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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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VoLTE service implementation in EPS-IMS networks
Baev, Mikhail ; Koton, Jaroslav (referee) ; Novotný, Vít (advisor)
Diplomová práce popisuje VoLTE službu, vývoj a nasazení LTE (zaváděcí fázi, skutečný LTE stav a výhledy do budoucna atd.), EPC-IMS architekturu (popis funkce uzlu, rozhraní atd.) Komunikace mezi uzly a funkce, rozhraní a protokoly jsou používány v průběhu signalizace (SIP SDP) a datový tok (RTCP RTP). Práce stručně popisuje základní toky hovorů, typy nosičů (GBR and N-GBR), a to vytvoření / mazaní nosičů během komunikace. Další část diplomové práce o implementaci volte, instalace a konfigurace IMS. Závěrečná část diplomové práce popisuje zkoušky sítě a, analýzu protokolu.
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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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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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Voice Dialog System in Web Browser for Demonstration Purposes
Vlček, Pavol ; Glembek, Ondřej (referee) ; Schwarz, Petr (advisor)
Cieľom práce je navrhnúť a vytvoriť hlasom ovládaného asistenta(voicebota), ktorý bude ľahko nasaditeľný na webovú stránku. Používateľom tak bude poskytnutý moderný spôsob, ako prirodzene komunikovať cez internetový prehliadač. Hlavný dôraz je kladený na synchronizáciu medzi hlasovým asistentom a obsahom na webovej stránke. Synchronizácia je dosiahnutá obojsmerným prenosom hlasu a textových príkazov medzi klientom a serverom. Na to je použitá technológia WebRTC v kombinácií so signalizačným protokolom SIP. Práca sa zaoberá oblasťami ako VoIP telefonovanie, počítačové siete a strojové učenie(proprietárne rečové technológie od Phonexie). Benefitom nasadenia hlasového asistenta je zníženie nákladov na odchádzajúce hovory pre klientov, odľahčenie agentov na call centrách pri odpovedaní na často kladené otázky a zvýšenie záujmu zákazníkov vďaka použitiu nových technológií.
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Implementation of Asterisk VoIP PBX
Schön, Martin ; Münster, Petr (referee) ; Szőcs, Juraj (advisor)
The thesis is divided into two parts, theoretical and practical. In theory, it analyzed the signaling protocols, the RTP protocol for data transmission, VoIP Asterisk PBX and security options. The practical part describes the installation of the Asterisk PBX, conducted demonstration call forwarding. PBX was associated with an LDAP directory server to manage users and ejabberd for text messaging protocol XMPP. It was configured translation SIP-Skype using the SiSky Gateway and with dialplan SIP-H.323 translation. The study was to develop Dialplan with IVR. It was measured bandwidth of the audio/video codecs in an encrypted/unencrypted connections.
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Adaptation of access networks for advanced networking technologies
Frollo, Martin ; Škorpil, Vladislav (referee) ; Novotný, Bohumil (advisor)
The bachelor thesis is focusing on the real-time services running in packet networks and problems that may arise while they are running. Specifically it is focused on the technology of VoIP service. There are many different signaling protocols to support this service. Among others, the SIP protocol is very widespread. In addition of the connection it is necessary to ensure the transmission of voice data. That is the task of the RTP protocol. This service and also others for which time is critical with increasing constraints on networks it needs to be frequently given the priority over traditional services like downloading files, and more. VoIP service has certain requirements for transmission parameters that must be followed for optimal function. They are for example, packet loss, one-way delay and delay variation. Most problematic is to ensure the optimal operation of the service on slower transmission routes. That is why it was necessary to establish mechanisms capable of real-time services to ensure sufficient network capacity to meet end user. These are, for example technologies of integrated services or differentiated services. At the present time, however, the technology of integrated services is not very widespread because of its inefficiency. On assembled telephone network made from available components in the laboratory, there are analyzed and confirmed some of the distinctive features and characteristics of the SIP protocol. Furthermore, measurements confirmed sufficient network capacity for VoIP service of the used network and parameters affecting security of the QoS have been evaluated.
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SIP Client for Windows Mobile
Regueyra, Philip ; Müller, Jakub (referee) ; Číka, Petr (advisor)
The main objective of this thesis is the design and implementation of client software, which will be able to conduct a phone connection through the packet network. Signalling protocol SIP together with RTP, the protocol for the transmission of multimedia, is used for this intention. Structure and functioning of the SIP protocol and RTP protocol is analyzed in this thesis. This is introducing with the issue, not a detailed study of these instruments. In addition, a brief overview of the Windows Mobile operating system for which the application is intended primarily is released. The characteristic of several libraries that have been considered for use in the application is briefly described in this thesis. Next the design of solution and the analysis of the program follow. In the last part the functioning of the program is described and its behavior is tested in various model situations.
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