Este trabalho tem como objetivo o estudo da tecnologia VoIP (Voz sobre IP) e a sua aplicação em redes corporativas, enfocando o planejamento de capacidade da rede de dados para absorver serviços VoIP. Serão apresentados tópicos sobre a fundamentação teórica de VoIP (Voz sobre IP), os requisitos de arquitetura de rede e QoS (Qualidade de Serviço) exigidos pelo serviço. Mostra-se também como a metodologia para planejamento de capacidade usado em telefonia convencional pode ser adaptada aos serviços VoIP em uma rede corporativa. Foi aplicada a metodologia adaptada através de um estudo de caso em uma rede corporativa real. / This work has as objective the study of capacity planning in corporate networks for the implementation of VoIP (Voice over IP) services. We will presents topics about the theorical background of VoIP, the requirements of architecture of network and QoS (Quality of Service) demanded by the service. It will also reveal how the methodology used for planning capacity in conventional telephony, could be adjusted to the VoIP services in a corporate network. The adjusted methodology was applied in a real corporate network.

A redução de custo com telefonia através do uso de voz sobre IP tem disparado a busca de soluções que transformem redes IP originalmente dedicadas a transporte de dados em redes para transporte de voz. Esta dissertação tem por objetivo apresentar uma metodologia para sistematizar a avaliação de redes para o tráfego de voz sobre IP de acordo com as possibilidades disponíveis no cenário a ser avaliado. Inicialmente é dada uma visão geral de voz sobre IP, apresentando os protocolos utilizados, os fatores que influenciam na qualidade da voz e os métodos de avaliação de qualidade da voz. Na seqüência são apresentados trabalhos correlatos a avaliação de qualidade de aplicações de voz sobre IP. E por fim descreve-se a proposta de uma metodologia para sistematizar a avaliação de redes com VoIP. / The use of voice over IP telephony was started with solutions to adapt existent data networks to carrier voice streams. The use of monitoring techniques, QoS and signaling protocols can be combined on a such design. Our goal is to present a methodology to evaluate and choose the probing points and the voice quality evaluation techniques to be used in network redesign. An overview about VoIP protocols and parameters that change the voice quality are presented as well as some related works on evaluating voice quality based on network parameters. A proposed methodology is presented, with a case study to show how one can choose the right combination of probing points with some voice quality measurement technique.

Este trabalho tem como objetivo o estudo da tecnologia VoIP (Voz sobre IP) e a sua aplicação em redes corporativas, enfocando o planejamento de capacidade da rede de dados para absorver serviços VoIP. Serão apresentados tópicos sobre a fundamentação teórica de VoIP (Voz sobre IP), os requisitos de arquitetura de rede e QoS (Qualidade de Serviço) exigidos pelo serviço. Mostra-se também como a metodologia para planejamento de capacidade usado em telefonia convencional pode ser adaptada aos serviços VoIP em uma rede corporativa. Foi aplicada a metodologia adaptada através de um estudo de caso em uma rede corporativa real. / This work has as objective the study of capacity planning in corporate networks for the implementation of VoIP (Voice over IP) services. We will presents topics about the theorical background of VoIP, the requirements of architecture of network and QoS (Quality of Service) demanded by the service. It will also reveal how the methodology used for planning capacity in conventional telephony, could be adjusted to the VoIP services in a corporate network. The adjusted methodology was applied in a real corporate network.

