IP Telephony in Third Generation Mobile Systems: Transport Layer Dimensioning

Tämä diplomityö käsittelee kolmannen sukupolven matkaviestinjärjestelmien kuljetuskerroksen mitoitusta. Nykyisten matkapuhelinverkkojen korvaajiksi suunnitellut kolmannen sukupolven matkaviestinjärjestelmät tulevat yhdistämään perinteisen puhelinviestinnän ja uudenlaiset datapalvelut. Uudet verkot tulevat perustumaan pakettivälitteiseen tiedonsiirtoon joka mahdollistaa molempien liikennetyyppien, puheen sekä datan, siirtämisen samassa verkossa. Tämän ratkaisun uskotaan tarjoavan paremmat mahdollisuudet uusien palvelujen luomiseen ja parantavan tiedonsiirtokapasiteettia. Siirtyminen pakettivälitteiseen tiedonsiirtoon aiheuttaa kuitenkin suuria muutoksia verkkoarkkitehtuurissa. Tässä diplomityössä tarkastellaan tulevien runkoverkkojen mitoitukseen liittyviä näkökohtia sekä muodostetaan alustavia kuljetuskerroksen mitoitusohjeita. Diplomityö on tehty osaksi diplomi-insinöörin tutkintoa Lappeenrannan teknillisessä korkeakoulussa. Työ on tehty Nokia Networksin palveluksessa Helsingissä, vuoden 2000 toisella puoliskolla.

Service Architecture in Mobile IP Telephony Network

Uudet palvelut ovat tarkeinta, mita asiakkaat odottavat uudelta teknologialta.Se on paaasiallinen syy siihen, etta asiakkaat ovat valmiita maksamaan uudesta teknologiasta ja kayttamaan sita. Sen vuoksi uuden verkon tuoma uusi palveluarkkitehtuuri on tarkea koko projektin onnistumiselle. Tama dokumentti keskittyy kolmannen sukupolven matkapuhelinverkkojen palveluarkkitehtuuriin, jonka viitemallista annetaan kuvaus. Verkon palvelut esitellaan ja kuvaillaan. Toteutukseen liittyvia asioita selostetaan. USA:n markkinoilla tarvittava WIN konsepti kuvataan ja sen toteutuksesta annetaan myos kuvaus. Lopussa kuvataan Pre-Paid tilaajien laskutustietojen kasittelya WIN konseptissa elvytystilanteessa.

Investigation of interface between Multimedia IP Network and Call Processing Server in 3G IP Telephony Core Network

The thesis presents an overview of third generation of IP telephony. The architecture of 3G IP Telephony and its components are described. The main goal of the thesis is to investigate the interface between the Call Processing Server and Multimedia IP Networks. The interface functionality, proposed protocol stack and a general description are presented in the thesis. To provide useful services, 3G IP Telephony requires a set of control protocols for connection establishment, capabilities exchange and conference control. The Session Initiation Protocol (SIP) and the H.323 are two protocols that meet these needs. In the thesis these two protocols are investigated and compared in terms of Complexity, Extensibility, Scalability, Services, Resource Utilization and Management.

Puhe IP-verkossa operaattorin näkökulmasta

Tässä diplomityössä tutkittiin operaattorin IP-verkossa toteutettavia puhepalveluita. Tutkimus perustui käytännön tarpeeseen. VoIP-tekniikalla toteutetuista puheluista on nopeasti tullut vakavasti otettava haastaja perinteiselle piirikytkentäiselle puhelintekniikalle. IP-tekniikka mahdollistaa data-, puhe- sekä videopalvelujen integroimisen yhteen verkkoon. Lisäksi IP-verkko on edullinen, laajalle levinnyt ja tehokas. Nämä ominaisuudet tekevät siitä houkuttelevan vaihtoehdon puhepalvelujen alustaksi. Verkkojen yhdistyminen mahdollistaa uudentyyppisen kommunikointiympäristön, jossa voidaan käyttää monenlaisia sovelluksia ja apuvälineitä ihmisten välisen kommunikoinnin helpottamiseksi. Tähän työhön sisältyi testilaitteiston hankkiminen ja asentaminen. Laitteistolla oli pystyttävä toteuttamaan operaattorin VoIP-järjestelmä, jolla oli kyettävä toteuttamaan usealle yritykselle IP-vaihdepalvelut. Laitteistoa testattiin itse aiheutetuilla virhetilanteilla sekä koekäyttäjillä. Testauksessa selvitettiin järjestelmän soveltuvuutta operaattorin tuotantokäyttöön.

