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

Orientador: Shusaburo Motoyama / Tese (doutorado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação / Made available in DSpace on 2018-08-17T02:39:28Z (GMT). No. of bitstreams: 1 Marques_LeandroBentoSena_D.pdf: 3039404 bytes, checksum: 84c76813ec2349c45ae1c49f833c31d4 (MD5) Previous issue date: 2010 / Resumo: Este trabalho apresenta um estudo de desempenho dos enlaces diretos dos sistemas CDMA 1xEVDO RA, UMTS/HSDPA e WiMAX com ênfase em escalonadores de dados e nos novos critérios de aceitação de tráfego handoff horizontal e vertical para redes 3G e 4G. Estes novos critérios de aceitação de tráfego handoff horizontal e vertical levam em conta a ocupação do enlace, a ocupação do buffer, a potência do sinal recebido (RSS) e o tamanho do quantum (DRR) como parâmetros para a decisão do processo de handoff. Além disso, o estudo considera os escalonadores de dados Max C/I (Maximum Carrier Interference), DRR (Deficit Round Robin), PF (Proportional Fair), Pr (Prioritário) e a nova proposta Pr/PF (Priority Proportional Fair). Os critérios combinados aos escalonadores são avaliados pormeio de métricas de QoS em função da chegada de tráfego HTTP interno ou em handoff. Os resultados mostraram que conforme o critério e o escalonador adotados, podem assegurar a QoS dos sistemas móveis e ainda aceitar uma boa quantidade de tráfego handoff. O estudo é baseado em simulações computacionais através da ferramenta de software Matlab / Abstract: This work presents a performance study of the forward links of CDMA 1xEV-DO RA, UMTS/HSDPA and WiMAX systems with emphasis on data schedulers and new criteria for horizontal and vertical handoff traffic acceptance in the 3G and 4G networks. These new criteria for horizontal and vertical handoff traffic acceptance take into account the link occupation, the buffer occupation, the received signal strength (RSS) and the size of quantum (DRR) as inputs for decision of handoff process. Moreover, the study considers the data schedulers Max C/I (Maximum Carrier Interference), DRR (Deficit Round Robin), PF (Proportional Fair), Pr (Priority) and the new proposal Pr/PF (Priority Proportional Fair). The criteria combined with the data schedulers are evaluated using QoS metrics in function of internal HTTP traffic or handoff traffic. The results showed that depending on the chosen criterion and scheduler, it is possible to assure the QoS of mobile systems and still accept a good amount of handoff traffic. The study is based on computer simulations through Matlab software tool / Doutorado / Telecomunicações e Telemática / Doutor em Engenharia Elétrica

