also an interesting option, creating the necessity of new studies. This work considers the interconnection between LonWorks control networks and IP-based networks. Four categories of applications are the focus: remote monitoring of LonWorks networks thorugh IP networks; remote management of LonWorks networks through IP networks; use of an IP network as a backbone to interconnect more than one LonWorks network; and the same use of an IP network for applications with real time requirements. Four questions establish the escope of this study: what kinds of solutions are available to provide interconnection between LonWorks and IP networks; what kinds of network parameters should be considered in a LonWorks/IP based solution and how can they be used to determine requirements for an application; which are the most common requirements for each one of the four categories of applications; and which solution does best fit to each category. The answers of these questions were based on theoretical studies and experiments. The notation used in different references was unified, and basic concepts and elements about the topic were presented. New explanations, examples and considerations were also introduced. A software application to emulate a control loop running over LonWorks/IP networks has been developed. This tool can be used for tests and even as a start point for real projects. The main contributions of this work are: identification of quantitative network parameters for establishement of requirements of LonWorks/IP applications; creation of a software application to run a control loop over LonWorks/IP networks; analysis of current knowledge about LonWorks technology; and identification of relationships between technological solutions for LonWorks/IP interconnection and categories of application of the technology.

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44

Garcia, Raul Murillo. "Development and evaluation of a novel tool for real-time control and remote monitoring over local area networks". Thesis, Glasgow Caledonian University, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.289677.

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Li, Yue. "Edge computing-based access network selection for heterogeneous wireless networks". Thesis, Rennes 1, 2017. http://www.theses.fr/2017REN1S042/document.

