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Communication systems based on IP telephony
Zimek, Josef ; Kapoun, Vladimír (referee) ; Herman, Ivo (advisor)
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.
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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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Analysis of Audio Codecs Applied in IP telephony
Hlavica, Michal ; Szőcs, Juraj (referee) ; Škorpil, Vladislav (advisor)
Issue of this diploma thesis is focused on analysis of audio codecs used within IP telephony. Attention of teoretical part is given mostly to audio codecs according to ITU-T recommendations, but also to signaling protocols used here. For practical part of analysis is chosen router Cisco 2821 and IP phones Cisco 7975G. Configuration is done over operating system Cisco IOS. Chosen signaling protocol is SCCP. For analysis itself are chosen 2 analysers – L-580FX and Fluke NetTool. These are used in combination with program Wireshark. Analysed parameters are latency, packet lost, bandwidth, jitter and mean opinion score. Measured values are presented in graphs and tables and they are discussed. Next output of the thesis is laboratory excercise, which deals with analysis of audio codecs.
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Analysis of VoIP implementation in the WAN network of Třinecké železárny a.s.
Pieniažek, Ireneusz ; Skořepa, Michal (referee) ; Molnár, Karol (advisor)
This diploma thesis is focused on the implementation of the technology VoIP in the remote branch offices of a big company. There are used the current WAN lines, which have been so far used only for transfer of data. Implementation of IP telephony is extending the utilization of those lines and at the same time the costs for running of the telephone services are reduced. There are described the models for proposal and also the proposal for solution of IP telephony for a specific company.
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Home VoIP
Bílek, Tomáš ; Ščuglík, František (referee) ; Ráb, Jaroslav (advisor)
The thesis explores practical applications of VoIP technology at home, in combination with the opensource private branch exchange Asterisk. A substantial part of this work is the documentation of the functioning of the main parts of public switched telephone networks (PSTN) and VoIP networks. Also described are the principles of voice digitization, commonly used codecs and several VoIP protocols. The last part of the thesis focuses on the design, practical realization and verification of two home VoIP installations.
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Speech signal transmission over unreliable networks
Rozman, Jiří ; Krajsa, Ondřej (referee) ; Polívka, Michal (advisor)
This thesis deals with analysis of internet telephony. It is fosuced on analysis of VoIP over metalic and wireless network using virtual telephonic switch-board with hardware and software phones and also alternative programs to transmit voice such as Skype and Windows Messenger. It also deals with simulating parameters that influance voice transmission like jitter, latency packet loss. WANem system was used for this simulation. In the end it is focused on the problem of providing flawless voice communication over internet, specially providing Quality of Service.
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VoIP Quality Analyzer
Krajíček, Michal ; Žádník, Martin (referee) ; Tobola, Jiří (advisor)
This thesis deals with the quality of the IP telephony and its measuring using the netflow technology. It describes individual factors influencing the quality from sampling and quantization over the impairment caused by codecs to the degradation during network transfers. Next part focuses on models allowing to regard quality of IP telephony with emphasis to the E-model and R-factor. It shortly describes the netflow technology and the quality measuring connected with it. Practical part describes the principle of how the quality is measured by the netflow probe nProbe together with offer and implementation of application measuring the quality. The comparison with commercial application is presented and the results are discussed.
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Automatic Attendant application using Asterisk PBX
Benýšek, Jiří ; Krajsa, Ondřej (referee) ; Kovář, Petr (advisor)
Application of IP telephony is still more and more popular. This technology is designed for large companies as well as for home or office use. If customer needs to use advanced features of the private branch exchanges, he had to purchase more expensive solutions than common equipment. These professional products may be replaced by open source solutions, which can fully replace them. Private branch exchange Asterisk is one example of these products. Bachelor’s thesis is focused on IP telephony, treats of basic principles and network protocols used by this service. This thesis then describes the public branch exchange Asterisk, its possibilities for use in call centers and administration of call queues using this PBX. After theoretical analysis, this thesis deals with installation and configuration of this PBX. It includes installation of necessary operating system, additional packages containing necessary libraries and modules needed for correct function of the PBX. Configuration of Asterisk such as creation of user accounts of SIP protocol and dialplan is described in the next chapter and then all of these functions are tested. Achieved practical results are presented in the last chapter of this bachelor’s thesis.
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DoS a DDoS útoky na SIP protokol
Staněk, Jan ; Peterka, Jiří (advisor) ; Vozňák, Miroslav (referee)
The aim of this diploma thesis is to get accustomed with the SIP protocol and with the problematics of attacks targeting this protocol, with the emphasis on DoS and DDoS attacks. The thesis focuses on detailed classification of the attacks, possibilities and forms of generation of the attacks and methodics of defense against them. The attacks of the flood type are especially stressed because they are easily generated and the SIP components are very prone to these attacks. Prototype implementations of the most important ideas concerning attack generation and protection against these attacks are also part of this thesis. Practical tests of the implementations performed in a simulated SIP environment are also included. 1
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