Adaptive optimization schemes for mobile VoIP applications Battery life and call quality aspects

In this thesis programmatic, application-layer means for better energy-efficiency in the VoIP application domain are studied. The work presented concentrates on optimizations which are suitable for VoIP-implementations utilizing SIP and IEEE 802.11 technologies. Energy-saving optimizations can have an impact on perceived call quality, and thus energy-saving means are studied together with those factors affecting perceived call quality. In this thesis a general view on a topic is given. Based on theory, adaptive optimization schemes for dynamic controlling of application's operation are proposed. A runtime quality model, capable of being integrated into optimization schemes, is developed for VoIP call quality estimation. Based on proposed optimization schemes, some power consumption measurements are done to find out achievable advantages. Measurement results show that a reduction in power consumption is possible to achieve with the help of adaptive optimization schemes.

An Upgrade of Network Traffic Recognition System for SIP/VoIP Traffic Recognition

The purpose of this project is to update the tool of Network Traffic Recognition System (NTRS) which is proprietary software of Ericsson AB and Tsinghua University, and to implement the updated tool to finish SIP/VoIP traffic recognition. Basing on the original NTRS, I analyze the traffic recognition principal of NTRS, and redesign the structure and module of the tool according to characteristics of SIP/VoIP traffic, and then finally I program to achieve the upgrade. After the final test with our SIP data trace files in the updated system, a satisfactory result is derived. The result presents that our updated system holds a rate of recognition on a confident level in the SIP session recognition as well as the VoIP call recognition. In the comparison with the software of Wireshark, our updated system has a result which is extremely close to Wireshark’s output, and the working time is much less than Wireshark. In the aspect of practicability, the memory overflow problem is avoided, and the updated system can output the specific information of SIP/VoIP traffic recognition, such as SIP type, SIP state, VoIP state, etc. The upgrade fulfills the demand of this project.

Älykäs viestintäjärjestelmä

TeliaSoneran SME-viestintäjärjestelmän tarkoituksena on toimia demo- ja kehitysalustana uusille viestintäpalveluille. Näitä palveluita ovat esimerkiksi tilatiedon seuraaminen ja muuttaminen sekä pikaviestien lähettäminen ja vastaanottaminen eri alustoilla, kuten matkapuhelimilla, PDA-laitteilla, PC:eillä. Järjestelmän tulee myös toimia helppona yhteysrajapintana mahdollisille tuleville tiedonsiirtoteille. Pikaviestien lähetys- ja puhelujensignalointiprotokollana järjestelmässä toimii SIP. Järjestelmän sisäisenä kommunikaatioprotokollana toimii CORBA. Työn osuus ja tämän dokumentin fokus ovat järjestelmäytimessä eli Core:ssa. Core:n avulla järjestelmän eri komponentit, kuten SIP-, WAP- ja WWW-viestinvälityspalvelimet sidotaan toisiinsa sekä tietokantaan. Ytimen tehtäviin kuuluu myös käyttäjien tilatietojen ylläpito, viestien reititys eri päätelaitteisiin, ryhmäviestintä sekä pikaviestikomentojen toiminnallisuus, eli sanalla sanoen älykkyys. Lisäksi työssä tarkastellaan ytimen käyttämiä rajapintoja ja niiden toteutustekniikoita, projektin toteutukseen käytettyjä työkaluja sekä järjestelmän tarjoamia palveluita yksityiskohtaisemmin ytimen kannalta. Lopuksi luodaan silmäys tulevaisuuden näkymiin ja järjestelmän nykytilaan sekä kokonaisuutena hyvin menneen projektin tavoitteisiin.

Puhelunohjaus Internetissä

Puhelunohjaus Internetissä ei sido käyttäjää tiettyyn paikkaan tai päätelaitteeseen. Käyttäjä voidaan paikantaa ja kommunikointi hänen kanssaan on mahdollista yhdistelemällä perinteisen puhelinverkon ja Internetin palveluita. Käyttäjä voi itse valita haluamansa palvelut aina tarpeensa mukaan. Tämä diplomityö esittelee Internet-puhelunohjauksen arkkitehtuurin ja tämän toteutuksen. Arkkitehtuurin perusosat kuvataan ja toteutuksen toimivuutta arvioidaan. Internet-puhelunohjaukseen liittyvät Internet-standardit esitellään ja niiden sopivuutta puhelunohjaukseen pohditaan. Aiheeseen liittyviä vaihtoehtoisia standardeja ja menetelmiä puhelunohjaukseen vertaillaan. Lisäksi työssä tutkitaan Necsom Oy:n Media Switch-reitittimen sopivuutta puhelunohjauksen laitealustaksi. Puhelunohjauksen arkkitehtuuri Internetissä mahdollistaa istuntojen muodostamisen ja ohjaamisen käyttäjäkohtaisen osoitteen perusteella. Muuta tietoa käyttäjästä ei tarvita istunnon aloitusvaiheessa. Työssä toteutettu puhelunohjausjärjestelmä on tehty Necsom Oy:n ja Lappeenrannan teknillisen korkeakoulun yhteishankkeena. Toteutuksesta saatujen tulosten perusteella Internet-puhelunohjaus on käyttökelpoinen tuotantokäytössä nyt ja tulevaisuudessa.

