Made available in DSpace on 2014-06-12T15:04:21Z (GMT). No. of bitstreams: 2 arquivo1112_1.pdf: 1031240 bytes, checksum: 02753136ced35de8f0a4387436937135 (MD5) license.txt: 1748 bytes, checksum: 8a4605be74aa9ea9d79846c1fba20a33 (MD5) Previous issue date: 2008 / As mudanças na forma de comunicação estão estreitamente associadas a saltos evolutivos na história da humanidade e, conseqüentemente, das organizações. Dentre essas transformações, destacam-se as decorrentes dos recentes e vertiginosos avanços proporcionados pelo uso da tecnologia de informação e comunicação, principalmente da Internet. Esse profícuo ambiente à troca de informações e conhecimentos das organizações é, atualmente, a principal plataforma das aplicações empresariais por se apresentar como o mais eficiente, de baixo custo e ilimitado meio para o desenvolvimento dessas soluções. Destaca-se, dentre estas, a tecnologia de voz sobre o protocolo da Internet (VoIP), por flexibilizar a comunicação mediada por computador, permitindo a conjugação de recursos comunicacionais através de um único meio, reduzindo custos. Apesar do uso da tecnologia VoIP ter crescido consideravelmente com o objetivo de reduzir a conta telefônica e interligar escritórios sem custo, sua utilização na comunicação interna das organizações ainda é pouco conhecida, o que dificulta um melhor aproveitamento de seu enorme potencial. Nesse contexto, o objetivo do presente estudo é proporcionar um maior entendimento das motivações dos gestores ao implantar a tecnologia VoIP na comunicação interna das organizações. Para elaborar o modelo de pesquisa, numa perspectiva exploratória, foi empreendido um estudo de caso

VoIP-system inom företag blir mer vanligt. Säkerheten bör då beaktas för att undvika hot som riskerar konfidentialitet, integritet och tillgänglighet. Denna rapport visar resultat från två olika VoIP-systems säkerhet med hjälp av praktiska penetrationstestscenarion i labbmiljö. En redogörelse över verktyg som är använda för säkerhetstesterna mot VoIP och tillvägagångssätt redovisas i rapporten med förklarande text och tabeller. / VoIP systems in enterprises are becoming more common. Security should then be followed to avoid threats against confidentiality, integrity and availability. This report shows the results from two different VoIP systems security using practical penetration test scenarios in a laboratory environment. A statement of tools that are used for safety tests on VoIP and methods for this, is presented in the report with explanatory text and tables.

Current Voice over IP (VoIP) services are regarded less secure than the traditional public switched telephone network (PSTN). This is due to the fact that VoIP services are frequently deployed in an relatively open environment so that VoIP infrastructures can be easily accessed by potential attackers. Furthermore, current VoIP services heavily rely on other public Internet infrastructures shared with other applications. Thus, the vulnerabilities of these Internet infrastructures can affect VoIP applications as well. Nevertheless, deployed in a closed environment with independent protocols, PSTN has never faced similar risks. The main goal of this licentiate thesis is the discussion of security issues of the Session Initiation Protocol (SIP), which serves as a signalling protocol for VoIP services. This work especially concentrates on the security risks of SIP related to performance. These risks can be exploited by attackers in two ways: either actively or passively. The throughput of a SIP proxy can be actively manipulated by attackers to reduce the availability of services. It is defined as Denial of Service (DoS) attacks. On the other hand, attackers can also profile confidential information of services (e.g., calling history) by passively observing the performance of a SIP proxy. It is defined as a timing attack. In this thesis, we carefully studied four concrete vulnerabilities existing in current SIP services, among which, three of them can lead to DoS attacks and one can be exploited for timing attacks. The results of our experiments demonstrate that these attacks can be launched easily in the real applications. Moreover, this thesis discusses different countermeasure solutions for the attacks respectively. The defending solutions have all in common that they are influencing the performance, by either enhancing the performance of the victim during a DoS attack, or abating the performance to obscure the time characteristic for a timing attack. Finally, we carefully evaluated these solutions with theoretical analyses and concrete experiments.

The significant growth in multimedia services and traffic (e.g. VoIP, video streaming and video gaming) in current and emerging mobile networks including the latest 4G Long-Term Evolution (LTE) networks and the rising user expectation for high Quality of Experience (QoE) for these services have posed real challenges to network operators and service providers. One of the key challenges is how to bring multimedia services to the end-user over resource-constrained mobile networks with a satisfactory QoE. Cost-effective solutions are needed for network operators to improve the bandwidth usage of these mobile networks. Therefore, scheduling schemes are of extreme importance in LTE, where scheduling algorithms are responsible for the overall efficiency of resource allocation in an LTE system. The aim of the project is to develop novel QoE-driven scheduling algorithms for improving system capacity in delivering multimedia services over downlink 3GPP LTE. This is to move away from traditional QoS-driven scheduling schemes to a QoE-driven scheme which guarantee end-user satisfaction in resource allocation. The main contributions of the thesis are threefold: 1. Performance of several existing scheduling algorithms for VoIP applications was evaluated thoroughly in terms of QoE metric (i.e. MOS), instead of QoS metrics (e.g. packet loss and delay). Using QoE metrics instead of QoS ones will facilitate the development of QoE-driven scheduling schemes in order to achieve optimised end-user experiences or optimised mobile system capacity. 2. A novel QoE-driven LTE downlink scheduling scheme for VoIP application was developed to maximize the number of users per cell at an acceptable MOS score. The proposed scheme achieved significant improvement in cell capacity at an acceptable quality (75% compared to MLWDF, and 250% compared to PF and EXP-PF in all three lower speed scenarios considered). 3. A QoE-driven LTE downlink scheduling scheme for multiservice multimedia applications was developed to improve the cell capacity with satisfactory QoE for both VoIP and video streaming services. The proposed algorithm performed well in a pedestrian scenario increasing cell capacity to double for video stream with ‘Rapid Movement’ (RM) content. For ‘Medium Movement’ (MM) video content, the capacity was increased about 20% compared to MLWDF and by 40% compared to EXP-PF. In a vehicular scenario, the proposed scheme managed to enhance the cell capacity for MM video stream case. The project has led to three publications (IEEE Globecom’12 – QoEMC Workshop, IEEE CCNC’15 and IEEE MMTC E-letter/May-2015). A journal paper is in preparation.