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Au cours de ces dernières décennies, les réseaux de télécommunications mobiles ont évolué de la 1G à la 4G. La 4G permet la coexistence de différents réseaux d'accès. Ainsi, les utilisateurs ont la capacité de se connecter à un réseau hétérogène, constitué de plusieurs réseaux d'accès. Toutefois, la sélection du réseau approprié n'est pas une tâche facile pour les utilisateurs mobiles puisque les conditions de chaque réseau d'accès changent rapidement. Par ailleurs, en termes d'usage, le streaming vidéo devient le service principal de transfert de données sur les réseaux mobiles, ce qui amène les fournisseurs de contenu et les opérateurs de réseau à coopérer pour garantir la qualité de la diffusion. Dans ce contexte, la thèse propose la conception d'une approche novatrice pour la prise de décision optimale de sélection de réseau et une architecture améliorant les performances des services de streaming adaptatif dans un réseau hétérogène. En premier lieu, nous introduisons un modèle analytique décrivant la procédure de sélection de réseau en ne considérant déjà qu'une seule classe de trafic. Nous concevons ensuite une stratégie de sélection basée sur des fondements de la théorie du contrôle optimal linéaire. Des simulations sous MATLAB sont effectuées pour valider l'efficacité du mécanisme proposé. Sur ce même principe, nous étendons ce modèle avec un modèle analytique général décrivant les procédures de sélection de réseau dans des environnements de réseaux hétérogènes avec de multiples classes de trafic. Le modèle proposé est ensuite utilisé pour dériver un mécanisme adaptatif basé sur la théorie du contrôle, qui permet non seulement d'aider à piloter dynamiquement le trafic vers l'accès réseau le plus approprié mais aussi de bloquer dynamiquement le trafic résiduel lorsque le réseau est congestionné en ajustant les probabilités d'accès optimales. Nous discutons aussi les avantages d'une intégration transparente du mécanisme proposé avec l'ANDSF, solution fonctionnelle normalisée pour la sélection de réseau. Un prototype est également implémenté dans ns-3. En second lieu, nous nous concentrons sur l'amélioration des performances de DASH pour les utilisateurs mobiles dans un environnement de réseau d'accès 4G uniquement. Nous introduisons une nouvelle architecture basée sur l'utilisation de serveurs distribués en périphérie de réseau suivant le standard MEC. Le mécanisme d'adaptation proposé, fonctionnant en tant que service MEC, peut modifier les fichiers de manifeste en temps réel, en réponse à la congestion du réseau et à la demande dynamique de flux de streaming. Ces modifications conduisent ainsi les clients à sélectionner des représentations vidéo de débit / qualité plus appropriées. Nous avons développé une plateforme de test virtualisée pour l'expérimentation de notre proposition. Les résultats ainsi obtenus démontrent ses avantages en terme de QoE comparés aux approches d'adaptation traditionnelles, purement pilotées par les clients, car notre approche améliore non seulement le MOS mais aussi l'équité face à la congestion. Enfin, nous étendons l'architecture proposée basée sur MEC pour supporter le service de streaming adaptatif DASH dans un réseau hétérogène multi-accès afin de maximiser la QoE et l'équité des utilisateurs mobiles. Dans ce scénario, notre mécanisme doit aider les utilisateurs à sélectionner la qualité vidéo et le réseau et nous le formulons comme un problème d'optimisation. Ce problème d'optimisation peut être résolu par l'outil IBM CPLEX, mais cela prend du temps et ne peut être envisagé à grande échelle. Par conséquent, nous introduisons une heuristique pour aborder la solution optimale avec moins de complexité. Ensuite, nous mettons en œuvre une expérimentation sur notre plateforme de tests. Le résultat démontre que, par rapport à l'outil IBM CPLEX, notre algorithme permet d'obtenir des performances similaires sur la QoE globale et l'équité, avec un gain de temps significatif
Telecommunication network has evolved from 1G to 4G in the past decades. One of the typical characteristics of the 4G network is the coexistence of heterogeneous radio access technologies, which offers end-users the capability to connect them and to switch between them with their mobile devices of the new generation. However, selecting the right network is not an easy task for mobile users since access network condition changes rapidly. Moreover, video streaming is becoming the major data service over the mobile network where content providers and network operators should cooperate to guarantee the quality of video delivery. In order to cope with this context, the thesis concerns the design of a novel approach for making an optimal network selection decision and architecture for improving the performance of adaptive streaming in the context of a heterogeneous network. Firstly, we introduce an analytical model (i.e. linear discrete-time system) to describe the network selection procedure considering one traffic class. Then, we consider the design of a selection strategy based on foundations from linear optimal control theory, with the objective to maximize network resource utilization while meeting the constraints of the supported services. Computer simulations with MATLAB are carried out to validate the efficiency of the proposed mechanism. Based on the same principal we extend this model with a general analytical model describing the network selection procedures in heterogeneous network environments with multiple traffic classes. The proposed model was, then, used to derive a scalable mechanism based on control theory, which allows not only to assist in steering dynamically the traffic to the most appropriate network access but also helps in blocking the residual traffic dynamically when the network is congested by adjusting dynamically the access probabilities. We discuss the advantages of a seamless integration with the ANDSF. A prototype is also implemented into ns-3. Simulation results sort out that the proposed scheme prevents the network congestion and demonstrates the effectiveness of the controller design, which can maximize the network resources allocation by converging the network workload to the targeted network occupancy. Thereafter, we focus on enhancing the performance of DASH in a mobile network environment for the users which has one access network. We introduce a novel architecture based on MEC. The proposed adaptation mechanism, running as an MEC service, can modify the manifest files in real time, responding to network congestion and dynamic demand, thus driving clients towards selecting more appropriate quality/bitrate video representations. We have developed a virtualized testbed to run the experiment with our proposed scheme. The simulation results demonstrate its QoE benefits compared to traditional, purely client-driven, bitrate adaptation approaches since our scheme notably improves both on the achieved MOS and on fairness in the face of congestion. Finally, we extend the proposed the MEC-based architecture to support the DASH service in a multi-access heterogeneous network in order to maximize the QoE and fairness of mobile users. In this scenario, our scheme should help users select both video quality and access network and we formulate it as an optimization problem. This optimization problem can be solved by IBM CPLEX tool. However, this tool is time-consuming and not scalable. Therefore, we introduce a heuristic algorithm to make a sub-optimal solution with less complexity. Then we implement a testbed to conduct the experiment and the result demonstrates that our proposed algorithm notably can achieve similar performance on overall achieved QoE and fairness with much more time-saving compared to the IBM CPLEX tool

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46

Grigorescu, Eduard. "Reducing internet latency for thin-stream applications over reliable transport with active queue management". Thesis, University of Aberdeen, 2018. http://digitool.abdn.ac.uk:80/webclient/DeliveryManager?pid=236098.