Contribution to Asterisk Open Source Project

With this final master thesis we are going to contribute to the Asterisk open source project. Asterisk is an open source project that started with the main objective of develop an IP telephony platform, completely based on Software (so not hardware dependent) and under an open license like GPL. This project was started on 1999 by the software engineer Mark Spencer at Digium. The main motivation of that open source project was that the telecommunications sector is lack of open solutions, and most of the available solutions are based on proprietary standards, which are close and not compatible between them. Behind the Asterisk project there is a company, Digum, which is the project leading since the project was originated in its laboratories. This company has some of its employees fully dedicated to contribute to the Asterisk project, and also provide the whole infrastructure required by the open source project. But the business of Digium isn't based on licensing of products due to the open source nature of Asterisk, but it's based on offering services around Asteriskand designing and selling some hardware components to be used with Asterisk. The Asterisk project has grown up a lot since its birth, offering in its latest versions advanced functionalities for managing calls and compatibility with some hardware that previously was exclusive of proprietary solutions. Due to that, Asterisk is becoming a serious alternative to all these proprietaries solutions because it has reached a level of maturity that makes it very stable. In addition, as it is open source, it can be fully customized to a givenrequirement, which could be impossible with the proprietaries solutions. Due to the bigness that is reaching the project, every day there are more companies which develop value added software for telephony platforms, that are seriously evaluating the option of make their software fully compatible withAsterisk platforms. All these factors make Asterisk being a consolidated project but in constant evolution, trying to offer all those functionalities offered by proprietaries solutions. This final master thesis will be divided mainly in two blocks totally complementaries. In the first block we will analyze Asterisk as an open source project and Asterisk as a telephony platform (PBX). As a result of this analysis we will generate a document, written in English because it is Asterisk project's official language, which could be used by future contributors as an starting point on joining Asterisk. On the second block we will proceed with a development contribution to the Asterisk project. We will have several options in the form that we do the contribution, such as solving bugs, developing new functionalities or start an Asterisk satellite project. The type of contribution will depend on the needs of the project on that moment.

Simulation and analysis of network traffic for efficient and reliable information transfer

With the growing commercial importance of the Internet and the development of new real-time, connection-oriented services like IP-telephony and electronic commerce resilience is becoming a key issue in the design of TP-based networks. Two emerging technologies, which can accomplish the task of efficient information transfer, are Multiprotocol Label Switching (MPLS) and Differentiated Services. A main benefit of MPLS is the ability to introduce traffic-engineering concepts due to its connection-oriented characteristic. With MPLS it is possible to assign different paths for packets through the network. Differentiated services divides traffic into different classes and treat them differently, especially when there is a shortage of network resources. In this thesis, a framework was proposed to integrate the above two technologies and its performance in providing load balancing and improving QoS was evaluated. Simulation and analysis of this framework demonstrated that the combination of MPLS and Differentiated services is a powerful tool for QoS provisioning in IP networks.

