With the constant development of wireless technologies, the usageof wireless devices tends to increase even more in the future.Wireless multi-hop networks (WMNs) have emerged as a keytechnology to numerous potential scenarios, ranging from disasterrecovery to wireless broadband internet access. The distributedarchitecture of WMNs enables nodes to cooperatively relay othernode's packets. Because of their advantages over other wirelessnetworks, WMNs are undergoing rapid progress and inspiringnumerous applications. However, many technical issues still existin this field. In this thesis we investigate how Voice over IP(VoIP) and peer-to-peer (P2P) application are influenced bywireless multi-hop network characteristics and how to optimizethem in order to provide scalable communication.We first consider the deployment of VoIP service in wirelessmulti-hop networks, by using the Session Initiation Protocol (SIP)architecture. Our investigation shows that the centralized SIParchitecture imposes several challenges when deployed in thedecentralized wireless multi-hop environment. We find that VoIPquality metrics are severely degraded as the traffic and number ofmultiple hops to the gateway increase. In the context ofscalability, we further propose four alternative approaches whichavoid current limitations.In the second part of this thesis we tackle the network capacityproblem while providing scalable VoIP service over wirelessmulti-hop networks. The performance evaluation shows the influenceof intra and inter-flow interference in channel utilization, whichdirect impacts the VoIP capacity. In order to avoid the small VoIPpacket overhead, we propose a new adaptive hop-by-hop packetaggregation scheme based on wireless link characteristics. Ourperformance evaluation shows that the proposed scheme can increasethe VoIP capacity by a two-fold gain.The study of peer-to-peer applicability over wireless multi-hopnetworks is another important contribution. A resource lookupapplication is realized through structured P2P overlay. We showthat due to several reasons, such as characteristics of wirelesslinks, multi-hop forwarding operation, and structured P2Pmanagement traffic aggressiveness the performance of traditionalP2P applications is rather low in wireless multi-hop environments.Therefore, we suggested that a trade-off between the P2P lookupefficiency and the P2P management traffic overhead can be achievedwhile maintaining the overlay network consistency in wirelessmulti-hop networks.

This document provides information about Cisco NetFlow technology and its usage to protect networks from different types of attacks. Part of the document is a summary of common security risks in term of their detection on network and transport layer. There are specified characteristics of NetFlow data containing samples of security risks. On the basis of these characteristics, an application for detection these risks is designed and implemented.

