Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

ADAPTIVE SPEECH QUALITY IN VOICE-OVER-IP COMMUNICATIONS

The quality of VoIP communication relies significantly on the network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage the voice-over-IP stream dynamically, changing parameters as needed to assure quality. The main objective of this dissertation is to develop an adaptive speech encoding system that can be applied to conventional (telephony-grade) and wideband voice communications. This comprehensive study includes the investigation and development of three key components of the system. First, to manage VoIP quality dynamically, a tool is needed to measure real-time changes in quality. The E-model, which exists for narrowband communication, is extended to a single computational technique that measures speech quality for narrowband and wideband VoIP codecs. This part of the dissertation also develops important theoretical work in the area of wideband telephony. The second system component is a variable speech-encoding algorithm. Although VoIP performance is affected by multiple codecs and network-based factors, only three factors can be managed dynamically: voice payload size, speech compression and jitter buffer management. Using an existing adaptive jitter-buffer algorithm, voice packet-size and compression variation are studied as they affect speech quality under different network conditions. This study explains the relationships among multiple parameters as they affect speech transmission and its resulting quality. Then, based on these two components, the third system component is a novel adaptive-rate control algorithm that establishes the interaction between a VoIP sender and receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average voice quality than traditional VoIP

Measuring difference, numbering normal: Setting the standards for disability in the interwar period

Measuring difference, numbering normal provides a detailed study of the technological construction of disability by examining how the audiometer and spirometer were used to create numerical proxies for invisible and inarticulable experiences. Measurements, and their manipulation, have been underestimated as crucial historical forces motivating and guiding the way we think about disability. Using measurement technology as a lens, this book draws together several existing discussions on disability, healthcare, medical practice, embodiment and emerging medical and scientific technologies at the turn of the twentieth century. As such, this work connects several important and usually separate academic subject areas and historical specialisms. The standards embedded in instrumentation created strict but ultimately arbitrary thresholds of normalcy and abnormalcy. Considering these standards from a long historical perspective reveals how these dividing lines shifted when pushed. The central thesis of this book is that health measurements are given artificial authority if they are particularly amenable to calculability and easy measurement. These measurement processes were perpetuated and perfected in the interwar years in Britain as the previously invisible limits of the body were made visible and measurable. Determination to consider body processes as quantifiable was driven by the need to compensate for disability occasioned by warfare or industry. This focus thus draws attention to the biopower associated with systems, which has emerged as a central area of concern for modern healthcare in the second decade of the twenty-first century

Measuring difference, numbering normal

Measuring difference, numbering normal provides a detailed study of the technological construction of disability by examining how the audiometer and spirometer were used to create numerical proxies for invisible and inarticulable experiences. Measurements, and their manipulation, have been underestimated as crucial historical forces motivating and guiding the way we think about disability. Using measurement technology as a lens, this book draws together several existing discussions on disability, healthcare, medical practice, embodiment and emerging medical and scientific technologies at the turn of the twentieth century. As such, this work connects several important and usually separate academic subject areas and historical specialisms. The standards embedded in instrumentation created strict but ultimately arbitrary thresholds of normalcy and abnormalcy. Considering these standards from a long historical perspective reveals how these dividing lines shifted when pushed. The central thesis of this book is that health measurements are given artificial authority if they are particularly amenable to calculability and easy measurement. These measurement processes were perpetuated and perfected in the interwar years in Britain as the previously invisible limits of the body were made visible and measurable. Determination to consider body processes as quantifiable was driven by the need to compensate for disability occasioned by warfare or industry. This focus thus draws attention to the biopower associated with systems, which has emerged as a central area of concern for modern healthcare in the second decade of the twenty-first century

Non-Intrusive Subscriber Authentication for Next Generation Mobile Communication Systems

