This diploma thesis focuses on development of graphical user interface for managing IVR applications. The work is more software oriented, it analyzes current state of the iPBX product belonging to the IPEX a.s. company, describes used technologies and introduces new concept of interactive user interface for generating IVR diagrams together with Asterisk dial plan generator.

Development in VoIP technology is connected with the rapid growth of quality and data rate of the Internet connection. The application of the VoIP technology has become one of the key areas in telecommunication and networking. The main task of this thesis was to explore the questions of IP telephony and voice transmission over data networks according to H.323, SIP and Cisco architectures. Next focus was on designing and implementing experimental workplaces of mentioned architectures and their converged solutions. The purpose of the first chapter is to introduce the theory background of architectures required for implementation of real-time services in data networks and their comparison. The key objectives of the second and third part are to design and implement experimental workplaces containing signaling and proxy servers and hardware or software IP phones of several architectures. Illustration examples of different network topologies are included to demonstrate designed and realized VoIP solutions. Several tests of established communication are done using a packet analyzer. These tests examine duplex data transmission over IP. This article also contains two laboratory exercises designed according to gained practical experiences. The laboratory exercises provide an overview of theory and clearly indicate the possibilities of IP telephony.

The topic of this diploma thesis has been chosen for a discussion and implementation of quality of service on the application and link layer of the OSI model. At the beginning of this thesis the general parameters and basic technologies of the OSI model network layer has been explained. This part was chosen as a resource for explanation of the basic parameters in the quality of services. Basic part is followed by the introduction to the Ethernet technology, which became as a ground for the real model in this diploma thesis. As a part of this section has been written a block which contains a analysis of quality of services in the Ethernet, which means the implementation of IEEE 802.1Q/p. This analysis is followed by structured descrition of the controlling and functionality of quality of services parameters on the link layer with the system tool in the Windows XP, description of Win XP link layer drivers and its modification with system tool tcmon. The end of the theoretical part is represented by introduction to the implementation of quality of service in the wireless networks especially the 802.11e standard. At the beginning of the practical part is specified the description of the topology design which is dedicated to the quality of services implementation. This section is developed to well-founded analysis of applications which were used during topology creating. The last segment of this thesis is dedicated to evolving the practical informations which were obtained during the measurement.

This diploma thesis deals with developing multimedia applications on Java Micro Edition platform. The aim of this work is to design and implement the application which could establish a call between two users. The first part of the work describes J2ME platform, its two configurations and profiles. Next part is focused on Session Initiation Protocol and Real-time Transport Protocol. The application design consists of choosing the suitable virtual machine JVM, SIP and RTP libraries. The main part of this work describes application structure, graphic user interface and installation packages creating. It also shows a way of customizing the media stack - Java Media Framework, version Cross Platform.

The aim of this work is to study methods of handover used in WiMAX and WLAN networks, next suggest the posibility of simulation methods for WLAN handover in the simulation environment OPNET Modeler. This work is focused primarily on obtaining the information about neighboring AP and subsequent selection of a new AP using such information. The work describes in detail each processes solutions that are used to implement newscanning methods

Open source PBX Kamailio and OpenSIPS diploma thesis covers familiarization with appointed SIP exchanges and with their power comparing. A detailed installation instructions on the operating system Ubuntu is the aim of this work too. The work includes the historical development of telephone exchanges with a focus on the latest generation. The following is SIP protocol basic description and components that can be composed SIP exchanges. Another part is devoted to the development of exchanges Kamailio and OpenSIPS. The thesis contain the archutecture and configuration file description. The practical part of the thesis deals with high-capacity switches, and comparing it in terms of memory and computational demands. Selected measurements are compared with the Asterisk PBX.

The thesis deals with the analysis and concept of the functionality optimization of a company information system. All parts of the information system are analysed and system modifications, already implemented in some cases, are proposed afterwards. The thesis also aims to evaluate the proposed solution with regard to further possible extensions and portability.

