H.323 is an umbrella recommendation from the ITU-T, that defines the protocols to provide audio-visual communication sessions on any packet network. It is currently implemented by various Internet real-time applications as NetMeeting and GnomeMeeting (the latter using the OpenH323 implementation). It is a part of the H.32x series of protocols which also address communications over ISDN and PSTN. A challenger to H.323 is SIP, a standard from the IETF. Both protocols are used in Voice over IP (VoIP, Internet Telephony, or IP Telephony).

One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services, needed to address business communication expectations. H.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks.

H.323 is suited for interworking scenarios between IP and ISDN, respectively between IP and QSIG. A call model, similar to the ISDN call model, eases the introduction of IP Telephony into existing networks of ISDN based PBX systems. A smooth migration towards IP based PBX systems becomes plannable.

Within the context of H.323, an IP based PBX is, simply spoken, a Gatekeeper plus supplementary services.

H.323 references many other ITU-T protocols like:

• H.225.0 protocol is used to describe call signaling, the media (audio and video), the stream packetization, media stream synchronization and control message formats.
• the H.245 describes the messages and procedures used for opening and closing logical channels for audio, video and data, capability exchange, control and indications.
• H.450 describes the Supplementary Services
• H.235 describes security in H.323

External links

• H.323 Information Site
• H.323 open source project
• VoIP information site