This dissertation considers packet-switched wireless networks for transmission of variable-rate layered hybrid video streams. Target applications are video streaming and broadcasting services. The work can be divided into two main parts. In the first part, a novel quality-scalable scheme based on coefficient refinement and encoder quality constraints is developed as a possible extension to the video coding standard H.264. After a technical introduction to the coding tools of H.264 with the main focus on error resilience features, various quality scalability schemes in previous research are reviewed. Based on this discussion, an encoder decoder framework is designed for an arbitrary number of quality layers, hereby also enabling region-of-interest coding. After that, the performance of the new system is exhaustively tested, showing that the bit rate increase typically encountered with scalable hybrid coding schemes is, for certain coding parameters, only small to moderate. The double- and triple-layer constellations of the framework are shown to perform superior to other systems. The second part considers layered code streams as generated by the scheme of the first part. Various error propagation issues in hybrid streams are discussed, which leads to the definition of a decoder quality constraint and a segmentation of the code stream to transmit. A packetization scheme based on successive source rate consumption is drafted, followed by the formulation of the channel code rate optimization problem for an optimum assignment of available codes to the channel packets. Proper MSE-based error metrics are derived, incorporating the properties of the source signal, a terminate-on-error decoding strategy, error concealment, inter-packet dependencies, and the channel conditions. The Viterbi algorithm is presented as a low-complexity solution to the optimization problem, showing a great adaptivity of the joint source channel coding scheme to the channel conditions. An almost constant image qualiity is achieved, also in mismatch situations, while the overall channel code rate decreases only as little as necessary as the channel quality deteriorates. It is further shown that the variance of code distributions is only small, and that the codes are assigned irregularly to all channel packets. A double-layer constellation of the framework clearly outperforms other schemes with a substantial margin. Keywords — Digital lossy video compression, visual communication, variable bit rate (VBR), SNR scalability, layered image processing, quality layer, hybrid code stream, predictive coding, progressive bit stream, joint source channel coding, fidelity constraint, channel error robustness, resilience, concealment, packet-switched, mobile and wireless ATM, noisy transmission, packet loss, binary symmetric channel, streaming, broadcasting, satellite and radio links, H.264, MPEG-4 AVC, Viterbi, trellis, unequal error protection

The aim of this thesis is to optimize the design of the Fat-IBC-based communication of a novel neuroprosthetic system in which a brain-machine interface is used to control a prosthetic arm. Fat-based intra-body communication (Fat-IBC) uses the fat tissue inside the body of the bearer as a transmission medium for low-power microwaves. Future projects will use the communication system and investigate ways to control the prosthetic arm directly from the brain. The finished system was able to individually control all movable joints of multiple prosthesis prototypes using information that was received wirelessly through Fat-IBC. Simultaneous transmission in the other direction was possible, with the control data then being replaced by sensor readings from the prosthesis. All data packets were encoded with the COBS/R algorithm and the wireless communication was handled by Digi Xbee 3 radio modules using the IEEE 802.15.4 protocol at a frequency of 2.45 GHz. The Fat-IBC communication was evaluated with the help of so-called "phantoms" which emulated the conditions of the human body fat channel. During said testing, packet loss measurements were performed for various combinations of packet sizes and time intervals between packets. The packet loss measurements showed that the typical amount of transmitted data could be handled well by the fat channel test setup. Although the transmission system was found to be well-functioning in its current state, increasing the packet size to achieve a higher granularity of the movement was perceived to be viable considering the findings from the packet loss measurements.

Syftet med detta examensarbete är att och utvärdera undersöka analysera och utvärdera prestandan hos WebRTC-protokollet, samt att utveckla en webbaserad klient med hjälp av JavaScript för distribuerade system och demonstrera protokollets användbarhet i ett verkligt scenario. Studien inkluderade användning av olika verktyg och bibliotek, såsom Socket.IO, Node.js, Express.js och PeerJS. De viktigaste prestandaindikatorerna som utvärderades var latens/tur- och returtid (RTT), jitter och paketförlust. Implementationen testades både lokalt och på distans. Prestandatestningen av applikationen utfördes med hjälp av webbplatserna "Chrome webrtc-internals" och "TestRTC", vilka erbjöd detaljerade insikter och statistik om WebRTC-prestanda. Resultaten indikerade att WebRTC erbjuder högpresterande och kostnadseffektiv realtidskommunikation som är kompatibel med andra applikationer som stöder protokollet. Protokollet visade sig ha robusta säkerhetsåtgärder, vara kompatibelt med distribuerade system och erbjuda stark prestanda när det gäller latens, jitter och paketförlust. Studien drog slutsatsen att WebRTC, med sin skalbarhet och förmåga att erbjuda kommunikation i realtid, är ett fördelaktigt val för distribuerade system och webbaserade videochattapplikationer. Resultaten uppmanar till ytterligare undersökningar inom områden som end-to-end-kryptering och integration av artificiell intelligens för att förbättra systemets prestanda och säkerhet. / The aim of this thesis is to analyse and evaluate the performance of the WebRTC protocol, develop a web-based client using JavaScript for distributed systems, and demonstrate the utility of the protocol in a real-world scenario. The study involved the use of various tools and libraries, including Socket.IO, Node.js, Express.js, and PeerJS. Key performance indicators evaluated were latency/round-trip time (RTT), jitter, and packet loss. The implementation was tested both locally and remotely. Performance testing of the application was conducted using the "Chrome webrtc-internals" and "TestRTC" websites, which provided detailed insights and statistics on WebRTC performance. The results indicated that WebRTC offers high-performance and cost-effective real-time communication that is compatible with other applications supporting the protocol. The protocol demonstrated robust security measures, compatibility with distributed systems, and strong performance in terms of latency, jitter, and packet loss. The study concluded that WebRTC, with its scalability and ability to provide real-time communication, is a beneficial choice for distributed systems and webbased video chat applications. The findings encourage further investigations in areas such as end-to-end encryption and the integration of artificial intelligence to enhance system performance and security.

