967 resultados para Voice over IP- VoIP


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Nowadays, networks must support applications such as: distance learning, electronic commerce, access to Internet, Intranets and Extranets, voice over IP (Internet Protocol) and many others. These new applications, employing data, voice, and video traffic, require high bandwidth and Quality of Service (QoS). The ATM (Asynchronous Transfer Mode) technology, together with dynamic resource allocation methods, offers network connections that guarantee QoS parameters, such as minimum losses and delays. This paper presents a system that uses Network Management Functions together with dynamic resource allocation for provision of the end-to-end QoS parameters for rt-VBR connections.

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New versions of SCTP protocol allow the implementation of handover procedures in the transport layer, as well as the supply of a partially reliable communication service. A communication architecture is proposed herein, integrating SCTP with the session initiation protocol, SIP, besides additional protocols. This architecture is intended to handle voice applications over IP networks with mobility requirements. User localization procedures are specified in the application layer as well, using SIP, as an alternative mean to the mechanisms used by traditional protocols, that support mobility in the network layer. The SDL formal specification language is used to specify the operation of a control module, which coordinates the operation of the system component protocols. This formal specification is intended to prevent ambiguities and inconsistencies in the definition of this module, assisting in the correct implementation of the elements of this architecture

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El desarrollo de las redes IP ha marcado un creciente interés por unificar todas las comunicaciones sobre una misma infraestructura aprovechando así el cableado existente. Con esta idea nació la tecnología conocida como VoIP (Voice Over Internet Protocol) que consiste en la trasmisión de la voz sobre paquetes IP. El objetivo principal de este proyecto es el diseño e implementación de una infraestructura de voz sobre IP que utilice una red de datos existente. La primera parte del proyecto estará formada por un estudio detallado de los factores que influyen en esta tecnología: codecs, protocolos y otros factores a tener en cuenta. Tras esta parte, aprovechando la experiencia adquirida durante casi tres años en una empresa integradora de servicios, se realizará un caso de estudio en el que, bajo las premisas impuestas por un supuesto cliente, se diseñará una solución que cumpla todos los requisitos y aporte un valor añadido sobre el sistema de telefonía que posee el cliente. El diseño de la mejor solución se hará utilizando productos del fabricante Cisco Systems y, además del coste económico, se valorarán los esfuerzos personales que costará montar dicha solución, incluyendo un valor añadido como es el dotar de buzón de voz y mensajería a todos los usuarios. La última parte del caso de estudio consistirá en la implementación de la infraestructura anteriormente diseñada adquiriendo conocimientos sobre virtualización de servidores utilizando productos de la compañía VMWare (especialista en virtualización), instalación y parametrización de aplicativos de Cisco y, finalmente, la interconexión con la red pública a través de gateways para poder hacer llamadas al exterior. El proyecto finalizará con la presentación de unas conclusiones finales y la exposición de unas líneas futuras de trabajo. ABSTRACT. The IP network development has marked a growing interest in unifying all communications over the same infrastructure taking advantage of the existing wiring. From this idea, a technology was born known as VoIP (Voice Over Internet Protocol) which consists of the transmission of voice packets over the IP network. The main goal of this project is the design and implementation of a VoIP infrastructure for transmitting voice packets over the existing wired network infrastructure on the client. The first part of the project will consist of a detailed study of the factors influencing this technology: codecs, protocols, and other factors to consider. After this part, drawing on the experience gained during nearly three years in an integrated services company, a case study will be made under the premises imposed for a supposed client. A solution that serves all the requirements will be designed and provide an added value on the customer’s telephone system. The design of the best solution will be done using Cisco Systems products and besides the economic cost of the whole solution, the personal effort cost will be valued. The added value named before will be provided with two important applications such as voice mail and instant messaging for all users. The last part of the case study will consist in the implementation of an infrastructure designed to acquire knowledge about virtualization, using VMWare company products (specialist in virtualization), installation and configuration of applications from Cisco Systems and finally the interconnection with the public network through gateway routers in order to make external calls. The project will end with the presentation of final conclusions and exposing future working lines.

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Bei der Übermittlung von Nachrichtensignalen in Kommunikationsnetzen treten neben der physikalischen Laufzeit weitere Verzögerungen auf, welche durch technische Bearbeitungsvorgänge (Quellencodierung, Verschlüsselung, Kanalcodierung, Leitungscodierung, Bedienungsvorgänge, Vermittlungsvorgänge) verursacht werden. In dieser Arbeit wird die mittlere EndezuEndeDurchlaufverzögerung zwischen zwei Endgeräten in Abhängigkeit von den Eigenschaften und Parametern der Endgeräte und des paketbasierten Kommunikationsnetzes allgemein berechnet. Insbesondere wird der Mindestwert der mittleren Verzögerung in einem paketbasierten Kommunikationsnetz für Sprache (Voice over Internet Protocol, VoIP) und Daten analysiert. Die Wirkung einer Priorisierung der Sprachpakete wird untersucht. Zur Berechnung der Verzögerungen in einem Vermittlungsknoten wird das Bedienungsmodell M / G / 1 / ∞ mit Priorisierung verwendet. Die Anwendung der theoretischen Ergebnisse erfolgt auf ein stark vereinfachtes ModellNetz mit typischen Parametern für die Netzkanten und Netzknoten. Dessen mittlere EndezuEndeDurchlaufverzögerung für Sprachsignale wird unter verschiedenen Randbedingungen (eine oder zwei Verkehrsklassen, mit oder ohne Priorisierung, unterschiedliche Parameter des Datenverkehrs) numerisch berechnet und graphisch veranschaulicht.