Este trabalho apresenta uma proposta de sistema para permitir o acesso à informações numa estrutura baseada em sistemas computacionais, visando monitoramento remoto. Desta forma, pessoas responsáveis por um ambiente eletronicamente monitorado poderão coletar informações em um ambiente computacional através de telefonia IP, síntese de voz e integração com a rede de telefonia pública comutada, obter informações passivamente ou ativamente de uma estrutura monitorada. Os recursos sugeridos para esse modelo proposto foram escolhidas conforme sua utilização no mercado, ou facilidade na aplicação prática. Sempre que possível, usando software livre (Open Source), de tal forma que o formato apresentado neste trabalho possa fomentar outros estudos e ser utilizado para soluções mais sofisticadas. É possível, também, haver aplicações relacionadas com a acessibilidade para deficientes visuais. Sistemas proprietários para acesso a informações remotas estão cada vez mais comuns, especialmente em soluções corporativas, porém acesso a informações através de voz não é uma prática comum. Sabendo que a telefonia cada vez se torna mais acessível e comum de ser acessada de qualquer local, é possível coletar informações em um sistema eletrônico, transformando-as em frases parametrizada e posteriormente, reproduzidas por um sistema computacional. Foram realizados ensaios práticos do sistema proposto com os elementos básicos de funcionamento. Os resultados apresentados foram satisfatórios e indicam que aplicações deste modelo podem ser realizadas em diversas estruturas, observando as adequações para cada uma delas. / This work presents a proposed system for accessing information in structure-based computer systems, aiming at remote monitoring. Persons responsible for an electronically monitored environment may collect information in a computer environment and through IP telephony, voice synthesis and integration with the PSTN (Public Switched Telephone Network), obtain information passively or actively from a monitored environment. The resources suggested for this model were chosen according to their use in the market or facility in practice. Whenever possible, using Open Source software, so that the format presented in this work could stimulate further studies and be used for more sophisticated solutions. You can also have applications related to accessibility for the visually impaired people. Proprietary systems for remote access to information are increasingly common, especially in enterprise solutions, but to access information through voice is not a common practice. Knowing that the phone becomes ever more accessible and common to be accessed from anywhere, it is possible to collect information in an electronic system, transforming them into sentences parameterized and subsequently reproduced by a computer system. Tests with the basic elements of operation were performed. The results were satisfactory and indicate that applications of this model can be realized in various structures, observing the adjustments for each one.

Mobile phones and Interactive Voice Response (IVR) applications are being progressively used in developing countries to collect voice-based reports about bad governance or poor public service delivery, reported by citizens. Such systems (e.g. Avaaj Otalo, Foroba Blon, etc.) can give an opportunity to rural users in developing countries to easily influence and participate in public affairs. Despite the ongoing efforts on using such solutions, the lack of an efficient system of administration can cause delays in broadcasting the collected reports as quickly as possible, to reach the relevant authorities. This thesis presents the results of a real-world deployment of an administrative functionality for an IVR system called U-Call, used in the Northern districts of Uganda. U-Call Administrative Functionality (UCAF) interacts with the U-Call administrators through mobile phones and gives the moderator access to the registered users. It allows administrators to easily publish and tag audio reports over the Web using their mobile phones. It also uses a semantic tagging module to increase findability and information categorization on the U-Call’s website. After an initial validation and successful evaluation of UCAF in the field, during a trip to Uganda, additional features were incorporated, such as multiple authentication process and dynamic tagging. UCAF and its additional features was succefully delivered to the end user, as part of the  U-Call reporting system. / People’s Voices: Developing Cross Media Services to Promote Citizens Participation in Local Governance Activities

Long Term Evolution (LTE) is the last step towards the 4th genera-tion of cellular networks. This revolution is necessitated by the un-ceasing increase in demand for high speed connection on LTE net-works. This thesis mainly focuses on performance evaluation of end-to end delay (E2E) for VoIP in the LTE networks. In the course of E2E performance evaluation, simulation approach is realized using simulation tool OPNET 16.0. Three scenarios have been created. The first one is the baseline network while among other two, one consists of VoIP traffic solely and the other consisted of FTP along with VoIP. E2E delay has been measured for both scenarios in various cases under the varying mobility speed of the node. Furthermore, packet loss for two network scenarios has been studied and presented in the same cases as for E2E delay measurement. Comparative performance analysis of the two networks has been done by the simulation output graphs. In light of the result analysis, the performance quality of a VoIP network (with and without the presence of additional network traffic) in LTE has been determined and discussed. The default parameters in OPNET 16.0 for LTE have been used during simulation.

