Communication over today’s IP-based networks are to some extent subject to packet loss. Most real-time applications, such as video streaming, need methods to hide this effect, since resending lost packets may introduce unacceptable delays. For IP-based video streaming applications such a method is referred to as a packet loss concealment scheme. In this thesis a recently proposed mixture model and least squares-based packet loss concealment scheme is implemented and evaluated together with three more well known concealment methods. The JM reference software is used as basis for the implementation, which is a public available software codec for the H.264 video coding standard. The evaluation is carried out by comparing the schemes in terms of objective measurements, subjective observations and a study with human observers. The recently proposed packet loss concealment scheme shows good performance with respect to the objective measures, and careful observations indicate better concealment of scenes with fast motion and rapidly changing video content. The study with human observers verifies the results for the case when a more sophisticated packetization technique is used. A new packet loss concealment scheme, based on joint modeling of motion vectors and pixels, is also investigated in the last chapter as an additional contribution of the thesis.

The Internet supports a diverse number of applications, which have different requirements for a number of services. Next generation networks provide high speed connectivity between hosts, which leaves the service provider to configure network devices appropriately, in order to maximize network performance. Service provider settings are based on best recommendation parameters, which give an opportunity to optimize these settings even further. This dissertation focuses on a packet discarding algorithm, known as random early detection (RED), to determine parameters which will maximize utilization of a resource. The two dominant traffic protocols used across an IP backbone are UDP and TCP. UDP traffic flows transmit packets regardless of network conditions, dropping packets without changing its transmission rates. However, TCP traffic flows concern itself with the network condition, reducing its packet transmission rate based on packet loss. Packet loss indicates that a network is congested. The sliding window concept, also known as the TCP congestion window, adjusts to the amount of acknowledgements the source node receives from the destination node. This paradigm provides a means to transmit data across the available bandwidth across a network. A well known and widely implemented simulation environment, the network simulator 2 (NS2), was used to analyze the RED mechanism. The network simulator 2 (NS2) software gained its popularity as being a complex networking simulation tool. Network protocol traffic (UDP and TCP) characteristics comply with theory, which verifies that the traffic generated by this simulator is valid. It is shown that the autocorrelation function differs between these two traffic types, verifying that the generated traffic does conform to theoretical and practical results. UDP traffic has a short-range dependency while TCP traffic has a long-range dependency. Simulation results show the effects of the RED algorithm on network traffic and equipment performance. It is shown that random packet discarding improves source transmission rate stabilization, as well as node utilization. If the packet dropping probability is set high, the TCP source transmission rates will be low, but a low packet drop probability provides high transmission rates to a few sources and low transmission rates to the majority of other sources. Therefore, an ideal packet drop probability was obtained to complement TCP source transmission rates and node utilization. Statistical distributions were fitted to sampled data from the simulations, which also show improvements to the network with random packet discarding. The results obtained contribute to congestion control across wide area networks. Even though a number of queuing management implementation exists, RED is the most widely used implementation used by service providers. / Dissertation (MEng)--University of Pretoria, 2013. / Electrical, Electronic and Computer Engineering / unrestricted

In this thesis, we investigate two different system infrastructures in underlay cognitive radio network, in which two popular techniques, cross-layer design and cooperative communication, are considered, respectively. In particular, we introduce the Aggressive Adaptive Modulation and Coding (A-AMC) into the cross-layer design and achieve the optimal boundary points in closed form to choose the AMC and A-AMC transmission modes by taking into account the Channel State Information (CSI) from the secondary transmitter to both the primary receiver and the secondary receiver. What’s more, for the cooperative communication design, we consider three different relay selection schemes: Partial Relay Selection, Opportunistic Relay Selection and Threshold Relay Selection. The Probability Density Functions (PDFs) of the Signal-to- Noise Ratio (SNR) in each hop for different selection schemes are provided, and then the exact closed-form expressions for the end-to-end packet loss rate in the secondary link considering the cooperation of the Decode-and-Forward (DF) relay for different relay selection schemes are derived.

