Using an automated service to access information via the telephone has become an important productivity enhancer in the developed world. However, such automated services are generally quite inaccessible to users who have had little technological exposure. There has been a widespread belief that speech-recognition technology can be used to bridge this gap, but little objective evidence for this belief has been produced. To address this situation, two interfaces, touchtone and speech-based, were designed and implemented as input modalities to a system that provides technologically unsophisticated users with access to an informational/transactional service. These interfaces were optimised and compared using transaction completion rates, time taken to complete tasks, error rates and user satisfaction. The speech-based interface was found to outperform the touchtone interface in terms of completion rate, error rate and user satisfaction. The data obtained on time taken to complete tasks could not be compared as the DTMF interface data were highly influenced by people who are not technologically unsophisticated. These results serve as a confirmation that speech-based interfaces are more effective and more satisfying and can therefore enhance information dissemination to people who are not well exposed to the technology. / Dissertation (MSc)--University of Pretoria, 2006. / Computer Science / unrestricted

Voice-enabled applications, applications that interact with a user via an audio channel, are used extensively today. Their use is growing as speech related technologies improve, as speech is one of the most natural methods of interaction. They can provide customer support as IVRs, can be used as an assistive technology, or can become an aural interface to the Internet. Given that the telephone is used extensively throughout the globe, the number of potential users of voice-enabled applications is very high. VoiceXML is a popular, open, high-level, standard means of creating voice-enabled applications which was designed to bring the benefits of web based development to services. While VoiceXML is an ideal language for creating these applications, VoiceXML gateways, the hardware and software responsible for interpreting VoiceXML applications and interfacing with the PSTN, are still expensive and so there is a need for a low-cost gateway. Asterisk, and open-source, TDM/VoIP telephony platform, can be used as a low-cost PSTN interface. This thesis investigates adding a VoiceXML service to Asterisk, creating a low-cost VoiceXML prototype gateway which is able to render voice-enabled applications. Following the Component-Based Software Engineering (CBSE) paradigm, the VoiceXML gateway is divided into a set of components which are sourced from the open-source community, and integrated to create the gateway. The browser requires a VoiceXML interpreter (OpenVXI), a Text-To-Speech engine (Festival) and a speech recognition engine (Sphinx 4). The integration of the components results in a low-cost, open-source VoiceXML gateway. System tests show that the integration of the components was successful, and that the system can handle concurrent calls. A fully compliant version of the gateway can be used in the real world to render voice-enabled applications at a low cost. / KMBT_363 / Adobe Acrobat 9.55 Paper Capture Plug-in

The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments. / KMBT_363 / Adobe Acrobat 9.54 Paper Capture Plug-in

Since the deregulation of Voice over Internet Protocol (VoIP) in 2005, many South African organizations are now attempting to leverage its cost saving and competitive values. However, it has been recently cited that VoIP is one of the greatest new risks to organizations and this risk is cited to increase Information Security insurance premiums in the near future. Due to the dynamic nature of the VoIP technology, regulatory and legislative concerns such as lawful interception of communications and privacy may also contribute to business risk. In order to leverage value from the VoIP implementation, an organization should implement the technology with knowledge of the potential risk of civil liability. This is further highlighted by the King III Report which indicates that the Directors of an organization should be ultimately responsible for Corporate Governance and, therefore, IT Governance and Information Security Governance. The report goes further to say that any newly implemented technology, such as VoIP, should comply with all South African legislation and regulations. This responsibility encourages the practice of both due care and due diligence. However, recent trends exercised by Information Security professionals, responsible for drafting Information Security policies and related procedures, often neglect the regulatory requirements and choose to only implement international best practices with no consideration of the risk of civil liability. Although these best practice frameworks may inadvertently comply with existing local legislation, a chance of an oversight is possible. Oversights may not only result in criminal sanctions, but also civil action due to losses or damages suffered. With regard to implementing VoIP, good Corporate Governance could potentially be ensured through the use of both identified regulations and relevant international best practices. This dissertation aims to aid organizations in avoiding or at least mitigating the risk of civil liability to better leverage VoIP’s value, through good Corporate Governance practices. This should aid in the exercise of due care and due diligence when implementing VoIP as a means of conducting business communication.

