|
|
Implementing of GSM modem in PBX Asterisk
Benýšek, Jiří ; Krajsa, Ondřej (referee) ; Šilhavý, Pavel (advisor)
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.
|
|
|
Cross-platform performance testing of open source PBX FreeSWITCH
Melichar, Ondřej ; Kováč, Dominik (referee) ; Šilhavý, Pavel (advisor)
This bachelor’s thesis deals with the open source private branch exchange FreeSWITCH. The thesis describes the process of installation and consecutive configuration of the telephone system on various distributions of platforms Windows and Linux. Emphasis is placed on comparison of the performance between these systems and also the quality of calls as well as transcoding functionality. Specifically, the performance of the telephone system is tested while using the SIP protocol. Thesis also contains an introduction into the subject and a brief description of the FreeSWITCH architecture and its functions and capabilities.
|
|
|
Proprietary VoIP protocols of PBX manufacturers
Bělík, David ; Daněček, Vít (referee) ; Šilhavý, Pavel (advisor)
This thesis focuses on the analysis of proprietary Voice over IP protocols. The first part describes the principles and VoIP technology types. The next section describes proprietary protocols Panasonic IP-PT from Panasonic Corporation and Siemens CorNet-IP from Siemens Corporation. Protocols which are being used in these systems are listed here. Further there is being described the process of registration of terminals to central province. Subsequently calling process and particular terminal functions including packet analysis are being decoded in detail. In conclusion H.323 and MGCP protocols are being compared.
|
|
|
Securing IP PBX against attacks and resistance testing
Kakvic, Martin ; Šedý, Jakub (referee) ; Šilhavý, Pavel (advisor)
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.
|
|
|
Asterisk VoIP private branch exchange and its distributions
Melichar, Ondřej ; Komosný, Dan (referee) ; Papež, Nikola (advisor)
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.
|
|
|
Modular web interface for Asterisk PBX
Moučka, Martin ; Rášo, Ondřej (referee) ; Krajsa, Ondřej (advisor)
This bachelor thesis is dealing with various ways of configuring Asterisk PBX and creating a modular web interface. This interface’s goal is to simplify the configuration of PBX and allow expanding its functions by adding new modules. Apart from a simple configuration, interface contains a self care zone where user can check his call history and keep his own phone book. As part of this work, there are two modules especially useful in environment of technical support and call centers. Those modules provide an opportunity to create interactive voice response menu, calling automat and queues management. Interface depends on Asterisk in version 13 which is the newest version with long term support. The application is secured by user account management with role assigning. The role can be modified by permitting only specified actions.
|
|
|
Kamailio and OpenSIPs open source PBX
Janeček, Václav ; Krkoš, Radko (referee) ; Šilhavý, Pavel (advisor)
Open source PBX Kamailio and OpenSIPS diploma thesis covers familiarization with appointed SIP exchanges and with their power comparing. A detailed installation instructions on the operating system Ubuntu is the aim of this work too. The work includes the historical development of telephone exchanges with a focus on the latest generation. The following is SIP protocol basic description and components that can be composed SIP exchanges. Another part is devoted to the development of exchanges Kamailio and OpenSIPS. The thesis contain the archutecture and configuration file description. The practical part of the thesis deals with high-capacity switches, and comparing it in terms of memory and computational demands. Selected measurements are compared with the Asterisk PBX.
|
|
|
Secured audio codec for Asterisk PBX
Jakubíček, Michal ; Sysel, Petr (referee) ; Krajsa, Ondřej (advisor)
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.
|
|
|
SIP/H.323 gateway with Asterisk PBX
Smékal, Lukáš ; Herman, Ivo (referee) ; Kovář, Petr (advisor)
This Bachelor’s thesis is engaged in problems of the internet telephony and its usage in today´s world, which offers many techniques how communicate among people. The internet telephony gives telephoning other format and shows new possibilities of IP protocol´s usage. The thesis is focused on the most used protocol corveying functions of VoIP technology, it is the standard H.323 and also the SIP protokol, which even if provides the same functions as H.323 is based on the different principle and because of its easy structure, it prepares good conditions for rise of new applications and services in internet telephony. In this thesis the Open Source private branch exchange Asterisk is also discussed. Asterisk, not only thanks to the usage of Voice over IP provides against the standard analogue private branch exchange many advantages and thanks to its modularity and expansibility also offers much more possibilities of configuration. The Session Initiation Protocol (SIP) is the basic protocol, with which Asterisk works. The standard H.323 is not implemented in the basic installation of Asterisk, therefore it is necessary to do its suitable installation and congiruation, in order to expand exchange´s functions for work with this standard and also to be able to realize translation between the protocol SIP and the standard H.323. The results, which I accomplished, are presented in the chapters „Dosažené výsledky“ and in the conclusion of the bachelor’s thesis.
|
|
|
SIP security
Tůma, Petr ; Šilhavý, Pavel (referee) ; Ježek, Jiří (advisor)
This bachelor thesis focuses on security issues of the SIP signalling protocol. The goal was to carry out three attacks and design defences against them. The chosen attacks were a flood attack, a modified message attack and a man-in-the-middle attack. The attacks were conducted against the Asterisk PBX and the results show that some attacks were able to prevent communication between the PBX and clients. Defensive measures are described for each attack in the mitigation subchapter.
|
guest ::
