440-2326/02 – Voice over IP I (VoIP I)
Gurantor department | Department of Telecommunications | Credits | 4 |
Subject guarantor | prof. Ing. Miroslav Vozňák, Ph.D. | Subject version guarantor | Ing. Filip Řezáč, Ph.D. |
Study level | undergraduate or graduate | Requirement | Optional |
Year | 3 | Semester | winter |
| | Study language | English |
Year of introduction | 2021/2022 | Year of cancellation | |
Intended for the faculties | FEI | Intended for study types | Bachelor |
Subject aims expressed by acquired skills and competences
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.
Teaching methods
Lectures
Tutorials
Experimental work in labs
Project work
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.
Compulsory literature:
M. Voznak, Voice over IP, scripts, VSB-TUO, 252 p., 2014
Recommended literature:
Way of continuous check of knowledge in the course of semester
Each student can get a maximum of 40 points during the semester, including:
• semestral project 25 points,
• laboratory exercises, 3x5 points.
E-learning
https://lms.vsb.cz
Other requirements
Knowledge of the TCP/IP protocol family, knowledge of the ISO/OSI network model.
Prerequisities
Subject has no prerequisities.
Co-requisities
Subject has no co-requisities.
Subject syllabus:
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.
Conditions for subject completion
Occurrence in study plans
Occurrence in special blocks
Assessment of instruction
Předmět neobsahuje žádné hodnocení.