<p>Providing carrier grade characteristics for Internet Protocol (IP) communication applications is a significant problem for IP application providers in order to offer integrated services that span IP&nbsp / and telecommunication networks. This thesis addresses the provision of life-cycle management, which is only one carrier grade characteristic, for a SoftBridge application, which is an example of IP communication applications. A SoftBridge provides semi-synchronous multi-modal IP-based communication. The work related to IP-Telecommunication integrated services and the SoftBridge is analyzed with respect to life-cycle management in a literature review. It is suggested to use an Application Server in a Next Generation Network (NGN) to provide life-cyclemanagement functionality for IP-Telecommunication applications. In this thesis, the Application Server is represented by a JAIN Service Logic Execution Environment(JSLEE), in which&nbsp / a SoftBridge application can be deployed, activated, deactivated, uninstalled and upgraded online.Two methodologies are applied in this research: exploratory prototyping, which evolves the development of a SoftBridge application, and empirical comparison, which is concerned with the empirical evaluation of a SoftBridge application in terms of carriergrade capabilities. A SoftBridge application called SIMBA&nbsp / provides a Deaf Telephony service similar to aprevious Deaf Telephony SoftBridge, However, SIMBA&rsquo / s SoftBridge design and implementation are unique to this thesis. In order to test the life-cycle&nbsp / management ability of SIMBA, an empirical evaluation is carried out including the experiments oflife-cycle management and call-processing performance. The final experimental results of the evaluation show that a JSLEE is able to provide life-cycle management for SIMBA without causing a significant decrease in performance. In conclusion, the life-cycle management can be provided&nbsp / or a SoftBridge application by using an Application Server such as a JSLEE. Futhermore, the results indicate that&nbsp / approach of using Application Server (JSLEE) integration should be&nbsp / sufficiently general to provide life cycle management, and indeed other carrier grade capabilities, for other IP communication applications. This allows IP communication applications to be&nbsp / &nbsp / &nbsp / integrated into an NGN.</p>

Traditionally, real-time communication, such as voice calls, has run on separate, closed networks. Of all the limitations that these networks had, the ability of malicious attacks to cripple communication was not a crucial one. This situation has changed radically now that real-time communication and data have merged to share the same network. The objective of this project is to investigate the securing of softswitches with functionality similar to Private Branch Exchanges (PBX) from malicious attacks. The focus of the project will be a practical investigation of how to secure ILANGA, an ASTERISK-based system under development at Rhodes University. The practical investigation that focuses on ILANGA is based on performing six varied experiments on the different components of ILANGA. Before the six experiments are performed, basic preliminary security measures and the restrictions placed on the access to the database are discussed. The outcomes of these experiments are discussed and the precise reasons why these attacks were either successful or unsuccessful are given. Suggestions of a theoretical nature on how to defend against the successful attacks are also presented.

