This master thesis deals with implementing triple-play services and providing it’s quality of services in heterogenous comunnication networks. The aim of thesis is to apply the- oretical methods in real case and existing network infrastructure. Practical part aims to create technical solution to prioritize network traffic based on classification of required services. The technical solution is created with conditions of being possible to add more functions and to provide scalability.

The main goal of this thesis is to design videoconferential system by Java Media Framework. This means to write two programs. One will capture multimedia data from webcam and will sent it into network. The second application which can receive this data is supposed to be on another computer. The multimedia data are transmitted by using Java Media Framework library. We suppose one lector and group of students. Video and audio channel is transmitted to all students. Communication between lector and student  is solved using chat. Lector has possibility open another audio or video channel with all students. Multimedia data will be transferred using Real-time Transport Protocol (RTP). The message including information about student is transferred by Transmission Control Protocol (TCP).

In recent years, there has been an exponential growth in the use of multimedia applications. A satellite system offers great potential for multimedia applications with its ability to broadcast and multicast a large amount of data over a very large area as compared to a terrestrial system. However, the limited transmission capacity along with the dynamically varying channel conditions impedes the delivery of good quality multimedia service in a satellite system which has resulted in research efforts for deriving efficient radio resource management techniques. This issue is addressed in this thesis, where the main emphasis is to design a CAC framework which maximizes the utilization of the scarce radio resources available in the satellite and at the same time increases the performance of the system for a UMTS based satellite system supporting unicast and multicast traffic. The design of the system architecture for a UMTS based satellite system is presented. Based on this architecture, a CAC framework is designed consisting of three different functionalities: the admission control procedure, the retune procedure and the pre-emption procedure. The joint use of these functionalities is proposed to allow the performance of the system to be maintained under congestion. Different algorithms are proposed for different functionalities; an adaptive admission control algorithm, a greedy retune algorithm and three pre-emption algorithms (Greedy, SubSetSum, and Fuzzy). A MATLAB simulation model is developed to study the performance of the proposed CAC framework. A GUI is created to provide the user with the flexibility to configure the system settings before starting a simulation. The configuration settings allow the system to be analysed under different conditions. The performance of the system is measured under different simulation settings such as enabling and disabling of the two functionalities of the CAC framework; retune procedure and the pre-emption procedure. The simulation results indicate the CAC framework as a whole with all the functionalities performs better than the other simulation settings.

QoS routing is one of the major building blocks for supporting QoS in communication networks and, hence, a necessary component of future communication networks. Bandwidth- Constrained Routing Algorithm (BCRA) may help to satisfy QoS requirements such as end-to-end delay, delay-jitter etc when WFQ-like (Weighted Fair Queuing) scheduling mechanisms are deployed. The existing algorithms for bandwidth constrained routing suffer from high message overhead and have a high computational and space complexity. The work presented in the thesis, therefore, focuses on the different techniques that an be used to reserve bandwidth for a unicast connection with low protocol overhead in terms of number of messages. We have compared the performance of the proposed routing algorithms using simulation studies with other bandwidth constrained routing algorithms. The call blocking ratio and message overhead have been used as the performance metric to compare the proposed algorithm with the existing ones. We present three source routing algorithms for unicast connections satisfying the band- width requirement. The first two routing algorithms are based on the partitioning of the network. The link-state broadcasts are limited to the partition. In the first algorithm, the source node queries the other partitions for the state information on a connection request and computes the path based on the information received from the other partitions. The second algorithm is based on state aggregation. The aggregated state of other partitions is maintained at every node. The source node finds a feasible path based on the aggregated information. The path is expanded in every partition, if required, at the time of resource reservation. The third QoS routing algorithm uses the Distance Vector Tables to find a route for a connection. If the shortest path satisfies the bandwidth requirement, then it is selected; otherwise a random deviation is taken at the point where bandwidth requirement is not satisfied and shortest path algorithm is again followed. In all the three algorithms presented, the packets carry the entire path information to the destination node. Therefore, no per connection information is required to be maintained at the intermediate nodes. Simulation results indicate that the proposed algorithms indeed help educing the protocol overhead considerably, and at the same time they give comparable or better performance in terms of resource utilization across a wide range of workloads.

