The voice activity detection (VAD) recorder and VAD network recorder : a thesis presented in partial fulfilment of the requirements for the degree of Master of Science in Computer Science at Massey University

The project is to provide a feasibility study for the AudioGraph tool, focusing on two application areas: the VAD (voice activity detector) recorder and the VAD network recorder. The first one achieves a low bit-rate speech recording on the fly, using a GSM compression coder with a simple VAD algorithm; and the second one provides two-way speech over IP, fulfilling echo cancellation with a simplex channel. The latter is required for implementing a synchronous AudioGraph. In the first chapter we introduce the background of this project, specifically, the VoIP technology, the AudioGraph tool, and the VAD algorithms. We also discuss the problems set for this project. The second chapter presents all the relevant techniques in detail, including sound representation, speech-coding schemes, sound file formats, PowerPlant and Macintosh programming issues, and the simple VAD algorithm we have developed. The third chapter discusses the implementation issues, including the systems' objective, architecture, the problems encountered and solutions used. The fourth chapter illustrates the results of the two applications. The user documentations for the applications are given, and after that, we analyse the parameters based on the results. We also present the default settings of the parameters, which could be used in the AudioGraph system. The last chapter provides conclusions and future work

Time delay estimation algoritms for echo cancellation

The following case study describes how to eliminate echo in a VoIP network using delay estimation algorithms. It is known that echo with long transmission delays becomes more noticeable to users. Thus, time delay estimation, as a part of echo cancellation, is an important topic during transmission of voice signals over packetswitching telecommunication systems. An echo delay problem associated with IP-based transport networks is discussed in the following text. The paper introduces the comparative study of time delay estimation algorithm, used for estimation of the true time delay between two speech signals. Experimental results of MATLab simulations that describe the performance of several methods based on cross-correlation, normalized crosscorrelation and generalized cross-correlation are also presented in the paper

Signalling in voice over IP Networks

Voice signalling protocols have evolved, keeping with the prevalent move from circuit to packet switched networks. Standardization bodies have provided solutions for carrying voice traffic over packet networks while the main manufacturers are already providing products in workgroup, enterprise, or operator portfolio. This trend will accrue in next years due to the evolution of UMTS mobile networks to an “all-IP” environment. In this paper we present the various architectures that are proposed for signalling in VoIP, mainly: H.323, SIP and MGCP. We also include a brief summary about signalling in classical telephone networks and, at the end, we give some ideas about the proposed “all-IP” architectures in UMTS 3G mobile networks.Publicad

Effect of Free Bandwidth on VoIP Performance in 802.11b WLAN Networks

In this paper we experimentally study the relationship between bandwidth utilization in the wireless LAN and the quality of VoIP calls transmitted over the wireless medium. Specifically we evaluate how the amount of free bandwidth decreases as the number of calls increases and how this influences transmission impairments (i.e. delay, loss and jitter) and thus degrades call quality. We show that the amount of free bandwidth is a good indicator for predicting VoIP call quality

Mobile VoIP : managing , scheduling and refining voice packets to and from mobile phones

This thesis report is submitted in partial fulfillment of the requirements for the degree of Bachelor of Science in Computer Science and Engineering, 2006.Cataloged from PDF version of thesis report.Includes bibliographical references (page 19).Voice over IP (VoIP) is rapidly gaining acceptance. VoIP has already become a multi billion dollar industry in the IT world. On the other hand the use of mobile phones in the world has exceeded all estimations. Given these two technologies, we are unfortunate that there is no VoIP support available for mobile phones. The bridge between VoIP and mobile phones is a long way, but an initiative like this paper would surely lay down the roadmap. Mobile phones have several annoying limitations. Limited data transfer rate, processor speed, lack of protocol implementation etc. are the relevant issues for VoIP. This paper addresses the challenges of VoIP and tries to explore ways to overcome those challenges in mobile phones. The language we used for our experiment is JavaME for its simplicity and wide acceptance.Mohammad Abdus SalamTapan Biswas,B. Computer Science and Engineerin

TABLE OF CONTENTS

Once you are aware of the benefits and applications of Voice over IP, it is too good to resist. Perhaps that is why vendors are flooding the market with VOIP products and services. The following paper analyzes the various issues in the evolving VOIP technology and the challenges in the development of VOIP products. It then presents the features of few VOIP Products offered by the leaders in this field, how well they handle the issues and som

An Algorithm to Evaluate the Echo Signal and the Voice Quality in VoIP Networks

Voice over the Internet Protocol (VoIP) has been increasingly popular, but reliability and voice quality remain important factors that limit the widespread adoption of VoIP systems. Providing good voice quality is of major importance for the transition from the PSTN to VoIP networks. There are several non-real-time algorithms that estimate the voice quality such as the PESQ and the E-model. In this thesis we propose a real-time fuzzy algorithm to estimate the echo quality component of the voice quality in VoIP networks. Differently from the existing algorithms, the proposed algorithm does not need a reference signal and has low computational complexity. For these reasons, the proposed algorithm can be embedded in every VoIP system of a network to monitor live calls, giving an estimate of the instantaneous voice quality to the network provider

An Investigation On An Efficient Approach Of Improving Quality Of Service Of Voip Over Satellite

Kepentingan Protokol Suara Melalui Internet (VoIP) semakin diperakui oleh industri telekomunikasi. Sejak dekad yang lalu banyak perisian VoIP telah dibangunkan yang menawarkan banyak faedah kepada kedua-dua pembekal perkhidmatan rangkaian dan telekomunikasi. The importance of Voice over Internet Protocol (VoIP) is slowly being recognized by the telecommunications industry. In the past decade, many VoIP applications have been developed, offering a wide range of benefits to both telecommunications and network service providers