12 research outputs found
Quality aspects of Internet telephony
Internet telephony has had a tremendous impact on how people communicate.
Many now maintain contact using some form of Internet telephony.
Therefore the motivation for this work has been to address the quality aspects
of real-world Internet telephony for both fixed and wireless telecommunication.
The focus has been on the quality aspects of voice communication,
since poor quality leads often to user dissatisfaction. The scope of the work
has been broad in order to address the main factors within IP-based voice
communication.
The first four chapters of this dissertation constitute the background
material. The first chapter outlines where Internet telephony is deployed
today. It also motivates the topics and techniques used in this research.
The second chapter provides the background on Internet telephony including
signalling, speech coding and voice Internetworking. The third chapter
focuses solely on quality measures for packetised voice systems and finally
the fourth chapter is devoted to the history of voice research.
The appendix of this dissertation constitutes the research contributions.
It includes an examination of the access network, focusing on how calls are
multiplexed in wired and wireless systems. Subsequently in the wireless
case, we consider how to handover calls from 802.11 networks to the cellular
infrastructure. We then consider the Internet backbone where most of our
work is devoted to measurements specifically for Internet telephony. The
applications of these measurements have been estimating telephony arrival
processes, measuring call quality, and quantifying the trend in Internet telephony
quality over several years. We also consider the end systems, since
they are responsible for reconstructing a voice stream given loss and delay
constraints. Finally we estimate voice quality using the ITU proposal PESQ
and the packet loss process.
The main contribution of this work is a systematic examination of Internet
telephony. We describe several methods to enable adaptable solutions
for maintaining consistent voice quality. We have also found that relatively
small technical changes can lead to substantial user quality improvements.
A second contribution of this work is a suite of software tools designed to
ascertain voice quality in IP networks. Some of these tools are in use within
commercial systems today
Ontwerp en evaluatie van content distributie netwerken voor multimediale streaming diensten.
Traditionele Internetgebaseerde diensten voor het verspreiden van bestanden, zoals Web browsen en het versturen van e-mails, worden aangeboden via één centrale server. Meer recente netwerkdiensten zoals interactieve digitale televisie of video-op-aanvraag vereisen echter hoge kwaliteitsgaranties (QoS), zoals een lage en constante netwerkvertraging, en verbruiken een aanzienlijke hoeveelheid bandbreedte op het netwerk. Architecturen met één centrale server kunnen deze garanties moeilijk bieden en voldoen daarom niet meer aan de hoge eisen van de volgende generatie multimediatoepassingen. In dit onderzoek worden daarom nieuwe netwerkarchitecturen bestudeerd, die een dergelijke dienstkwaliteit kunnen ondersteunen. Zowel peer-to-peer mechanismes, zoals bij het uitwisselen van muziekbestanden tussen eindgebruikers, als servergebaseerde oplossingen, zoals gedistribueerde caches en content distributie netwerken (CDN's), komen aan bod. Afhankelijk van de bestudeerde dienst en de gebruikte netwerktechnologieën en -architectuur, worden gecentraliseerde algoritmen voor netwerkontwerp voorgesteld. Deze algoritmen optimaliseren de plaatsing van de servers of netwerkcaches en bepalen de nodige capaciteit van de servers en netwerklinks. De dynamische plaatsing van de aangeboden bestanden in de verschillende netwerkelementen wordt aangepast aan de heersende staat van het netwerk en aan de variërende aanvraagpatronen van de eindgebruikers. Serverselectie, herroutering van aanvragen en het verspreiden van de belasting over het hele netwerk komen hierbij ook aan bod