Internet Protocol Television (IPTV) transmits the television signal over the TCP/IP family of protocols. Its advantages are for example that the transmitting is not only one-way as in the “classical” TV broadcasting, but it can provide feedback such as interactivity. There are also some problems which avoids development, for example low channel capacity of access networks. That is why new methods , for example how to get more efficiency in IPTV transmission, are proposed. The main task of this diploma thesis is to visualize tree structure of relations between nodes in the network, based on understanding of principles of the hierarchical summarization and IPTV transmitting. The nodes in the tree structure provide computing and summarizing of the data in back-way channel. There is the data from the end users in this channel. In the first part of this diploma thesis there is explained the principle of IPTV and its differences as compared with classical TV broadcasting. The part is also aimed for the supported services, advantages and disadvantages. There is explained the compressing data with the MPEG-2 and MPEG-4 standards and problems with transport networks called “last mile problem.” To transmitting data IPTV uses Source Specific Multicast – every user connects to the multicast session with requested TV program. Feedback is provided by unicast. Feedback network uses the hierarchical summarization principle to reduce the data. This problem, connected with RTP, RTCP and TTP protocols, is in the work described as well. There is an international experimental network called PlanetLab mentioned in theoretical part of this work. Proposed structure of new protocol and applications including the visualization for IPTV broadcast, is tested in that network. In the practical part of this work there are discussed possibilities and methods for the visualization and data storage. Because of high availability and flexibility, there were chosen web technologies, such as MySQL for data storage. The tree model is executed by Java. The visualization is solved by web technologies, source code for visualization is dynamically generated by scripts in JSP (Java Server Pages). Graphical output is provided by the vector format SVG (Scalable Vector Graphics) which is created for graphical expression on the internet and in the mobile phones. There were created interactive web application thanks its ability to cooperation with Javascript technology. This application visualizes relation-tree structure of nodes. In this work there are explained basics of all used technologies, there are also given reasons for chosen methods and formats. Examples and interesting parts of solution are mentioned as well.

ITC/USA 2009 Conference Proceedings / The Forty-Fifth Annual International Telemetering Conference and Technical Exhibition / October 26-29, 2009 / Riviera Hotel & Convention Center, Las Vegas, Nevada / The Central Test and Evaluation Investment Program (CTEIP) Integrated Network Enhanced Telemetry (iNET) program is currently developing new standards for wired-wireless local area networking (LAN-WLAN) using the Internet Protocol (IP), for use in telemetry (TM) channels, under the umbrella of the Telemetry Network System (TmNS). Some advantages of TmNS are real-time command and control of instrumentation, quick-look acquisition, data retransmission and recovery ('gapless TM' or 'PCM backfill'), data segmentation, etc. The iNET team is developing and evaluating prototypes, based on commercial 802.x and other technologies, in conjunction with Range Commander's Council (RCC) Inter-Range Instrumentation Group (IRIG) standards and standards developed under the iNET program.

Nowadays the Learning Management System (LMS) platforms provide limited bidirectional, interactive mechanisms that they are competent to handle personal or small-scale distance learning systems. These mechanisms are designed for one to many online tutorials, and the technology utilizes single-input by single-output video stream technology, the video and audio data need to be coupled with one or many Multipoint Control Units (MCU) to mix or convert them into a single output media stream. In this platform MCU is critical to LMS, however, such system is expensive, lack of capacity and difficult to be massively deployed. Access-Grid (AG), an Open Source program, offers users capability to watch online multimedia audio-video contents from all the interconnected nodes of LMS through Multicast protocol, and supports groups-to-group high quality interactive distance learning. It requires all the networks to support the Multicast protocol. The MBONE (Multicast Backbone) can be used to connect different Multicast groups via Unicast communication. However, if the number of groups involving in the distance learning are large, the host computers or routers of the network will be heavily loaded because they need to handle the delivering of the media packets. To use a QuickBridge for aggregating and delivering packages is an alternative of LMS and requires (N-1) *N *BW bandwidth . For example, if there is a 15 nodes online conference and each node uses 800kbps data rate to transmit audio-video contents, then the demanded bandwidth of the aggregation is 168 Mbps. The way of dispersing and controlling the data flow becomes important factors and will greatly affect the quality of the AG online conference. This thesis modifies the procedure of AG and QuickBridge, and allows all the AG Clients to be able to transmit Unicast and Multicast packets in the online conference. It offers a Meeting Management Server to dynamically adjust topology and hub points, and achieves better elasticity to the system. By modifying VIC and RAT procedure, the system controls the outbound audio-video data flow from each nodes of online conference, and reduces the demand of bandwidth. The system can directly provide end-to-end conferencing, using Unicast communication to connect the nodes in different Multicast groups, or using the Multicast on the backbone and then using unicast communication to the local nodes. The functionality of the LMS can be improved and capable of supporting multi-windows to multi-user interactive online conference for the users. The results of this thesis can be applied upon real-time interactive distance learning, online video conferencing and interactive online TV. It also helps to lower the cost of the system and reduce the requirement of network bandwidth.