Analysis of EnterpriseVoIP Traffic from aWire line IMS System

With the rapid development of telecommunication industry, the IP multimedia Subsystem (IMS) could very well be the panacea for most telecom operators. It is originally defined as the core network for 3G mobile systems by the 3rd Generation Partnership Project (3GPP), the more recent development is merging between fixed line network and wireless networkd This report researchs the characteristic of the IMS data and proposes an IMS characterization analysis. We captured the IMS traffic data with 10 tousands users for about 41 hours. By analyzing the characteristics of the IMS, we know that the most important application in the IMS is VoIP call. Then we use the tool designed by Tsinghua University & Ericsson Company to recognize the data, and the results we got can be used to build the traffic models. From the results of the traffic models, I will get some reasons and conclusion. The traffic model gives out the types of session and types of VoIP call. I bring into a concept—busy hour. This concept is very important because it can help us to know which period is the peak of the VoIP call. The busy hour is from 10:00 to 11:00 in the morning. I also bring into another concept—connection ratio. This concept is significant because it can evaluate whether the VoIP call is good when it use IMS network. By comparing the traffic model with other one’s models, we found the different results from them, both the accuracy and the busy hour. From the contract, we got the advantages of our traffic models.

Statistical study of talk spurts and silence lengths with relation to various languages and speakers for advanced silence suppression in Next Generation Wireless Communication Devices

As wireless network technologies evolve towards an All-IP framework, Next Generation Wireless Communication Devices demand better use of spectral resources by employing advanced techniques of silence suppression. This paper presents an analysis of VoIP call data and compares the statistical results based on observed patterns of talk spurts and silence lengths to those achieved by a modified on-off voice model for silence suppression in wireless networks. As talk spurts and silence lengths are sensitive to varying word lengths, temporal structure and other prosodic aspects of speech, the impact of the use of various languages, dialects and gender of speakers on these results is also assessed.

Modelling data and voice traffic over IP networks using continuous-time Markov models

Common approaches to IP-traffic modelling have featured the use of stochastic models, based on the Markov property, which can be classified into black box and white box models based on the approach used for modelling traffic. White box models, are simple to understand, transparent and have a physical meaning attributed to each of the associated parameters. To exploit this key advantage, this thesis explores the use of simple classic continuous-time Markov models based on a white box approach, to model, not only the network traffic statistics but also the source behaviour with respect to the network and application. The thesis is divided into two parts: The first part focuses on the use of simple Markov and Semi-Markov traffic models, starting from the simplest two-state model moving upwards to n-state models with Poisson and non-Poisson statistics. The thesis then introduces the convenient to use, mathematically derived, Gaussian Markov models which are used to model the measured network IP traffic statistics. As one of the most significant contributions, the thesis establishes the significance of the second-order density statistics as it reveals that, in contrast to first-order density, they carry much more unique information on traffic sources and behaviour. The thesis then exploits the use of Gaussian Markov models to model these unique features and finally shows how the use of simple classic Markov models coupled with use of second-order density statistics provides an excellent tool for capturing maximum traffic detail, which in itself is the essence of good traffic modelling. The second part of the thesis, studies the ON-OFF characteristics of VoIP traffic with reference to accurate measurements of the ON and OFF periods, made from a large multi-lingual database of over 100 hours worth of VoIP call recordings. The impact of the language, prosodic structure and speech rate of the speaker on the statistics of the ON-OFF periods is analysed and relevant conclusions are presented. Finally, an ON-OFF VoIP source model with log-normal transitions is contributed as an ideal candidate to model VoIP traffic and the results of this model are compared with those of previously published work.

Language independent on-off voice over IP source model with lognormal transitions

The recent explosive growth of voice over IP (VoIP) solutions calls for accurate modelling of VoIP traffic. This study presents measurements of ON and OFF periods of VoIP activity from a significantly large database of VoIP call recordings consisting of native speakers speaking in some of the world's most widely spoken languages. The impact of the languages and the varying dynamics of caller interaction on the ON and OFF period statistics are assessed. It is observed that speaker interactions dominate over language dependence which makes monologue-based data unreliable for traffic modelling. The authors derive a semi-Markov model which accurately reproduces the statistics of composite dialogue measurements. © The Institution of Engineering and Technology 2013.

