This master’s thesis delves into the possibilities of the open-source Private Branch Exchange Asterisk, elaborates on its features and compares it with several other distros. The term SIP stack is explained here with the mention of two of its representatives. Further in the thesis, the security risks of the VoIP technology are explained, and specific attacks are described and then realized. As a part of the testing process, the possibilities of a custom module and its following implementation are explored, as well as the portability between the individual distros and its proper functioning.

This thesis deals with the possibilities of video conferencing in Asterisk PBX and their use in practice. They also described the contingencies and how its configuration. Particular attention is paid to the protocols SIP, IAX and H.323, which are described in one of the chapters. The thesis was created by the Asterisk PBX, which demonstrates cooperation with videoconferencing clients. The thesis describes the configuration files so that the central set. Conclusion the work assesses the use of codecs for different clients.

This study examines the use of SS7 signaling system over IP networks by using the open source PBX YATE. At first it starts with describing the SS7 followed by an explanation of the function of each of its levels and the messages that are used within the SS7 network. The study then sheds some light on the ways of using SS7 inside IP network with the use of some protocols. It also discusses the architecture of YATE and its files, and how it is installed in Linux operating system. Finally, it describes the important files for delivering this task. The study was commenced by using two virtual machines that have two different open source PBX's which are YATE and Asterisk, and after acquiring some results by establishing communication between them via the means of SIP trunk, furthermore the study was extended to the laboratory in order to test it over real servers that have TDM cards, in order to apply the study by the means of SS7 protocols, SIGTRAN, MGCP gateway and SIP-T. The experiments have almost delivered successful communications after conducting a configuration for the files on multiple sides.

This diploma thesis focuses on attacks on PBX Asterisk, FreeSWITCH and Yate in LTS versions. In this work was carried out two types of attacks, including an attack DoS and the attack Teardown. These attacks were carried out using two different protocols, SIP and IAX. During the denial of service attack was monitored CPU usage and detected if its possible to establish call and whether if call can be processed. The Security of PBX was build on two levels. As a first level of security there was used linux based firewall netfilter. The second level of security was ensured with protocols TLS and SRTP.

The aim of this thesis is to install and to configure three Open source PBXes Asterisk, Freeswitch and YATE. Furthermore, the aim is to realize the performance test and stability tests on three different HW configurations with the tester Spirent Abacus 5000. The scripts in bash were created to monitor PBX performance. Another part of the study is to analyze and to compare PBX security and to compare the Open Source PBX with a proprietary PBX Alcatel-Lucent OXE.

This diploma thesis examines and compares several selected libraries of SIP protocol, performance, stability, security and impact of their configuration. The main functions of the signalling protocol are briefly named at the beginning. The following chapters describe the tested PBXs and several stacks for SIP protocol are theoretically compared. The practical part deals with measurements conducted on the load generator Spirent TestCenter C1 which is used for all the performed tests on exchanges. All the mentioned SIP libraries, PBXs and the operating system on which the PBXs were running are open source software.

Workshop "Netz- und Service-Infrastrukturen" Das Inter-Asterisk eXchange (IAX)-Protokoll ermöglicht eine unproblematische Kommunikation zwischen IAX-fähigen VoIP-Systemen. In der Präsentation zu dem Vortrag werden das Protokoll vorgestellt und die Vorteile von IAX skizziert.

Submitted by Patrícia Cerveira (pcerveira1@gmail.com) on 2017-06-13T15:41:09Z No. of bitstreams: 1 Marcelo Dissertação.pdf: 13477166 bytes, checksum: e46f91138593316f4c2c1504aaf297a3 (MD5) / Rejected by Biblioteca da Escola de Engenharia (bee@ndc.uff.br), reason: Boa tarde, Patrícia! Favor acertar o resumo. Atenciosamente, Catarina Ribeiro Bibliotecária BEE - Ramal 5992 on 2017-06-29T16:10:33Z (GMT) / Submitted by Patrícia Cerveira (pcerveira1@gmail.com) on 2017-06-29T19:26:55Z No. of bitstreams: 1 Marcelo Dissertação.pdf: 13477166 bytes, checksum: e46f91138593316f4c2c1504aaf297a3 (MD5) / Approved for entry into archive by Biblioteca da Escola de Engenharia (bee@ndc.uff.br) on 2017-07-03T12:53:35Z (GMT) No. of bitstreams: 1 Marcelo Dissertação.pdf: 13477166 bytes, checksum: e46f91138593316f4c2c1504aaf297a3 (MD5) / Made available in DSpace on 2017-07-03T12:53:35Z (GMT). No. of bitstreams: 1 Marcelo Dissertação.pdf: 13477166 bytes, checksum: e46f91138593316f4c2c1504aaf297a3 (MD5) / O serviço ”Móvel de Voz sobre IP” é a convergência natural da tecnologia de voz sobre IP (VoIP) e a comunicação sem fio, e pode impulsionar o aumento da popularidade da primeira bem como promover constantes avanços da última. Embora seja possível encontrar várias aplicações que oferecem serviço de VoIP na Internet e muitos dispositivos que implementam VoIP em hardware, uma implementação aberta e não-proprietária pode ser integrada aos serviços legados - como PABXs institucionais - o que proporciona uma contribuição significativa. Este trabalho descreve a implementação do serviço Móvel de Voz sobre IP no Instituto Federal Fluminense e destaca os desafios a serem enfrentados em seu gerenciamento e operação, enfatizando a segurança contra ataques. Os principais resultados indicam que este serviço oferece flexibilidade, conforto, redução de custos e mobilidade para o serviço de voz. / A “Mobile Voice over IP” service is the natural convergence of Voice over IP (VoIP) technology and wireless communication, and can leverage the increading popularity of the former and the constant advances of the latter. Although we can find various applications that o er VoIP service on the Internet, and in e ect many devices that implement VoIP in hardware, an open, non-propietary implementation that can be integrated with legacy services such as institutional PABXs is a welcome addition. This works describes the implementation of the Mobile Voice over IP service in the Instituto Federal Fluminense, and the challenges of its management and operation, with emphasis in security against attacks. This services brings flexibility, confort, cost reduction and mobility to the voice service.

My master’s thesis is focused on designing and creating communication network, which provides communication between two independent networks through encrypted tunnel. My solution is based on routers formed by older personal computers with FreeBSD like a operating system. Between routers is created static encrypted tunnel by using IPSec protocol. Voice services provides packet oriented exchange Asterisk with support of signaling protocol SIP. This solution can be used eg. for connecting remote branch to headquarters of company and then can branch utilize shrared resources. To headquarters can connect also remote workers from their home. In this case are used SSL certificates to authentication of user. This scenario is very required today.

This master's thesis describes how to develop loable kernel module for operating system Linux. Module can be use like driver for concrete hardware device. In this case for telecommunication hardware. The second part of this thesis describes how to implement support for this hardware in Asterisk PBX. Support in Asterisk is realized by channel module. In that channel module are implemented functions for process incoming and dialed calls.