Voice over IP

The area that this thesis covers is Voice over IP (or IP Telephony as it is sometimes called) over Private networks and not over the Internet. There is a distinction to be made between the two even though the term is loosely applied to both. IP Telephony over Private Networks involve calls made over private WANs using IP telephony protocols while IP Telephony over the Internet involve calls made over the public Internet using IP telephony protocols. Since the network is private, service is reliable because the network owner can control how resources are allocated to various applications, such as telephony services. The public Internet on the other hand is a public, largely unmanaged network that offers no reliable service guarantee. Calls placed over the Internet can be low in quality, but given the low price, some find this solution attractive. What started off as an Internet Revolution with free phone calls being offered to the general public using their multimedia computers has turned into a telecommunication revolution where enterprises are beginning to converge their data and voice networks into one network. In retrospect, an enterprise\u27s data networks are being leveraged for telephony. The communication industry has come full circle. Earlier in the decade data was being transmitted over the public voice networks and now voice is just another application which is/will be run over the enterprises existing data networks. We shall see in this thesis the problems that are encountered while sending Voice over Data networks using the underlying IP Protocol and the corrective steps taken by the Industry to resolve these multitudes of issues. Paul M. Zam who is collaborating in this Joint Thesis/project on VoIP will substantiate this theoretical research with his practical findings. On reading this paper the reader will gain an insight in the issues revolving the implementation of VoIP in an enterprises private network as well the technical data, which sheds more light on the same. Thus the premise of this joint thesis/project is to analyze the current status of the technology and present a business case scenario where an organization will be able to use this information

Design and implementation of an on-line demonstrator for a video telephony system over heterogeneus networks

In recent times, Next Generation Mobile Networks (NGMN) enable user's mobility, not needing to be in a fixed place anymore. For this service to be successful, seamless transitions between the different technologies become essential, in order to make possible the always best connected goal. This brings an additional problematic, which is the impairments resulting from a handover between two networks. In order to succesfully plan and continue the development of always on services and mobility management, the approach must be based on user's perception of phenomena such us packet loss, the so-called Quality of Experience (QoE). This is the context in which Mobisense was born, intending a better understanding of NGMN transmission phenomena and resulting quality. Hence, Mobisense project is focused on the evaluation of the quality of service from user's point of view and on the seamless switch provision between video codecs when connections are transferred between two networks. On that purpose, a NGMN test environment was developed for real-time multimedia services. In this environment, specific network conditions can be associated with user's assessments within a realistic model. Mobisense project creates the foundations for the employment of advance prediction methods for real-time mobility management, to make decisions depending on the characteristics measured in the network and the predictions of quality resulting from them. The present degree final project gathers the work carried out to develop an extension for Mobisense testbed, in order to deploy it in a real environment of network technologies, as well as the integration with Quality of Service (QoS) algorithms. Therefore, the aim of this project consists on the development of the software required for the creation of a video thelephony system over NGMN, taking Mobisense and MultiRAT testbeds as starting point. Mobisense brings adaptation in the application layer and user's perception, and MultiRAT provides QoS adaptation and new wireless technologies. Both testbeds combined to explore more QoE aspects in wireless networks of tomorrow.En los últimos tiempos, las NGMN posibilitan la movilidad del usuario, sin ser ya necesaria su permanencia en un lugar fijo. Para el éxito de este servicio se hacen indispensables transiciones continuas entre las diferentes tecnologías, de manera que sea posible el objetivo de "siempre la mejor conexión". Esto lleva consigo una problemática adicional, que son las de ciencias resultantes del handover entre dos redes. Para planificar y continuar satisfactoriamente el desarrollo de servicios always on y la gestión de la movilidad, el enfoque debe ser en base a la percepción del usuario de fenómenos tales como la pérdida de paquetes, la llamada QoE. En este contexto nació el proyecto Mobisense, buscando una mejor comprensión del fenómeno de transmisión NGMN y la calidad resultante. El proyecto Mobisense se centra, por tanto, en la evaluación de la calidad de servicio desde el punto de vista del usuario y en la provisión de cambios continuos entre codecs de vídeo al transmitirse conexiones entre dos redes. Para tal propósito, un entorno de pruebas NGMN fue desarrollado para servicios multimedia en tiempo real. En este entorno, pueden asociarse determinadas condiciones en la red con valoraciones de calidad por parte del usuario en un modelo realista. El proyecto Mobisense sienta las bases para el empleo de métodos avanzados de predicción para la gestión de movilidad en tiempo real, para tomar decisiones dependientes de las características de la red medidas y de las predicciones de calidad derivadas a partir de éstas. El presente proyecto de fin de carrera recoge el trabajo realizado para desarrollar una extensión del testbed Mobisense, de cara a desplegarlo en un entorno real de tecnologías de red, así como la integración de algoritmos de QoS. Por tanto, el objeto de este proyecto consiste en el desarrollo software requerido para la creación de un sistema de videotelefonía sobre NGMN, tomando los testbeds Mobisense y MultiRAT como punto de partida. Mobisense aporta adaptación en la capa de aplicación y la percepción de usuario, y MultiRAT proporciona adaptación QoS y nuevas tecnologías de red. Ambos testbeds se combinan para la exploración más amplia de los aspectos de QoE en las redes inalámbricas del mañana.Ingeniería de Telecomunicació

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

Convivo communicator : an interface-adaptive voice over IP system for poor quality networks

Thesis (S.M.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2002.Includes bibliographical references (leaves 100-102).This thesis presents Convivo, a VoIP system designed to provide reliable voice communication for poor quality networks, especially those found in rural areas of the developing world. Convivo introduces an original approach to maintain voice communication interaction in presence of poor network performance: an Interface-Adaptation mechanism that suggests adjusting the application user interface to conform to one of three voice communication modalities (full duplex, half duplex, and voice messaging). The thesis proposes that changes in communication modality are an option to sustain voice communication interaction despite poor network performance. The goals of the changes in communication modality are to reduce the impact of high latency and low bandwidth on voice communication interaction, to facilitate turn taking for a high latency connection, and to sustain voice communication for extremely low bandwidth or high error links. The system was tested via a user study in Bohechio, a small village in the Dominican Republic. The study found that Interface-Adaptation helped users to maintain voice communication interaction when network performance degrades. Transitions from full duplex to voice messaging were found particularly valuable. Initial results suggest that as users get more experience with the application they would like to manually control transitions based on feedback provided by the application and their own perceived voice quality.by Marco Antonio Escobedo Gonzalez Maiz.S.M