VoIP: Making Secure Calls and Maintaining High Call Quality

Modern multimedia communication tools must have high security, high availability and high quality of service (QoS). Any security implementation will directly impact on QoS. This paper will investigate how end-to-end security impacts on QoS in Voice over Internet Protocol (VoIP). The QoS is measured in terms of lost packet ratio, latency and jitter using different encryption algorithms, no security and just the use of IP firewalls in Local and Wide Area Networks (LAN and WAN). The results of laboratory tests indicate that the impact on the overall performance of VoIP depends upon the bandwidth availability and encryption algorithm used. The implementation of any encryption algorithm in low bandwidth environments degrades the voice quality due to increased loss packets and packet latency, but as bandwidth increases encrypted VoIP calls provided better service compared to an unsecured environment.

Implementación de balanced scorecard en el call center de Andinatel S.A.

La presente investigación tiene como objetivo aplicar los conocimientos adquiridos en el Programa de Maestría en Dirección de Empresas, tomando en consideración lo aprendido en las materias de Metodología de la Investigación, Procesos, Indicadores de Gestión y Estrategia Operacional a lo largo de dos años académicos. Con estos elementos la pregunta central es: ¿Es posible implementar un BSC en el Call Center de Andinatel S.A.? Es mi interés aplicar la herramienta Balanced Scorecard (BSC) en la Gerencia de Call Center de Andinatel S.A., debido a que al momento no existe en esta área un instrumento de medición de indicadores que posibilite tomar acciones de mejora; así también, al inicio de la carrera realicé investigaciones sobre como mejorar los procesos y cuales serían las alternativas de solución para satisfacer al cliente a través de este punto de contacto. Esto me ha permitido mejorar mi formación profesional presentado propuestas, mismas se han servido de aporte para lograr varios objetivos de mi Empresa en el área de estudio, la cual atiende al 70% de clientes, un promedio de 900.000 usuarios, distribuidos en 12 provincias de cobertura, maneja cerca de 6’000.000 millones de minutos al mes con la intervención de una plataforma tecnológica VoIP (voz sobre IP) y 300 agentes telefónicos que trabajan en turnos de 6 horas los 365 días del año. He solicitado a ANDINATEL S.A., lugar en donde laboro desde hace seis años, me permita realizar un estudio en una sus áreas funcionales, para lo cual utilizaré varios datos referenciales que servirán para este fin académico. La metodología Balanced Scorecard fue desarrollada por los académicos KAPLAN y NORTON de la Universidad de Harvard que consiste en organizar, difundir y controlar la ejecución de la estrategia de las organizaciones. Al concluir el presente trabajo haré uso de los resultados obtenidos en el Cuadro de Mando Integral del Call Center de Andinatel S.A. para lograr que el sistema de indicadores presentado posibilite crear un Proyecto de Mejora Continua y de esta manera aportar para la obtención de incremente de rentabilidad y satisfacción del cliente.

Uma ferramenta para projeto de sistemas de diálogos para call center baseados em asterisk

Este trabalho apresenta o aplicativo DialogBuilder, uma ferramenta de código aberto escrita em Java que disponibiliza ao usuário uma interface para projeto de sistemas de diálogos e exportação destes para implantação no software Asterisk, o mais popular framework VoIP. O DialogBuilder disponibiliza um wizard para que o usuário leigo possa projetar seu sistema sem precisar aprender a programar para Asterisk. O software separa a fase de concepção do diálogo de sua codificação e se posiciona para tornar técnica e economicamente viável, mesmo para pequenas empresas, construir e manter sistemas de diálogo para aplicações telefônicas.

On Advertisement Call Of The Poison Frog Ameerega Berohoka (dendrobatidae, Anura) From The Brazilian Cerrado.

The poison frog genus Ameerega (Dendrobatidae) currently contains 32 species. They are distributed from central Brazil into western Amazonia to the lower Andean versant. In addition, three trans-Andean species have been allocated to Ameerega (Andrade et al. 2013; Frost 2014). Ameerega berohoka (Vaz-Silva & Maciel 2011) was described based on specimens from central Brazil (type-locality: Arenópolis, GO) and it is assumed to occur in parts of western and southwestern state of Goiás (Frost 2014). More recently, Andrade et al. (2013) extended its distribution to the state of Mato Grosso. Here we re-describe the advertisement call of A. berohoka, providing additional information regarding its temporal structure and spectral traits. Our observations also consist of a new distribution record for this species to the state of Mato Grosso.

Geographic call variation and further notes on habitat of Ameerega flavopicta (Anura, Dendrobatidae)

We describe habitat and inter-populational call variation of the dendrobatid frog Ameerega flavopicta. Data were collected in the Brazilian states ofminas Gerais and Goias. Principal component analysis separated the Goias population from others because of its higher call rates and shorter calls. The Paranaiba Rivermay represent themajor geographic barrier. We recognize the cephalic amplexus as themain type for the species. Although habitat disturbances increased since 1990, we did not notice differences in the density of callingmales at Serra do Cipo. Ameerega flavopicta appears to be quite resistant to alterations in its natural habitats caused by human activities.