A measurement-based approach to service modeling and bandwidth estimation in IEEE 802.11 wireless networks
[no abstract
Uma abordagem para análise de desempenho de fluxos VoIP em redes de serviços diferenciados
This work presents a viability analysis of the use of a synthetic VoIP control flow to
infer about the performance of individual flows of a flow aggregate belonging to an EF PHB
in a DiffServ network. The proposed approach aims to establish a simple performance
verification of SLA accomplishment related to the some of the VoIP flow requirements. The
results should be used to feed requirements specifications for tool design, for example, to
capacity planning activities and management actions. We classify the VoIP traffic as
homogeneous (all flow packets created by a same codec type) and heterogeneous (packets
originated from more than one codec type) to carry out the experiments. The experiments
checked the hypothesis that the control flow performance can be somehow related to the
performance of individual flows of a flow aggregate under the agreed assumptions and
metrics. The metrics one-way delay, jitter and packet loss were evaluated by simulation for
both homogeneous and heterogeneous traffic at several network-controlled load. The results
let us conclude about the viability of the approach to evaluate one-way delay and with
confidence limitations, also the jitter, depending on the traffic type (heterogeneous) and codec type.O presente trabalho apresenta uma análise de viabilidade do emprego de um fluxo de
controle sintético VoIP para inferir sobre a performance de fluxos individuais de um fluxo
agregado pertencente a um EF PHB em uma rede de serviços diferenciados. A abordagem
proposta visa estabelecer através de simples verificação de performance quanto ao
atendimento do SLA relacionado a alguns requisitos do fluxo VoIP. Os resultados poderão ser
utilizados para alimentar especificações e requisitos para o projeto de ferramentas, por
exemplo, para capacitar atividades de planejamento e ações de gerência de rede. O tráfego
VoIP foi classificado como homogêneo (todos os pacotes do fluxo são criados pelo mesmo
tipo de codec) e como heterogêneo (pacotes originados por mais de um tipo de codec) durante
a realização dos experimentos. Os experimentos verificaram a hipótese de que a performance
do fluxo de controle possa ser relacionada de alguma forma com a performance dos fluxos
individuais de um fluxo agregado sob as suposições e métricas definidas. As métricas retardo,
jitter e perda de pacotes foram estimadas por simulação tanto para o tráfego homogêneo
quanto para o tráfego heterogêneo, em diversas condições de carga controlada. Os resultados
permitem concluir quanto a viabilidade da abordagem para estimar o retardo e com limitações
de confiança, quanto ao jitter, dependendo do tipo de tráfego (heterogêneo) e tipo de codec
Analysis of generic discrete-time buffer models with irregular packet arrival patterns
De kwaliteit van de multimediadiensten die worden aangeboden over de huidige breedband-communicatienetwerken, wordt in hoge mate bepaald door de performantie van de buffers die zich in de diverse netwerkele-menten (zoals schakelknooppunten, routers, modems, toegangsmultiplexers, netwerkinter- faces, ...) bevinden. In dit proefschrift bestuderen we de performantie van een dergelijke buffer met behulp van een geschikt stochastisch discrete-tijd wachtlijnmodel, waarbij we het geval van meerdere uitgangskanalen en (niet noodzakelijk identieke) pakketbronnen beschouwen, en de pakkettransmissietijden in eerste instantie één slot bedragen. De grillige, of gecorreleerde, aard van een pakketstroom die door een bron wordt gegenereerd, wordt gekarakteriseerd aan de hand van een algemeen D-BMAP (discrete-batch Markovian arrival process), wat een generiek kader creëert voor het beschrijven van een superpositie van dergelijke informatiestromen. In een later stadium breiden we onze studie uit tot het geval van transmissietijden met een algemene verdeling, waarbij we ons beperken tot een buffer met één enkel uitgangskanaal.
