The diploma thesis deals with the system OpenWRT. Installing this system on the router Siemens Gigaset SX762. Describes how to compile and upload the simple package helloworld into this system. The package was tasked invitation simple phrase "Hello World" to the system console of OpenWRT. Package only serve to show that the system OpenWRT allows users to customize it according to their needs. After that it was installed PBX Asterisk into the system OpenWRT. Proper functioning of PBX Asterisk has been verified to make a call between two software phones ZoiPer. Furthermore, the work described telephony application programming interface (TAPI). Some of its fun- ctions, interfaces and packages needed to communicate with the system OpenWRT. In conclusion, the presented test topology and verify the operation of making calls between analog as well as softphones.

As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise.

Semestral project describes signaling system number 7, it's settings options and options of signaling over IP networks, especially two transport protocols SCTP and BICC for signaling SS7 over IP. Analyses kinds of implemetations of this signaling system to the Asterisk PBX with TDM E1 card support. Further part of this work is dedicated to the open source implementations libss7 of Digium and chan_ss7 which is currently developed by Dicea. Describes in detail their installation to the open source PBX Asterisk including testing of both and comparing these two open source solutions. Last part is focused on realization of gateway which converts communication from TDM network to IP network. For this part, three computers are used. First as SS7 signalling end softswitch, second as SIP signalling end softswitch and last as gateway between them. This gate works as interface between SS7 signalling and SIP signalling. Testing call was realized successfully for both directions.

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 topic of this work is verification of support WebRTC communication through selected Open Source PBX. This work examine demands for WebRTC communications and describes configuration of branch centers for this type of communication. In the theoretical part is reader acquainted with the term WebRTC and with protocols related to this kind of communications. The purpose of this part of the work is to bring the reader closer look to the principles of functioning to ensuring support for this kind of communications. This is also connected with Description of basic interfaces of WebRTC applications. Further the reader finds the configuration of the selected Open Source PBX so that they can make audio-video call between WebRTC clients. This section is divided into three subchapters, each of it deals with the same problems for one of the aforementioned PBX. At the end of each chapter where the PBX PBX is configured step-by-step, test calls are made. These calls are captured by the Wireshark packet analyzer and serve as a demonstration of the WebRTC configuration functionality. At the end of this section, PBXs are compared against each other about WebRTC support. Practical part is dealing with laboratory task for students which are studying subject telecommunication and information systems. In the task students will be configuring WebRTC for PBX Asterisk. The task contains brief description of WebRTC and comments for all steps for configuration. All steps and facts are demonstrated by exemplary configuration files.

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.

Short Message Service (shortly SMS) is the most widely used type of communication systems. The main advantages are that allow a fast exchange of messages between devices, a very good availability through GSM and a reasonable price. Nowadays the SMS service support has expanded to include other technologies such as a service of the information navigation and the remote connection. The master‘s thesis concentrates on the Short Message Service, deals with basic principles and statements using by this service. The topic of the thesis is software PBX Asterisk and its possibility of SMS implementation, especially verification of SMS processing goes through the PSTN. After the basic introduction the master‘s work deals with the installation and configuration of the server. The main focus is on an installation of the operating system with an additional pack including necessary libraries and modules for a correct working of the server. The following section is paying attention to the Asterisk server configuration, especially a hardware card installation which is necessary for a connection with analog telephones, done by Bluetooth connections, set up user’s profiles of the SIP protocol and create a dial plan. This is followed by a verification of SMS option of the implementation and communication with GSM modem which is used as a gate for an exchange SMS between PSTN and GSM network. The last chapter of this master‘s thesis comes with the aimed results.

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 thesis compares native SIP stack with PJSIP stack in the open source telephone private branch exchange (PBX) Asterisk. First, there are described both SIP protocol and Asterisk application. Subsequently, the architecture, new function support and the stacks setting possibilities are explored. For different exchange scenarios several commented configuration files are presented. The stacks are tested using Spirent TestCenter C1 software thereafter. In conclusion, selected properties are assessed and new PJSIP stack benefits are summarized. In addition, the laboratory assignment is attached.