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.
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.