The formal definition of network coding by Ahlswede et al. in 2000 has led to several breakthroughs in information theory, for example solving the bottleneck problem in butterfly networks and breaking the min-cut max-flow theorem for multicast communication. Especially promising is the usage of network coding as a packet-level Forward Error Correction (FEC) scheme to increase the robustness of a data stream against packet loss, also known as intra-session coding. Yet, despite these benefits, network coding-based FEC is still rarely deployed in real-world networks. To bridge this gap between information theory and real-world usage, this cumulative thesis will present our contributions to the integration, evaluation, and optimization of network coding-based FEC. The first set of contributions introduces and evaluates efficient ways to integrate coding into UDP-based IoT protocols to speed up bulk data transfers in lossy scenarios. This includes a packet-level FEC extension for the Constrained Application Protocol (CoAP) [P1] and one for MQTT for Sensor Networks (MQTT-SN), which levels the underlying publish-subscribe architecture [P2]. The second set of contributions addresses the development of novel evaluation tools and methods to better quantify possible coding gains. This includes link ’em, our award-winning link emulation bridge for reproducible networking research [P3], and also SPQER, a word recognition-based metric to evaluate the impact of packet loss on the Quality of Experience of Voice over IP applications [P5]. Finally, we highlight the impact of padding overhead for applications with heterogeneous packet lengths [P6] and introduce a novel packet-preserving coding scheme to significantly reduce this problem [P4]. Because many of the shown contributions can be applied to other areas of network coding research as well, this thesis does not only make meaningful contributions to specific network coding challenges, but also paves the way for future work to further close the gap between information theory and real-world usage.

Control systems utilizing wireless sensor and actuator networks can be severely affectedby the properties of the communication links. Radio fading and interferencemay cause communication losses and outages in situations when the radio environmentis noisy and low transmission power is desirable. This thesis proposes amethod to compensate for such unpredictable losses of data in the feedback controlloop by introducing a predictive outage compensator (POC). The POC is a filter tobe implemented at the receiver sides of networked control systems where it generatesartificial samples when data are lost. If the receiver node does not receive thedata, the POC suggests a command based on the history of past data. It is shownhow to design, tune and implement a POC. Theoretical bounds and simulationresults show that a POC can improve the closed-loop control performance undercommunication losses considerably. We provide a deterministic and a stochasticmethod to synthesize POCs. Worst-case performance bounds are given that relatethe closed-loop performance with the complexity of the compensator. We also showthat it is possible to achieve good performance with a low-order implementationbased on Hankel norm approximation. Tradeoffs between achievable performance,communication loss length, and POC order are discussed. The results are illustratedon a simulated example of a multiple-tank process. The thesis is concludedby an experimental validation of wireless control of a physical lab process. Herethe controller and the physical system are separated geographically and interfacedthrough a wireless medium. For the remote control we use a hybrid model predictivecontroller. The results reflect the difficulties in wireless control as well as theyhighlight the flexibility and possibilities one obtains by using wireless instead of awired communication medium. / VR, SSF, VINNOVA via Networked Embedded Control Systems, EU Sixt Framework Program via HYCON and SOCRADES

A packet buffer for the protocol processor is a large memory space that holds incoming data packets for an application. Data packets for each application are stored in the form of FIFO queues in the packet buffer. Packets are dropped when the buffer is full. An efficient buffer management algorithm is required to manage the buffer space among the different FIFO queues and to avoid heavy packet loss. This thesis develops a simulation model for the packet buffer and studies the performance of conventional buffer management algorithms when applied to packet buffer. This thesis proposes a new buffer management algorithm, Dynamic Algorithm with Different Thresholds (DADT) to improve the packet loss ratio. This algorithm takes advantage of the different packet sizes for each application and proportionally allocates buffer space for each queue. The performance of the DADT algorithm is dependent upon the packet size distribution in a network traffic load. Three different network traffic loads are considered for our simulations. For the average network traffic load, the DADT algorithm shows an improvement of 6.7 % in packet loss ratio over the conventional dynamic buffer management algorithm. For the high and actual network traffic loads, the DADT algorithm shows an improvement of 5.45 % and 3.6 % in packet loss ratio respectively. Based on the simulation results, the DADT algorithm outperforms the conventional buffer management algorithms for various network traffic loads.