Unified Communications solutions bring together several communication modes, technologies, and applications in order to answer businesses’ and individuals’ growing need for simpler, faster, and more effective communications means.  Although many hardware-based products allow the integration of telephony within a computer network environment, telephony features of software-based unified communications solutions are seldom used, which limits their effectiveness or requires another solution to be used jointly. This master’s thesis project aims to demonstrate that unified communications solutions based on Microsoft Lync Server 2013 can effectively address a wide variety of business scenarios, including a traditional telephony system replacement.  The first part of this master’s thesis introduces background knowledge about unified communications and associated technologies, as well as the different components of the selected unified communication solution. The case study presented in this thesis is the first large-scale Lync 2013 deployment with a complete telephony offering in France. The presentation follows the complete deployment process, starting from the analysis of the client’s needs to the solution design, construction, and validation. This project demonstrated the suitability of Lync 2013 as a telephony system replacement. However, the transition from a classic telephony solution to a unified communications solution can be a technical challenge. An essential step in making this transition successful was to take the users’ needs into account. It was also essential to accompany these users throughout the transition. / Samordnad kommunikation (engelska: unified communications) lösningar sammanföra flera kommunikationssätt, teknik och tillämpningar för att besvara företags och individers växande behovet av enklare, snabbare och mer effektivt kommunikationsmedel. Även många hårdvara-baserade produkter tillåter integration av telefoni inom ett datornätverk miljö, telefoni funktioner mjukvarubaserad Samordnad kommunikation-lösningar används sällan, vilket begränsar deras effektivitet eller kräver en annan lösning för att användas gemensamt.  Detta examensarbete syftar till att visa att samordnad kommunikation lösningar baserade på Microsoft Lync Server 2013 kan effektivt ta itu med en mängd olika scenarier. Den första delen av detta examensarbete introducerar bakgrundskunskap om samordnad kommunikation och tillhörande teknologier liksom de olika komponenterna i den valda samordnad kommunikation lösning. Fallstudien som presenteras i denna avhandling är den första storskaliga Lync 2013 utplacering med en komplett telefoni erbjuder i Frankrike. Den presentationen följer hela implementeringsprocessen, från analys av kundens kraven till utformning, konstruktion, och validering. Detta projekt visade tillförlitligheten i Lync 2013 som telefoni ersättning men intyga att även övergången från en klassisk telefoni lösning på ett samordnad kommunikation-lösning kan vara en teknisk utmaning, ta användarnas behov i beaktande och medföljande användare genom övergången är kritisk. / Les solutions de communications unifiées rassemblent différents modes de communications, technologies, et applications pour répondre aux besoin croissant des entreprises et individus de méthodes de communications plus simples, rapides et efficaces. Bien que de nombreuses solutions matérielles permettent l’intégration de la téléphonie à un réseau informatique, les fonctions de téléphonie des solutions logicielles sont rarement utilisées, ce qui limite leur efficacité ou nécessite l’utilisation conjointe d’autres solutions. Ce projet a pour but de démontrer l’efficacité des solutions de communications unifiées basées sur Microsoft Lync 2013 à répondre à une grande variété de besoins professionels, dont le remplacement d’un système de téléphonie traditionnel. La premiére partie de ce mémoire introduit les notions nécessaires sur les communications unifiées et les technologies associées, ainsi que les différents composants de la solution de communications unifiées choisie. L’étude de cas présentée décrit le premier déploiement majeur de Lync Server 2013 comportant une offre de téléphonie complète en France, et suit le processus de déploiement complet, de l’analyse des besoins client à la validation du projet, en passant par la conception, la construction et le test. Ce projet démontre l’aptitude de Lync en temps que sysème de téléphonie complet. Cependant la transition d’un système traditionnel à une solution de communications unifiées peut présenter des défis techniques, et il est essentiel de prendre en compte les besoins utilisateurs ainsi que de les accompagner durant la transition.

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Informationen des Universitätsrechenzentrums mit Jahresrückblick 2008 zu den aktuellen Projekten und Diensten des URZ:Jahresrückblick 2008 Softwareausstattung der Ausbildungspools Kurzinformationen Software-News

Les réseaux maillés sans fil (RMSF), grâce à leurs caractéristiques avantageuses, sont considérés comme une solution efficace pour le support des services de voix, vidéo et de données dans les réseaux de prochaine génération. Le standard IEEE 802.16-d a spécifié pour les RMSF, à travers son mode maillé, deux mécanismes de planifications de transmission de données; à savoir la planification centralisée et la planification distribuée. Dans ce travail, on a évalué le support de la qualité de service (QdS) du standard en se focalisant sur la planification distribuée. Les problèmes du système dans le support du trafic de voix ont été identifiés. Pour résoudre ces problèmes, on a proposé un protocole pour le support de VoIP (AVSP) en tant qu’extension au standard original pour permettre le support de QdS au VoIP. Nos résultats préliminaires de simulation montrent qu’AVSP offre une bonne amélioration au support de VoIP. / Wireless mesh networks (WMNs), because of their advantageous characteristics, are considered as an effective solution to support voice services, video and data in next generation networks. The IEEE 802.16-d specified for WMNs, through its mesh mode, two mechanisms of scheduling data transmissions; namely centralized scheduling and distributed scheduling. In this work, we evaluated the support of the quality of service (QoS) of the standard by focusing on distributed scheduling. System problems in the support of voice traffic have been identified. To solve these problems, we proposed a protocol for supporting VoIP, called Assisted VoIP Scheduling Protocol (AVSP), as an extension to the original standard to support high QoS to VoIP. Our preliminary simulation results show that AVSP provides a good improvement to support VoIP.