This thesis project is on helping a Voice over Internet Protocol (VoIP) service provider by improving server side of Opticall AB's Dial over Data solution. Nowadays, VoIP is becoming more and more popular. People use VoIP to call their family and friends every day. It is cheap, especially when users are abroad, because that they do need to pay any roaming fee. Many companies also like their employees to use VoIP, not only because the cost of calling is cheap, but using VoIP means that the company does not need a hardware Private Branch eXchange (PBX) -- while potentially offering all of the same types of services that such a PBX would have offered. As a result the company can replace their hardware PBX with a powerful PC which has Private Branch eXchange PBX software to connect all the employees and their VoIP provider. At the VoIP provider’s side, the provider can provide cheap calls for all users which are connected by Internet. The users can initialize and tear down a session using a VoIP user agent, but how can they place a VoIP call from a mobile device or other devices without a VoIP user agent? Users want to place cheap VoIP call everywhere. VoIP providers want to provide flexible solution to attract and keep users. So they both want to the users to be able to place cheap VoIP call everywhere. Although VoIP user agent are available for many devices as a software running on a computer, a hardware VoIP phone, and even in some mobile devices. However, there are some practical problems with placing a VoIP call from everywhere. The first problem is that not every device can have a VoIP user agent. But if you do not have a VoIP user agent on your device, then it would seem to be difficult to place a VoIP call. The second problem is that you have to connect to a network (probably Internet) to signal that you want to place a call. Thus at a minimum your device has to support connecting to an appropriate network. If your device is connecting to a mobile network, you can send signaling to set up a VoIP call through General Packet Radio Service (GPRS). However, the bandwidth and delay of the GPRS networks of some mobile operators is not suitable for the transfer of encoded voice data, additionally, some mobile operators charge high fees for using GPRS. All of these problems make placing VoIP calls via a mobile device difficult. However, if your mobile device has a VoIP user agent and you have suitable connectivity, then you can easily use VoIP from your mobile device[.] To provide a flexible solution to VoIP everywhere -- even to devices that do not or can not have a VoIP user agent, Opticall AB has designed Dial over Data (DoD) solution. By using this solution, you can place a VoIP call from your mobile device or even fixed phone -- without requiring that the device that you use have a VoIP user agent. This solution also provides a central Internet Protocol-Private Branch eXchange (IP-PBX) which can connect call to and from to Session Initiation Protocol (SIP) phones. Both individuals and companies can use this solution for call cost savings. Max Weltz created the existing DoD solution in an earlier thesis project. This thesis [1] provides a good description of the existing DoD solution. As a result of continued testing and user feedback, Opticall AB has realized that their DoD solution needs to be improved in several area. This thesis project first identified some of the places where improvement was needed, explains why these improvements are necessary, and finally designs, implements, and evaluates these changes to confirm that they are improvements. An important result of this thesis project was a clear demonstration of improvements in configuration of the solution, better presentation of call data records, correct presentation of caller ID, and the ability to use a number of different graphical user interfaces with the improve DoD solution. These improvements should make this solution more attractive to the persons who have to maintain and operate the solution. / Detta examensarbete behandlar förbättringar i serversidan av OptiCall ABs “Dial over Data” (DoD) lösning som tillhandahålls för tjänsteleverantörer av VoIP. VoIP blir mer och mer populärt. Människor använder VoIP för att ringa till sin familj och vänner varje dag. Det är billigt, särskilt när användaren är utomlands, eftersom de inte behöver betala någon roamingavgift. Många företag vill också att deras anställda skall använda IP-telefoni, inte bara därför att kostnaden för att ringa oftast är lägre, utan för att bolaget kan ersätta sin traditionella företagsväxel (PBX) med en kraftfull dator som har PBX programvara för att även ansluta alla anställda till deras VoIP leverantör. VoIP leverantören kan erbjuda billiga samtal till alla användare som också är anslutna via Internet. Användarna kan hantera VoIP samtal med en VoIP user agent, men hur kan de ringa ett VoIP-samtal från en mobil enhet eller andra enheter utan VoIP user agent? Användare vill kunna ringa billiga VoIP-samtal överallt. VoIP-leverantörer vill erbjuda en flexibel lösning för att locka och behålla användare. Även VoIP user agent finns utvecklade till många enheter som en programvara som körs på en dator, en hårdvara VoIP-telefon, och även i vissa mobila enheter. Men det finns vissa praktiska problem med att ringa ett VoIP-samtal från alla platser. Det första problemet är att inte varje enhet kan ha en VoIP user agent. Det andra problemet är att den måste ansluta till ett nätverk (troligen Internet) för att signalera att den vill ringa ett samtal. Om din enhet ansluter till ett mobilnät, kan du skicka signalerar att upprätta ett VoIP-samtal via General Packet Radio Service (GPRS). Dock är bandbredden och fördröjningen i GPRS-nät i vissa operatörers nät inte lämpliga för överföring av tal, dessutom tar vissa mobiloperatörer ut höga avgifter för att använda GPRS. Alla dessa problem gör det svårt att hantera VoIP-samtal via en mobil enhet. Men om din mobila enhet har en VoIP user agent och du har lämplig nätanslutning så kan du enkelt använda VoIP från din mobiltelefon[.] För att erbjuda en flexibel VoIP lösning överallt - även på enheter som inte kan ha en VoIP user agent har OptiCall AB utformad “Dial over Data” (DoD). Genom att använda denna lösning kan du initiera ett VoIP-samtal från din mobiltelefon eller fast telefon - utan att kräva att den enhet som du använder har en VoIP user agent. Denna lösning inkluderar också en central Internet Protocol-Private Branch Exchange (IP-PBX) som kan koppla samtal till och från Session Initiation Protocol (SIP) telefoner. Både privatpersoner och företag kan använda denna lösning för att minska samtalskostnader. Max Weltz vidareutvecklade den befintliga DoD lösning i ett tidigare examensarbete. Denna avhandling [1] ger en god beskrivning av den befintliga DoD lösning. Som ett resultat av fortsatt testning samt synpunkter från användarna har OptiCall AB insett att deras DoD lösning måste förbättras på flera områden. Detta examensarbete har i första hand identifierat några områden där förbättringar behövdes, förklarat varför dessa förbättringar är nödvändiga, och slutligen utvecklat och utvärderat dessa förändringar. Ett viktigt resultat av detta examensarbete visades av en tydlig demonstration av förbättrad utformning av lösningen. Gränssnittet fick bla en bättre presentation av samtalshistorik, mer korrekt nummerpresentation. Dessa förbättringar bör göra denna lösning mer attraktivt för de personer som skall använda och underhålla lösningen.