Magister Scientiae - MSc / Providing carrier grade characteristics for Internet Protocol (IP) communication applications is a significant problem for IP application providers in order to offer integrated services that span IP and telecommunication networks. This thesis addresses the provision of life-cycle management, which is only one carrier grade characteristic, for a SoftBridge application, which is an example of IP communication applications. A SoftBridge provides semi-synchronous multi-modal IP-based communication. The work related to IP-Telecommunication integrated services and the SoftBridge is analyzed with respect to life-cycle management in a literature review. It is suggested to use an Application Server in a Next Generation Network (NGN) to provide life-cyclemanagement functionality for IP-Telecommunication applications. In this thesis, the Application Server is represented by a JAIN Service Logic Execution Environment(JSLEE), in which a SoftBridge application can be deployed, activated, deactivated, uninstalled and upgraded online.Two methodologies are applied in this research: exploratory prototyping, which evolves the development of a SoftBridge application, and empirical comparison, which is concerned with the empirical evaluation of a SoftBridge application in terms of carriergrade capabilities. A SoftBridge application called SIMBA provides a Deaf Telephony service similar to aprevious Deaf Telephony SoftBridge, However, SIMBA’s SoftBridge design and implementation are unique to this thesis. In order to test the life-cycle management ability of SIMBA, an empirical evaluation is carried out including the experiments oflife-cycle management and call-processing performance. The final experimental results of the evaluation show that a JSLEE is able to provide life-cycle management for SIMBA without causing a significant decrease in performance. In conclusion, the life-cycle management can be provided or a SoftBridge application by using an Application Server such as a JSLEE. Futhermore, the results indicate that approach of using Application Server (JSLEE) integration should be sufficiently general to provide life cycle management, and indeed other carrier grade capabilities, for other IP communication applications. This allows IP communication applications to be integrated into an NGN.Providing carrier grade characteristics for Internet Protocol (IP) communication applications is a significant problem for IP application providers in order to offer integrated services that span IP and telecommunication networks. This thesis addresses the provision of life-cycle management, which is only one carrier grade characteristic, for a SoftBridge application, which is an example of IP communication applications. A SoftBridge provides semi-synchronous multi-modal IP-based communication. The work related to IP-Telecommunication integrated services and the SoftBridge is analyzed with respect to life-cycle management in a literature review. It is suggested to use an Application Server in a Next Generation Network (NGN) to provide life-cyclemanagement functionality for IP-Telecommunication applications. In this thesis, the Application Server is represented by a JAIN Service Logic Execution Environment(JSLEE), in which a SoftBridge application can be deployed, activated, deactivated, uninstalled and upgraded online.Two methodologies are applied in this research: exploratory prototyping, which evolves the development of a SoftBridge application, and empirical comparison, which is concerned with the empirical evaluation of a SoftBridge application in terms of carriergrade capabilities. A SoftBridge application called SIMBA provides a Deaf Telephony service similar to aprevious Deaf Telephony SoftBridge, However, SIMBA’s SoftBridge design and implementation are unique to this thesis. In order to test the life-cycle management ability of SIMBA, an empirical evaluation is carried out including the experiments oflife-cycle management and call-processing performance. The final experimental results of the evaluation show that a JSLEE is able to provide life-cycle management for SIMBA without causing a significant decrease in performance. In conclusion, the life-cycle management can be provided or a SoftBridge application by using an Application Server such as a JSLEE. Futhermore, the results indicate that approach of using Application Server (JSLEE) integration should be sufficiently general to provide life cycle management, and indeed other carrier grade capabilities, for other IP communication applications. This allows IP communication applications to be integrated into an NGN. / South Africa