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An increasing number of network applications use reliable transport protocols. Applications with constant data transmission recover from loss without major performance disruption, however, applications that send data sporadically, in small packets, also called thin-streams, experience frequently high latencies due to 'Bufferbloat', that reduce the application performance. Active Queue Management mechanisms were proposed to dynamically manage the queues in routers by dropping packets early and reduce these, hence reducing latency. While their deployment to the internet remains an open issue, the proper investigation into how their functioning mechanism impacts latency is the main focus of this work and research questions have been devised to investigate the AQM impact on latency. A range of AQM mechanisms has been evaluated by the research, exploring performance of the methods for latency sensitive network applications. This has explored new single queue AQM mechanisms such as Controlled Delay (CODEL) and Proportional Integral Enhanced (PIE) and Adaptive RED (ARED). The evaluation has shown great improvements in queuing latency when AQM are used over a range of network scenarios. Scheduling AQM algorithms such as FlowQueue CODEL (FQ-CODEL) isolates traffic and minimises the impact of Bufferbloat on flows. The core components of FQ-CODEL, still widely misunderstood at the time of its inception, have been explained in depth by this study and their contribution to reducing latency have been evaluated. The results show significant reductions in queuing latency for thin streams using FQ-CODEL. When TCP is used for thin streams, high application latencies can arise when there are retransmissions, for example after dropping packets by an AQM mechanism. This delay is a result of TCP's loss-based congestion control mechanism that controls sender transmission rate following packet loss. ECN, a marking sender-side improvement to TCP reduces applicationlayer latency without disrupting the overall network performance. The thesis evaluated the benefit of using ECN using a wide range of experiments. The findings show that FQ-CODEL with ECN provides a substantial reduction of application latency compared to a drop-based AQM. Moreover, this study recommends the combination of FQ-CODEL with other mechanisms, to reduce application latency. Mechanisms such as ABE, have been shown to increase aggregate throughput and reduce application latency for thin-stream applications.

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47

Sfairopoulou, Anna. "A cross-layer mechanism for QoS improvements in VoIP over multi-rate WLAN networks". Doctoral thesis, Universitat Pompeu Fabra, 2008. http://hdl.handle.net/10803/7563.

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In IEEE 802.11 WLANs, Link Adaptation mechanisms, which choose the transmission rate of each node, provoke unexpected and random variations on the effective channel capacity. When these changes are towards lower bitrates, inelastic flows, such as VoIP, can suffer from sudden congestion, which results on higher packet delays and losses. In this thesis, a VoIP codec adaptation algorithm is proposed as a solution, based on a cross-layer feedback from RTCP packets and the MAC layer, which can adapt the codecs of active calls to adjust them to the multirate scenario. A combination of this algorithm with a call admission control mechanism is also studied. The results show an important improvement in terms of the QoS of the already active flows as also in the total hotspot's capacity. Additionally, by defining a new Grade of Service related parameter, the Q-Factor, which captures the trade-off between dropping and blocking ratio and perceived speech quality, the codec adaptation algorithm can be tuned to achieve maximum capacity without severely penalizing any of those variables, and hence satisfying both technical and user quality requirements. Finally, a new QoS-enabled AP, which implements these enhancements is designed.
En las redes inalámbricas del estándar IEEE 802.11, los mecanismos de adaptación de enlace que eligen la tasa de transmisión de cada nodo, pueden provocar variaciones aleatorias e inesperadas en la capacidad efectiva del canal. Cuando estos cambios son hacia tasas de transmisión mas bajas, los flujos inelásticos, tales como los de VoIP, pueden de repente sufrir congestión, lo que se traduce en aumento de retrasos y pérdidas de paquetes. En esa tesis, se propone un algoritmo de adaptación de codificadores de voz como solución, basado en técnicas multinivel (cross-layer) que combinan el uso de información de diferentes capas, como los paquetes RTCP y la capa MAC, y que puede adaptar los codecs de las llamadas activas para ajustarlos al escenario "multi-rate". Adicionalmente, la combinación de este algoritmo con un mecanismo de control de admisión de llamadas (CAC) se ha estudiado. Los resultados muestran una importante mejora en términos de QoS de los flujos activos como también en la capacidad total del hotspot. Además, mediante la definición de un nuevo factor, el Q-Factor, que puede captar la compensación entre la tasa de corte y de bloqueo de llamadas y de la calidad percibida por esas, el algoritmo de adaptación de codecs se puede ajustar para lograr la máxima capacidad sin penalizar severamente ninguna de esas variables y así satisfacer los requisitos técnicos de calidad y los usuarios. Por último, un nuevo punto de acceso (AP) habilitado para ofrecer calidad de servicio, ha sido diseñado que lleva a cabo estas mejoras.