Les réseaux maillés sans fil (RMSF), grâce à leurs caractéristiques avantageuses, sont considérés comme une solution efficace pour le support des services de voix, vidéo et de données dans les réseaux de prochaine génération. Le standard IEEE 802.16-d a spécifié pour les RMSF, à travers son mode maillé, deux mécanismes de planifications de transmission de données; à savoir la planification centralisée et la planification distribuée. Dans ce travail, on a évalué le support de la qualité de service (QdS) du standard en se focalisant sur la planification distribuée. Les problèmes du système dans le support du trafic de voix ont été identifiés. Pour résoudre ces problèmes, on a proposé un protocole pour le support de VoIP (AVSP) en tant qu’extension au standard original pour permettre le support de QdS au VoIP. Nos résultats préliminaires de simulation montrent qu’AVSP offre une bonne amélioration au support de VoIP. / Wireless mesh networks (WMNs), because of their advantageous characteristics, are considered as an effective solution to support voice services, video and data in next generation networks. The IEEE 802.16-d specified for WMNs, through its mesh mode, two mechanisms of scheduling data transmissions; namely centralized scheduling and distributed scheduling. In this work, we evaluated the support of the quality of service (QoS) of the standard by focusing on distributed scheduling. System problems in the support of voice traffic have been identified. To solve these problems, we proposed a protocol for supporting VoIP, called Assisted VoIP Scheduling Protocol (AVSP), as an extension to the original standard to support high QoS to VoIP. Our preliminary simulation results show that AVSP provides a good improvement to support VoIP.

Submitted by Gisele Isaura Hannickel (gisele.hannickel@fgv.br) on 2012-04-24T18:53:49Z No. of bitstreams: 1 120423 Master Thesis FGV.pdf: 3697873 bytes, checksum: f3ccbecacf430eaee0f0606f9daa9f36 (MD5) / Approved for entry into archive by Eliene Soares da Silva (eliene.silva@fgv.br) on 2012-04-25T15:52:36Z (GMT) No. of bitstreams: 1 120423 Master Thesis FGV.pdf: 3697873 bytes, checksum: f3ccbecacf430eaee0f0606f9daa9f36 (MD5) / Made available in DSpace on 2012-04-25T15:54:46Z (GMT). No. of bitstreams: 1 120423 Master Thesis FGV.pdf: 3697873 bytes, checksum: f3ccbecacf430eaee0f0606f9daa9f36 (MD5) Previous issue date: 2012-04-23 / Internet Telephony (VoIP) is changing the telecommunication industry. Oftentimes free, VoIP is becoming more and more popular amongst users. Large software companies have entered the market and heavily invest into it. In 2011, for instance, Microsoft bought Skype for 8.5bn USD. This trend increasingly impacts the incumbent telecommunication operators. They see their main source of revenue – classic telephony – under siege and disappear. The thesis at hand develops a most-likely scenario in order to determine how VoIP is evolving further and it predicts, based on a ten-year forecast, the impact it will have on the players in the telecommunication industry.The paper presents a model combining Rogers’ diffusion and Christensen’s innovation research. The model has the goal of explaining the past evolution of VoIP and to isolate the factors that determine the further diffusion of the innovation. Interviews with industry experts serve to assess how the identified factors are evolving.Two propositions are offered. First, VoIP operators are becoming more important in international, corporate, and mobile telephony. End-to-end VoIP (IP2IP) will exhibit strong growth rates and increasingly cannibalize the telephony revenues of the classic operators. Second, fix-net telephony in SMEs and at home will continue to be dominated by the incumbents. Yet, as prices for telephony fall towards zero also they will implement IP2IP in order to save costs. By 2022, up to 90% of the calls will be IP2IP. The author recommends the incumbents and VoIP operators to proactively face the change, to rethink their business strategies, and to even be open for cooperation.

