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VoIP

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

Course Unit Code440-4106/03
Number of ECTS Credits Allocated6 ECTS credits
Type of Course Unit *Compulsory
Level of Course Unit *Second Cycle
Year of Study *First Year
Semester when the Course Unit is deliveredWinter Semester
Mode of DeliveryFace-to-face
Language of InstructionCzech
Prerequisites and Co-Requisites There are no prerequisites or co-requisites for this course unit
Name of Lecturer(s)Personal IDName
VOZ29prof. Ing. Miroslav Vozňák, Ph.D.
REZ106Ing. Filip Řezáč, Ph.D.
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.
Learning Outcomes of the Course Unit
Understand the technology VoIP.

Learning outcomes are set so that the students are able to identify tasks in the field of VoIP.
Course Contents
Lectures:
1. Real Time Protocol RTP, RTCP, SRTP, speech coding and decoding, speech bandwith requirements for RTP.
2. Standard H.323, protocol model, basic elements - GK, TE, GW, MCU, Signaling H.225.0 RAS, Q.931 and H.245.
3. Call models GRC and DRC, Fast Connect, H.245 tunelling, fax standard T.38.
4. H.323 open solution, GnuGK network design and configuration, GW for PSTN networking.
5. SIP/SDP protocol, description of elements - User Agent, Registrar, Redirect and Proxy server, SIP methods a answers, SDP protocol, transactions and dialogs, offer/answer model.
6. SIPp - SIP sessions generator, SIP/SDP grammar, practical SIP/SDP scenario with authentication.
7. Asterisk, SW PBX, applications, dial plan, extensions, practical using SIP and IAX.
8. SIP and IAX trunk, their application , security on SIP trunk, authorization of access on trunk, rules for modification of dialed umbers.
9. Asterisk - advances services, Presence and Instant Messaging, calendars, practial using presence with Jabber.
10. Asterisk billing - traffic tarification, CDR reports, practical implementation for Asterisk.
11. Kamailio, syntax, structure of configuration, modules, static routing, DB - interoperability, design and configuration, REGISTRAR modul, SIP NAT traversal, RTPPROXY, NATHELPER.
12. Quality of IP telephony - MOS, PESQ, E-model and R-factor, fragmentation, packet losses, delay and jitter, network requirements Intserv, Diffserv.
13. MGCP and Megaco/H.248, WebRTC and new trends in IP telephony.
14. Benchmarking and Penetration tests in VoIP infrastructure.

Exercises:

1. Codecs, RTP - introduction in the lab, software and hardware H.323 and SIP clients. W1
2. Introduction to H.323 - H.323 analysis using Wireshark. W2
3. Asterisk - installation, introduction, dialplan, extensions. W7
4. Asterisk - SIP Trunk configuration. W8
5. Asterisk - advanced services, IVR, Instant Messaging, Call Center. W9
6. Asterisk - billing, traffic accounting, CDR records, configuration, billing centers. W10
7. Quality of service in IP - MOS E-Model, R-factor. Project consulting. W12
8. Presentation of the semester project. W14


Labs:
1. GnuGK - introduction, analysis H.225 and H.245 - 4 points. W3
2. GnuGK - H.323 trunk configuration, DRC and GRC models - 2 points. W4
3. Analysis of SIP signaling and SIP headers - 6 points. W5
4. Introduction to Sipp + semestral project assignment 25p - 3 points. W6
5. Kamailio - Introduction to Kamailio, basic configuration, SIP proxy, database connection. W11
6. WebRTC, HTML5 - an introduction, configuration WebRTC client with Asterisk. W13

Projects:
Semestral project, design of VoIP network with one of the open source
solutions as GnuGK, Kamailio or Asterisk.
Recommended or Required Reading
Required Reading:
M. Voznak, Voice over Internet Protocol, college book, VSB-TUO, 137 p., 2012
Lectures in Moodle
M. Voznak, Voice over IP, VŠ skripta, VŠB-TUO, 252 str., 2014
Přednášky dostupné v Moodle
Recommended Reading:
CAMP, K:IP Telephony Demistified, McGraw-Hill, 2003, New York, ISBN 0-07-140670-0.
HARDY,W.: VoIP service quality, McGraw-Hill, 2003, New York, ISBN 0-07-141076-7.
Sinnreich, H.:Internet Communications Using SIP, Wiley Computer Publishing, New York, 2001, ISBN 0-471-41399-2

Voznak, M.: Introduction to Communication Systems.Issued by VSB-Technical University of Ostrava, 121p., 2012.
VOZŇÁK, M., ŘEZÁČ, F., ASTERISK: teorie a praxe, 2011

Planned learning activities and teaching methods
Lectures, Tutorials, Experimental work in labs
Assesment methods and criteria
Task TitleTask TypeMaximum Number of Points
(Act. for Subtasks)
Minimum Number of Points for Task Passing
Exercises evaluation and ExaminationCredit and Examination100 (100)51
        Exercises evaluationCredit40 15
        ExaminationExamination60 11