Σκοπός της παρούσας διπλωματικής εργασίας ήταν η μελέτη των παραμέτρων που επηρεάζουν την ποιότητα εξυπηρέτησης (QoS – Quality of Service) των εφαρμογών VoIP μέσω των διαφόρων τεχνολογιών ευρυζωνικής πρόσβασης. / Aim of the present diplomatic assigment was the study of parameters that affect the quality of service (QoS) of VoIP applications via the various broadband access technologies.

Made available in DSpace on 2016-08-17T14:52:53Z (GMT). No. of bitstreams: 1 Ricardo Henrique Bezerra Azoubel.pdf: 588389 bytes, checksum: 40c0c99f808c27d0a6967b9d18c9c22a (MD5) Previous issue date: 2004-09-24 / This work presents a platform of tests (testbed) inexpensive, constructed in a controlled environment composed by microcomputers and free softwares. It is implemented, in such platform, the differentiated service model (DiffServ), with expedited forwarding (PHB EF). It is basically considered, from the collection of metrics main of QoS (delay, jitter, loss and throughtput), the performance analysis of voice characteristic traffics, when submitted to experimental tests in some scenes and conditions. Initially, in an environment capable to differentiate traffics, flows generated by standardized voice coder/decoder (G.711 and G.726) are modeled, in which the packets size and the amount of aggregate flows are varied, in scenes with and without QoS. It is compared, after that, the behavior of flows generated by activity-silence (ON-OFF) and continuous (CBR) sources. Can be perceived in this study how much the packets size variation influence in the performance of the most priority packets. It is carried, in the sequence, a specific analysis of the aggregation factor in flows generated by ON-OFF sources, in which can be observed the action of the basic principle of the model DiffServ, where aggregate flows receive differentiated treatment. It is studied, finally, through the use of transport protocols (UDP and TCP) and of elastic flows of FTP type, how much the best effort traffic confuses the performance of voice modeled flows. / Este trabalho apresenta uma plataforma de testes (testbed) sem custos, construída num ambiente controlado composto por microcomputadores e softwares livres. Implementa-se, em tal plataforma, o modelo de serviço diferenciado (DiffServ), com encaminhamento expresso (PHB EF). Propõe-se, fundamentalmente, a partir da obtenção das principais métricas de QoS (atraso, jitter, perda e vazão), a análise do desempenho de tráfego característico de voz, quando submetido a testes experimentais em vários cenários e condições. Inicialmente, num ambiente capaz de diferenciar tráfego, modelam-se fluxos gerados por codificadores/decodificadores de voz padronizados (G.711 e G.726), em que se varia o tamanho dos pacotes e a quantidade de fluxos agregados, em cenários com e sem QoS. Compara-se, em seguida, o comportamento de fluxos gerados por fontes atividadesilêncio (ON-OFF) e contínuas (CBR). Pode-se perceber nesse estudo o quanto a variação do tamanho dos pacotes impacta no desempenho dos pacotes mais prioritários. Realiza-se, na seqüência, uma análise específica do fator agregação em fluxos gerados por fontes ONOFF e observa-se a atuação do princípio básico do modelo DiffServ, onde fluxos agregados recebem tratamento diferenciado. Estuda-se, por fim, através da utilização de protocolos de transporte (UDP e TCP) e de fluxos elásticos do tipo FTP, o quanto o tráfego de melhor esforço impacta no desempenho de fluxos modelados de voz.

Multiprotocol Label Switching (MPLS) is a new paradigm in routing architectures which has changed the way Internet Protocol (IP) packet is transferred in a Network. MPLS ensures the reliability of the communication minimizing the delays and enhancing the speed of packet transfer. One important feature of MPLS is its capability of providing Traffic Engineering (TE) which plays a vital role for minimizing the congestion by efficient load, balancing and management of the network resources. The performance evaluation is done considering the network parameters latency, jitter, packet end to end delay, and packet delay variation. Integration of QoS with the MPLS-TE network may enhance the performance of the network. Various scheduling algorithms can be used for implementing QoS on a network, which may vary the performance of the network. In our study, QoS is implemented on top of the MPLS-TE network using Differentiated Service (DiffServ) architecture. Different basic scheduling algorithms are used for the implementation of QoS and to check their impact on the network and to identify the suitable one among them. Performance evaluation is done considering the network parameters latency, jitter, packet end-to-end delay, and Packet Delay Variation. The simulation was done using OPNET modeler 16.0 and the results were analyzed. The simulation result shows that using TE along with QoS in MPLS network decreases the latency, jitter, packet delay variation and end to end packet delay compared to using TE alone for voice traffic. / +46738732963

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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.

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Informationen des Universitätsrechenzentrums: - Die Projekte Campusnetz II und IP-Telefonie (VoIP) - PROWeb - Ein neuer Dienst für Projekt-WWW-Server - Unterstützte Linux-Distributionen - WUSCH - Windows-Update-Service an der TU Chemnitz - Informationen des URZ zur "Rahmenvereinbarung zum Einkauf von Standard-PC-Technik" -Umstellung des Lokalsystems der UB auf LIBERO 5 - Elektronisches Publizieren an der TU Chemnitz - 10 Jahre MONARCH - Kurzinformationen - Software-News

Informationen des Universitätsrechenzentrums: - Die Projekte Campusnetz II und IP-Telefonie (VoIP) - PROWeb - Ein neuer Dienst für Projekt-WWW-Server - Unterstützte Linux-Distributionen - WUSCH - Windows-Update-Service an der TU Chemnitz - Informationen des URZ zur 'Rahmenvereinbarung zum Einkauf von Standard-PC-Technik' -Umstellung des Lokalsystems der UB auf LIBERO 5 - Elektronisches Publizieren an der TU Chemnitz - 10 Jahre MONARCH - Kurzinformationen - Software-News

In automatic speech recognition systems (ASRs), training is a critical phase to the system?s success. Communication media, either analog (such as analog landline phones) or digital (VoIP) distort the speaker?s speech signal often in very complex ways: linear distortion occurs in all channels, either in the magnitude or phase spectrum. Non-linear but time-invariant distortion will always appear in all real systems. In digital systems we also have network effects which will produce packet losses and delays and repeated packets. Finally, one cannot really assert what path a signal will take, and so having error or distortion in between is almost a certainty. The channel introduces an acoustical mismatch between the speaker's signal and the trained data in the ASR, which results in poor recognition performance. The approach so far, has been to try to undo the havoc produced by the channels, i.e. compensate for the channel's behavior. In this thesis, we try to characterize the effects of different transmission media and use that as an inexpensive and repeatable way to train ASR systems.