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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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Voice Service in Integrated Networks
Yeftsifeyeu, Aliaksandr ; Polívka, Michal (referee) ; Škorpil, Vladislav (advisor)
The master’s thesis presents routing protocols, SIP handshake, describes modern VoIP networks and its features. Thesis is primary focused on networks with integrated MPLS. The most known advantages and disadvantages were described of modern network based on such technology. Further, networks and a router with installed Cisco Call Manager Express 7.1 were configured, which is provided by CISCO. For recreating of fully working telephone network IP PBX10 has been used from SMC Networks. For connection between different phones within these integrated networks, SIP trunk was configured on SMC PBX10 as well as on router, which is connecting three configured Cisco IP phones. Each phone has its own number according to the dial plan. For calls from one network to another the special pattern has been established, so the number of each caller can be easily identified by added digit. On one of the CISCO routers was shown a configuration of MPLS and OSPF protocols. With an analyzer VePAL TX300e, the lab network has been measured to analyze QoS parameters of the network according to standard RFC 2544. Thesis also gives references to lab devices, which have been used accordingly to the work.
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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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Network Protection Using NetFlow Data
Sedlář, Petr ; Žádník, Martin (referee) ; Tobola, Jiří (advisor)
This document provides information about Cisco NetFlow technology and its usage to protect networks from different types of attacks. Part of the document is a summary of common security risks in term of their detection on network and transport layer. There are specified characteristics of NetFlow data containing samples of security risks. On the basis of these characteristics, an application for detection these risks is designed and implemented.
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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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