In recent years dramatic technology developments have exploited the development of better transmission media and allowed for broad internet penetration. This in turn has fostered the growth of IP telephony calls, i.e., Voice over IP (VoIP). New VoIP products are introduced almost daily, each seeking an opportunity in the market. Some of these products are free - thus putting pressure on other vendors. A good example of a commercial VoIP product is Skype. It is possibly the most important one as it has gained more than 3 millions users in approximately 2 years time. In contrast, Minisip is a non-commercial implementation of SIP developed by students at KTH. These programs are based on different architectures. While Skype is said to be based on a peer-to-peer protocol, Minisip utilized the Session Initiation Protocol (SIP) protocol. The aim of this thesis was to evaluate these two VoIP programs not only in terms of development, but also in terms of the quality of service and user perceived voice quality. The study of efficiency, usability, and installation of both are also in the scope of this thesis. The devices used for the evaluation included a HP iPAQ 5550, two PCs running in RedHat Linux 9, and a laptop running Microsoft Windows XP. / På senare år har den dramatiska teknikutvecklingen exploaterat utvecklingen av bättre överföringsmedia samt möjliggjort för en bred Internetpenetrering. Det i sin tur har gynnat ökningen av telefonsamtal med IP-telefoner, d.v.s. Voice over IP (VoIP). Nya VoIP produkter introduceras nästan dagligen och varje produkt söker sin möjlighet på marknaden. Vissa av dessa produkter är gratis och sätter alltså press på andra försäljarna. Ett bra exempel på en VoIP produkt är Skype. Det är möjligtvis den viktigaste produkten då den har fått tre miljoner användare på ungefär två års tid. I kontrast till detta finns Minisip som är en icke kommersiell implementation av SIP, utvecklad av studenter från KTH. Dessa program är baserade på olika arkitekturer. Medan Skype är baserat på ett peer-to-peer protokoll, utnyttjar Minisip protokollet Session Initiation Protocol (SIP) Protocol. Syftet med denna avhandling var att evaluera dessa två VoIP program, inte bara i termer av utveckling utan också i termer av ”quality of service” och hur användaren uppfattar röstkvaliteten. Studien innefattar också effektivitet, användarvänlighet och installation av de båda programmen. Enheterna som användes under denna evaluering var en HP iPAQ 5550, två pc:s med Linux Red Hat 9 samt en bärbar pc med Windows XP.