Session Initiation Protocol (SIP) arises as a new standard of establishing and releasing connections for vast variety of multimedia applications. The protocol may be used for voice calls, video calls, video conferencing, gaming and many more. The 3GPP (3rd Generation Partnership Project) suggests SIP as the signalling solution for 3rd generation telephony. Thereby, this purely IP-centric protocol appears as a promising alternative to older signalling systems such as H.323, SS7 or analog signals in PSTN. In contrast to them, SIP does not focus on communication with PSTN network. It is more similar to HTTP than to any of the mentioned protocols. The main standardisation body behind Session Initiation Protocol is The Internet Engineering Task Force (IETF). The most recent paper published on SIP is RFC 3261 [5]. Moreover, there are working groups within IETF that publish suggestions and extensions to the main standard. One of those extensions is “Caller Preferences for the Session Initiation Protocol (SIP)” [1]. This document describes a set of new rules that allow a caller to express preferences about request handling in servers. They give ability to select which Uniform Resource Identifiers (URI) a request gets routed to, and to specify certain request handling directives in proxies and redirect servers. It does so by defining three new request header fields, Accept-Contact, Reject-Contact, and Request-Disposition, which specify the caller preferences. [1]. The aim of this project is to extend the existing software with caller preferences and evaluate the new functionality.

Statistical distributions are used to model behaviour of real VoIP traffic. We investigate call holding and inter-arrival times as well as speech patterns. The consequences of using an inappropriate model for network dimensioning are briefly discussed. Visual examination is used to compare well known distributions with empirical data. Our results support the general opinion that the Exponential distribution is not appropriate for modelling call holding time. We find that the distribution of talkspurt periods is well modelled by the Lognormal distribution and the silence periods by the generalized Pareto distribution. It is also observed that the call inter-arrival times tend to follow a heavy tailed distribution.

In recent times, the usage of Voice and Video over Internet Protocol (VoIP) services has increased tremendously. There are many signaling protocols such as Bearer Independent Call Control (BICC), H.323, Media Gateway Control Protocol (MGCP), Session Initiation protocol (SIP) etc., that are used for establishment of sessions and to carry out voice and video data services. SIP has become popular because of its easy implementation, flexibility and good scalability. Choice of Open Source SIP server software (OS-SIP Server Software) is important when deploying in a VoIP based network. So we need to evaluate the performance of OS-SIP Server Softwares in an ideal condition before it is deployed to real environment. This document evaluates and compares the performance of three OS-SIP Server Softwares which are quite popular. A SIPp traffic generator tool is used to generate scenarios namely Registration with authentication, Registration without authentication, Session establishment, and Session establishment with response delay at User Agent Server (UAS) side. Using these scenarios, the performance of OS-SIP Server Softwares is evaluated with respect to parameters such as Registrations per second, Calls per second, Response delay, and Percentage of successful calls and registrations. From the experimental results, we observed that there is a significant performance difference among the SIP server softwares. OpenSIPS is the best OS-SIP Server Software for the scenarios Registration with authentication, Registration without authentication and Session establishment with response delay. Asterisk server is the best OS-SIP Server Software when compared with the other two (OpenIMSCore and OpenSIPS) servers for the scenario Session establishment.

This report presents an investigation of the possibilities to implement voice over IP (VoIP) in Sony Ericsson cellular phones. The results from this investigation show that it is partially possible to implement such a solution. The best option for doing so is to make use of the support for the Session Initiation Protocol and the Real-time Transport Protocol offered by the architecture. Another goal is to evaluate if Bluetooth is able to handle the requirements needed for the solution. The whole concept is proven by implementing a prototype. Measurements on this prototype show that Bluetooth will be able to handle the requirements of most IP-based voice communication, i.e., in respect to latency and bandwidth.