Under de senaste åren har Internettelefonin varit på frammarsch, och i takt med att tekniken mognat har fler och fler börjat se den som ett alternativ till att ringa via telefonnätet. Förutom att det är billigare att ringa över det förstnämnda, så erbjuder Internettelefonin också en rad revolutionerande tjänster. Det är dock troligt att telefonnätet kommer att få tjänstgöra i många år till, och det erbjuder fortfarande överlägset bäst stabilitet och har stor acceptans. Om de två telefoninätverken ska existera sida vid sida, med varsina användarbaser är det lämpligt om de kan fås att samverka, så att användare av det ena kan ringa användare av det andra, och vice versa. Detta kan göras med en VoIP/PSTN-gateway, som översätter kontrollinformation och rösttrafik mellan de två nätverken. Uppsatsen handlar om det arbete vi har utfört år TietoEnator i Karlstad. Uppgiften bestod i att utveckla en prototyp av en VoIP/PSTN-gateway. Från början var det avsett att systemet skulle klara uppringning från endera en ”vanlig” telefon, eller en så kallad IP-telefon. Därtill skulle rösttrafiken överföras genom ändamålsenlig hårdvara. För att utföra arbetet behövde vi först studera relevanta kommunikationsprotokoll både för telefonnätet och för Internet, för att se hur dessa kunde fås att samverka. Vi behövde också lära oss tillgängliga system, bibliotek och verktyg för att förstå hur vi skulle skapa vårt eget system i den efterkommande implementeringsfasen. På grund av en lång inläsningsperiod och inledande tekniska problem, samt att nödvändig hårdvara för översättning av rösttrafiken inte anlände i tid begränsades arbetet till att innefatta samtal initierade från den vanliga telefonen till ip-telefonen, utan röstöverföring. Likväl har ett resultatgivande arbete utförts, och det beskrivs i detalj i rapporten. / During the past few years Internet telephony has advanced rapidly, and as the technology has evolved, more and more have come to consider it an alternative to making phone calls through the telephone network. Besides being cheaper, Internet telephony also provides several revolutionary services. It is likely though that the telephone network will remain in use for several years to come, and it still offers by far the best stability and is accepted by most people. If the two networks are to coexist, with their respective users, it would be useful if they could be made to interact, so that users of one network can call users of the other, and vice versa. This can be done with a VoIP/PSTN gateway, which translates control information and voice traffic between the two networks. Our dissertation is about the work we have performed for TietoEnator in Karlstad. The assignment was to develop a prototype of a VoIP/PSTN gateway. Initially the system was meant to support phone calls initiated either from an “ordinary” phone or from an IP telephone. Also the voice traffic was supposed to be translated with the use of appropriate hardware. To manage this we first needed to study all the relevant protocols for communication used in the telephone network and on the Internet, to get an idea of how these could be made to interact. We also had to learn existing systems, libraries and tools in order to see how we could create our own system. Due to a long learning period and technical problems in the beginning, and because the necessary hardware equipment for translation of voice traffic did not arrive in time, the assignment was limited to include only calls initiated from the ordinary phone to the IP telephone, without voice transmission. Never the less, the efforts have produced results, and our work is explained in detail in this dissertation.

We estimate own-price elasticities for fixed network voice telephony access and (national) calls services for private users and cross-price elasticities to mobile using time series data from 2002-2007 from the Austrian market. Using instrumental variable estimates and taking into account the possibility of cointegration we find that access is inelastic while calls are elastic. We conclude that the retail market for national calls of private users can probably be deregulated due to sufficient competitive pressure from mobile. Access-substitution on the other hand does not seem to be strong enough to justify de-regulation. / Series: Working Papers / Research Institute for Regulatory Economics

This thesis focuses on enterprise VoIP telephony network security. Introduces brief comparison of old analog and digital voice networks and IP telephone networks with special focus on VoIP system security. The goal of the thesis is to identify the risks of implementation and operation of VoIP technologies in enterprise environment and so thesis brings some conclusion how to minimalize or avoid these risks. First two chapters briefly introduce the development of telephony technologies with differentiation of enterprise telephone network from public telephone networks. Further it describes individual technologies, digitalization of voice, processing the signal and VoIP protocols and components. Third chapter focuses on infrastructure of telephony networks with special interest for architecture of IP telephony and ways of establishing call processing. It describes data flows for further security risk analysis, which this technology came with. Fifth chapter is about enterprise security standards in common and is trying to describe information security management system (ISMS) adopting VoIP technology. Individual security threats and risks are described in sixth chapter, along with known methods how to avoid them. Final parts of thesis concludes of two real situation studies of threats and risks of VoIP technologies implemented in environment of small commercial enterprise and medium size enterprise, in this example represented by University of economics. These chapters conclude theoretical problems shown on practical examples.

This master’s thesis dissertate the possibilities of the linkage between Asterisk softswitch and the 4th generation private branch exchange. This should create a new generation’s network, so-called NGN, by the convergence of existing telecommunication networks with an IP computer network. This master’s thesis is divided into several chapters. In introduction is described the evolution of the private branch exchanges as well as the principles of the voice digitizing, codecs and signaling commonly used in both TDM and VoIP networks. The main aim of this project is the configuration of Asterisk software exchange for connection with PBX Alcatel 4400 as well as public phone network PSTN. Another goal of this master’s thesis was the configuration of Alcatel PBX and diagnostics of CCS and CAS signaling on E1 interface. In conclusion there are summarized advantages of NGN networks and their utilization in the future.