Implementing telephony signalling over IP networks using SIGTRAN standards

Diplomityössä tutkitaan yhteiskanavamerkinantoverkon signalointiliikenteen siirtoa IP-pohjaisiin verkkoihin käyttäen IETF:n aliorganisaation, SIGTRAN:in, määrittelemiä standardeja. Tutkimuksen lisäksi työssä suoritetaan myös toteutus, joka mahdollistaa em. verkkojen yhdistymisen. Ensin työssä käsitellään yhteiskanavamerkinantoverkon tulevaisuus ja alan yleiset näkymät, jotka luovat pohjan työn toteutuksen ymmärtämiseen. Toiseksi työssä esitellään toteutuksessa käytetyt teknologiat. Esiteltäviin teknologioihin kuuluvat yhteiskanavamerkinantoverkko, IP-pohjainen verkko, SIGTRAN:in määrittelemät standardit, toteutustyökalut sekä ympäristö, johon toteutus liitetään. Toteutukseen olennaisesti liittyvät asiat ovat painotettuina teknologioiden esittelyssä. Diplomityön loppuosassa kuvataan merkinantoyhdyskäytävän toteutus suunnittelun ja toteutettujen toiminnallisuuksien osalta. Diplomityön tuloksena saatiin uusi testattu merkinantosovellus, joka täyttää yhteiskanavamerkinantoverkolle asetetut vaatimukset. Toteutus toimii osana Intellitel(TM) ONE palvelualustaa. Toteutuksen kehitys tulee jatkumaan.

Disseny i implementació d’un sistema experimental de telefonia IP per a la Unitat de Tecnologies de la Informació i Comunicació de l’Institut Català de la Salut a Girona

L'objectiu d'aquest projecte és dissenyar i implementar un servei de telefonia IP per a la Unitat de Tecnologies de la Informació i Comunicació (UTIC) de l'Institut Català de la Salut (ICS) a Girona. En primer lloc, es descriu el funcionament actual del servei de telefonia de l'ICS a Girona i la seva infraestructura de xarxa. A continuació s’estudia els fonaments de la VoIP. Es segueix amb el disseny, les eines escollides i la implementació realitzada, per acabar amb les conclusions

Implantació de la telefonia IP en un entorn empresarial

L’objectiu d’aquest treball és dissenyar un model general d’un sistema de telefonia IP per a una petita o mitjana empresa. El model ha de tenir en compteles característiques actuals de la xarxa de l’empresa i proposar una solució adient. Un altre requeriment és la utilització de software de lliure distribució, des del sistema operatiu fins al relatiu a VoIP, i més concretament, el software de centraleta VoIP Asterisk sobre GNU/Linux.En primer lloc s’estudiaran els conceptes bàsics de la telefonia IP (protocols,codificadors, servidors, etc.). En segon lloc, s’analitzaran els diferents escenarispossibles i es proposaran solucions adequades per cadascun d’ells. Després s’estudiarà el funcionament de les centraletes Asterisk i la seva configuració encada escenari. Finalment s’aplicarà aquest estudi a una empresa concreta

Implantació de la telefonia IP en un entorn empresarial

L’objectiu d’aquest treball és dissenyar un model general d’un sistema de telefonia IP per a una petita o mitjana empresa. El model ha de tenir en compteles característiques actuals de la xarxa de l’empresa i proposar una solució adient. Un altre requeriment és la utilització de software de lliure distribució, des del sistema operatiu fins al relatiu a VoIP, i més concretament, el software de centraleta VoIP Asterisk sobre GNU/Linux. En primer lloc s’estudiaran els conceptes bàsics de la telefonia IP (protocols, codificadors, servidors, etc.). En segon lloc, s’analitzaran els diferents escenaris possibles i es proposaran solucions adequades per cadascun d’ells. Després s’estudiarà el funcionament de les centraletes Asterisk i la seva configuració encada escenari. Finalment s’aplicarà aquest estudi a una empresa concreta

Disseny i implementació d’un sistema experimental de telefonia IP per a la Unitat de Tecnologies de la Informació i Comunicació de l’Institut Català de la Salut a Girona