Internet Protocol multimedia subsystem (IMS) is an architecture for services such as voice over Internet Protocol (VoIP) in IP based communication systems. IMS is standardized by the 3GPP standardization forum, and was first released in 2002. Since then IMS has not had the wide adoption by operators as first anticipated. As 3G already supported voice and video, the operators could not justify the expense of IMS. The current emergence of the fourth generation mobile communication system named Long Term Evolution (LTE) has, however, increased the need for knowledge of IMS and of creating services for it. LTE networks are IP only networks that provide low latency. In order to use LTE for making phone calls, VoIP technologies are needed. IMS is the architecture intended to be used for Voice over LTE (VoLTE). The need for tools for education within IMS was seen in 2006 by Enea Experts in Linköping, Sweden. The author of this thesis designed an IMS for educational purposes, but the project was never fully completed. This thesis will reexamine the design decisions previously made by the author. The requirements stated by the customer remain: that an IMS with basic signaling and logging should be easy to install, maintain, and evolve at a low cost. A literature study of IMS and VoLTE is presented to contribute with knowledge in these areas. The previous design and implementation made by the author is presented and analyzed. The third-party software that the previous implementation was based on is reexamined. Existing open source components are analyzed in order to identify how they can be used to solve the problem and to identify what remains to be developed in order to fulfill the requirements. New design suggestions, presented in today´s context, are proposed and verified using analytical reasoning and experiments. The outcome of the final work is new verified design decisions for the customer to use when implementing a new IMS for educational purposes. The thesis should also provide useful insights which instructors and students can use to teach and learn more about IMS. / Internet Protocol multimedia subsystem (IMS) är en arkitektur för tjänster, som IP-telefoni (Voice over Internet Protocol, VoIP), i IP baserade kommunikationssystem. IMS standardi¬seras av standardiseringsforumet 3GPP och första utgåvan släpptes år 2002. IMS fick dock inte det breda genomslag bland operatörer som förväntats. Eftersom 3G redan hade stöd för tal och video kunde operatörerna inte se skäl till ytterligare utgifter för IMS. Den fjärde generationens mobila kommunikationssystem, Long Term Evolution (LTE) är helt IP-baserat och ger lägre fördröjningar i nätet. För att kunna ringa telefonsamtal via LTE krävs VoIP-teknik. IMS är en arkitektur avsedd för att användas för Voice over LTE (VoLTE). Den nuvarande utvecklingen av LTE har därför ökat behovet av kunskap om IMS och av utveckling av IMS-tjänster. Enea Experts i Linköping insåg behovet av verktyg för utbildning inom IMS år 2006. Författaren av det här examensarbetet designade därför ett IMS för utbildningssyfte. Projektet slutfördes dock aldrig. Syftet med examensarbetet är att ompröva de tidigare designbesluten. Kundens krav kvarstår: att ett IMS med grundläggande signalering och loggning bör vara enkelt att installera, enkelt att underhålla och möjligt att utveckla till en låg kostnad. Arbetet innehåller en litteraturstudie av IMS och VoLTE för att ge en inblick i dessa områden. Den tidigare designen och implementationen presenteras och analyseras. Tredjeparts mjukvara, som den tidigare implementationen baserades på, omprövas. Befintliga programvaror med öppen källkod analyseras i syfte att kartlägga hur de kan användas för att lösa uppgiften, samt att identifiera vad som återstår att utveckla för att uppfylla kraven. Nya beslut kring design presenteras och besluten verifieras med experiment och analytiskt resonemang. Resultatet av detta examensarbete innefattar nya verifierade beslut kring design som kunden kan använda vid utveckling av ett nytt IMS för utbildningssyfte. Arbetet erbjuder också värdefulla insikter som instruktörer och elever kan använda för att undervisa samt för att lära sig mer om IMS.

Les services de conversation Web en temps réel permettent aux utilisateurs d'avoir des appels audio et vidéo et de transférer directement des données sur Internet. Les opérateurs OTT (OTT) tels que Google, Skype et WhatsApp proposent des services de communication économiques avec des fonctionnalités de conversation évoluées. Avec l'introduction de la norme de Web Real Time Communication (WebRTC), n'importe quelle page Web peut désormais offrir des services d'appel. WebRTC est utilisé comme technologie sous-jacente pour déployer de nouvelles plateformes de communication centrées sur le Web. Ces plates-formes visent à offrir de nouvelles méthodes modernes de contact et de communication sur le web. Contrairement aux réseaux de télécommunication traditionnels, les identités sur le Web sont basées sur des profils d'utilisateur et des informations d'identification auto-affirmés. Par conséquent, les opérateurs Web sont