A utilização das redes sem fio nos mais diversos segmentos vem crescendo vertiginosamente nos últimos anos. Aliado ao crescimento da utilização das redes, está a diversificação de aplicações utilizadas por todos os usuários, sejam eles corporativos ou domésticos. Isto tem contribuído significativamente para o desenvolvimento de tecnologias que ofereçam mecanismos de qualidade de serviço, de forma a permitir o uso satisfatório de aplicações como voz e vídeo em tempo real. Este trabalho estuda os efeitos que a existência de tráfego de voz e dados em redes sem fio padrão IEEE 802.11 causa no desempenho da rede e por consequência no desempenho das aplicações. Para realizar este estudo foram executadas simulações baseadas em uma rede sem fio com topologia ad hoc, com variações no número de estações e quantidade de tráfego gerado. Foram simulados os padrões 802.11g e 802.11e com o objetivo de analisar o desempenho dos mecanismos de qualidade de serviço e os benefícios que estes mecanismos geram para a transmissão de tráfego em tempo real em redes sem fio padrão IEEE 802.11. Verificou-se que o padrão IEEE 802.11e apresenta um ganho de desempenho para aplicações de tempo real, porém ele apresenta limitações que devem ser consideradas nos projetos de redes sem fio. / The use of wirelles networks in most various sectors has been growing drastically in past years Allied to the wireless networks use, the diversification of applications and services provided can be directly verified whether by home or corporate users. This alliance contributes significantly to the needs of technology development which offer the quality of service mechanisms, allowing satisfactory use of real-time applications like voice and video This paper studies the effects that coexistent voice and data traffic on a IEEE 802.11 standard wireless network cause in the network performance and, consequently, in the applications performance. To make this study a reality, it was necessary to run simulations of a wireless ad hoc topology network, with variations in the number of workstations and the quantity of generated traffic. The 802.11g and 802.11e standards were used in the simulations with the purpose of analyzing the performance of quality of service mechanisms and the benefits they create for the real-time transmissions in IEEE 802.11 standard wireless networks. It was verified that the IEEE 802.11e standard presents a perfomance gain for the real-time applications, but it has limitations that should be considered in wireless networks design.

A utilização das redes sem fio nos mais diversos segmentos vem crescendo vertiginosamente nos últimos anos. Aliado ao crescimento da utilização das redes, está a diversificação de aplicações utilizadas por todos os usuários, sejam eles corporativos ou domésticos. Isto tem contribuído significativamente para o desenvolvimento de tecnologias que ofereçam mecanismos de qualidade de serviço, de forma a permitir o uso satisfatório de aplicações como voz e vídeo em tempo real. Este trabalho estuda os efeitos que a existência de tráfego de voz e dados em redes sem fio padrão IEEE 802.11 causa no desempenho da rede e por consequência no desempenho das aplicações. Para realizar este estudo foram executadas simulações baseadas em uma rede sem fio com topologia ad hoc, com variações no número de estações e quantidade de tráfego gerado. Foram simulados os padrões 802.11g e 802.11e com o objetivo de analisar o desempenho dos mecanismos de qualidade de serviço e os benefícios que estes mecanismos geram para a transmissão de tráfego em tempo real em redes sem fio padrão IEEE 802.11. Verificou-se que o padrão IEEE 802.11e apresenta um ganho de desempenho para aplicações de tempo real, porém ele apresenta limitações que devem ser consideradas nos projetos de redes sem fio. / The use of wirelles networks in most various sectors has been growing drastically in past years Allied to the wireless networks use, the diversification of applications and services provided can be directly verified whether by home or corporate users. This alliance contributes significantly to the needs of technology development which offer the quality of service mechanisms, allowing satisfactory use of real-time applications like voice and video This paper studies the effects that coexistent voice and data traffic on a IEEE 802.11 standard wireless network cause in the network performance and, consequently, in the applications performance. To make this study a reality, it was necessary to run simulations of a wireless ad hoc topology network, with variations in the number of workstations and the quantity of generated traffic. The 802.11g and 802.11e standards were used in the simulations with the purpose of analyzing the performance of quality of service mechanisms and the benefits they create for the real-time transmissions in IEEE 802.11 standard wireless networks. It was verified that the IEEE 802.11e standard presents a perfomance gain for the real-time applications, but it has limitations that should be considered in wireless networks design.

