Internet telephony is the transport of telephone calls over the Internet. The process of supporting voice calls over the Internet using the Internet communications protocol IP, is known as Voice over IP (VoIP). The first business customer implementations of VoIP were long distance calls over an IP WAN as an alternative to traditional PSTN trunk facilities. Network gateway servers had converted PBX TDM/PCM signals into IP format for transport through a router network. VoIP offered no new features or functions, merely an alternative means of transport. VoIP was not used for station-to-station on-premises calls, and IP control signaling was not used for call set-up or feature activation. As the new technology evolved to include PBX system desktop call control signaling and communications over an IP network infrastructure, ToIP gradually started to replace VoIP. VoIP is still the most commonly used term, although ToIP is being used more to describe the workings of an IP-PBX system.
There are several sets of VoIP communications protocols that can be used by an IP-PBX system for call control signaling and communications transmission. Circuit switched PBXs were based on proprietary call control signaling to the digital desktops and used TDM/PCM standards for transmission across internal switching networks and to network with other communications systems over PSTN trunk carrier facilities. It was the intention of the early IP-PBX system developers to use open standard communications protocols over packet switched networks as a counter to the closed, proprietary nature of circuit switched PBXs. The communications protocol of choice used by most first-generation IP-PBX system designers was the ITU-T H.323 series of protocols. A competing standards body, the IEFT, developed Session Initiation Protocol (SIP). Many IP-PBX manufacturers are planning to offer SIP versions of their current systems based on H.323, but the first product offerings will not be commercially available until later this year. SIP may offer several advantages over the ITU-T’s recommendation, but the momentum of H.323 will delay saturation of the IEFT’s solution to VoIP communications systems for several years according to the major IP-PBX manufacturers.
ITU-T H.323
ITU-T H.323 is a set of protocols for voice, video, and data conferencing over packet switched networks such as Ethernet LANs and the Internet that do not provide a guaranteed QoS. The H.323 protocol stack is designed to operate above the transport layer of the underlying network. H.323 uses the IP for internetwork conferencing.
H.323 was originally developed as one of several videoconferencing recommendations issued by the ITU-T. The H.323 standard is designed to allow clients on H.323 networks to communicate with clients on other videoconferencing networks. The first version of H.323 was issued in 1996 and designed for use with Ethernet LANs. H.323 Version 1 borrowed much of its multimedia conferencing aspects from other H.32x series recommendations. H.323 is part of a larger series of communications standards that enable videoconferencing across a range of networks. This series also includes H.320 and H.324, which address ISDN and PSTN communications, respectively.
H.323 is known as a broad and flexible recommendation. Although H.323 specifies protocols for real-time point-to-point voice communication between two terminals on a packet switched network, it also includes support of internetwork multipoint conferencing among terminals that support not only audio (voice) but also video and data communications.
H.323 recommendations can be summarized as followed.
Point-to-Point and Multipoint Conferencing Support
H.323 conferences may be set up between two or more clients without any specialized multipoint control software or hardware. If an MCU is used, H.323 supports a flexible topology for multipoint conferences. A multipoint conference can be centralized so that new participants can join all the others in the conference. A multipoint conference may also be decentralized so that new participants can elect to join one or more participants, but not all participants, in the conference. The centralized approach is a star topology; the decentralized one is a mesh topology.
Internetwork Interoperability
H.323 clients are interoperable with clients on circuit switched networks such as those based on recommendations H.320 (ISDN), H.321 (ATM), and H.324 (PSTN/Wireless). For example, it is possible to call from an H.323 client to a regular telephone on a PSTN. At the corporate level, this internetworking capability allows enterprises to migrate voice and video from existing networks to their own data networks.
Heterogeneous Client Capabilities
An H.323 client must support audio communication. Support of video and data communications is optional. During call set-up, capabilities are exchanged and communication established based on the lowest common denominator.
Audio and Video Codecs
H.323 specifies a required audio and video codec, but there is no restriction on the use of other codecs. Clients are allowed to decide which codec to use.
Management and Accounting Support
H.323 calls can be restricted on a network based on the number of calls already in progress, bandwidth limitations, or time restrictions. Policy management guidelines are used for H.323 traffic control. H.323 also provides accounting facilities that can be used for billing purposes.
Supplementary Services
Recommendation H.323 provides a basic framework for the development of application services, similar to the call processing features typically supported by a PBX system. This effort began with H.323 Version 2, which standardized a few services with recommendation H.450, including call transfer and call forwarding.
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