The increased use of computer networks has lead to the adoption of Internet-based solutions for reducing telephony costs. This has proved to be a boon to callers who can reach the other party directly via the Internet. Unfortunately numerous business persons still need to call to and from mobile phones which are currently a domain where the customers are generally tightly bound to their operators. To provide a simple solution to this problem for companies, Opticall AB has designed an integrated system called the Dial over Data solution, coupling a mobile interface with a low-rate communication channel, which allows calls to be originated remotely at the best price, exploiting the customer company's existing network. This scheme allows the customer company to easily control telecommunications costs, to monitor their employees' efficiency, and more generally speaking to claim a central role in the communications of their employees. The proposed solution allows distant callers (usually employees of the customer company) to benefit from the company's internal network, which is usually more cost effective and offering connectivity to more networks than a cell phone. The Dial over Data solution enables communication between any phone accessible from the customer company's telephony network (such as SIP clients, landline phones, and mobile phones) at a lower cost.</p> This thesis project analyzes existing technologies and compares them to the pre-existing prototype to ascertain the validity of the method and of the components used. This project also explains the improvements brought to the features offered by the DoD solution: the initial prototype has been developed into a stable and functional product, and has been tested internally. Prompted by a need for scalability and additional features, the replacement of Asterisk for the handling of SIP calls by other SIP servers has also been considered and tested. / Den nuvarande ökningen av datanätverk har lett till adoptionen av Internetbaserade lösningar för att förminskar kostnader inom telefoni. Tyvärr behöver åtskilliga affärsmän fortfarande ringa till och ifrån mobiler som återstår som ett område där kunderna är fastkedjade till deras operatörer. För att tillföra en enkel lösning till detta problem för kundföretag har Opticall AB planlagt ett integrerat system som kallas Dial over Data som kopplar ihop ett mobilt gränssnitt, med en billig kommunikationsmedel, som tillåter telefonsamtal påbörjas avlägset på det billigaste priset tack vare företagets nätverk. Det ger möjligheten till kundföretaget att vara centralt för sina personals kommunikationer. Det medger ett enkelt sätt att kontrollera kostnader samt övervaka personalens effektivitet. Den Dial over Data lösningen är en lösning som tillåter avlägsen besökarna med kundföretagets inre nätverk kommer att dra nytta av eftersom det är mer kostnadseffektiv och flexibel än en blott mobiltelefon. Denna möjliggör kommunikation mellan SIP-klienter, fast telefoni och mobiltelefoni för en lägre kostnad till företaget utan att framkalla besvär för sina anställda. Konnektiviteten till företagets inre nätverk samt en låg besvärlighetsnivå är garanterade respektive genom konfigurationsförmågan av produkten och ett praktiskt gränssnitt som är redo för korporativkontaktlistor och visar alla informationen som är relevanta till förbrukarnas erfarenhet.</p> Det här avhandlingsprojektet analyserar existerande teknologier och sätter dem i relation till den sedan tidigare framtagna prototypen för att utröna validiteten hos metoden och beståndsdelar. Projektet förklarar även de förbättringar som gjorts till de egenskaper som erbjuds av DOD-lösningen: prototypen har utvecklats till en stabil och funktionell produkt och har testats internt. Driven av behovet för skalabilitet och ytterligare egenskaper har ersättandet av Asterisk för hantering av SIP samtal av andra SIP servers övervägts och testats.

Master's Thesis is engaged on possibilities connect SIP server to telephone network on FIT. The main reason is that employees can call to university, when they are out of faculty. For resolution we will use SIP server Asterisk, which will be serving as authorization server for users. Next Asterisk will ensure transmission numbers to SIP address with ENUM. In the practical part we will verify the functionality.

Many researchers and speech user interface practitioners assert that interactive voice response (IVR) menus must be relatively short due to constraints of the human memory system. These individuals commonly cite Miller's (1956) paper to support their claims. The current paper argues that these authors commonly misuse the information provided in Miller's paper and that hypotheses drawn from modern theories of working memory (e.g., Baddeley and Hitch, 1974) would lead to the opposite conclusion – that reducing menu length by creating a greater number of menus and a deeper structure will actually be more demanding on users' working memories and will lead to poorer performance and poorer user satisfaction. The primary purpose of this series of experiments was to gain a greater understanding of the role of working memory in speech-enabled IVR use. The experiments also sought to determine whether theories of visual search and signal detection theory (SDT) could be used to predict auditory search behavior. Results of this experiment indicate that creating a deeper structure with shorter menus is detrimental to performance and satisfaction and more demanding of working memory resource. Further the experiment provides support for arguments developed from Macgregor, Lee, and Lam's dual criterion decision model and is a first step toward applying SDT to the IVR domain.