incapables d'assurer la fiabilité de leurs abonnés. Les services de communication Web restent exposés à des menaces dans lesquelles le contexte social entre les parties communicantes est manipulé. Un attaquant se définit comme une entité de confiance pour transmettre de fausses informations à l'utilisateur ciblé. Les menaces typiques contre le contexte social comprennent la fausse représentation d'identité, l’hameçonnage, le spam et la distribution illégale de contenu. Afin d'assurer la sécurité sur les services de communication Web, la confiance entre les parties communicantes doit être établie. La première étape consiste à permettre aux utilisateurs d'identifier leurs participants communicants afin de savoir avec qui ils parlent. Cependant, l'authentification seule ne peut garantir la fiabilité d'un appelant. De nouvelles méthodes d'estimation de la réputation de l'appelant devraient également être intégrées dans les services d'appel Web. Par conséquent, dans cette thèse, nous présentons un nouveau cadre de confiance qui fournit des informations sur la fiabilité des appelants dans les réseaux de communication Web. Notre approche est organisée en quatre parties. Premièrement, nous décrivons la notion de confiance dans la communication web en temps réel. Un modèle de confiance est présenté pour identifier les relations de confiance nécessaires entre les entités d'un système de communication. Les paramètres requis pour calculer la confiance dans les services de communication Web sont officiellement introduits. Deuxièmement, nous montrons comment les protocoles Single-Sign-On (SSO) peuvent être utilisés pour authentifier les utilisateurs d'une manière Peer-to-Peer (P2P) sans dépendre de leur fournisseur de service. Nous présentons une comparaison entre trois protocoles d'authentification appropriés (OAuth, BrowserID, OpenID Connect). La comparaison montre que OpenID Connect est le meilleur candidat en termes de confidentialité des utilisateurs. Troisièmement, un modèle de calcul de confiance est proposé pour mesurer la fiabilité des appelants dans un réseau de communication. La légitimité et l'authenticité d'un appelant sont calculées à l'aide de recommandations, tandis que la popularité d'un appelant est estimée en utilisant son comportement de communication. Un abonné d'un service de communication sera capable de visualiser la confiance calculée d'autres membres avant d'initier ou d'accepter une demande d'appel. Enfin, la réputation d'un appelant est utilisée pour lutter contre les appels nuisibles générés sur les réseaux de communication. Les appels de nuisance sont décrits comme des appels de spam non sollicités en masse générés sur un réseau de communication à des fins de marketing et de tromperie. Les enregistrements de données d'appel et les commentaires reçus par les parties communicantes sont utilisés pour déterminer la réputation de l'appelant. La réputation évaluée est utilisée pour différencier les spammeurs et les appelants légitimes du réseau / Real-time web conversational services allow users to have audio and video calls over the Internet. Over-The-Top operators such as Google and Facebook offer cost-effective communication services with advanced conversational features. With the introduction of WebRTC standard, any website or web application can now have built-in communication capabilities. WebRTC technology is expected to boost Voice-Over-IP by making it more robust, flexible and accessible. Telco operators also intend to use the underlying technology to offer communication services to their subscribers over the web. Emerging web-centric communication platforms aims to offer modern methods of contacting and communicating over the web. However, web operators are unable to ensure the trustworthiness of their subscribers, since identities are based on self-asserted user profiles and credentials. Thus, they remain exposed to many social threats in which the context between communicating parties is manipulated. An attacker usually misrepresents himself to convey false information to the targeted victim. Typical social threats include phishing, spam, fraudulent telemarketing and unlawful content distribution. To ensure user security over communication networks, trust between communicating parties needs to be established. Communicating participants should be able to verify each other’s identity to be sure of whom they are talking to. However, authentication alone cannot guarantee the trustworthiness of a caller. New methods of estimating caller’s reputation should also be built in web calling services. In this thesis, we present a novel trust framework that provides information about the trustworthiness of callers in web communication networks. Our approach is organized in four parts. Firstly, we describe the notion of trust in real-time web communication services. A trust model approach is presented to