The International Telecommunications Union (ITU) which sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS) has made H.323 as an umbrella recommendation that provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. H.323 includes parts of H.225.0 - RAS, Q.931, H.245 RTP/RTCP and audio/video codecs, such as the audio codecs ( G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263) that compress and decompress media streams. Media streams are transported on RTP/RTCP. [para]RTP carries the actual media and RTCP carries status and control information. The signalling is transported dependably over TCP. The H.323 standards are important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communication as these networks dictate today's corporate desktops and include packet-switched TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies.
H.323 is an umbrella 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, NetMeeting and X-Meeting, and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks.
It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over Integrated Services Digital Network (ISDN), Public Switched Telephone Network (PSTN) or Signaling System 7 (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 Private Branch eXchanges (PBXs).
Within the context of H.323, an IP-based PBX might be an H.323 Gatekeeper or other call control element that 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.
While H.323 excels at providing basic telephony functionality and interoperability, H.323’s strength lies in multimedia communication functionality designed specifically for IP networks.
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, commonly referred to as "H.323v6", was published in 2006.
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.
H.323 is a system specification that describes the use of several ITU-T and IETF protocols. The H.323 standard consists of following components and protocols:
* Call Signaling : H.225
* Media Control : 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.
* Audio Codecs : G.711, G.722, G.723, G.728, G.729
* Video Codecs : H.261, H.263
* Data Sharing : T.120
* Media Transport : RTP which is used for sending or receiving multimedia and RTCP for quality feedback.
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 utilizes various IETF Request for Comments (RFCs) for media transport and media packetization, including the Real-time Transport Protocol (RTP).