Development of an advanced web application for managing videoconference

El objetivo del proyecto es realizar un cliente para videoconferencias de alta definición basado en Web. Se ha utilizado SIP como protocolo de señalización (establecimiento, finalización) de videoconferencias, para la gestión la lista de los contactos, la presencia, y la negociación de las capacidades multimedia.Existen numerosas aplicaciones que ofrecen al usuario la posibilidad de realizar videoconferencias en alta definición sobre Internet. Estas aplicaciones centran sus esfuerzos en la transmisión de contenido de alta calidad sobre Internet, pero dejan sin resolver la gestión del establecimiento, finalización de llamadas, aceptación, rechazo de invitaciones, suscripción y recepción de notificaciones del servicio de presencia (aparición de usuarios, cambios de estado, etc). El objetivo del proyecto es realizar un cliente para videoconferencias de alta definición basado en Web. Se ha utilizado SIP como protocolo de señalización (establecimiento, finalización) de videoconferencias, para la gestión la lista de los contactos, la presencia, y la negociación de las capacidades multimedia La aplicación desarrollada permite que, un usuario a partir de una web, sea capaz de ver todos los usuarios conectados al servicio, conocer sus características y poder establecer videoconferencias de alta calidad, utilizando el entorno de videoconferencia negociado (por ejemplo dvts, ultragrid). La aplicación actúa como un gateway HTTP-SIP, traduciendo las peticiones SIP a peticiones HTTP y viceversa. Las peticiones HTTP son enviadas hacia el cliente para que la trate y realice los cambios necesarios en la interfaz web y en su modelo de datos. La interfaz de web del usuario se ha desarrollado con Google Web Toolkit, un toolkit de Google para el desarrollo de aplicaciones AJAX en lenguaje de programación Java. El servidor se comunica con tres módulos. Con el mundo SIP para la señalización de videoconferencia, con el agente de presencia para gestionar la lista y con el cliente AJAX para comunicarse con los usuarios de la aplicación. El proyecto explica las fases de definición de requerimientos, diseño y arquitectura de la aplicación. Expone el estado del arte de las tecnologías y comenta algunos detalles de la implementación, finalizando con una planificación temporal y unos resultados finales del proyecto

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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Wireless triple play system

Dissertação para obtenção do Grau de Mestre em Engenharia Electrotécnica e ComputadoresTriple play is a service that combines three types of services: voice, data and multimedia over a single communication channel for a price that is less than the total price of the individual services. However there is no standard for provisioning the Triple play services, rather they are provisioned individually, since the requirements are quite different for each service. The digital revolution helped to create and deliver a high quality media solutions. One of the most demanding services is the Video on Demand (VoD). This implicates a dedicated streaming channel for each user in order to provide normal media player commands (as pause, fast forward). Most of the multimedia companies that develops personalized products does not always fulfil the users needs and are far from being cheap solutions. The goal of the project was to create a reliable and scalable triple play solution that works via Wireless Local Area Network (WLAN), fully capable of dealing with the existing state of the art multimedia technologies only resorting to open-source tools. This project was design to be a transparent web environment using only web technologies to maximize the potential of the services. HyperText Markup Language (HTML),Cascading Style Sheets (CSS) and JavaScript were the used technologies for the development of the applications. Both a administration and user interfaces were developed to fully manage all video contents and properly view it in a rich and appealing application, providing the proof of concept. The developed prototype was tested in a WLAN with up to four clients and the Quality of Service (QoS) and Quality of Experience (QoE) was measured for several combinations of active services. In the end it is possible to acknowledge that the developed prototype was capable of dealing with all the problems of WLAN technologies and successfully delivery all the proposed services with high QoE

Delay-centric handover in SCTP

The introduction of the Stream Control Transmission Protocol (SCTP) has opened the possibility of a mobile aware transport protocol. The multihoming feature of SCTP negates the need for a solution such as Mobile IP and, as SCTP is a transport layer protocol, it adds no complexity to the network. Utilizing the handover procedure of SCTP, the large bandwidth of WLAN can be exploited whilst in the coverage of a hotspot, and still retain the 3G connection for when the user roams out of the hotspot’s range. All this functionality is provided at the transport layer and is transparent to the end user, something that is still important in non-mobile-aware legacy applications. However, there is one drawback to this scenario - the current handover scheme implemented in SCTP is failure-centric in nature. Handover is only performed in the presence of primary destination address failure. This dissertation proposes a new scheme for performing handover using SCTP. The handover scheme being proposed employs an aggressive polling of all destination addresses within an individual SCTP association in order to determine the round trip delay to each of these addresses. It then performs handover based on these measured path delays. This delay-centric approach does not incur the penalty associated with the current failover-based scheme, namely a number of timeouts before handover is performed. In some cases the proposed scheme can actually preempt the path failure, and perform handover before it occurs. The proposed scheme has been evaluated through simulation, emulation, and within the context of a wireless environment

Self-Verification Of Public-Key Agreement Over Voip Using Random Fusion Scheme

Telefoni Internet, yang dikenali juga sebagai Suara melalui Protokol Internet (VoIP), menjadi salah satu alternatif telekomunikasi yang popular disebabkan penggunaan Internet yang semakin meluas. Internet memperkaya cara sistem telefoni digunakan, tetapi dalam masa yang sama menimbulkan pelbagai kebimbangan, terutamanya keselamatan Internet telephony, also known as Voice over Internet Protocol (VoIP), has become one of popular alternatives in telecommunication due to the widespread of the Internet usage. The Internet enriches the way of telephony system is used, but in the meantime it elevates many concerns, particularly security