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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.

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O advento de novas formas multimídia tem atraído uma clientela exigente, onde preocupação não é somente com o serviço, mas também, com a qualidade que esse serviço pode ser oferecido. As WLAN (Wireless Local Area Networks) tornaram-se a forma mais comum de roteamento de Internet, devido ao seu baixo custo e facilidade de implementação. Para realizar um bom roteamento é necessário um planejamento, utilizando-se modelos. Os modelos de propagação existentes na literatura fazem a predição da intensidade do sinal, mas algumas vezes não contemplam a previsão de um bom serviço. Nesse sentido a presente dissertação propõe-se a elaborar um modelo de propagação empírico indoor multi-andar que não só prediz a potência recebida, mas também faz uma previsão para algumas métricas de QoS (Quality of Service) de chamadas VoIP (Voice over Internet Protocol). Para a elaboração do modelo proposto foram feitas campanhas de medição, em um prédio de dois andares, em pisos distintos mantendo-se a posição do ponto de acesso (PA) fixa. Estudos de geometria analítica para a contagem e agregação de perdas em pisos e paredes. Os resultados do modelo proposto foram comparados com um modelo da literatura que tem um comportamento similar, onde é possível verificar o melhor desempenho do modelo proposto, e para efeito de estudo um andar completamente simulado foi introduzido para avaliação.

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Arguing that Baz Luhrmann's "Australia" (2008) is a big-budget, non-independent film espousing a left-leaning political ideology in its non-racist representations of Aborigines on film, this paper suggests the addition of a 'fourth formation' to the 1984 Moore and Muecke model is warranted. According to their theorising, racist "first formation" films promote policies of assimilation whereas "second formation" films avoid overt political statements in favour of more acceptable multicultural liberalism. Moore and Muecke's seemingly ultimate "third formation films", however, blatantly foreground the director's leftist political dogma in a necessarily low budget, independent production. "Australia", on the other hand, is an advance on the third formation because its feminised Aboriginal voice is safely backed by a colossal production budget and indicates a transformation in public perceptions of Aboriginal issues. Furthermore, this paper argues that the use of low-cost post-production techniques such as voice-over narration by racially appropriate individuals and the use of diegetic song in Australia work to ensure the positive reception of the left-leaning message regarding the Stolen Generations. With these devices Luhrmann effectively counters the claims of right-wing denialists such as Andrew Bolt and Keith Windschuttle.

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Session Initiation Protocol (SIP) is developed to provide advanced voice services over IP networks. SIP unites telephony and data world, permitting telephone calls to be transmitted over Intranets and Internet. Increase in network performance and new mechanisms for guaranteed quality of service encourage this consolidation to provide toll cost savings. Security comes up as one of the most important issues when voice communication and critical voice applications are considered. Not only the security methods provided by traditional telephony systems, but also additional methods are required to overcome security risks introduced by the public IP networks. SIP considers security problems of such a consolidation and provides a security framework. There are several security methods defined within SIP specifications and extensions. But, suggested methods can not solve all the security problems of SIP systems with various system requirements. In this thesis, a Kerberos based solution is proposed for SIP security problems, including SIP authentication and privacy. The proposed solution tries to establish flexible and scalable SIP system that will provide desired level of security for voice communications and critical telephony applications.

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Access to quality higher education is challenging for many Western Australians that live outside the metropolitan area. In 2010, the School of Education moved to flexible delivery of a fully online Bachelor of Education degree for their non -metropolitan students. The new model of delivery allows access for students from any location provided they have a computer and an internet connection. A number of academic staff had previously used an asynchronous environment to deliver learning modules housed within a learning management system (LMS) but had not used synchronous software with their students. To enhance the learning environment and to provide high quality learning experiences to students learning at a distance, the adoption of synchronous software (Elluminate Live) was introduced. This software is a real-time virtual classroom environment that allows for communication through Voice over Internet Protocol (VoIP) and videoconferencing, along with a large number of collaboration tools to engage learners. This research paper reports on the integration of a live e-learning solution into the current LMS environment. Qualitative data were collected from academic staff through informal interviews and participant observation. The findings discuss (i) perceived level of support; (ii) identification of strategies used to create an effective online teacher presence; (iii) the perceived impact on the students' learning outcomes; and (iv) guidelines for professional development to enhance pedagogy within the live e-learning environment.

