My work deals with integration of complementary multimedia services into virtual private network in the company VÁHOSTAV-SK, a.s. It covers the possibilities of voice transfer and videoconferencing integration into the system.

The diploma thesis focuses on the VoIP technology optimization and performance management. The diploma thesis presents the theoretical basis of IP telephony and measurement of its quality. The thesis primarily deals with practical measurements of VoIP calls quality. Asterisk softswitch, various types of IP phones and simulated degradation of signal using Linux software router are used for measurements. Procedural diagram of VoIP technology real deployment is designed based on these measurements.

Voice over IP (VoIP) is the group of techniques for delivering voice communications over Internet Protocol (IP) networks. It has mainly served as the possible substitution for regular PSTN over the last decades, but has recently gained an increased interest in various areas such as alarm applications and customer service. Acoustic echo is the situation were a distorted version of the sent signal is transmitted back to the sender, due to acoustic feedback between loudspeaker and microphone. There already exists several algorithms to solve this problem, and this thesis provides a study of the performance in relation to the computational complexity of the algorithms. This is in order to indicate which approaches are better suited for implementation in an embedded system, where resources are limited. During the thesis a number of algorithms were tested, including variations of the LMS algorithm, some other approaches utilizing the correlation between echo and signal, and the RLS algorithm. They were first tested in MATLAB, on speech signals recorded at Syntronic and distorted by adding echo, then tested by implementation in C, and run on speech signals recorded in a simulated VoIP system at Syntronic. The results were then evaluated in terms of efficiency and computational complexity.

Tese de mestrado integrado. Engenharia Electrotécnica e de Computadores - Major Telecomunicações. Faculdade de Engenharia. Universidade do Porto. 2008

Estágio realizado na NovaBase e orientado pelo Eng.º Pedro Rondulha Gomes / Tese de mestrado integrado. Engenharia Informática e Computação. Faculdade de Engenharia. Universidade do Porto. 2008

Transmitting real-time audio or video applications over the Internet is a challenge in the current technology. The motivation for deploying this technology is the reduction in voice communication overheads and the enhancement of services. Voice over Internet Protocol (VoIP) provides improved features like flexible call routing, unified messaging and call center and network multimedia applications which in turn provide reduced costs and improvised services for distance learning, customer support, and remote sales presentations. The integration of voice, video, and data encounters a variable amount of jitter and delay. Typical packet loss ranges from 0 to20 percent and one-way delay from 5 to 500 milliseconds. Reducing jitter delay involves buffering of audio packets at the receiver so that the slower packets arrive sequentially on time at the destination. Adaptive jitter buffering at the receiver improves the quality of voice connections on the Internet. In this thesis, the existing jitter buffer model was further enhanced by proposing a model to change the codecs from a higher bit rate to a lower bit rate during an established call session thus reducing the packet loss and improving the call performance. A simulation model is shown to support this proposal, leading to the development of a new protocol. Various tests were conducted to analyze the performance of a number of calls and bandwidths by varying one and keeping the other constant. / "Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer Engineering. / Includes bibliographic references (leaves 55-57) / "July 2006."

With the network popularization, applications on network are increasing day by day. VoIP (Voice over IP) is one of exciting developments on network applications, and its voice quality has been as good as traditional phones. Furthermore, IP phone users do not afford expensive communication costs, but go to the expense of broadband service. Therefore, more and more people start to use IP phone. After meeting certain requirements of basic communication, users could ask for more additional services. In the thesis, a voicemail service is built under SIP (Session Initiation Protocol). Users can keep missing call by recording voice with digital files, and then, one can receive recorded files from E-mail. Besides, small social networks can communicate with IP phones by using pear-to-pear telephony, which costs less money and labor due to no cost on building servers.

The rapid development of the Internet and Wireless Local Area Network cause VoIP plays an important role among the emerging communication products. The purpose of this study is to explore the VoIP acceptance of the Internet user from the model of the Unified Theory of Acceptance and Use of Technology (UTAUT). This study designed a questionnaire that regarded with the degree of the acceptance of VoIP according to the review of the literatures. After the research, this study indicates that there are three components that influenced the degree of acceptance of VoIP include perceived easy of use, performance expectancy and social influence and different factors under different demography significantly influence the intention of the user. Related to these three components, this study first indicates that sex has a significant influence in the expectation of the achievement and the result shows that male is more significant than female. Second, age has no significant influence. Third, different education level has significant influence in the component of perceived easy of use, especially the user who has Master and PHD degree is more significant than the user only has graduated degree. Forth, different using experience and Voluntariness of Use have different significant influence among these three components.

VoIP(Voice over IP) is one of the most important applications on the Internet. As the voice coding enhancing, VoIP provides a good voice quality with low bandwidth. Therefore, the IP telephony services have a price advantage with PSTN. In the near future, VoIp may replace with the PSTN to provide a better telephony service. In this paper, we discuss how to design and implement an embedded VoIp user agent system. With the help of DSP(Digital Signal Processor), we speed up the voice data processing. We use Linux as the embedded operating system and other open source library to implement a VoIP user agent base on SIP(Session Initiation Protocol) standard. With the hardware and software co-design, we build up an embedded VoIP Integrated Access Device.

This study aims to find possible effects from the use of Voice over internet protocol software together with computer gaming on young adults English output. The study also tries to find effects of media usage on output and if there exists differences between boys and girls. This is done through analysis based on data collected from 28 informants, age 16-18. The data consists of recordings of individual stories coupled with a survey. The empirical data is analyzed in both quantitative and qualitative way. The results show that while English activity has a positive effect, it is hard to attribute this to computer gaming and VOIP alone.