Voice over IP (VoIP) is a key enabling technology for the migration of circuit-switched PSTN architectures to packet-based IP networks. However, this migration is successful only if the present problems in IP networks are addressed before deploying VoIP infrastructure on a large scale. One of the important issues that the present VoIP networks face is the problem of unwanted calls commonly referred to as SPIT (spam over Internet telephony). Mostly, these SPIT calls are from unknown callers who broadcast unwanted calls. There may be unwanted calls from legitimate and known people too. In this case, the unwantedness depends on social proximity of the communicating parties. For detecting these unwanted calls, I propose a framework that analyzes incoming calls for unwanted behavior. The framework includes a VoIP spam detector (VSD) that analyzes incoming VoIP calls for spam behavior using trust and reputation techniques. The framework also includes a nuisance detector (ND) that proactively infers the nuisance (or reluctance of the end user) to receive incoming calls. This inference is based on past mutual behavior between the calling and the called party (i.e., caller and callee), the callee's presence (mood or state of mind) and tolerance in receiving voice calls from the caller, and the social closeness between the caller and the callee. The VSD and ND learn the behavior of callers over time and estimate the possibility of the call to be unwanted based on predetermined thresholds configured by the callee (or the filter administrators). These threshold values have to be automatically updated for integrating dynamic behavioral changes of the communicating parties. For updating these threshold values, I propose an automatic calibration mechanism using receiver operating characteristics curves (ROC). The VSD and ND use this mechanism for dynamically updating thresholds for optimizing their accuracy of detection. In addition to unwanted calls to the callees in a VoIP network, there can be unwanted traffic coming into a VoIP network that attempts to compromise VoIP network devices. Intelligent hackers can create malicious VoIP traffic for disrupting network activities. Hence, there is a need to frequently monitor the risk levels of critical network infrastructure. Towards realizing this objective, I describe a network level risk management mechanism that prioritizes resources in a VoIP network. The prioritization scheme involves an adaptive re-computation model of risk levels using attack graphs and Bayesian inference techniques. All the above techniques collectively account for a domain-level VoIP security solution.

Gemeinsamer Workshop von Universitaetsrechenzentrum und Professur Rechnernetze und verteilte Systeme (Fakultaet fuer Informatik) der TU Chemnitz. Workshop-Thema: Infrastruktur der ¨Digitalen Universitaet¨ Der Vortrag zeigt Grundlagen und aktuelle Entwicklungen auf dem Gebiet der IP-Telephony sowie Arbeiten des Autors an der TU Darmstadt zu diesem Gebiet.

An increasing number of people in Sweden are having problems with their hearing ability. The three major tools to aid hearing-impaired and deaf individuals are: hearing aids, special telephony, and alerting systems. Both hearing aids and telephony have seen a huge technical development. Hearing aids have gone from huge ponderous devices to small delicate in-ear devices. Simple text telephones have evolved into total conversation telephones with audio, video, and text all operating in real time. Although smart lamps and other alerting services not specifically made for hearing-impaired individuals do exist, the development of alerting system is unsatisfactory. The gap in technology is a huge problem and integration between modern products and alerting systems is getting harder. This thesis explores how to close this gap. The result of this thesis project is a prototype that provides the missing technological link between an alerting systems and modern smart devices. An eventual product should support all kinds of services, but the prototype is limited to solving the problem of connecting an alerting system to a modern total conversation telephones. The prototype was evaluated and based on the evaluation data a timeline was created. An overall positive response towards the product exists and the timeline had adding more third party services (such as Skype and FaceTime) as a high priority. The complete timeline as well as adding Signal Initiation Protocol support is left as future work. / I Sverige har antalet personer med hörselskada ökat de senaste åren. För att hjälpa de med hörselproblem finns det tre viktiga hjälpmedel: hörapparater, special telefoner och varseblivningssystem. Stora teknologiska framsteg har skett för både hörapparater och special telefoner. Hörapparater har gått från stora otympliga apparater till små nätta anordningar som man har i örat. Enkla texttelefoner är idag komplexa system som stödjer både video, ljud och text i realtid. Även fast smarta lampor och andra varseblivningsprodukter existerar så är utveckling för varseblivning speciellt gjorda hörselskadade och döva undermåliga. Gapet som skapats mellan moderna varseblivningsprodukter och varseblivning som hjälpmedel växer sig allt större. Denna rapport ska undersöka detta gap. Resultatet av detta projekt är en prototyp som tillhandahåller den teknologin som ska länka modern varseblivning och varseblivning som hjälpmedel. Den tänkta produkten kan användas för många olika tjänster men i detta projekt är den begränsad till total konversations telefoner. Prototypen har blivit utvärderad och en tidslinje, baserad på utvädringen, har skapats. Tidslinjen ska beskriva kommande tjänster och enheter som skall kunna användas tillsammans med prototypen. Det visar sig att den skapade prototypen blev positivt mottagen och att tjänster som Skype och Facetime skulle ha hög prioritering på tidslinjen.

Cloud computing has evolved from cutting-edge technology to a best practice for businesses across industries. However, compliance with regulatory mandates and addressing security challenges in the cloud environment remain significant concerns. This thesis aims to explore the compliance, regulatory, and security challenges associated with cloud computing, with a particular focus on the differences in regulatory frameworks between an Asian country (India) and a European country (Sweden). Additionally, the study delves into the forensic investigation challenges in terms of evidence collection in the cloud environment. The research methodology involves studying the available literature on regulatory rules and cloud forensics, conducting surveys with cloud customers, experts, and cloud service provider (CSP) professionals, and proposing possible solutions and recommendations to overcome the identified challenges. By addressing these issues, this research contributes to a comprehensive understanding of the impacts of compliance regulations on cloud and IP Telephony services and the security and forensic investigation challenges in cloud platforms.