Made available in DSpace on 2015-04-20T12:33:26Z (GMT). No. of bitstreams: 1 Leandro.pdf: 2831865 bytes, checksum: 5804d85c95f338cf4054c799f4dfd45d (MD5) Previous issue date: 2011-10-07 / Voice calls based on Voice over Internet Protocol (VoIP) technology are liable to several impairments from both application and network layer, such as codec compression, end-to-end delay, and packet loss. For years, this problem has been challenging researchers and practitioners, who have been designing and improving QoS control mechanisms for VoIP applications. Such mechanisms aim to make optimum use of network and terminal resources so as to minimize the effects of network impairments on voice quality. Among the several proposed QoS control mechanisms for VoIP, some of them seek to adapt the voice flow or other VoIP-related parameters in accordance with significant changes in the network, end users preferences, or service providers requirements. VoIP systems are particularly likely to require a dynamic adaptation solution for dealing with the complex trade-off between speech quality and impairments, because of the decentralized control nature of IP networks and the stochastic nature of data packet delivery. Although the existing adaptive solutions for QoS control of VoIP show some performance improvement and exhibit some sort of feedback, they do not provide explicit focus on the control loop. This document shows the current progress of our thesis, which addresses the adjustment of internal parameters of VoIP terminals (at application layer) that affect the voice flow, with the aim of improving speech quality in response to changes in network conditions. It is not in the scope of the thesis to propose adaptive solutions that focus exclusively on signaling, billing, security issues, or operate at the network layer. Therefore, this thesis addresses the problem of how adjust encoding parameters in response to variations in delay and packet loss, in order to optimize speech quality. The objective is to optimize user-perceptible attributes of speech, under the perspective of self-adaptive software systems. The emphasis is not to develop new audio codecs, but to build a control loop in the core of sender and receiver terminals to adapt voice flow settings according to network conditions. The main contributions of this thesis are the following: determination of user s perception during codec switching; parametrization of codec precedence for supporting codec switching decision; explicit design of a monitoring analysis planning execution control loop as the core of the adaptation process; and efficiency analysis of feedback message exchanging. / Chamadas de voz baseadas na tecnologia VoIP (Voice over Internet Protocol) estão suscetíveis a degradações diversas, provenientes tanto da camada de aplicação, como da camada de rede, tais como compressão do codec, atraso fim a fim e perda de pacotes. Durante anos, esse problema tem desafiado pesquisadores e profissionais, que têm concebido e melhorado mecanismos de controle de QoS para aplicações VoIP. Tais mecanismos visam otimizar a utilização dos recursos da rede e do terminal VoIP de modo a minimizar os efeitos deletérios da rede subjacente sobre a qualidade de voz. Entre as várias propostas de mecanismos de controle de QoS para VoIP, alguns deles procuram adaptar o fluxo de voz ou outros parâmetros VoIP de acordo com mudanças significativas na rede, preferências de usuário, ou requisitos dos provedores de serviços VoIP. Sistemas VoIP particularmente exigem soluções de adaptação dinâmica para lidar com a complexa relação de compromisso entre qualidade de voz e fatores de degradação, por causa da natureza descentralizada e estocástica das redes IP na entrega de pacotes de voz. Embora as soluções adaptativas existentes para controle de QoS em VoIP mostrem alguma melhora de desempenho e apresentem algum tipo de feedback, elas não fornecem foco explícito na ciclo de controle (control loop). Este documento mostra o progresso atual da nossa tese, que aborda o ajuste de parâmetros internos de terminais VoIP (camada de aplicação) que afetam o fluxo de voz, com o objetivo de melhorar a qualidade da fala em resposta a mudanças nas condições da rede. Não faz parte do escopo da tese abordar soluções adaptativas que se concentram exclusivamente em sinalização, bilhetagem, problemas de segurança, ou que operam no nível da camada de rede. Portanto, esta tese aborda o problema da concepção e avaliação de estratégias adaptativas que explorem as relações de compromisso entre qualidade da fala e os seguintes fatores de degradação: compressão do codec, atraso fim a fim e perda de pacotes. A finalidade é otimizar atributos da fala perceptíveis aos usuário, sob a perspectiva de sistemas de software autoadaptativo. A ênfase não reside em desenvolver novos codecs de áudio, mas sim em desenvolver um ciclo de controle como entidade central de um terminal VoIP, que possa adaptar as configurações do fluxo de voz de acordo com as condições da rede. As principais contribuições desta tese são as seguintes: determinação da percepção do usuário durante a comutação de codec; parametrização de precedência de codecs para suporte de decisão de comutação de codec; enfoque no ciclo de controle baseado nas atividades de monitoramento análise planejamento execução como núcleo do processo de adaptação; e análise de eficiência de troca de mensagens de feedback.

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