Merged with duplicate record 10026.1/753 on 14.03.2017 by CS (TIS)The last decade has witnessed massive growth in both the technological development, and the consumer adoption of mobile devices such as mobile handsets and PDAs. The recent introduction of wideband mobile networks has enabled the deployment of new services with access to traditionally well protected personal data, such as banking details or medical records. Secure user access to this data has however remained a function of the mobile device's authentication system, which is only protected from masquerade abuse by the traditional PIN, originally designed to protect against telephony abuse. This thesis presents novel research in relation to advanced subscriber authentication for mobile devices. The research began by assessing the threat of masquerade attacks on such devices by way of a survey of end users. This revealed that the current methods of mobile authentication remain extensively unused, leaving terminals highly vulnerable to masquerade attack. Further investigation revealed that, in the context of the more advanced wideband enabled services, users are receptive to many advanced authentication techniques and principles, including the discipline of biometrics which naturally lends itself to the area of advanced subscriber based authentication. To address the requirement for a more personal authentication capable of being applied in a continuous context, a novel non-intrusive biometric authentication technique was conceived, drawn from the discrete disciplines of biometrics and Auditory Evoked Responses. The technique forms a hybrid multi-modal biometric where variations in the behavioural stimulus of the human voice (due to the propagation effects of acoustic waves within the human head), are used to verify the identity o f a user. The resulting approach is known as the Head Authentication Technique (HAT). Evaluation of the HAT authentication process is realised in two stages. Firstly, the generic authentication procedures of registration and verification are automated within a prototype implementation. Secondly, a HAT demonstrator is used to evaluate the authentication process through a series of experimental trials involving a representative user community. The results from the trials confirm that multiple HAT samples from the same user exhibit a high degree of correlation, yet samples between users exhibit a high degree of discrepancy. Statistical analysis of the prototypes performance realised early system error rates of; FNMR = 6% and FMR = 0.025%. The results clearly demonstrate the authentication capabilities of this novel biometric approach and the contribution this new work can make to the protection of subscriber data in next generation mobile networks.Orange Personal Communication Services Lt

Enhanced Quality of Experience Based on Enriched Network Centric and Access Control Mechanisms

In the digital world service provisioning in user satisfying quality has become the goal of any content or network provider. Besides having satisfied and therefore, loyal users, the creation of sustainable revenue streams is the most important issue for network operators [1], [2], [3]. The motivation of this work is to enhance the quality of experience of users when they connect to the Internet, request application services as well as to maintain full service when these users are on the move in WLAN based access networks. In this context, the aspect of additional revenue creation for network operators is considered as well. The enhancements presented in this work are based on enriched network centric and access control mechanisms which will be achieved in three different areas of networks capabilities, namely the network performance, the network access and the network features themselves. In the area of network performance a novel authentication and authorisation method is introduced which overcomes the drawback of long authentication time in the handover procedure as required by the generic IEEE 802.1X process using the EAP-TLS method. The novel sequential authentication solution reduces the communication interruption time in a WLAN handover process of currently several hundred milliseconds to some milliseconds by combining the WPA2 PSK and the WPA2 EAP-TLS. In the area of usability a new user-friendly hotspot registration and login mechanisms is presented which significantly simplifies how users obtain WLAN hotspot login credentials and logon to a hotspot. This novel barcode initiated hotspot auto-login solution obtains user credentials through a simple SMS and performs an auto-login process that avoids the need to enter user name and password on the login page manually. In the area of network features a new system is proposed which overcomes the drawback that users are not aware of the quality in which a service can be provided prior to starting the service. This novel graceful denial of service solution informs the user about the expected application service quality before the application service is started

Analytic Assessment of Telephone Transmission Impact on ASR Performance Using a Simulation Model

This paper addresses the impact of telephone transmission channels on automatic speech recognition (ASR) performance. A real-time simulation model is described and implemented, which allows impairments that are encountered in traditional as well as modern (mobile, IP-based) networks to be flexibly and efficiently generated. The model is based on input parameters which are known to telephone network planners; thus, it can be applied without measuring specific network characteristics. It can be used for an analytic assessment of the impact of channel impairments on ASR performance, for producing training material with defined transmission characteristics, or for testing spoken dialogue systems in realistic network environments. In the present paper, we present an investigation of the first point. Two speech recognizers which are integrated into a spoken dialogue system for information retrieval are assessed in relation to controlled amounts of transmission degradations. The measured ASR performance degradation is compared to speech quality degradation in human-human communication. It turns out that different behavior can be expected for some impairments. This fact has to be taken into account in both telephone network planning as well as in speech and language technology development