IP telephony is defined as the transport of telephony calls over an IP network. IP telephony exploits the integration of voice and data networks. However, enterprises are still reluctant to deploy IP telephony despite the potential increase in productivity and reduction of costs. The principal concerns are: can IP telephony provide the same level of performance in terms of security, reliability, and scalability as traditional telephony? If so, are its proclaimed benefits such as flexibility and mobility cost-effective? The aim of this thesis is to analyze how to deploy IP telephony in large corporations - while providing the necessary security and facilitating mobility. Through the different parts of this thesis, we will analyze the applicable technologies, along with their integration and management. We will focus on the essential requirements for an enterprise of scalability, reliability, flexibility, high-availability, and cost-effectiveness. The massive changes brought about due to the deregulation of telecommunications in nearly all countries, the increasingly global nature of business, and the progressively affordable and power technology underlying information and communication technologies have lead to increasing adoption of IP telephony by residential and commercial users. This thesis will examine these technologies in the context of a very large distributed corporation. / IP telefoni är definierat som transporten av telefon samtal genom ett IP nätverk. IP telefoni utnyttjar integrationen av tal och data nätverk. Dock är affärsföretag fortfarande motsträviga till att införa IP telefoni trots potentiell ökning i produktivitet och minskade kostnader. Huvud bekymren är: kan IP telefoni tillhandahålla samma nivå av prestanda med avseende på säkerhet, tillförlitlighet, och skalbarhet som traditionell telefoni? Och i så fall, är dom proklamerade fördelarna flexibilitet och rörlighet kostnadseffektiva? Målet för detta examensarbete är att analysera hur IP telefoni kan införas i stora affärsföretag - medan samtidigt tillhandahålla nödvändig säkerhet och främja rörlighet. Genom olika delar av detta examensarbete, analyserar vi tillämpliga teknologier, inklusive deras integrering och skötsel. Vi kommer att fokusera på de grundläggande kraven för ett affärsföretag gällande skalbarhet, tillförlitlighet, flexibilitet, hög tillgänglighet, och kostnadseffektivitet. Dom massiva förändringarna frambringade i och med avregleringen av telekommunikation i stort sett alla länder, affärsverksamhetens alltmer globala natur, och de progressivt kostnadseffektiva och kraftfulla underliggande teknologier bakom informations och kommunikations system har lett till ökande adoptering av IP telefoni av både privata och kommersiella användare. Detta examensarbete undersöker relevanta teknologier i samband med mycket stora utbredda affärsföretag. / <p>Exchange student from Centro Politecnico Superior (University of Zaragoza, Spain).</p>

Echo in telephones is generally undesirable but inevitable. There are two possible sources of echo in a telephone system. The impedance mismatch in hybrids generates network (electric) echo. The acoustic coupling between loudspeaker and microphone, in hands-free telephones, produces acoustic echo. Echo cancelers are used to control these echoes. In this thesis, we analyze the Least Mean Squares (LMS), Normalized LMS (NLMS), Recursive Least Squares (RLS), and Subband NLMS (SNLMS) algorithms, and evaluate their performance as acoustic and network echo cancelers. The algorithms are compared based on their convergence rate, steady state echo return loss (ERL), and complexity of implementation. While LMS is simple, its convergence rate is dependent on the eigenvalue spread of the signal. In particular, it converges slowly with speech as input. This problem is mitigated in NLMS. The complexity of NLMS is comparable to that of LMS. The convergence rate of RLS is independent of the eigenvalue spread, and it has the fastest convergence. On the other hand, RLS is highly computation intensive. Among the four algorithms considered here, SNLMS has the least complexity of implementation, as well as the slowest rate of convergence. Switching between the NLMS and SNLMS algorithms is used to achieve fast convergence with low computational requirements. For a given computational power, it is shown that switching between algorithms can give better performance than using either of the two algorithms exclusively, especially in rooms with long reverberation times. We also discuss various implementation issues associated with an integrated echo cancellation system, such as double-talk detection, finite precision effects, nonlinear processing, and howling detection and control. The use of a second adaptive filter is proposed, to reduce near-end ambient noise. Simulation results indicate that this approach can reduce the ambient noise by about 20 dB. A configuration is presented for the real time single-chip DSP implementation of acoustic and network echo cancelers, and an interface between the echo canceler and the telephone is proposed. Finally, some results obtained from simulations and implementations of individual modules, on the TMS320C31 and ADSP 2181 processors, are reported. The real time NLMS DSP implementations provide 15 dB of echo return loss. / Master of Science