This diploma dissertation is dealing with the VoIP software exchange Asterisk. In the dissertation there are described its abilities and possible ways of its configuration. Special attention is given to the signalling protocol SIP, which is described in one of the chapters. Within this dissertation a dial plan, which demonstrates the technique of dial plan creating, was created. Within the boundaries of the dialplan following services could be found: a voicemail, conference, Interactive Voice Response and call queues. Configuration files, with the help of which the exchange is configurated, are described in my dissertation as well. Finally, three laboratory assignments for purposes of the subject Multimedia Services are mentioned. Their main aim is to familiarise students with the creation of SIP accounts in the exchange, their mutual connections, defining the Interactive Voice Response and forming a new call centre.

This master’s thesis deals with security properties of protocols used for VoIP systems. In the first part, there is a description of most commonly used protocols and structure of VoIP systems. This part also discuss signaling and transport protocols. The second part of this paper describes techniques of ensuring quality of services. The next part presents SIP messages and communication. Last part of this paper overviews security risks of VoIP protocol. Practical part of this thesis describes creation of a testing VoIP network, on which several attacks has been made, fallowed by securing of mentioned VoIP network. Result of this thesis is evaluation of security risks connected to VoIP communication.

Main goal of creating the Open Source project and GPL licence are free sources and applications available for a wide public. Competent communities are responsible for support and upgrade of Open source based applications and softwares, which are created on a voluntary bases. Due to this fact an implementation depends on plenty others publicly available libraries and applications, which sometimes complicate the installation process itself. Successfully created VoIP connection is two-phase based process. Signalization is necessary in the first place, which might be supported with H.323 or SIP. After call parameter negotiation – voice codec, cipher code, ports etc, the second phase takes over to transfer voice. Theoretical part of this thesis describes SIP, H.323, MGCP, RTP and IAX protocols, as well as secure ways of signalization and voice stream part of the call. These might be SIPS, SRTP, ZRTP and IPsec. In thesis Open Source Asterisk PBX is well described, when mentioning its options, features and community support. I put near options available for particular releases and introduce attacks and abuses which are possible to perform on the VoIP system in general, together with available, no cost and working tools to perform the attacks with. Practical part focuses on possibilities to generate experimental attacks on individual systen parts with exact definition of what the consequences are. Based on the overall analyse of achieved results I conclude three solutions as autoinstallation linux packages. These „deb“ packages consist of specific Asterisk release required to meet the security needs, ready-to-test configuration and guide to follow with correct options to set. Final security possibilities requires hardening on application layer, where Iptables takes its part. „Linux firewall“ as some express Iptables are configured to reflect VoIP system parameters and protect from DoS attacks.

The Master´s thesis „Traffic management in wireless networks“ discusses how to ensure Quality of Service in these networks. A term „traffic management“ is connected mainly with certain restrictions and prioritization of some services and traffics in network. The thesis contains an overview of the most used wireless technologies and describes the approach of these technologies to ensure QoS. Theoretical part of the thesis follows with the definition of general principles of traffic management in IP networks and provides an overview of network parameters used in evaluating the quality of different types of communications services. Practical part is focused on the most widely used wireless technology of standard 802.11 and also on implementation of remote administration system on a small SOHO routers. These systems allow the use of queueing disciplines to manage QoS. Designed and implemented HTB discipline is tested on a real traffic network model.

Main goal of creating this diploma thesis is existence of increasingly amount of potential threats against IP voice networks (VoIP). The thesis is devoted to testing of various types of attacks and provides some possible solutions for this systems as well. The work points out to a various types of current attacks against either insecure or very little secure structures. The theoretical part is dedicated to analyse and description of wide spectrum of VoIP protocols including signaling protocols (SIP, IAX2), transport protocols (RTP, RTCP) and security protocols (SRTP, ZRTP, IPsec, SDES). Further attention is dedicated to the one of possible open source IP PBX solutions called Asterisk. There is shown a variety of possible attacks against this system due to its openness, because open systems always tend to be more susceptible for various attacks as they need an advanced administration and endless need for searching of new trends in area of security. The last block of the theoretical part is focused on common threats and types of attacks against VoIP networks. The practical part is about design and creation of web application called ,,VoIP Hacks using PHP” written in PHP scripting language and ist main task is to execute three basic attacks: eavesdropping, call drop and call flood. There is also a possibility of port scanning of selected network which is added as supplementary part of this application. The application can be comfortably managed from web browser user interface. All captured data can be displayed directly into the web browser. Tests of the application were performed on Google Chrome and Mozzila Firefox browsers. There is an accent placed on cooperation between the application and terminal linux programmes such as Tshark, BYE Teardown, INVITE flooder or Nmap, which all accept commands from web interface and interpret gained output values back to the web browser.

This thesis describes architecture of IP telephony networks based on signaling protocol SIP and transport protocol RTP. Also tools used for analyzing VoIP traffic are described. Main objective is to design and implement a system for the detection of calls and extraction of voice payloads from  the captured packets. The system first recognizes the signaling messages of SIP protocol. These messages are analyzed afterwards. Output statistics are generated based on gathered data. Voice data will be stored in form, which is suitable for further processing.