Lectures:
Real Time Protocol RTP, RTCP, SRTP, speech coding and decoding, speech bandwith requirements for RTP.
Standard H.323, protocol model, basic elements - GK, TE, GW, MCU, Signaling H.225.0 RAS, Q.931 and H.245
Call models GRC and DRC, Fast Connect, H.245 tunelling, fax standard T.38
Open solution H.323, network design and configuration GnuGK.
GW for PSTN networking, comparison of interfaces and capabilities - FXS, FXO, EM, ISDN PRI and BRI.
SIP/SDP protocol, description of elements - User Agent, Registrar, Redirect and Proxy server, SIP methods a answers, RFC 3261, SDP protocol fro media description of SIP session.
SIP headers parsing, transactions and dialogs, next methods not included in SIP core, offer/answer model, scenarios of media negotiations, using DNS record for IP telephony, ENUM.
SIPp – SIP sessions generator, SIP/SDP grammar, practical SIP/SDP scenario with authentication.
Asterisk, SW PBX, applications, dial plan, extensions, practical using SIP and IAX.
SIP and IAX trunk, their application , security on SIP trunk, authorization of access on trunk, rules for modification of dialed umbers.
Asterisk – advances services, Presence and Instant Messaging, calendars, practial using presence with Jabber.
Kamailio, syntax, structure of configuration, modules, static routing, DB - interoperability, design and configuration, REGISTRAR modul, SIP NAT traversal, RTPPROXY, NATHELPER.
Quality of IP telephony - MOS, PESQ, E-model and R-factor, fragmentation, packet losses, delay and jitter, network requirements Intserv, Diffserv).
MGCP and Megaco/H.248, webRTC and new trends in IP telephony.
Exercises:
Codecs, RTP – introduction to the lab, software H.323 and SIP clients. W1
Introduction into H.323 – hardware H.323 and SIP clients. W2
Introduction into H.323 - H.323 analysis. W3
Asterisk introduction, diaplan and extensions + project assigment - 25 points. W9
Asterisk trunk and security. W10
Asterisk advances services. W11
Quality of service in IP telephony – consultation of the project. W13
Trends in IP telephony - webRTC,ENUM, best practices + discussion – project presentation. W14
Labs:
GNuGK – H.225 and H.245 – rated exercise with maximum 4 points. W4
Gnugk trunk - rated exercise with maximum 2 points. W5
SIP - SIP signalization - rated exercise with maximum 3 points. W6
SIP - SIP header- rated exercise with maximum 3 points. W7
Sipp - rated exercise with maximum 3 points. W8
Kamailio – introduction into Kamailio + work on the project. W12
Projects:
Semestral project, design of VoIP network with one of the open source solutions as GnuGK or Kamailio or Asterisk.
Real Time Protocol RTP, RTCP, SRTP, speech coding and decoding, speech bandwith requirements for RTP.
Standard H.323, protocol model, basic elements - GK, TE, GW, MCU, Signaling H.225.0 RAS, Q.931 and H.245
Call models GRC and DRC, Fast Connect, H.245 tunelling, fax standard T.38
Open solution H.323, network design and configuration GnuGK.
GW for PSTN networking, comparison of interfaces and capabilities - FXS, FXO, EM, ISDN PRI and BRI.
SIP/SDP protocol, description of elements - User Agent, Registrar, Redirect and Proxy server, SIP methods a answers, RFC 3261, SDP protocol fro media description of SIP session.
SIP headers parsing, transactions and dialogs, next methods not included in SIP core, offer/answer model, scenarios of media negotiations, using DNS record for IP telephony, ENUM.
SIPp – SIP sessions generator, SIP/SDP grammar, practical SIP/SDP scenario with authentication.
Asterisk, SW PBX, applications, dial plan, extensions, practical using SIP and IAX.
SIP and IAX trunk, their application , security on SIP trunk, authorization of access on trunk, rules for modification of dialed umbers.
Asterisk – advances services, Presence and Instant Messaging, calendars, practial using presence with Jabber.
Kamailio, syntax, structure of configuration, modules, static routing, DB - interoperability, design and configuration, REGISTRAR modul, SIP NAT traversal, RTPPROXY, NATHELPER.
Quality of IP telephony - MOS, PESQ, E-model and R-factor, fragmentation, packet losses, delay and jitter, network requirements Intserv, Diffserv).
MGCP and Megaco/H.248, webRTC and new trends in IP telephony.
Exercises:
Codecs, RTP – introduction to the lab, software H.323 and SIP clients. W1
Introduction into H.323 – hardware H.323 and SIP clients. W2
Introduction into H.323 - H.323 analysis. W3
Asterisk introduction, diaplan and extensions + project assigment - 25 points. W9
Asterisk trunk and security. W10
Asterisk advances services. W11
Quality of service in IP telephony – consultation of the project. W13
Trends in IP telephony - webRTC,ENUM, best practices + discussion – project presentation. W14
Labs:
GNuGK – H.225 and H.245 – rated exercise with maximum 4 points. W4
Gnugk trunk - rated exercise with maximum 2 points. W5
SIP - SIP signalization - rated exercise with maximum 3 points. W6
SIP - SIP header- rated exercise with maximum 3 points. W7
Sipp - rated exercise with maximum 3 points. W8
Kamailio – introduction into Kamailio + work on the project. W12
Projects:
Semestral project, design of VoIP network with one of the open source solutions as GnuGK or Kamailio or Asterisk.