Creation of a carrier-grade telco based on a voice over IP backbone

En aquesta memòria l'autor, fent servir un enfoc modern, redissenya i implementa la plataforma que una empresa de telecomunicacions del segle 21 necessita per poder donar serveis de telefonia i comunicacions als seus usuaris i clients. Al llarg d'aquesta exposició es condueix al lector des d'una fase inicial de disseny fins a la implementació i posada en producció del sistema final desenvolupat, centrant-nos en solucionar les necessitats actuals que això implica. Aquesta memòria cubreix el software, hardware i els processos de negoci associats al repte de fer realitat aquest objectiu, i presenta al lector les múltiples tecnologies emprades per aconseguir-ho, fent emfàsi en la convergència actual de xarxes cap al concepte de xarxes IP i basant-se en aquesta tendència i utilitzant aquesta tecnologia de veu sobre IP per donar forma a la plataforma que finalment, de forma pràctica, es posa en producció.

Voice over IP 2.0: an analysis of limits and potential of IP2IP telecommunication

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.

Studio ed analisi della sicurezza del protocollo SIP nelle reti Voice over IP

Nell'era di Internet e della digitalizzazione, anche la telefonia ha avuto la possibilità di evolversi, e grazie alle tecnologie Voice-over-IP è stato possibile realizzare servizi di comunicazione avanzata su reti di dati. Anche se la comunicazione vocale è l'aspetto chiave di questi sistemi, le reti VoIP supportano altri tipi di servizi, tra cui video, messaggistica istantanea, condivisione di file, ecc. Il successo di questa nuova tipologia di rete è dovuto ad una migliore flessibilità rispetto ai vecchi sistemi analogici, grazie ad architetture aperte e implementazioni a livello software, e soprattutto ad un minor costo legato alle apparecchiature ed ai collegamenti utilizzati, ed ai nuovi modelli di business e di consumo sempre più orientati allo sfruttamento della connettività a banda larga. Tuttavia, l'implementazione dei sistemi VoIP rappresenta anche un grado di complessità maggiore in termini di architetture di rete, di protocolli, e di implementazione, e con questo ne segue un incremento delle possibili vulnerabilità. Una falla nella sicurezza in questi sistemi può portare a disservizi e violazione della privacy per gli utenti con conseguenti ripercussioni economiche per i relativi gestori. La tesi analizza la sicurezza delle reti VoIP concentrandosi sul protocollo che sta alla base dei servizi multimediali, il protocollo SIP. SIP è un protocollo di livello applicativo realizzato per creare, modificare e terminare delle sessioni multimediali tra due o più utenti. Dopo un'introduzione alle generalità del protocollo, vengono esaminate le classi di vulnerabilità delle reti VoIP e gli attacchi a SIP, e vengono presentate alcune contromisure attuabili. Viene mostrato un esempio di come vengano attuati alcuni dei principali attacchi a SIP tramite l'utilizzo di appositi strumenti. L'eborato conclude con alcune considerazioni sulle minacce al protocollo e sugli obiettivi futuri che la comunità scientifica dovrebbe perseguire.

Language independent on-off voice over IP source model with lognormal transitions

The recent explosive growth of voice over IP (VoIP) solutions calls for accurate modelling of VoIP traffic. This study presents measurements of ON and OFF periods of VoIP activity from a significantly large database of VoIP call recordings consisting of native speakers speaking in some of the world's most widely spoken languages. The impact of the languages and the varying dynamics of caller interaction on the ON and OFF period statistics are assessed. It is observed that speaker interactions dominate over language dependence which makes monologue-based data unreliable for traffic modelling. The authors derive a semi-Markov model which accurately reproduces the statistics of composite dialogue measurements. © The Institution of Engineering and Technology 2013.

La voix-over dans le cinéma documentaire

La création cinématographique de l’étudiant qui accompagne ce mémoire sous la forme d’un DVD est disponible à la Médiathèque de la Bibliothèque des lettres et sciences humaines sous le titre : D'où, chose étonnante, rien ne s'en suit.(http://atrium.umontreal.ca/notice/UM-ALEPH002343857)

Performance analysis and deployment of VoLTE mechanisms over 3GPP LTE-based networks

Long Term Evolution based networks lack native support for Circuit Switched (CS) services. The Evolved Packet System (EPS) which includes the Evolved UMTS Terrestrial Radio Access Network (E-UTRAN) and Evolved Packet Core (EPC) is a purely all-IP packet system. This introduces the problem of how to provide voice call support when a user is within an LTE network and how to ensure voice service continuity when the user moves out of LTE coverage area. Different technologies have been proposed for the purpose of providing a voice to LTE users and to ensure the service continues outside LTE networks. The aim of this paper is to analyze and evaluate the overall performance of these technologies along with Single Radio Voice Call Continuity (SRVCC) Inter-RAT handover to Universal Terrestrial Radio Access Networks/ GSM-EDGE radio access Networks (UTRAN/GERAN). The possible solutions for providing voice call and service continuity over LTE-based networks are Circuit Switched Fall Back (CSFB), Voice over LTE via Generic Access (VoLGA), Voice over LTE (VoLTE) based on IMS/MMTel with SRVCC and Over The Top (OTT) services like Skype. This paper focuses mainly on the 3GPP standard solutions to implement voice over LTE. The paper compares various aspects of these solutions and suggests a possible roadmap that mobile operators can adopt to provide seamless voice over LTE.

Quality of experience evaluation of voice communication: an affect-based approach

Voice communication systems such as Voice-over IP (VoIP), Public Switched Telephone Networks, and Mobile Telephone Networks, are an integral means of human tele-interaction. These systems pose distinctive challenges due to their unique characteristics such as low volume, burstiness and stringent delay/loss requirements across heterogeneous underlying network technologies. Effective quality evaluation methodologies are important for system development and refinement, particularly by adopting user feedback based measurement. Presently, most of the evaluation models are system-centric (Quality of Service or QoS-based), which questioned us to explore a user-centric (Quality of Experience or QoE-based) approach as a step towards the human-centric paradigm of system design. We research an affect-based QoE evaluation framework which attempts to capture users' perception while they are engaged in voice communication. Our modular approach consists of feature extraction from multiple information sources including various affective cues and different classification procedures such as Support Vector Machines (SVM) and k-Nearest Neighbor (kNN). The experimental study is illustrated in depth with detailed analysis of results. The evidences collected provide the potential feasibility of our approach for QoE evaluation and suggest the consideration of human affective attributes in modeling user experience.

Tradução e legendagem do documentário Fárria

Trabalho de Projeto Apresentado ao Instituto de Contabilidade e Administração do Porto para a obtenção do grau de Mestre em Tradução e Interpretação Especializadas, sob orientação da Mestre Graça Chorão e coorientação da Mestre Paula Almeida

Realização de uma arquitectura diffServ com configuração transparente de QOS

Dissertação apresentada na Faculdade de Ciências e Tecnologia da Universidade Nova de Lisboa para a obtenção do grau de Mestre em Engenharia Electrotécnica e de Computadores