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H.323

Audio-visual communication signaling protocol

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  • 6 chapters

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Through Audiovisual, Packet switching and Internet

It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and video services over IP networks.

It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over ISDN, the PSTN or SS7, and 3G mobile networks.

H.323 call signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDN-based PBX systems, including transitions to IP-based PBXs.

Within the context of H.323, an IP-based PBX might be a gatekeeper or other call control element which provides service to telephones or videophones. Such a device may provide or facilitate both basic services and supplementary services, such as call transfer, park, pick-up, and hold.

History

The first version of H.323 was published by the ITU in November 1996 with an emphasis of enabling videoconferencing capabilities over a local area network (LAN), but was quickly adopted by the industry as a means of transmitting voice communication over a variety of IP networks, including WANs and the Internet (see VoIP).

Over the years, H.323 has been revised and re-published with enhancements necessary to better enable both voice and video functionality over packet-switched networks, with each version being backward-compatible with the previous version. Recognizing that H.323 was being used for communication not only on LANs, but over WANs and within large carrier networks, the title of H.323 was changed when published in 1998. The title, which has since remained unchanged, is "Packet-Based Multimedia Communications Systems." The current version of H.323 was approved in 2009.

One strength of H.323 was the relatively early availability of a set of standards not only defining the basic call model, but also the supplementary services needed to address business communication expectations.

H.323 was the first VoIP standard to adopt the Internet Engineering Task Force (IETF) standard Real-time Transport Protocol (RTP) to transport audio and video over IP networks.

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