This work presents a novel Peer to Peer (P2P) framework for Session Initiation Protocol (SIP) on Mobile Ad Hoc Network (MANET). SIP is a client-server model of computing which can introduce a single point of failure problem. P2P SIP addresses this problem by using a distributed implementation based on a P2P paradigm. However, both the traditional SIP and P2P SIP architectures are not suitable for MANETs because they are initially designed for infrastructured networks whose most nodes are static. We focus on distributed P2P resource lookup mechanisms for SIP which can tolerate failures resulting from the node mobility. Our target application is SIP-based multimedia communication in a rapidly deployable disaster emergency network. To achieve our goal, we provide four contributions as follows. The first contribution is a novel P2P lookup architecture based on a concept of P2P overlay network called a Structured Mesh Overlay Network (SMON). This overlay network enables P2P applications to perform fast resource lookups in the MANET environment. SMON utilizes a cross layer design based on the Distributed Hashing Table (DHT) and has direct access to OLSR routing information. Its cross layer design allows optimizing the overlay network performance during the change of network topology. The second contribution is a distributed SIP architecture on MANET providing SIP user location discovery in a P2P manner which tolerates single-point and multiple-point of failures. Our approach extends the traditional SIP user location discovery by utilizing DHT in SMON to distribute SIP object identifiers over SMON. It offers a constant time on SIP user discovery which results in a fast call setup time between two MANET users. From simulation and experiment results, we find that SIPMON provides the lowest call setup delay when compared to the existing broadcast-based approaches. The third contribution is an extended SIPMON supporting several participating MANETs connected to Internet. This extension (SIPMON+) provides seamless mobility support allowing a SIP user to roam from an ad hoc network to an infrastructured network such as Internet without interrupting an ongoing session. We propose a novel OLSR Overlay Network (OON), a single overlay network containing MANET nodes and some nodes on the Internet. These nodes can communicate using the same OLSR routing protocol. Therefore, SIPMON can be automatically extended without modifying SIPMON internal operations. Through our test-bed experiments, we prove that SIPMON+ has better performance in terms of call setup delay and handoff delay than MANET for Network Mobility (MANEMO). The fourth contribution is a proof-of-concept and a prototype of P2P multimedia communication based on SIPMON+ for post disaster recovery missions. We evaluate our prototype and MANEMO-based approaches through experimentation in real disaster situations (Vehicle to Infrastructure scenarios). We found that our prototype outperforms MANEMO-based approaches in terms of call setup delay, packet loss, and deployment time.

This study represents an investigation into the effect of multiple production techniques on the recall and retention of information of university students. The purpose of the study was to determine how the ‘Voice of God’ narration affects recall and retention of information compared to an on-camera interview. In documentary video, conveying information is one of the key goals of the director. In most cases, the conveyance of information in itself is insufficient. The director seeks the emotional participation of the audience so that they can become aware of a specific issue. In this mediated environment, the producer’s attempts at reaching the audience are interwoven with the producer’s capability to facilitate the audience’s recall of information. By using the ‘Voice of God’ narration, a director can enhance the narrative and make the information more understandable. Theory suggests that the use of multiple production techniques can either have a negative or positive impact on the processing of information. This statement is based on different experiments that were done by researchers on how various production techniques affect the information processing of an individual. The theoretical basis of the study is rooted in the metatheory, cybernetics. Within cybernetics, the narrative theory explains the structure of the story and how it is conveyed to an audience. Voice-over in documentary video is situated in the narrative theory because the structure of the information the voice-over gives to an audience is of utmost importance. From the theoretical basis, this study uses Lang’s (2000) limited capacity model of mediated message processing to investigate the effect of narration (voice-over) as embedded in documentary video. Specifically, it addresses the mediator’s (in documentary video, the producer’s) goal of maximum information recall by the receiver of the message. The application of Lang’s model is outlined in an empirical design that explores recall of message content and the retention of information in two ways; the recall and retention of information presented through an on-camera interview and the recall and retention of information presented by a narrator whilst images are shown that do not include an image of the narrator him/herself (Voice of God narration). Two experiments were designed for the purpose of this study in which 37 students from the North-West University’s Potchefstroom campus participated. The students were divided into two groups; group one watched the video where the information is presented by an on-camera-interview, and group two watched the video where the information is presented by a ‘Voice of God’ narrator. Two questionnaires were given to the groups at two different times. The results obtained suggest that there is no significant difference in the production techniques and the recall and retention of information. Based on the results, certain recommendations are made for future research, which include modifying the message and research design. / MA (Communication Studies), North-West University, Potchefstroom Campus, 2014

O projeto VoIPFix surgiu da necessidade de uma ferramenta que complementasse as demais existentes no ramo de análise de redes de computadores para telefonia IP. Ele foi construído para ser uma ferramenta de gerenciamento eficiente e exclusiva para VoIP, com funcionalidades necessárias para dar suporte ao profissional de rede de computadores e telefonia IP a observar e diagnosticar problemas de VoIP. / The VoIPFix project arose from the need for a tool to complement similar tools in the analysis of computer networks for IP telephony. It was built to be an efficient and unique management tool for VoIP, with advanced features required to support the computer network and IP telephony professionals to observe and diagnose problems related to VoIP.

