440-4106/04 – VoIP (VoIP)
Gurantor department | Department of Telecommunications | Credits | 6 |
Subject guarantor | prof. Ing. Miroslav Vozňák, Ph.D. | Subject version guarantor | prof. Ing. Miroslav Vozňák, Ph.D. |
Study level | undergraduate or graduate | Requirement | Choice-compulsory |
Year | 1 | Semester | winter |
| | Study language | English |
Year of introduction | 2015/2016 | Year of cancellation | 2022/2023 |
Intended for the faculties | FEI | Intended for study types | Follow-up Master |
Subject aims expressed by acquired skills and competences
Understand the technology VoIP.
Learning outcomes are set so that the students are able to identify tasks in the field of VoIP.
Teaching methods
Lectures
Tutorials
Experimental work in labs
Summary
This course is directed towards the students of study program Information and communication technology. The aim is to acquaint students with technologies and standards of voice transmission in IP network with communication protocols H.323, SIP, MGCP and with elements enabling an implementation of voice services in IP network. A significant part is focused on area of Quality of Service. Laboratory works are oriented on protocol analyzing and students can choose semestral project from three themes based on open source VoIP solution as Asterisk, GnuGK or OpenSER. Communication standards are been already formed with pursuit to network design with integrated service which are able to transfer a data, voice or video. The next generation networks are using in considerable amount these techniques which are called as Voice over IP and VoIP is the significant direction in next evolution of communications.
Compulsory literature:
M. Voznak, Voice over Internet Protocol, college book, VSB-TUO, 137 p., 2012
Lectures in Moodle
Recommended literature:
Additional study materials
Way of continuous check of knowledge in the course of semester
Conditions for credit:
Student can obtain max. 40 points, during the semester he obtains points for lab works max. 15 p. and for semestral project max 25 p., there is required to get minimally 15 p.
E-learning
Other requirements
Knowledge of computer networks are required
Prerequisities
Subject has no prerequisities.
Co-requisities
Subject has no co-requisities.
Subject syllabus:
Lectures:
Real Time Protocol RTP, RTCP, SRTP, speech coding and decoding, speech bandwith requirements for RTP.
Standard H.323, protocol model, basic elements - GK, TE, GW, MCU, Signaling H.225.0 RAS, Q.931 and H.245
Call models GRC and DRC, Fast Connect, H.245 tunelling, fax standard T.38
Open solution H.323, network design and configuration GnuGK.
GW for PSTN networking, comparison of interfaces and capabilities - FXS, FXO, EM, ISDN PRI and BRI.
SIP/SDP protocol, description of elements - User Agent, Registrar, Redirect and Proxy server, SIP methods a answers, RFC 3261, SDP protocol fro media description of SIP session.
SIP headers parsing, transactions and dialogs, next methods not included in SIP core, offer/answer model, scenarios of media negotiations, using DNS record for IP telephony, ENUM.
SIPp – SIP sessions generator, SIP/SDP grammar, practical SIP/SDP scenario with authentication.
Asterisk, SW PBX, applications, dial plan, extensions, practical using SIP and IAX.
SIP and IAX trunk, their application , security on SIP trunk, authorization of access on trunk, rules for modification of dialed umbers.
Asterisk – advances services, Presence and Instant Messaging, calendars, practial using presence with Jabber.
Kamailio, syntax, structure of configuration, modules, static routing, DB - interoperability, design and configuration, REGISTRAR modul, SIP NAT traversal, RTPPROXY, NATHELPER.
Quality of IP telephony - MOS, PESQ, E-model and R-factor, fragmentation, packet losses, delay and jitter, network requirements Intserv, Diffserv).
MGCP and Megaco/H.248, webRTC and new trends in IP telephony.
Exercises:
Codecs, RTP – introduction to the lab, software H.323 and SIP clients. W1
Introduction into H.323 – hardware H.323 and SIP clients. W2
Introduction into H.323 - H.323 analysis. W3
Asterisk introduction, diaplan and extensions + project assigment - 25 points. W9
Asterisk trunk and security. W10
Asterisk advances services. W11
Quality of service in IP telephony – consultation of the project. W13
Trends in IP telephony - webRTC,ENUM, best practices + discussion – project presentation. W14
Labs:
GNuGK – H.225 and H.245 – rated exercise with maximum 4 points. W4
Gnugk trunk - rated exercise with maximum 2 points. W5
SIP - SIP signalization - rated exercise with maximum 3 points. W6
SIP - SIP header- rated exercise with maximum 3 points. W7
Sipp - rated exercise with maximum 3 points. W8
Kamailio – introduction into Kamailio + work on the project. W12
Projects:
Semestral project, design of VoIP network with one of the open source solutions as GnuGK or Kamailio or Asterisk.
Conditions for subject completion
Occurrence in study plans
Occurrence in special blocks
Assessment of instruction
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