The work discusses about the PBX Yate, describes its configuration from the initial installation, through the basic setup and description of the way through the logs of user registration, the syntax for defining the calling plan, including examples, and additional services, including demonstration of practical verification of the functionality in the computer lab. Are also created two samples job for more familiar with the principles of operation and the Yate PBX settings.

This master's thesis deals with open source PBX security against security attacks. In the theoretical part is detailed description of problematic about attacks that could be used on VoIP systems with high focus on the Denial of Service attack. Furthermore are in theoretical part described methods of security of initialization protocol SIP. Individual chapter is devoted to intrusion detection and prevention of IDS and IPS systems, focusing on Snort and OSSEC. In the practical part of the work was created generator of attacks against various PBX systems, which was subsequently used for detailed testing. Special tests of PBX system are then used against DoS attacks, for which was created protection in form of active elements consisting of IDS Snort & OSSEC. These are capable to provide protection in real-time. The protection was tested on particular PBX systems and in matter of comparison were measured possibilities before and after of security implementation. The output of this work is attacks generator VoIPtester and creation of configuration rules for Snort and OSSEC.

Parlay is an open standard that focuses on opening up the telecommunication networks for new services and new service providers. It encapsulates the underlying protocols and signaling layers and provides a framework for services so that the applications do not need to handle signaling and network related tasks. Parlay is also created to integrate the public telephone network with wireless and packet based networks. Java APIs for Integrated Networks (JAIN) is the Java implementation of the Parlay specification and includes a Java framework for network access and call control, among others. I have in this thesis used the Parlay and JAIN technology to design a Virtual Private Branch Exchange (VPBX), which is a software based system for the telecommunications domain. The VPBX provides functionality such as attendant control, routing and re-routing of calls, telephone queue handling and calls on hold. All use of the system is made from mobile phones, even the attendant client application is designed for a handheld device. I have in this thesis presented an architectural design for the VPBX and an evaluation of the design.

I detta arbete har vanligt förekommande krav på IP-telefonilösningar identifierats via litteraturstudier. Därefter har en testmiljö skapats där ett urval av mjukvaror har utvärderats med avseende på funktionalitet och dokumentation. Kraven som ställts relaterar till säkerhet och kvalitet samt den funktionalitet som krävs för att motsvara den hos det vanliga telefonnätet. Mjukvarorna som presenteras är 3CX, Asterisk, sipXecs och Switchvox. Urvalet av mjukvaror är baserade på antingen öppen källkod eller proprietära motsvarigheter. Faktorer som möjlighet till support och användarvänlighet har också belysts. De mjukvaror som presterar bäst utifrån givna parametrar är 3CX och Asterisk. Arkitekturen mellan de två mjukvarorna olika och båda kan ha fördelar beroende på struktur och kompetens hos det specifika företaget.

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 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.

This thesis is focused on the design of secured audio codec for Asterisk PBX. The first chapter is focused on the basic division of traditional PBX producers and the open source PBX. The second chapter explains the structure of Asterisk PBX and its fundamental difference from a traditional PBX. Asterisk is based on components called modules, therefore the work also deals with the most important modules for operation of exchanges and their division of terms of support and dividing by the type of application and their properties. In this chapter there are described in more detail audio codec A-law and u-law. The third chapter contains simple instructions to get your orientation in the construction of the module for Asterisk PBX and this guide is accompanied by a simple example of creating a module demonstration of his method of translation, commissioning and loaded into Asterisk. Simulation of voice security is in the fourth chapter which provides a description of the proposed security solutions with subsequent implementation in Simulink. This simulation verifies the functionality of the solution proposed security phone call. In the fifth chapter outlines the historical use of encryption techniques primarily mirroring the spectrum and time division signal and comparing them with current modern digital technics. In the last sixth chapter is the actual implementation audio codec module with encryption.

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.