Thesis (Ph. D.)--University of North Texas, Dec., 2007. / Title from title page display. Includes bibliographical references.

Thesis (M.S. in Computer Science)--Naval Postgraduate School, December 2002. / Thesis advisor(s): Geoffrey Xie, Burt Lundy. Includes bibliographical references (p. 91-95). Also available online.

Thesis (Ph.D.) - Carleton University, 2007. / Includes bibliographical references (p. 194-2007). Also available in electronic format on the Internet.

Thesis (M. Eng.)--Carleton University, 2001. / Includes bibliographical references (p. 111-113). Also available in electronic format on the Internet.

Thesis (M.S.)--Villanova University, 2007. / Computer Science Dept. Includes bibliographical references.

Magister Scientiae - MSc / Location-based service (LBS) is a geographical location-related service that provides highly personalized services for users. It is a platform for network operators to provide new and innovative ways of increasing profits from new services. With the rapidly growing trend toward LBS, there is a need for standard LBS protocols. This thesis started with introducing the Internet Engineering Task Force GEOPRIV working group, which endeavors to provide standard LBS protocols capable of transferring geographic location information for diverse location-aware applications. Through careful observation, it was found that Session Initiation Protocol (SIP) is well suited to the GEOPRIV requirements. The aim of this research was therefore to explore the possibility of the integration of LBS and the SIP protocol and, to some extent fulfill the GEOPRIV requirements. / South Africa

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. / <p>Also identified as KTH publication TRITA-EE 2009:025</p> / Real-time router / Vinnova SIBED

This thesis presents a domain-independent approach to dynamic system management using goals and policies. A goal is a general, high-level aim a system must continually work toward achieving. A policy is a statement of how a system should behave for a given set of detectable events and conditions. Combined, goals may be realised through the selection and execution of policies that contribute to their aims. In this manner, a system may be managed using a goal-directed, policy-based approach. The approach is a collection of related techniques and tools: a policy language and policy system, goal definition and refinement via policy selection, and conflict filtering among policies. Central to these themes, ontologies are used to model application domains, and incorporate domain knowledge within the system. The ACCENT policy system (Advanced Component Control Enhancing Network Technologies, http://www.cs.stir.ac.uk/accent) is used as a base for the approach, while goals and policies are defined using an extension of APPEL (Adaptable and Programmable Policy Environment and Language, http://www.cs.stir.ac.uk/appel). The approach differs from existing work in that it reduces system state, goals and policies to a numerical rather than logical form. This is more user-friendly as the goal domain may be expressed without any knowledge of formal methods. All developed techniques and tools are entirely domain-independent, allowing for reuse with other event-driven systems. The ability to express a system aim as a goal provides more powerful and proactive high-level management than was previously possible using policies alone. The approach is demonstrated and evaluated within this thesis for the domains of Internet telephony and sensor network/wind turbine management.

Thesis (MScEng (Electrical and Electronic Engineering))--University of Stellenbosch, 2010. / ENGLISH ABSTRACT: Telephony has developed substantially over the years, but the fundamental auditory model of mixing all the audio from di erent sources together into a single monaural stream has not changed since the telephone was rst invented. Monaural audio is very di cult to follow in a multiple-source situation such as a conference call. Sound originating from a speci c point in space will travel along a slightly di erent path to each ear. Although we are not consciously aware of it, our brain processes these spatial cues to help us to locate sounds in space. It is this spatial information that allows us to focus our attention and listen to a single speaker in an environment where many di erent sources may be active at the same time; a phenomenon known as the \cocktail party e ect". It is possible to reproduce these spatial cues in a sound recording, using Head-Related Transfer Functions (HRTFs) to allow a listener to experience localised audio, even when sound is reproduced through a headset. In this thesis, spatial audio is implemented in a telephony application as well as in a virtual world. Experiments were conducted which demonstrated that spatial audio increases the intelligibility of speech in a multiple-source environment and aids active speaker identi cation. Resource usage measurements show that these bene ts are, however, not without a cost. In conclusion, spatial audio was shown to be an improvement over the monaural audio model traditionally implemented in telephony. / AFRIKAANSE OPSOMMING: Telefonie het ansienlik ontwikkel oor die jare, maar die basiese ouditiewe model waarin die klank van alle verskillende bronne bymekaar gemeng word na een enkelouditoriese stroom het nie verander sedert die eerste telefoon gebou is nie. Enkelouditoriese klank is baie moeilik om te volg in 'n meervoudigebron situasie, soos byvoorbeeld in 'n konferensie oproep. Klank met oorsprong by 'n sekere punt in die ruimte sal 'n e ens anderse pad na elke oor volg. Selfs is ons nie aktief bewus hiervan nie, verwerk ons brein hierdie ruimtelike aanduidinge om ons te help om klanke in die ruimte te vind. Dit is hierdie ruimtelike inligting wat ons toelaat om ons aandag te vestig en te luister na 'n enkele spreker in 'n omgewing waar baie verskillende bronne terselfdertyd aktief mag wees, 'n verskynsel wat bekend staan as die \skemerkelkiepartytjiee ek". Dit is moontlik om hierdie ruimtelike leidrade na 'n klank te reproduseer met behulp van hoofverwandeoordragfunksies (HRTFs) en om daardeur 'n luisteraar gelokaliseerde klank te laat ervaar, selfs wanneer die klank deur middel van oorfone gespeel word. In hierdie tesis word ruimtelike klank ge mplementeer in 'n telefonieprogram, sowel as in 'n virtuelew^ereld. Eksperimente is uitgevoer wat getoon het dat ruimtelike klank die verstaanbaarheid van spraak in 'n meerderebronomgewing verhoog en help met aktiewe spreker identi kasie. Hulpbrongebruiks metings toon aan dat hierdie voordele egter nie sonder 'n koste kom nie. Ter afsluiting, dit is bewys dat ruimtelike klank 'n verbetering tewees gebring het oor die enkelouditorieseklankmodel wat tradisioneel in telefonie gebruik het.

Location-based service (LBS) is a geographical location-related service that provides highly personalized services for users. It is a platform for network operators to provide new and innovative ways of increasing profits from new services. With the rapidly growing trend toward LBS, there is a need for standard LBS protocols. This thesis started with introducing the Internet Engineering Task Force GEOPRIV working group, which endeavors to provide standard LBS protocols capable of transferring geographic location information for diverse location-aware applications. Through careful observation, it was found that Session Initiation Protocol (SIP) is well suited to the GEOPRIV requirements. The aim of this research was therefore to explore the possibility of the integration of LBS and the SIP protocol and, to some extent fulfill the GEOPRIV requirements.