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.
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.