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48

Vittorias, Iason [Verfasser], Sandra [Akademischer Betreuer] Hirche y Eckehard [Akademischer Betreuer] Steinbach. "Human-centered control design for haptic telepresence systems over communication networks / Iason Vittorias. Gutachter: Eckehard Steinbach ; Sandra Hirche. Betreuer: Sandra Hirche". München : Universitätsbibliothek der TU München, 2012. http://d-nb.info/1024161218/34.

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Ahmed, Sabbir. "Performance of Multi-Channel Medium Access Control Protocol incorporating Opportunistic Cooperative Diversity over Rayleigh Fading Channel". Thesis, Blekinge Tekniska Högskola, Avdelningen för signalbehandling, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-6171.

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This thesis paper proposes a Medium Access Control (MAC) protocol for wireless networks, termed as CD-MMAC that utilizes multiple channels and incorporates opportunistic cooperative diversity dynamically to improve its performance. The IEEE 802.11b standard protocol allows the use of multiple channels available at the physical layer but its MAC protocol is designed only for a single channel. The proposed protocol utilizes multiple channels by using single interface and incorporates opportunistic cooperative diversity by using cross-layer MAC. The new protocol leverages the multi-rate capability of IEEE 802.11b and allows wireless nodes far away from destination node to transmit at a higher rate by using intermediate nodes as a relays. The protocol improves network throughput and packet delivery ratio significantly and reduces packet delay. The performance improvement is further evaluated by simulation and analysis.
sabbir@linuxmail.org

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50

Kalfas, Georgios. "Medium-transparent MAC protocols for converged optical wireless networks". Doctoral thesis, Universitat Politècnica de Catalunya, 2017. http://hdl.handle.net/10803/406358.