Les communications téléphoniques en voix sur IP souffrent de la perte de paquets causée par les problèmes d'acheminement dus aux nœuds du réseau. La perte d'un paquet de voix induit la perte d'un segment de signal de parole (généralement 10ms par paquet perdu). Face à la grande diversité des codeurs de parole, nous nous sommes intéressés dans le cadre de cette thèse à proposer une méthode de masquage de pertes de paquets générique, indépendante du codeur de parole utilisé. Ainsi, le masquage de pertes de paquets est appliqué au niveau du signal de parole reconstruit, après décodage, s'affranchissant ainsi du codeur de parole. Le système proposé repose sur une modélisation classique de type « modèles de Markov cachés » afin de suivre l'évolution acoustique de la parole. À notre connaissance, une seule étude a proposé l'utilisation des modèles de Markov cachés dans ce cadre [4]. Toutefois, Rødbro a utilisé l'utilisation de deux modèles, l'un pour la parole voisée, l'autre pour les parties non voisées, posant ainsi le problème de la distinction voisée/non voisée. Dans notre approche, un seul modèle de Markov caché est mis en œuvre. Aux paramètres classiques (10 coefficients de prédiction linéaire dans le domaine cepstral (LPCC) et dérivées premières) nous avons adjoint un nouvel indicateur continu de voisement [1, 2]. La recherche du meilleur chemin avec observations manquantes conduit à une version modifiée de l'algorithme de Viterbi pour l'estimation de ces observations. Les différentes contributions (indice de voisement, décodage acoutico-phonétique et restitution du signal) de cette thèse sont évaluées [3] en terme de taux de sur et sous segmentation, taux de reconnaissance et distances entre l'observation attendue et l'observation estimée. Nous donnons une indication de la qualité de la parole au travers d'une mesure perceptuelle : le PESQ. / Packet loss due to misrouted or delayed packets in voice over IP leads to huge voice quality degradation. Packet loss concealment algorithms try to enhance the perceptive quality of the speech. The huge variety of vocoders leads us to propose a generic framework working directly on the speech signal available after decoding. The proposed system relies on one single "hidden Markov model" to model time evolution of acoustic features. An original indicator of continuous voicing is added to conventional parameters (Linear Predictive Cepstral Coefficients) in order to handle voiced/unvoiced sound. Finding the best path with missing observations leads to one major contribution: a modified version of the Viterbi algorithm tailored for estimating missing observations. All contributions are assessed using both perceptual criteria and objective metrics.

My master’s thesis is focused on designing and creating communication network, which provides communication between two independent networks through encrypted tunnel. My solution is based on routers formed by older personal computers with FreeBSD like a operating system. Between routers is created static encrypted tunnel by using IPSec protocol. Voice services provides packet oriented exchange Asterisk with support of signaling protocol SIP. This solution can be used eg. for connecting remote branch to headquarters of company and then can branch utilize shrared resources. To headquarters can connect also remote workers from their home. In this case are used SSL certificates to authentication of user. This scenario is very required today.

This work deals with accessible communications technologies and their possible usage in firms. It is necessary to have a survey about types and trends of communications technologies at proposal of communications network. It is important to choose the combination that is suitable with its reliability, achievement and price.

Die vorliegende Arbeit beschreibt SIP Clients für Linux und Windows. Die Programme funktionieren in einem leistungsstarken Netzwerk, wie z.B. dem Intranet der Technischen Universität Chemnitz, problemlos. Alle Funktionen wurden in einer VoIP-Umgebung getestet. Die Sprachübertragungqualität über das Internet mittels dieser Clients ist jedoch nicht Gegenstand der Arbeit und somit auch nicht erprobt worden. In vielen Unternehmensbereichen fällt häufig der Begriff Echtzeitkommunikation in Verbindung mit einer geeigneten Infrastruktur, ausgelöst durch den Zuwachs der verfügbaren Netzwerkbandbreite und somit immer realistischer werdender Sprach- und Videoübertragungen. Ein Ansatz für die Anwendungen ist das Protokoll SIP (Session Initiation Protocol), welches für die Signalisierung der Video- und Sprachübertragung verwendet wird.

Das Ziel dieser Arbeit ist es, ein Handover-Verfahren für ein Siemens Handset zu entwickeln. Die Entwicklungsumgebung beruht auf den Wireless-LAN Standards 802.11 der IEEE (Institute of Electrical and Electronics Engineers). Dabei liegen die Schwerpunkte auf den Standardisierungen 802.11f und 802.11i, wobei sich eine neue Arbeitsgruppe (IEEE 802.11r) direkt mit dem Thema "Handover" beschäftigen wird. Das Handset soll selbständig die Verwaltung und Einleitung des Handovers übernehmen und lediglich insofern vom Access Point unterstützt werden, dass dieser als Informationssammler dient und somit Entscheidungshilfen geben kann.