L'objectiu d'aquest projecte és dissenyar i implementar un servei de telefonia IP per a la Unitat de Tecnologies de la Informació i Comunicació (UTIC) de l'Institut Català de la Salut (ICS) a Girona. En primer lloc, es descriu el funcionament actual del servei de telefonia de l'ICS a Girona i la seva infraestructura de xarxa. A continuació s’estudia els fonaments de la VoIP. Es segueix amb el disseny, les eines escollides i la implementació realitzada, per acabar amb les conclusions

Voice over IP 2.0: an analysis of limits and potential of IP2IP telecommunication

Internet Telephony (VoIP) is changing the telecommunication industry. Oftentimes free, VoIP is becoming more and more popular amongst users. Large software companies have entered the market and heavily invest into it. In 2011, for instance, Microsoft bought Skype for 8.5bn USD. This trend increasingly impacts the incumbent telecommunication operators. They see their main source of revenue – classic telephony – under siege and disappear. The thesis at hand develops a most-likely scenario in order to determine how VoIP is evolving further and it predicts, based on a ten-year forecast, the impact it will have on the players in the telecommunication industry.The paper presents a model combining Rogers’ diffusion and Christensen’s innovation research. The model has the goal of explaining the past evolution of VoIP and to isolate the factors that determine the further diffusion of the innovation. Interviews with industry experts serve to assess how the identified factors are evolving.Two propositions are offered. First, VoIP operators are becoming more important in international, corporate, and mobile telephony. End-to-end VoIP (IP2IP) will exhibit strong growth rates and increasingly cannibalize the telephony revenues of the classic operators. Second, fix-net telephony in SMEs and at home will continue to be dominated by the incumbents. Yet, as prices for telephony fall towards zero also they will implement IP2IP in order to save costs. By 2022, up to 90% of the calls will be IP2IP. The author recommends the incumbents and VoIP operators to proactively face the change, to rethink their business strategies, and to even be open for cooperation.

Implementation of a Control Loop Experiment in a Network-Based Control System With LonWorks Technology and IP Networks

Considering the increasing popularity of network-based control systems and the huge adoption of IP networks (such as the Internet), this paper studies the influence of network quality of service (QoS) parameters over quality of control parameters. An example of a control loop is implemented using two LonWorks networks (CEA-709.1) interconnected by an emulated IP network, in which important QoS parameters such as delay and delay jitter can be completely controlled. Mathematical definitions are provided according to the literature, and the results of the network-based control loop experiment are presented and discussed.

Randomized comparison of the quality of realtime fetal ultrasound images transmitted by ISDN and by IP videoconferencing

We compared the quality of realtime fetal ultrasound images transmitted using ISDN and IP networks. Four experienced obstetric ultrasound specialists viewed standard recordings in a randomized trial and rated the appearance of 30 fetal anatomical landmarks, each on a seven-point scale. A total of 12 evaluations were performed for various combinations of bandwidths (128, 384 or 768 kbit/s) and networks (ISDN or IF). The intraobserver coefficient of variation was 2.9%, 5.0%, 12.7% and 14.7% for the four observers. The mean overall ratings by each of the four observers were 4.6, 4.8, 5.0 and 5.3, respectively (a rating of 4 indicated satisfactory visualization and 7 indicated as good as the original recording). Analysis of variance showed that there were no significant interobserver variations nor significant differences in the mean scores for the different types of videoconferencing machines used. The most significant variable affecting the mean score was the bandwidth used. For ISDN, the mean score was 3.7 at 128 kbit/s, which was significantly worse than the mean score of 4.9 at 384 kbit/s, which was in turn significantly worse than the mean score of 5.9 at 768 kbit/s. The mean score for transmission using IP was about 0.5 points lower than that using ISDN across all the different bandwidths, but the differences were not significant. It appears that IP transmission in a private (non-shared) network is an acceptable alternative to ISDN for fetal tele-ultrasound and one deserving further study.