Šio darbo tema - Internetinės telefonijos teisinis reglamentavimas Lietuvoje. Šiuolaikinis Internetinės telefonijos populiarumas ne tik sukelia vartotojų, bet ir reguliuotojų suinteresuotumą. Būtent ji pastaruoju metu kelia daugybę diskusijų tarptautiniuose bei nacionalinėse forumuose, kurių vienas pagrindinių aspektų – tinkamo Internetinės telefonijos reguliavimo sukūrimas. Pagrindinis baigiamojo magistrinio darbo tikslas – išnagrinėti Internetinės telefonijos reguliavimą tarptautiniu ir Lietuvos mastu, bei apžvelgti su juo susijusias problemas. Darbe nagrinėjama užsienio šalių praktika, remiantis kuria iškeliamos pagrindinės Lietuvos IP telefonijos teisinio reguliavimo gairės. Būtent: telefono numerių skyrimas, numerio perkeliamumas, skambučiai į pagalbos tarnybas, skambučiai kitais telefono numeriais, IP telefonijos skambučių saugumas, bei aprašomos su jų įgyvendinimu susijusios problemos. / The topic of the paper is Legal Reglamentation of VoIP Telephony in Lithuania. VoIP telephony’s nowadays spread and popularity scores an interest and debates not only between it’s consumers but also between legal regulators. VoIP is the main and rather often discussed topic of international and national forums, which aim to develop its proper regulation. The main aim of this paper is to internationally analyse VoIP’s legal regulation and to survey its associated problems. Foreign countries’ experience helps to formulate basic guidelines of Lithuanian VoIP legal regulation. Namely: numbering, numbers portability, calls to emergency services, calls to other telephone numbers, safety of VoIP calls and its associated problems.

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.

<p>SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.</p>

Neste trabalho, uma versão de RBM (Restricted Boltzmann Machine) tendo uma camada de classificação é adaptada a fim de permitir o seu uso com dados definidos num domínio contínuo. Essa adaptação dá origem a uma variante do modelo para o qual são desenvolvidas as regras de atualização de parâmetros dos treinamentos discriminativo, generativo e híbrido. A aplicação da variante como classificador no problema de VAD (Voice Activity Detection) é então investigada. Por meio de simulações envolvendo o corpus NOIZEUS e empregando como entradas do classificador tanto MFCCs (Mel-Frequency Cepstral Coefficients) quanto FBEs (Filter-Bank Energies), são obtidos resultados comparáveis aos de detectores considerados como estado da arte, com um menor custo computacional. A variante de RBM é comparada também com as SVMs (Support Vector Machines) lineares e com núcleo gaussiano. Com treinamento discriminativo, a RBM fornece desempenhos intermediários entre as duas versões de SVM, porém um custo computacional que é consideravelmente inferior aos de ambas. Adicionalmente, um conjunto de medidas do áudio que tiveram seu uso em VAD proposto recentemente são avaliadas com o emprego da RBM com treinamento discriminativo. Embora os resultados não sejam conclusivos, os desempenhos conseguidos indicam que essas medidas não são vantajosas quando comparadas com os tradicionais MFCCs. / In this work, a type of Restricted Boltzmann Machine (RBM) having a classification layer is adapted to allow its use with data defined in a continuous domain. Such adaptation gives rise to a variant of the model for which the parameter update rules are developed for the discriminative, generative and hybrid types of training. The application of the variant as a classifier to the Voice Activity Detection (VAD) problem is then investigated. By means of simulations involving the corpus NOIZEUS and employing Mel-Frequency Cepstral Coefficients (MFCCs) or Filter-Bank Energies (FBEs) as classifier inputs, results comparable to those of state-of-the-art detectors are achieved with a lower computational cost. The RBM variant is also compared to the linear and Gaussian kernel Support Vector Machines (SVMs). With the discriminative training, the RBM provides intermediate performances between the two SVM types, but a computational cost that is considerably lower than theirs. Additionally, a set of measures from the audio whose application in VAD has been recently proposed are evaluated by employing the RBM with discriminative training. Although the results are not conclusive, the performances obtained indicate that the measures are not advantageous when compared to the traditional MFCCs.

Novas demandas por serviços de valor agregado têm surgido constantemente em sistemas de telefonia celular. Somente o tráfego de voz não tem sido mais suficiente para suprir as necessidades dos usuários, que hoje clamam também por serviços de dados que sejam rápidos, eficientes, baratos, com mobilidade e que atendam às mais diversas aplicações, como correio eletrônico, vídeo-conferência ou acesso à Internet. Este trabalho apresenta um método de determinação das taxas mínimas necessárias nos canais de tráfego para atender a essas aplicações. Utilizando-se da tecnologia CDMA2000, será feita uma análise do impacto na interface aérea da alocação dos canais de tráfego determinados. A mesma análise será feita nos casos em que se determinem taxas mínimas e/ou máximas alocadas para cada usuário, comparando-se também tempo necessário para a transmissão dos dados, eficiência na célula (taxa útil / taxa total transmitida), etc. As conclusões deste trabalho podem indicar estratégias para se atender aos usuários de serviços de dados em redes de telefonia móvel sob diferentes aspectos, como melhor relação custo-benefício, nichos específicos (concentrar-se em serviços que demandem maiores ou menores taxas de transmissão, por exemplo) ou até mesmo prever a quantidade de recursos de rede necessária para atender a uma determinada base de usuários. / Mobile telephony systems have been constantly asked for new demands of value added services. Voice traffic only is not enough anymore to support the new needs of mobile users because nowadays they ask for fast, cost-efficient, cheap and mobile services to support electronic mail, videoconference or Internet access. This work presents a method of how to determine minimum rates on traffic channels to support these services. Considering CDMA2000 technology, an impact analysis will be made in the air interface for these specific traffic channels. Same analysis will be made when minimum or maximum rates are reserved for each user, also comparing the time needed for data transmission, efficiency of the cell (useful rate / maximum rate) and others. The conclusions of this work may allow strategies to support data users in mobile telephony networks in such different ways, as better cost-effective solution, specific segment of the market (i.e., to concentrate in services with higher or lower transmission rates) or even to foresee the amount of resources needed in a network to support a forecasted number of users.