High data rate wireless packet data networks have made real-time IP based services available through mobile devices. At the same time, differences in the characteristics of radio technologies (802.11/WiFi and 3G networks) make seamless handoff across heterogeneous wireless networks difficult. Despite this, many believe that the ultimate goal of next generation networks (often referred to as the fourth generation) is to allow convergence of such dissimilar heterogeneous networks. Supporting voice over Internet Protocol in next-generation wireless systems is thought by some to require support for mobility and quality of service features. Currently a mobile node can experience interruptions or even sporadic disconnections of an on going real-time session due to handovers between both networks of different types and networks of the same type. Many tests have already been done in this area and one may wonder why it is worth spending even more time investigating it? This thesis focuses on the important problem of providing session security despite handovers between networks (be they operated by the same operator or different operators and be they the same link technologies or different). One of the goals in this thesis is to investigate how an ongoing speech session can continue despite a change in transmission media1. Additionally, a number of security threats that could occur due to the handover will be identified and presented. Finally, the most suitable solution to address these threats will be tested in a real environment. Eventual shortcomings and weaknesses will be identified and presented; along with suggestions for future work.  1 When utilizing IP over carriers such as wired Ethernet, WLAN, and 3G. / Trådlösa hög-hastighets datanät har möjliggjort appliceringen av realtids tjänster på mobil utrustning över IP. Samtidigt har skillnaderna i de olika radioteknologierna (802.11/WiFi och 3G näten) introducerat nya problem med att upprätthålla trådlösa kommunikationen tvärs den heterogena trådlösa accessen. Många tror att slutmålet för nästa generations nätverk (ofta refererade som fjärde generationens nätverk) är att tillåta konvergensen av dessa olika heterogena nätverk. Stödet för Voice over Internet Protokollet (VoIP) i nästa generations trådlösa nät tror somliga kräver ett inslag av kombination mellan mobilitet samt upprätthållandet av kvaliteten. För närvarande kan den mobila noden (MN) råka ut för störningar och även sporadiska avbrott av en pågående realtidssessionen på grund av övergångar mellan samma eller olika typer av medier. Många tester har redan gjorts inom det här området och man kan fråga sig varför det är värt att lägga ner ännu mer tid på att undersöka det här? Det här examensarbetet fokuserar på det viktiga problemet som handlar om att kunna erbjuda sessions säkerhet trots övergångar mellan näten (oavsett om dessa drivs av samma eller olika operatörer samt oavsett om de är av samma eller olika nätverks typ). Ett av målen för det här examensarbetet är att undersöka hur en pågående talsession behålls vid byte av transmissionsmedia2. Vidare kommer olika säkerhetsaspekter och hot som kan tänkas uppstå vid bytet att identifieras och presenteras. Slutligen kommer den mest lämpade lösningen till problemet att testas i verklig miljö. Eventuella brister och svagheter kommer att identifieras och redovisas i slutet av rapporten tillsammans med förslag på framtida arbete. 2 Då man nyttjar IP bärare som trådbundet Ethernet, WLAN och 3G.