formally introduce the trust computation parameters and relationships in a communication system. Secondly, we detail the mechanism of identity provisioning that allows communicating participants to verify each other’s identity in a Peer-to-Peer fashion. The choice of authentication protocol highly impacts user privacy. We showed how OpenID Connect used for Single-Sign-On authentication purposes can be effectively used for provisioning identities while preserving user privacy. Thirdly, a trust computational model is proposed to measure the trustworthiness of callers in a communication network. The legitimacy and genuineness of a caller’s identity is computed using recommendations from members of the network. On the other hand, the popularity of a caller is estimated by analyzing its behavior in the network. Each subscriber will be able to visualize the computed trust of other members before initiating or accepting a call request. Lastly, the reputation of a caller is used to combat nuisance calls generated over communication networks. Nuisance calls are described as unsolicited bulk spam phone calls generated for marketing and deceptive purposes. Caller’s reputation is computed using the diversity of outgoing calls, call duration, recommendations from called participants, reciprocity and repetitive nature of calls. The reputation is used to differentiate between legitimate and nuisance calls generated over the network

Video conferencing has been a part of many communication platforms over the years. Over the last decades users have moved from dedicated telephony networks to the Internet, and recently to the Web. With the introduction of Web Real-Time Communication (WebRTC) it is now possible to make voice- and video calls simply by visiting a web page, without having to install any additional software. Services that enable multi-user conferences are quite common. However existing solutions such as the Multipoint Control Unit (MCU) inherently do not scale and can be a single point of failure, due to its centralized architecture. This can lead to high maintenance costs and poor service availability.To solve the scalability- and availability problems of video-conferencing services, a decentralized alternative to the MCU is proposed. A decentralized conferencing system uses the distributed resources of its users instead of relying on a central server. This means that the system can handle an increasing number of users without having to upgrade any server infrastructure. Additionally, failures are only partial and can happen regularly without affecting the rest of the system. This report presents the development of a protocol built on top of WebRTC that enables completely decentralized multi-user conferencing. It includes a distributed algorithm for voice-activated switching to reduce the computation and network resources used. A load-balancing technique based on media stream relays is used to distribute the resource requirements of the conference participants. The protocol is implemented as a Javascript library that can be included in a web application. A proof-of-concept web application is developed using the library and its performance is evaluated. The performance data is analyzed and the results are used to make incremental improvements to the protocol and implementation. Although not all features of the protocol are implemented, the tests show promising results. The application allows multiple users to participate in high-definition video conferences, with no server infrastructure aside from a Mini PC that hosts a web server and a WebRTC signaling server. / Videokonferenser har varit en del av många olika kommunikationsplattformar genom åren. Tekniken har yttats från dedikerade telefonnnät,, till Internet, och på senare tid till webben. I och med introduktionen av WebRTC (Web Real-Time Communication) är deti dag möjligt att enkelt deltaga i röst- och videosamtal genom att gå till en webbsida utan att behöva installera någon programvara annat an en webbläsare. De flesta existerande konferenstjänster är byggda med en centraliserad arkitektur, vilket kan leda till tekniska problem när antalet användare ökar eller när fel uppstår i systemens centrala servrar. Dessa problem kan leda till driftstopp och skada tjänstens tillgänglighet för användarna. Den här rapporten täcker utvecklingen av ett protokoll som tillsammans med WebRTC kan användas för att bygga en helt decentraliserad konferenstjänst. Målet är att tjänsten ska vara oberoende av centrala servrar, och på så vis lösa problemen med skalbarhet och tillgänglighet. Protokollet implementeras i en webbapplikation som testas och utvärderasöver flera iterationer för att hitta nya förbättringar. Testerna visar lovande resultat. Slutsatsen dras det är fullt möjligt att bygga en konferenstjänst på detta sätt, och möjligheter för framtida optimeringar och testfall föreslås.