De analyse van deze wachtlijnmodellen gebeurt hoofdzakelijk aan de hand van een particuliere wiskundig-analytische aanpak waarbij uitvoerig gebruik gemaakt wordt van probabiliteitsgenererende functies, die er toe leidt dat de diverse performantiematen (min of meer expliciet) kunnen worden uitgedrukt als functie van de systeemparameters. Dit resul-teert op zijn beurt in efficiënte en accurate berekeningsalgoritmen voor deze grootheden, die op relatief eenvoudige wijze geïmplementeerd kunnen worden
Improved learning automata applied to routing in multi-service networks
Multi-service communications networks are generally designed, provisioned and configured, based on source-destination user demands expected to occur over a recurring time period. However due to network users' actions being non-deterministic, actual user demands will vary from those expected, potentially causing some network resources to be under- provisioned, with others possibly over-provisioned. As actual user demands vary over the recurring time period from those expected, so the status of the various shared network resources may also vary. This high degree of uncertainty necessitates using adaptive resource allocation mechanisms to share the finite network resources more efficiently so that more of actual user demands may be accommodated onto the network. The overhead for these adaptive resource allocation mechanisms must be low in order to scale for use in large networks carrying many source-destination user demands. This thesis examines the use of stochastic learning automata for the adaptive routing problem (these being adaptive, distributed and simple in implementation and operation) and seeks to improve their weakness of slow convergence whilst maintaining their strength of subsequent near optimal performance. Firstly, current reinforcement algorithms (the part causing the automaton to learn) are examined for applicability, and contrary to the literature the discretised schemes are found in general to be unsuitable. Two algorithms are chosen (one with fast convergence, the other with good subsequent performance) and are improved through automatically adapting the learning rates and automatically switching between the two algorithms. Both novel methods use local entropy of action probabilities for determining convergence state. However when the convergence speed and blocking probability is compared to a bandwidth-based dynamic link-state shortest-path algorithm, the latter is found to be superior. A novel re-application of learning automata to the routing problem is therefore proposed: using link utilisation levels instead of call acceptance or packet delay. Learning automata now return a lower blocking probability than the dynamic shortest-path based scheme under realistic loading levels, but still suffer from a significant number of convergence iterations. Therefore the final improvement is to combine both learning automata and shortest-path concepts to form a hybrid algorithm. The resulting blocking probability of this novel routing algorithm is superior to either algorithm, even when using trend user demands
XIII Jornadas de ingenierÃa telemática (JITEL 2017)
Las Jornadas de IngenierÃa Telemática (JITEL), organizadas por la Asociación de Telemática (ATEL), constituyen un foro propicio de reunión, debate y divulgación para los grupos que imparten docencia e investigan en temas relacionados con las redes y los servicios telemáticos. Con la organización de este evento se pretende fomentar, por un lado el intercambio de experiencias y resultados, además de la comunicación y cooperación entre los grupos de investigación que trabajan en temas relacionados con la telemática.
En paralelo a las tradicionales sesiones que caracterizan los congresos cientÃficos, se desea potenciar actividades más abiertas, que estimulen el intercambio de ideas entre los investigadores experimentados y los noveles, asà como la creación de vÃnculos y puntos de encuentro entre los diferentes grupos o equipos de investigación. Para ello, además de invitar a personas relevantes en los campos correspondientes, se van a incluir sesiones de presentación y debate de las lÃneas y proyectos activos de los mencionados equiposLloret Mauri, J.; Casares Giner, V. (2018). XIII Jornadas de ingenierÃa telemática (JITEL 2017). Editorial Universitat Politècnica de València. http://hdl.handle.net/10251/97612EDITORIA
Actas de las XIV Jornadas de IngenierÃa Telemática (JITEL 2019) Zaragoza (España) 22-24 de octubre de 2019
En esta ocasión, es la ciudad de Zaragoza la encargada de servir de anfitriona a las XIV Jornadas de IngenierÃa Telemática (JITEL 2019), que se celebrarán del 22 al 24 de octubre de 2019. Las Jornadas de IngenierÃa Telemática (JITEL), organizadas por la Asociación de Telemática (ATEL), constituyen un foro propicio de reunión, debate y divulgación para los grupos que imparten docencia e investigan en temas relacionados con las redes y los servicios telemáticos. Con la organización de este evento se pretende fomentar, por un lado el intercambio de experiencias y resultados, además de la comunicación y cooperación entre los grupos de investigación que trabajan en temas relacionados con la telemática. En paralelo a las tradicionales sesiones que caracterizan los congresos cientÃficos, se desea potenciar actividades más abiertas, que estimulen el intercambio de ideas entre los investigadores experimentados y los noveles, asà como la creación de vÃnculos y puntos de encuentro entre los diferentes grupos o equipos de investigación. Para ello, además de invitar a personas relevantes en los campos correspondientes, se van a incluir sesiones de presentación y debate de las lÃneas y proyectos activos de los mencionados equipos