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The quality of an online university degree is paramount to the student, the reputation of the university and most importantly, the profession that will be entered. At the School of Education within Curtin University, we aim to ensure that students within rural and remote areas are provided with high quality degrees equal to their city counterparts who access face-to-face classes on campus.In 2010, the School of Education moved to flexible delivery of a fully online Bachelor of Education degree for their rural students. In previous years, the degree had been delivered in physical locations around the state. Although this served the purpose for the time, it restricted the degree to only those rural students who were able to access the physical campus. The new model in 2010 allows access for students in any rural area who have a computer and an internet connection, regardless of their geographical location. As a result enrolments have seen a positive increase in new students. Academic staff had previously used an asynchronous environment to deliver learning modules housed within a learning management system (LMS). To enhance the learning environment and to provide high quality learning experiences to students learning at a distance, the adoption of synchronous software was introduced. This software is a real-time virtual classroom environment that allows for communication through Voice over Internet Protocol (VoIP) and videoconferencing, along with a large number of collaboration tools to engage learners. This research paper reports on the professional development of academic staff to integrate a live e-learning solution into their current LMS environment. It involved professional development, including technical orientation for teaching staff and course participants simultaneously. Further, pedagogical innovations were offered to engage the students in a collaborative learning environment. Data were collected from academic staff through semi-structured interviews and participant observation. The findings discuss the perceived value of the technology, problems encountered and solutions sought.

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Australia is a vast land and access to quality higher education is challenging for many Australians that live outside the larger metropolitan areas. In 2010, the School of Education at an Australian university (Curtin University in Western Australia) moved to flexible delivery of a fully online Bachelor of Education degree for their rural students. The new model of delivery allows access for students from any location provided they have a computer and an internet connection.A number of teaching staff had previously used an asynchronous environment to deliver learning modules housed within a learning management system (LMS) but had not used synchronous software with their students. To enhance the learning environment and to provide high quality learning experiences to students learning at a distance, the adoption of synchronous software (Elluminate Live) was introduced. This software is a real-time virtual classroom environment that allows for communication through Voice over Internet Protocol (VoIP) and video conferencing, alongside a large number of collaboration tools to engage learners.This research paper reports on the integration of a live e-learning solution into the current Learning Management System (LMS) environment. Staff were interviewed about their perceptions and a questionnaire was administered to a sample of students to identify their experience with the synchronous software in order to inform future practice.

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Tese dout., Engenharia Electrónica e Computação, Universidade do Algarve, 2009

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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)

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Voz sobre IP (VoIP) é uma tecnologia que permite a digitalização e a codificação da voz e o empacotamento em pacotes de dados IP para a transmissão em uma rede que utilize o protocolo TCP/IP. Devido ao volume de dados gerados por uma aplicação VoIP, esta tecnologia se encontra em funcionamento, em redes corporativas privadas. Mas se a rede base para o transporte desta aplicação for a Internet, certamente, não deve ser utilizada para fins profissionais, pois o TCP/IP não oferece padrões de QoS (Qualidade de Serviço) comprometendo desta forma a qualidade da voz. A qualidade da voz fica dependente do tráfego de dados existentes no momento da conversa. Para realizar um projeto de VoIP é necessário conhecer todo o tráfego existente na rede e verificar o quanto isto representa em relação à banda total da rede. Também se deve conhecer o tipo de aplicação que se deseja implantar, verificando a banda a ser utilizada por esta, e então projetar como a rede deverá ser estruturada. Para auxiliar no projeto de VoIP, pretende-se mostrar o que está sendo desenvolvido para que o protocolo TCP/IP ofereça QoS e uma ferramenta para análise do tráfego de voz sobre redes TCP/IP e também análises dos resultados obtidos em experimentos simulando diversas situações práticas.

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Com o intuito de utilizar uma rede com protocolo IP para a implementação de malhas fechadas de controle, este trabalho propõe-se a realizar um estudo da operação de um sistema de controle dinâmico distribuído, comparando-o com a operação de um sistema de controle local convencional. Em geral, a decisão de projetar uma arquitetura de controle distribuído é feita baseada na simplicidade, na redução dos custos e confiabilidade; portanto, um diferencial bastante importante é a utilização da rede IP. O objetivo de uma rede de controle não é transmitir dados digitais, mas dados analógicos amostrados. Assim, métricas usuais em redes de computadores, como quantidade de dados e taxa de transferências, tornam-se secundárias em uma rede de controle. São propostas técnicas para tratar os pacotes que sofrem atrasos e recuperar o desempenho do sistema de controle através da rede IP. A chave para este método é realizar a estimação do conteúdo dos pacotes que sofrem atrasos com base no modelo dinâmico do sistema, mantendo o sistema com um nível adequado de desempenho. O sistema considerado é o controle de um manipulador antropomórfico com dois braços e uma cabeça de visão estéreo totalizando 18 juntas. Os resultados obtidos mostram que se pode recuperar boa parte do desempenho do sistema.