Next generation wireless systems have an IP-based infrastructure with the support of heterogeneous access technologies. One research challenge for next generation all-IP based wireless systems is to design intelligent mobility management techniques that take advantage of IP-based technologies to achieve global roaming between various access networks. To support global roaming, next generation wireless systems require the integration and interoperation of heterogeneous mobility management techniques. Mobility in a hierarchical structure or multilayered environment should be supported. The objective of this study is to develop new mobility management techniques for global roaming support in next generation all-IP based wireless systems. More specifically, new schemes for location management and paging in Mobile IP for network layer mobility support, and new schemes for location management and handoff management in heterogeneous overlay networks for link layer mobility support are proposed and evaluated. For network layer mobility support, a distributed and dynamic regional location management mechanism for Mobile IP is proposed. Under the proposed scheme, the signaling burden is evenly distributed and the regional network boundary is dynamically adjusted according to the up-to-date mobility and traffic load for each terminal. Next, a user independent paging scheme based on last-known location and mobility rate information for Mobile IP is proposed. The proposed scheme takes the aggregated behavior of all mobile users as the basis for paging. For link layer mobility support, an IP-based system architecture for the integration of heterogeneous mobility management techniques is proposed. Three location management schemes under this IP-based architecture are proposed. All the three schemes support user preference call delivery which is a very important feature of next generation wireless communications. A threshold-based enhancement method is also proposed to further improve the system performance. Finally, a hybrid resource allocation scheme for handoff management in wireless overlay networks is proposed. Under this scheme, the overall system resources can be optimally allocated when mobile users are covered by multiple overlay networks.

<p>The subject of the thesis is to examine what communication tools the students of Kungsmadskolan in Växjö uses to communicate on the Internet and what are the motives that lie behind this choice and continued use. The thesis focus on five groups of communication tools: chat, Internet forum, community, Internet telephony and e-mail. The purpose of the study is to investigate and understand the motive that lies behind the choice of Internet-based communication tools done by high school students. We have chosen a qualitative method to conduct our research. We have collected empirical data through discussions with two focus groups divided into one with only boys and one with only girls from Kungsmadskolan in Växjö. We then analyze this data and draw conclusions from it. Prior to the focus group discussions, we created a theoretical framework. This framework has been used to steer and support the discussions and as a theoretical support in our analysis. The framework consists of five parts: communication, communication tools, motivation, choice and acceptance. Through this study we have been able to establish that the most important influence on the students choice of communication tools comes from their friends. This means that the strongest motivation to their choice comes from the search for social communion. Communication tools that the students use are: MSN, Flashback.com, Forumet.nu, Fuska.nu, Bilddagboken.se, Lunarstorm, Community.se, Helgon.nu, Facebook, Hotmail, school-mail, work-mail and Spraymail.</p> / <p>Ämnet för uppsatsen är att undersöka vilka kommunikationsverktyg gymnasieelever på Kungsmadskolan i Växjö använder sig av för att kommunicera på Internet, vilka motiv som ligger bakom valet av kommunikationsverktyg och det fortsatta användandet. Uppsatsen inriktar sig på fem olika typer av kommunikationsverktyg: chatt, Internetforum, community, Internettelefoni och e-post. Undersökningens syfte är föreståelseinriktat, vilket innebär att vi ämnar att undersöka och förstå vilka motiv som avgör gymnasieelevernas val. Vi har valt en kvalitativ metod för vår undersökning. Vi har genom diskussion med två stycken fokusgrupper indelade i en med bara killar och en med bara tjejer från Kungsmadskolan i Växjö samlat in empirisk data. Denna data har vi sedan analyserat och dragit slutsatser utifrån. Inför våra diskussioner med fokusgrupperna har vi skapt en teoretisk referensram. Denna referensram har fungerat både som underlag för att styra och stödja diskussionerna och som teoretiskt stöd vid analysen av resultatet. Referensramen är uppdelad i fem avsnitt: kommunikation, kommunikationsverktyg, motivation, val och acceptans. Vi kom fram till att vänner i den direkta omgivningen har störst påverkan på vilken/vilka kommunikationsverktyg gymnasieeleverna använder sig av och väljer att börja använda. Detta innebär också att den starkaste motivationsfaktorn i valet är sökandet efter social gemenskap. Kommunikationsverktygen gymnasieeleverna använder är: MSN, Flashback.info, Forumet.nu, Fuska.nu, Bilddagboken.se, Lunarstorm, Community.se, Helgon.nu, Facebook, Hotmail, skolmail, jobbmail och Spraymail.</p>

