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Performance limits, reliability and security of open source PBX
Bednár, Jakub ; Šedý, Jakub (referee) ; Šilhavý, Pavel (advisor)
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.
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Monitoring VoIP Networks
Drozda, Tomáš ; Marek, Marcel (referee) ; Matoušek, Petr (advisor)
This thesis deals with the monitoring VoIP networks. The main objective of this work was to design and then to program monitoring systems for IP telephony network using SIP signaling protocol, primarily for billing purposes. Another important objective was to explore the possibility of obtaining geographic location of the communicating stations. The theoretical part of the thesis deals with the description of the SIP protocol. The rest of the thesis is focused to the design and implementation of the system. The system was implemented as a set of console applications that are complemented by a web application that is used to display the obtained data. Geolocation was solved with the help of IPInfoDb services that provides API for obtaining geolocation information.
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Increasing the efficiency of the handover in real network environment
Michalec, Richard ; Sobotka, Jiří (referee) ; Vajsar, Pavel (advisor)
The aim of this work is to study methods of handover used in WiMAX and WLAN networks, next suggest the posibility of simulation methods for WLAN handover in the simulation environment OPNET Modeler. This work is focused primarily on obtaining the information about neighboring AP and subsequent selection of a new AP using such information. The work describes in detail each processes solutions that are used to implement newscanning methods
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SIP implementations in Asterisk open source PBX
Bednář, Vít ; Krajsa, Ondřej (referee) ; Šilhavý, Pavel (advisor)
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.
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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.
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Deployment of SIP Server at the FIT for IP Telephony
Hýbner, Lukáš ; Ráb, Jaroslav (referee) ; Matoušek, Petr (advisor)
Master's Thesis is engaged on possibilities connect SIP server to telephone network on FIT. The main reason is that employees can call to university, when they are out of faculty. For resolution we will use SIP server Asterisk, which will be serving as authorization server for users. Next Asterisk will ensure transmission numbers to SIP address with ENUM. In the practical part we will verify the functionality.
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Laboratory of Quality of Service
Eltahir Ahmadon, Ahmed ; Kolka, Zdeněk (referee) ; Polívka, Michal (advisor)
The Bachelor's work, you are reading, deals with the issue of VoIP calling and measurement its quality. In the following paragraphs the discussion will be about the basics of networking protocols such as TCP, UPD, IP and SIP, after that a preliminary introduction to QoS issues - ways of measuring voice quality for instance MOS and E-Model. Importance of implementing QoS for VoIP, and QoS parameters used in the proposed services (will be called "mechanisms"), such as Best effort service, Integrated and Differentiated services. Based on these fundamentals points with designed laboratory for quality of service. The laboratory will be aimed at testing the quality of codecs with regard to bandwidth, jitter, delay and packet loss. The solution lies in the fact that, which codecs are actually suitable for the service.
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A Tool for Testing VoIP Security
Mazálek, Pavel ; Skokanová, Jana (referee) ; Matoušek, Petr (advisor)
This work is focused on the security of SIP. It describes basic principles of signaling protocol and it outlines the necessary protocols for voice data transfer. It describes some basic attacks on device working with SIP and provides examples and applications. The last chapter is devoted to descrie the program can be used for penetration testing of devices.
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Speaker Recognition in the VoIP Environment
Remeš, Jan ; Pešán, Jan (referee) ; Plchot, Oldřich (advisor)
Tato práce popisuje použití systémů pro rozpoznávání mluvčího v~prostředí VoIP, úspěšnost systému a přístupy k jejímu zlepšení. Popisuje architekturu těchto systémů, metriky pro vyhodnocení jejich úspěšnosti a klíčové komponenty VoIP z hlediska rozpoznávání mluvčího. Je zde popsáno vytvoření simulace VoIP prostředí, úspěšnost systému je vyhodnocena na datech pocházejících z různých druhů VoIP prostředí a výsledky jsou demostrovány. Adaptace a kalibrace systému je provedena a jejich přínosy zhodnoceny.
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VoIP Security
Černý, Michal ; Číka, Petr (referee) ; Kovář, Petr (advisor)
Internet telephony becomes still more popular option to common telephone service. Reason may be it's price as well as availability. With growing popularity grows the risk of unauthorized use of transferred data too. There are several ways to secure VoIP communication. This bachelor's thesis talk about one of many, which is use of Secure RTP. There will be made it's implementation in the software telephone exchange Asterisk and tested functionality by trial talks and analysis of network traffic as well.
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