The target of master's thesis is implementation solution of communication VoIP technologies and connection recording system and CRM system in order to saving costs for company INTERNET TRADING s.r.o..

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.

Unified Communications solutions bring together several communication modes, technologies, and applications in order to answer businesses’ and individuals’ growing need for simpler, faster, and more effective communications means.  Although many hardware-based products allow the integration of telephony within a computer network environment, telephony features of software-based unified communications solutions are seldom used, which limits their effectiveness or requires another solution to be used jointly. This master’s thesis project aims to demonstrate that unified communications solutions based on Microsoft Lync Server 2013 can effectively address a wide variety of business scenarios, including a traditional telephony system replacement.  The first part of this master’s thesis introduces background knowledge about unified communications and associated technologies, as well as the different components of the selected unified communication solution. The case study presented in this thesis is the first large-scale Lync 2013 deployment with a complete telephony offering in France. The presentation follows the complete deployment process, starting from the analysis of the client’s needs to the solution design, construction, and validation. This project demonstrated the suitability of Lync 2013 as a telephony system replacement. However, the transition from a classic telephony solution to a unified communications solution can be a technical challenge. An essential step in making this transition successful was to take the users’ needs into account. It was also essential to accompany these users throughout the transition. / Samordnad kommunikation (engelska: unified communications) lösningar sammanföra flera kommunikationssätt, teknik och tillämpningar för att besvara företags och individers växande behovet av enklare, snabbare och mer effektivt kommunikationsmedel. Även många hårdvara-baserade produkter tillåter integration av telefoni inom ett datornätverk miljö, telefoni funktioner mjukvarubaserad Samordnad kommunikation-lösningar används sällan, vilket begränsar deras effektivitet eller kräver en annan lösning för att användas gemensamt.  Detta examensarbete syftar till att visa att samordnad kommunikation lösningar baserade på Microsoft Lync Server 2013 kan effektivt ta itu med en mängd olika scenarier. Den första delen av detta examensarbete introducerar bakgrundskunskap om samordnad kommunikation och tillhörande teknologier liksom de olika komponenterna i den valda samordnad kommunikation lösning. Fallstudien som presenteras i denna avhandling är den första storskaliga Lync 2013 utplacering med en komplett telefoni erbjuder i Frankrike. Den presentationen följer hela implementeringsprocessen, från analys av kundens kraven till utformning, konstruktion, och validering. Detta projekt visade tillförlitligheten i Lync 2013 som telefoni ersättning men intyga att även övergången från en klassisk telefoni lösning på ett samordnad kommunikation-lösning kan vara en teknisk utmaning, ta användarnas behov i beaktande och medföljande användare genom övergången är kritisk. / Les solutions de communications unifiées rassemblent différents modes de communications, technologies, et applications pour répondre aux besoin croissant des entreprises et individus de méthodes de communications plus simples, rapides et efficaces. Bien que de nombreuses solutions matérielles permettent l’intégration de la téléphonie à un réseau informatique, les fonctions de téléphonie des solutions logicielles sont rarement utilisées, ce qui limite leur efficacité ou nécessite l’utilisation conjointe d’autres solutions. Ce projet a pour but de démontrer l’efficacité des solutions de communications unifiées basées sur Microsoft Lync 2013 à répondre à une grande variété de besoins professionels, dont le remplacement d’un système de téléphonie traditionnel. La premiére partie de ce mémoire introduit les notions nécessaires sur les communications unifiées et les technologies associées, ainsi que les différents composants de la solution de communications unifiées choisie. L’étude de cas présentée décrit le premier déploiement majeur de Lync Server 2013 comportant une offre de téléphonie complète en France, et suit le processus de déploiement complet, de l’analyse des besoins client à la validation du projet, en passant par la conception, la construction et le test. Ce projet démontre l’aptitude de Lync en temps que sysème de téléphonie complet. Cependant la transition d’un système traditionnel à une solution de communications unifiées peut présenter des défis techniques, et il est essentiel de prendre en compte les besoins utilisateurs ainsi que de les accompagner durant la transition.