Airplanes have different ways to connect to the ground, including satellite air-to-ground communication (SA2GC) and direct air-to-ground communication (DA2GC). Each connection/link offers a different varying amount of transmission capacity over flight time. The traffic generated in the airplane must be forwarded/sent to ground over the available links. It is however not clear how the traffic should be forwarded so that traffic quality of service (QoS) requirements are met. The thesis at hand considers this question, and implements an algorithm handling the forwarding decision with three different forwarding schemes. Those consider traffic parameters in calculating a value assigned to each traffic flow, over a combination of priority, delay requirement and the number of times a traffic flow is dropped. The forwarding algorithm relies on proposed in-flight broadband connectivity (IFBC) network traffic and air-to-ground (A2G) link models, which aim at approximating the network environment of future IFBC networks. It is shown that QoS requirements of traffic flows in terms of packet loss and delay cannot be satisfied with capacities offered by current DA2GC and SA2GC technology. For a future scenario, with higher assumed link capacities, the QoS requirements are met to a higher extent. This is shown in lower packet loss and delay experienced by the respective traffic flows. Further, it is shown that the performance can be improved with specific forwarding schemes used by the forwarding algorithm. It is also investigated how a web cache can be used as a fallback technology. For this a required web cache hit rate is found, which should be high enough to offload the network with content served from the cache. Overall, the thesis aims at proposing an efficient traffic forwarding technique, and at giving insight into an alternative if this technique fails. / Flygplan har olika sätt att ansluta till marken, inklusive satellit-mark-kommunikation (SA2GC) och direkt luft till markkommunikation (DA2GC). Varje anslutning/länk erbjuder en annan varierande mängd överföringskapacitet under flygtid. Den trafik som genereras i flygplanet måste vidarebefordras/skickas till marken över de tillgängliga länkarna. Det är emellertid inte klart hur trafiken ska vidarebefordras så att trafiksäkerhetskvaliteten (QoS) uppfylls. Avhandlingen handlar om denna fråga och implementerar en algoritm som hanterar vidarebefordringsbeslutet med tre olika vidarebefordringssystem. De betraktar trafikparametrar vid beräkning av ett värde som tilldelas varje trafikflöde, över en kombination av prioritet, fördröjningskrav och antalet gånger ett trafikflöde tappas. Vidarebefordringsalgoritmen är beroende av föreslagna bredbandsförbindelser (IFBC) i nätverk och A2G-länkmodeller, som syftar till att approximera nätverksmiljön för framtida IFBC-nätverk. Det visas att QoS-krav på trafikflöden när det gäller paketförlust och fördröjning inte kan tillgodoses med kapacitet som erbjuds av nuvarande DA2GC- och SA2GC-teknik. För ett framtida scenario, med högre antagna länkkapacitet, uppfylls QoS-kraven i högre utsträckning. Detta visas med lägre paketförlust och fördröjning som upplevs av respektive trafikflöden. Vidare är det visat att prestanda kan förbättras med specifika vidarekopplingsscheman som används av vidarebefordringsalgoritmen. Det undersöks också hur en webbcache kan användas som en återgångsteknik. För detta hittas en obligatorisk webbcache-träfffrekvens, som bör vara tillräckligt hög för att ladda upp nätverket med innehåll som serveras från cacheminnet. Sammanfattningsvis syftar uppsatsen till att föreslå en effektiv trafiköverföringsteknik och att ge insikt om ett alternativ om denna teknik misslyckas.

This thesis is focused on network behavior analysis (NBA) designed to detect network attacks. The goal of the thesis is to increase detection accuracy of obfuscated network attacks. Methods and techniques used to detect network attacks and network traffic classification were presented first. Intrusion detection systems (IDS) in terms of their functionality and possible attacks on them are described next. This work also describes principles of selected attacks against IDS. Further, obfuscation methods which can be used to overcome NBA are suggested. The tool for automatic exploitation, attack obfuscation and collection of this network communication was designed and implemented. This tool was used for execution of network attacks. A dataset for experiments was obtained from collected network communications. Finally, achieved results emphasized requirement of training NBA models by obfuscated malicious network traffic.

This thesis discuss about the problem with mobility of stations in IP networks, concretely protocol Mobile IP and about the problems with this protocol in MANET networks, with the scope on Motorola MESH. First part is about design integration of protocol Mobile IP in this networks with usage Tropos 5210 MetroMesh routers and Cisco components (router, switch) to design and configure wireless MESH network, than connect this network with Cisco components and try to implement Mobile IP into this network. Second part is about design and configure wireless network from Cisco Wireless Access Points and about succesfully implementation of Mobile IP protocol into this network. In the last part is some tests of the function and quality of this topology.