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In order to address the explosive demand for high-capacity and omnipresent wireless access, modern cell-based wireless networks are slowly adopting two major solution roadmaps. The first is the employment of small-cell formations in order to increase the overall spectral efficiency, whereas the second is the employment of higher frequency bands, such as the mm-wave 60GHz band, that offers vast amounts of bandwidth. Depending on the specific application, the above solutions inevitably require the installation and operational management of large amounts of Base Stations (BSs) or Access Points (APs), which ultimately diminishes the overall cost-effectiveness of the architecture. In order to reduce the system cost, Radio over Fiber (RoF) technology has been put forward as an ideal candidate solution, due to the fact that it provides functionally simple antenna units, often termed as Remote Antenna Units (RAUs) that are interconnected to a central managing entity, termed as the Central Office (CO), via an optical fiber. Although extensive research efforts have been dedicated to the development of the physical layer aspects regarding RoF technologies, such as CO/RAU physical layer design and radio signal transport techniques over fiber, very limited efforts have con-centrated on upper layer and resource management issues. In this dissertation, we are concerned with access control and resource management of RoF-based mm-wave network architectures targeting the exploitation of the dual medium and its centralized control properties in order to perform optimal optical/wireless/time resource allocation. In this dissertation, we propose a Medium-Transparent MAC (MT-MAC) protocol that concurrently administers the optical and wireless resources of a 60GHz RoF based network, seamlessly connecting the CO to the wireless terminals through minimal RAU intervention. In this way, the MT-MAC protocol forms extended reach 60GHz WLAN networks offering connectivity amongst wireless devices that are attached to the same or different RAUs under both Line of Sight (LOS) and non LOS conditions. The notion of medium-transparency relies on two parallel contention periods, the first in the optical domain and the second in the wireless frequency and time domains, with nested dataframe structures. The MT-MAC operation is based on a proposed RAU design that allows for wavelength selectivity functions, thus being compatible with completely passive optical distribution network implementations that are predominately used by telecom operators today. Two variants of the MT-MAC protocol are considered. The first offers dynamic wavelength allocation with fixed time windows, whereas the second targets fairness-sensitive applications by offering dynamic wavelength allocation with dynamic transmission opportunity window sizes, based on the number of active clients connected at each RAU. Both variants of the protocol are evaluated by both simulation and analytical means. For the latter part, this thesis introduces two analytical models for calculating saturation throughput and non-saturation packet delay for the converged MT-MAC protocol. Finally, this thesis presents an extensive study regarding the network planning and formation of 60GHz Gigabit WLAN networks when the latter are deployed over existing Passive Optical Network (PON) infrastructures. Three possible architectures where studied: i) the RoF approach, ii) the Radio & Fiber approach and iii) the hybrid RoF-plus-R&F approach that combines the properties of both the aforementioned architectures. During the elaboration of this thesis, one major key conclusion has been extracted. The work proposed in this thesis considers that there is a fundamental requirement for implementing new converged optical/wireless MAC protocols, that have the complete overview of both available resources in order to effectively administer the hybrid Radio-over-Fiber networks.
A fin de atender la demanda explosiva de alta capacidad y acceso inalámbrico omnipresente, las redes inalámbricas basadas en celdas están poco a poco adoptando dos principales guías de solución. La primera es el empleo de formaciones de celdas pequeñas con el fin de aumentar la eficiencia espectral global, mientras que la segunda es el empleo de bandas de frecuencia superior, como la banda de 60GHz, la cual ofrece una gran cantidad de ancho de banda. Dependiendo de la aplicación en específico, las soluciones anteriores inevitable-mente requieren de una instalación y una gestión operativa de grandes cantidades de Estaciones Base o Puntos de Acceso, que en última instancia disminuye la rentabilidad de la arquitectura. Para reducir el coste, la tecnología radioeléctrica por fibra (RoF) se presenta como una solución ideal debido al hecho de que proporciona unidades de antenas de sim-ple funcionamiento, a menudo denominadas Unidades de Antenas Remotas (RAUs), las cuales están interconectadas a una entidad central de gestión, denominada Oficina Central (CO), a través de la fibra óptica. A pesar de que se han dedicado muchos esfuerzos de investigación al desarrollo de varios aspectos de las capas física con respecto a las tecnologías RoF, muy pocos esfuerzos se han concentrado en la capa superior y cuestiones de gestión de recursos. En esta tesis, nos enfocando en el control de acceso y gestión de recursos de arquitecturas RoF y comunicaciones milimétricas, con el fin de aprovechar y explotar el medio dual y las propiedades para realizar una óptima asignación de los recursos ópticos, inalámbricos y temporales. Nosotros proponemos un protocolo Transparente al Medio MAC (MT-MAC) que simultáneamente administre los recursos ópticos e inalámbricos de una red RoF a 60GHz, conectando a la perfección el CO a los terminales inalámbricos a través de una mínima intervención RAU. El protocolo MT-MAC forma unas redes WLAN 60GHz de alcance extendido, ofreciendo así conectividad entre los dispositivos inalámbricos que están conectados al mismo o diferentes RAUs bajo con o sin Línea de Vista (condiciones LOS o NLOS) respectivamente. La noción de transparencia al medio se basa en dos períodos de contención para-lelos, el primero en el dominio óptico y el segundo en la frecuencia inalámbrica y dominio del tiempo, con estructuras de datos anidados. La operación MT-MAC se basa en proponer un diseño RAU que permita la selectividad de funciones de longitud de onda. Dos variantes del protocolo MT- MAC son considerados; el primer ofrece asignación de longitud de onda dinámica con ventanas de tiempo fijo, mientras que la segunda tiene como objetivo entornos de aplicaciones sensibles ofreciendo asignación de longitud de onda con tamaño de ventana de oportunidad de transmisión dinámico, basado en el número de clientes conectados en cada RAU. Ambas variantes del protocolo están evaluadas tanto por medios analíticos como de simulación. En la segunda parte, esta tesis introduce dos modelos analíticos para calcular el rendimiento de saturación y no saturación del retardo de paquetes para el protocolo MT-MAC convergente. Finalmente, esta tesis presenta un extenso estudio de la planificación de red y la formación de redes 60GHz Gigabit WLAN cuando esta se encuentra desplegada sobre las ya existente infraestructuras de Redes Ópticas Pasivas (PONs). Tres posibles arquitecturas han sido estudiadas: i) el enfoque RoF, ii) el enfoque Radio y Fibra , y iii) el enfoque híbrido, RoF más R&F el cual combina las propiedades de ambas arquitecturas anteriormente mencionadas. Durante la elaboración de esta tesis, se ha extraído una importante conclusión: hay un requerimiento fundamental para implementar nuevos protocolos ópticos/inalámbricos convergentes, que tengan una completa visión de ambos recursos disponibles para poder administrar efectivamente las redes de tecnología RoF.

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Tesis sobre el tema "Control over networks"

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