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The Internet of Musical Things (IoMusT) is an emerging paradigm that envisions ecosystems of interconnected smart musical instruments and devices dedicated to the production and reception of musical content. Realizing the IoMusT vision requires addressing key challenges in real-time audio streaming over networks, including ultra-low latency, high reliability, and perceptual audio quality. One of the central components of the IoMusT paradigm is represented by Networked Music Performance (NMP) systems, which aim at enabling geographically displaced musicians to play together in a realistic way over the network. Prior to this thesis, there was a lack of comprehensive studies evaluating the ability of fourth-generation (4G), fifth-generation (5G), and transitional 4G/5G networks to support the strict Quality of Service (QoS) requirements of NMP and IoMusT applications in realistic multi-user scenarios. On the device side, a knowledge gap existed in quantitatively comparing the real-time audio performance of different embedded Linux architectures for IoMusT devices. Furthermore, tools and frameworks were needed to systematically evaluate and compare Packet Loss Concealment (PLC) algorithms, which are crucial for maintaining audio quality under inevitable network losses, especially considering the unique challenges posed by musical signals. The fundamental objective of this thesis is to address these research gaps in order to advance the state-of-the-art in real-time audio streaming over Internet Protocol (IP) networks for musical applications. The key research questions investigated include: Can 4G, 5G, and mixed 4G/5G networks meet the demanding latency, reliability, and quality requirements of IoMusT applications in multi-user scenarios?How do different embedded Linux architectures compare in terms of real-time audio performance metrics relevant for IoMusT devices?How can PLC algorithms be systematically evaluated and compared, taking into account the perceptual aspects of musical signals?Can perceptual aspects be taken into account to design better metrics and loss functions for PLC in NMP?To answer these questions, this thesis conducts realistic network simulations, develops opensource tools, and explores novel techniques. The outcomes demonstrate that 5G significantly improves IoMusT support compared to 4G, provide quantitative insights to guide platform selection for IoMusT devices, deliver a modular framework for evaluating PLC methods for NMP systems, and propose perceptually-motivated approaches to enhance concealment algorithm design. Overall, the research presented in this thesis advances the understanding of real-time audio streaming over modern wireless networks and provides concrete tools and techniques to support the realization of the IoMusT vision, enabling new forms of distributed musical experiences and collaborations.

Vehicular communication systems are integral for efficient highway operational management and for mitigating severe traffic congestion. While vehicular ad hoc networks (VANET) are reliable and provide avenues to minimal reliance on existing infrastructure, they can experience high communication overhead and network disruptions. Vehicular micro clouds (VMCs) provide a promising solution to overcome the challenges of VANET by reducing communication latency through cooperative and collaborative resource allocation and data offloading. This thesis offers a comparative performance analysis of freeway incident management and vehicle platooning, comparing VANET communications versus stationary and platoon-based dynamic VMCs. Specifically, it studies speed and lane-changing advisories in addition to freeway platooning. To further enhance the analysis, the performance of both communication architectures is evaluated using communication protocols of DSRC versus cellular technologies of C-V2X, 4G LTE, and 5G NR for latency, bandwidth, range, and deployment considerations. The system-level features, such as driving safety and vehicular mobility are measured to evaluate the efficacy of the communication systems under incident-induced traffic conditions. The study uses the AIMSUN microscopic traffic simulator to model and analyze the performance of the proposed systems. Key performance indicators include communication latency and packet loss ratio. In addition, the stationary and dynamic cloud systems show advantages in reducing travel time delay, even at high penetration rates of the connected vehicles, whilst reducing collision risks. On average, we observe improvements in travel time by 10% by implementing vehicular clouds over traditional ad-hoc networks. From a communications standpoint, the overall latency delay and packet loss are reduced by 7% and 11%, respectively, with the implementation of cloud models. The findings also delineate the benefits of dynamic cloud models, given their improved manoeuvrability, can maximize the computational capabilities of CVs, even at high market penetrations in large-scale freeway demands. The results suggest a shift towards more reliance on connected vehicular clouds to minimize the risks associated with message interference and system overload, whilst fostering advancements in intelligent freeway traffic management systems. / Thesis / Master of Applied Science (MASc)

Reliable and efficient data transfer on the Internet is an important issue. Since late 70's the protocol responsible for that has been the de facto standard TCP, which has proven to be successful through out the years, its self-managed congestion control algorithms have retained the stability of the Internet for decades. However, the variety of existing new technologies such as high-speed networks (e.g. fibre optics) with high-speed long-delay set-up (e.g. cross-Atlantic links) and wireless technologies have posed lots of challenges to TCP congestion control algorithms. The congestion control research community proposed solutions to most of these challenges. This dissertation adds to the existing work by: firstly tackling the highspeed long-delay problem of TCP, we propose enhancements to one of the existing TCP variants (part of Linux kernel stack). We then propose our own variant: TCP-Gentle. Secondly, tackling the challenge of differentiating the wireless loss from congestive loss in a passive way and we propose a novel loss differentiation algorithm which quantifies the noise in packet inter arrival times and use this information together with the span (ratio of maximum to minimum packet inter arrival times) to adapt the multiplicative decrease factor according to a predefined logical formula. Finally, extending the well-known drift model of TCP to account for wireless loss and some hypothetical cases (e.g. variable multiplicative decrease), we have undertaken stability analysis for the new version of the model.