Made available in DSpace on 2015-05-14T12:36:36Z (GMT). No. of bitstreams: 1 ArquivoTotalElenilson.pdf: 4384305 bytes, checksum: 9256e23f2f52ecd3f4d8262f8ada2941 (MD5) Previous issue date: 2013-03-22 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior - CAPES / The Information and Communications Technology (ICT) field has changed due to both universalization of connectivity and popularization of devices. Such events have provided for the appearance of new networks which are able to send larges amounts of data and also on a faster way. The broadband networks support the development of new technologies and then more data can be generated and sent by these networks. When these technologies are applied to medical or military domains, for example, one of the main issues is the integrity and confidentiality of the data being transmitted over the network. This work presents a privacy and security strategy based on authentication, verification of users authenticity and distribution of encrypted streams. This paper is divided into two parts: the first one describes a strategy for multimedia reflectors which send AES-encrypted streams and are able to perform many types of media encoding. The second part describes a security server which authenticates, creates user groups and distributes group keys only to authenticated and permitted users. It uses Federated Authentications provided by the Federated Academic Community (CAFe). For the matter of comparison of the present strategy with previous studies, a systematic mapping was done to discuss the techniques for authentication and media transmission. In order to validate the work, the whole strategy was integrated to the Arthron - a tool for multimedia transmissions. The testing scheme used the 2k factorial designs and it analyzed the influence of the processing, encrypting and coding in the time transmission. / A área de Tecnologia da Informação e Comunicação (TIC) tem sofrido uma notável transformação caracterizada pela universalização das formas de conectividade e popularização de dispositivos. Tais fatores propiciaram o surgimento de novas redes de transmissão para lidar com grandes volumes de dados e com grande poder de transmissão , o q ue leva o rápido desenvolvimento de várias tecnologias multimídia onde cada vez mais dados são gerados e transmitidos pela rede. Quando aplicadas a domínios médicos, comerciais e militares, por exemplo, um dos principais pontos a serem considerados é a int egridade e confidencialidades d esses dados . Assim, este trabalho apresenta uma estratégia de segurança com ênfase em confidencialidade , baseada na autenticação e verificação de autenticidade de usuários, além da distri buição de fluxos criptografados . Para atingir tais objetivos, foi dividido em duas partes : uma que descreve uma estratégia de refletores para transmissão de fluxos com criptografia AES e múltiplas codificações e outra que demonstra um servidor de segurança para fazer autenticação, criação de g rupos e distribuição de chaves de grupos para usuários autenticados e com permissão para acessar tais informações , além da utilização de autenticação federada através da Comunidade Acadêmica Federada (CAFe) . Um Mapeamento Sistemático foi feito o que permit iu a descoberta de diversas técnicas que puderam ser aplicadas ou comparadas tanto nos refletores quanto no servidor de segurança, além do destacamento de possíveis lacunas na literatura que permitem su a exploração. Como parte dos resultados, a estratégia foi aplicada à Arthron 3.0 e possibilitou realização de transmissões com diversos fluxos de vídeo em alta definição

In the pursuit for ever-increasing compute power, and with Moore's law slowly coming to an end, high-performance computing started to scale-out to larger systems. Alongside the increasing system size, the interconnection network is growing to accommodate and connect tens of thousands of compute nodes. These networks have a large influence on total cost, application performance, energy consumption, and overall system efficiency of the supercomputer. Unfortunately, state-of-the-art routing algorithms, which define the packet paths through the network, do not utilize this important resource efficiently. Topology-aware routing algorithms become increasingly inapplicable, due to irregular topologies, which either are irregular by design, or most often a result of hardware failures. Exchanging faulty network components potentially requires whole system downtime further increasing the cost of the failure. This management approach becomes more and more impractical due to the scale of today's networks and the accompanying steady decrease of the mean time between failures. Alternative methods of operating and maintaining these high-performance interconnects, both in terms of hardware- and software-management, are necessary to mitigate negative effects experienced by scientific applications executed on the supercomputer. However, existing topology-agnostic routing algorithms either suffer from poor load balancing or are not bounded in the number of virtual channels needed to resolve deadlocks in the routing tables. Using the fail-in-place strategy, a well-established method for storage systems to repair only critical component failures, is a feasible solution for current and future HPC interconnects as well as other large-scale installations such as data center networks. Although, an appropriate combination of topology and routing algorithm is required to minimize the throughput degradation for the entire system. This thesis contributes a network simulation toolchain to facilitate the process of finding a suitable combination, either during system design or while it is in operation. On top of this foundation, a key contribution is a novel scheduling-aware routing, which reduces fault-induced throughput degradation while improving overall network utilization. The scheduling-aware routing performs frequent property preserving routing updates to optimize the path balancing for simultaneously running batch jobs. The increased deployment of lossless interconnection networks, in conjunction with fail-in-place modes of operation and topology-agnostic, scheduling-aware routing algorithms, necessitates new solutions to solve the routing-deadlock problem. Therefore, this thesis further advances the state-of-the-art by introducing a novel concept of routing on the channel dependency graph, which allows the design of an universally applicable destination-based routing capable of optimizing the path balancing without exceeding a given number of virtual channels, which are a common hardware limitation. This disruptive innovation enables implicit deadlock-avoidance during path calculation, instead of solving both problems separately as all previous solutions.

The goal of this thesis was to design two complex scenarios with focus on different kinds of transmission in packet-switched networks. First scenario is about ARQ (Automatic Repeat Request) protocols. It consists of introduction to Stop-and-Wait, Go Back N and Selective Repeat protocols and their comparison. Second scenario compares unicast, multicast, broadcast and anycast transmission methods. Both scenarios use applications which can simulate particular methods or protocols. These applications along with virtual environments are used for demonstration of characteristics of these methods/protocols.