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VSB-TUO ›Study ›Exchange students ›Erasmus+ ›Course catalogue for Erasmus+ students ›Faculty of Electrical Engineering and Computer Science

Faculty of Electrical Engineering and Computer Science

ECTS Course Overview



Voice over IP I

* Exchange students do not have to consider this information when selecting suitable courses for an exchange stay.

Course Unit Code440-2326/02
Number of ECTS Credits Allocated4 ECTS credits
Type of Course Unit *Optional
Level of Course Unit *First Cycle
Year of Study *
Semester when the Course Unit is deliveredWinter Semester
Mode of DeliveryFace-to-face
Language of InstructionEnglish
Prerequisites and Co-Requisites Course succeeds to compulsory courses of previous semester
Name of Lecturer(s)Personal IDName
VOZ29prof. Ing. Miroslav Vozňák, Ph.D.
REZ106Ing. Filip Řezáč, Ph.D.
Summary
The graduates will understand VoIP technology and its protocols. They will also be introduced to open-source programmable platforms that provide VoIP services and offer complete management of IP telephony infrastructure.
Learning Outcomes of the Course Unit
Learning outcomes are set so that students are able to identify, apply and solve tasks in the field of configuration of VoIP services and analysis of IP telephone protocols.
Course Contents
1. RTP protocol, RTCP, Voice encoding and decoding methods, bandwidth calculation for RTP.
2. Signaling protocols in IP telephony - H.323, SIP / PJSIP, SCCP, IAX, MGCP, connection with PSTN.
3. SIP protocol - addressing, SIP URI, ENUM, description of elements User Agent, Registrar, Redirect, Proxy server, B2BUA, SIP methods and responses, transactions and dialogs.
4. SIP protocol - early media, fields and header parameters, (de) registration, routing, tags, SIP and NAT.
5. SIP protocol - advanced methods and answers - re-INVITE, UPDATE, redirection, click2dial, call acceptance, timers, SIP identity.
5. SDP protocol - fields and parameters, attributes to the stream media, offer / answer model, SDP XML and JSON.
6. SIPp - SIP session generator, SIP / SDP grammar, compilation of a practical scenario with authentication
7. WebRTC - description of APIs and modules, Websockets and encapsulation principle, signaling and media transmission protocols, Peer2peer vs. WebRTC / SIP gateway, practical examples.
8. Asterisk - description of SIP vs. PJSIP, applications, dial plan, extensions, regular expressions, contexts and prefixes.
9. Asterisk - SIP call, SIP trunk, voicemail, IVR, queues, CDR records, billing.
10. Kamailio - description, modules, synthesis, configuration structure, kamctl, usage.
11. Monitoring and management of SIP sessions - CDR, HEP protocol, HOMER.
12. Videoconferencing - MCU, media GW.
13. Quality in IP telephony - MOS, PESQ, E-model and R-factor, fragmentation, packet loss, delay and jitter, network requirements (Intserv, Diffserv).
14. Introduction to VoIP security - SIP over TLS, S / MIME, SRTP, ZRTP.
Recommended or Required Reading
Required Reading:
M. Voznak, Voice over IP, scripts, VSB-TUO, 252 p., 2014
M. Voznak, Voice over IP, VŠ skripta, VŠB-TUO, 252 str., 2014
M. Voznak, Voice over IP, scripts, VSB-TUO, 252 p., 2014
Recommended Reading:
Alan B Johnston, SIP: Understanding the Session Initiation Protocol, Fourth edition, 2015, ISBN: 978-1608078639
Jim Van Meggelen, Russell Bryant, Leif Madsen, Asterisk: The Definitive Guide, 5th Edition, 2019, ISBN: 978-1492031604
Planned learning activities and teaching methods
Lectures, Tutorials, Experimental work in labs, Project work
Assesment methods and criteria
Tasks are not Defined