Tactical radio communications used in military applications hasstrict requirements regarding security and has to be operable inrough environments in which there may be disturbances and disruptionson a radio link. The performance of the Secure CommunicationInteroperability Protocol (SCIP) operating in an asynchronouscommunication network with various levels of packet loss isinvestigated and found inadequate mainly due to problems withcryptographic synchronization between the transmitting and receivingunits. The introduction of additional counter data to each datapacket remedies this problem and allows the receiving units to fillthe holes left by packet losses with filler packets, maintainingsynchronization. The audio quality can then be measured using thePerceptual Evaluation of Speech Quality (PESQ) algorithm.Measurements are performed in an emulated radio link with aconfigurable packet loss ratio developed by Saab. The results showthat parts of SCIP can be used alongside the counter solution withoutimpacting the audio quality. The insertion of filler packets is shownto have a positive effect on the audio quality, while aggregation ofpackets to conserve transmission data rate is shown to have anegative effect.

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There is a demand for video quality measurements in modern video applications specifically in wireless and mobile communication. In real time video streaming it is experienced that the quality of video becomes low due to different factors such as encoder and transmission errors. HEVC/H.265 is considered as one of the promising codecs for compression of ultra-high definition videos. In this research, full reference based video quality assessment is performed. The raw format reference videos have been taken from Texas database to make test videos data set. The videos are encoded using HM9 reference software in HEVC format. Encoding errors has been set during the encoding process by adjusting the QP values. To introduce packet loss in the video, the real-time environment has been created. Videos are sent from one system to another system over UDP protocol in NETCAT software. Packet loss is induced with different packet loss ratios into the video using NETEM software. After the compilation of video data set, to assess the video quality two kind of analysis has been performed on them. Subjective analysis has been carried on different human subjects. Objective analysis has been achieved by applying five quality matrices PSNR, SSIM, UIQI, VFI and VSNR. The comparison is conducted on the objective measurement scores with the subjective and in the end results deduce from classical correlation methods.

<p>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 <em>packet loss concealment </em>scheme.</p><p>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.</p><p>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.</p><p>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.</p>

In wireless/mobile networks, users freely change their service point while they are communicating with other users. In order to support the mobility of mobile nodes, Mobile IPv6 (MIPv6) is proposed in IETF, in which an MN must inform its home agent the binding of its home address and the current care-of address (CoA). The home agent forwards packets to CoA when it receives packets for the MN. There is a significant problem in MIPv6 due to its inability to support micro-mobility cause by long delay and high packet loss during handover. Hierarchical mobile IPv6 (HMIPv6) is proposed to separate mobility into micro-mobility (within one domain) and macro-mobility (between domains). HMIPv6 introduced a new protocol element called Mobility Anchor Point (MAP) to manage the mobility. HMIPv6 can reduce the delay and the amount of signaling during handover. However HMIPv6 only improves micro mobility problem where the significant delay still occurs in macro mobility management. Duplicate address detection and the transmission time during the handoff operation could cause high delays. This paper considers handover operations. By simulations, we show that the proposed scheme can realize low handoff delay and packet loss during handover.

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.

In a video-on-demand streaming system, each user watch dufferent video frame according to their arrival time. In the thesis, the concept of multi-source and the forward error correction scheme are implemented to decrease the workload bandwidth of server and reduce the packet loss probability of each peer in the peer-to-peer video-on-demand system. Here, in order to share data to some peers arrive later, each peer must cache part of video data that recently viewed. Each new arrival peer needs to contact multiple sources who have initial part of video data to get video streaming data, and each source transmits different part of video streaming. Once receiving all of these partial stream, the peer will get completely video data. Simulation shows that a suitable preserve time of peers in the system can be used. Thus, the workload bandwidth of server used and memory used by peers can be reduced, too. Furthermore, the packet loss probability is decreased by the sources diversity and the FEC recovery.

Today’s technology has a strong tendency towards automation. As a matter of fact, the tremendous improvement of science in the recent years brought new ideas regarding to accelerate the scientific process that is not separated of automation. This paper also deals with automation of manual tests that were used to analyze packet loss and packet loss duration in a network. These two metrics were chosen based on their importance they have in the communication technology, also based on the weak points that were found in the manual processes. This experiment is done in the Ixia platform that was an appropriate test bed to design an automation framework.After a comprehensive research on network and communication we could choose packet loss and packet loss duration as two important parameters that are under test several times per day. Therefore, based on the properties that are used for automation, these two metrics had the priority compare to other metrics. We could create a framework that works correspond to the manual test process. For this purpose, Tcl programming language is used. The script that was written with this high-level language can communicate with the graphical user interface, configuring all the connected devices, measuring mentioned metrics and ultimately save the result in a csv file.Finally, we could reach to the main objective of this project which was to show how positively automatic method can affect on the quality of test in terms of accuracy, time and manpower saving.

This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality.