H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. 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.
Protocols H.323 is a system specification that describes the use of several ITU-T and IETF protocols. The protocols that comprise the core of almost any H.323 system are:
H.225.0 Registration, Admission and Status (RAS), which is used between an H.323 endpoint and a Gatekeeper to provide address resolution and admission control services. H.225.0 Call Signaling, which is used between any two H.323 entities in order to establish communication. (Based on Q.931) H.245 control protocol for multimedia communication, which describes the messages and procedures used for capability exchange, opening and closing logical channels for audio, video and data, control and indications. Real-time Transport Protocol (RTP), which is used for sending or receiving multimedia information (voice, video, or text) between any two entities. Many H.323 systems also implement other protocols that are defined in various ITU-T Recommendations to provide supplementary services support or deliver other functionality to the user. Some of those Recommendations are:
H.235 series describes security within H.323, including security for both signaling and media. H.239 describes dual stream use in videoconferencing, usually one for live video, the other for still images. H.450 series describes various supplementary services. H.460 series defines optional extensions that might be implemented by an endpoint or a Gatekeeper, including ITU-T Recommendations H.460.17, H.460.18, and H.460.19 for Network address translation (NAT) / Firewall (FW) traversal. In addition to those ITU-T Recommendations, H.323 implements various IETF Request for Comments (RFCs) for media transport and media packetization, including the Real-time Transport Protocol (RTP).
Codecs H.323 utilizes both ITU-defined codecs and codecs defined outside the ITU. Codecs that are widely implemented by H.323 equipment include:
Audio codecs: G.711, G.729 (including G.729a), G.723.1, G.726, G.722, G.728, Speex, AAC-LD Text codecs: T.140 Video codecs: H.261, H.263, H.264, H.265 All H.323 terminals providing video communications shall be capable of encoding and decoding video according to H.261 QCIF. All H.323 terminals shall have an audio codec and shall be capable of encoding and decoding speech according to ITU-T Rec. G.711. All terminals shall be capable of transmitting and receiving A-law and μ-law. Support for other audio and video codecs is optional.
Architecture The H.323 system defines several network elements that work together in order to deliver rich multimedia communication capabilities. Those elements are Terminals, Multipoint Control Units (MCUs), Gateways, Gatekeepers, and Border Elements. Collectively, terminals, multipoint control units and gateways are often referred to as endpoints. H.323 uses TCP port number 1720. While not all elements are required, at least two terminals are required in order to enable communication between two people. In most H.323 deployments, a gatekeeper is employed in order to, among other things, facilitate address resolution.
H.323 network elements
Terminals
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