The subject of the thesis is to examine what communication tools the students of Kungsmadskolan in Växjö uses to communicate on the Internet and what are the motives that lie behind this choice and continued use. The thesis focus on five groups of communication tools: chat, Internet forum, community, Internet telephony and e-mail. The purpose of the study is to investigate and understand the motive that lies behind the choice of Internet-based communication tools done by high school students. We have chosen a qualitative method to conduct our research. We have collected empirical data through discussions with two focus groups divided into one with only boys and one with only girls from Kungsmadskolan in Växjö. We then analyze this data and draw conclusions from it. Prior to the focus group discussions, we created a theoretical framework. This framework has been used to steer and support the discussions and as a theoretical support in our analysis. The framework consists of five parts: communication, communication tools, motivation, choice and acceptance. Through this study we have been able to establish that the most important influence on the students choice of communication tools comes from their friends. This means that the strongest motivation to their choice comes from the search for social communion. Communication tools that the students use are: MSN, Flashback.com, Forumet.nu, Fuska.nu, Bilddagboken.se, Lunarstorm, Community.se, Helgon.nu, Facebook, Hotmail, school-mail, work-mail and Spraymail. / Ämnet för uppsatsen är att undersöka vilka kommunikationsverktyg gymnasieelever på Kungsmadskolan i Växjö använder sig av för att kommunicera på Internet, vilka motiv som ligger bakom valet av kommunikationsverktyg och det fortsatta användandet. Uppsatsen inriktar sig på fem olika typer av kommunikationsverktyg: chatt, Internetforum, community, Internettelefoni och e-post. Undersökningens syfte är föreståelseinriktat, vilket innebär att vi ämnar att undersöka och förstå vilka motiv som avgör gymnasieelevernas val. Vi har valt en kvalitativ metod för vår undersökning. Vi har genom diskussion med två stycken fokusgrupper indelade i en med bara killar och en med bara tjejer från Kungsmadskolan i Växjö samlat in empirisk data. Denna data har vi sedan analyserat och dragit slutsatser utifrån. Inför våra diskussioner med fokusgrupperna har vi skapt en teoretisk referensram. Denna referensram har fungerat både som underlag för att styra och stödja diskussionerna och som teoretiskt stöd vid analysen av resultatet. Referensramen är uppdelad i fem avsnitt: kommunikation, kommunikationsverktyg, motivation, val och acceptans. Vi kom fram till att vänner i den direkta omgivningen har störst påverkan på vilken/vilka kommunikationsverktyg gymnasieeleverna använder sig av och väljer att börja använda. Detta innebär också att den starkaste motivationsfaktorn i valet är sökandet efter social gemenskap. Kommunikationsverktygen gymnasieeleverna använder är: MSN, Flashback.info, Forumet.nu, Fuska.nu, Bilddagboken.se, Lunarstorm, Community.se, Helgon.nu, Facebook, Hotmail, skolmail, jobbmail och Spraymail.

Indiana University-Purdue University Indianapolis (IUPUI) / In today's internet world with such a high traffic, it becomes inevitable to have multiple servers representing a single logical server to share enormous load. A very common network configuration consists of multiple servers behind a load balancer. The load balancer determines which server would service a clients request or incoming load from the client. Such a hardware is expensive, runs a fixed policy or algorithm and is a single point of failure. In this paper, we will implement and analyze an alternative load balancing architecture using OpenFlow. This architecture acquires flexibility in policy, costs less and has the potential to be more robust. This paper also discusses potential usage of OpenFlow based load balancing for media gateway selection in SIP-PSTN networks to improve VoIP performance.