H.323

H.323 is part of a broader family of standards developed by ITU describing how audio, video and data communications occur between terminals, network equipment and services on IP networks that do not provide guaranteed QoS--most notably, the Internet. H.323 defines four major components for networked communications:

• Terminals, which are LAN client end points that provide two-way communications.

• Gateways, which provide for real-time, two-way communications between H.323 terminals on an IP network and other ITU terminals on a switched-based network or to another H.323 gateway, functioning as a translator.

• Gatekeepers, which act as central points for calls in their zones and provide services to registered end points.

• MCUs (multipoint control units), which are end points that provide the capability for three or more terminals and gateways to participate in a multipoint conference.

Session Initiation Protocol

SIP, an application-layer control protocol proposed by the IETF (RFC 2543), overcomes H.323's shortcomings. It can establish, modify and terminate multimedia sessions or calls, such as conferences, distance learning, Internet telephony and similar applications. SIP enables VoIP gateways, client end points, PBXes, and other communications systems and devices to communicate with each other. It provides a lightweight protocol that enables scalable call control and a platform for applications.

SIP handles user location, user capabilities, user availability, call setup and handling. It supports name mapping and redirection services that facilitate the implementation of ISDN and Intelligent Network telephony services that allow for mobility. It can also initiate multiparty calls using an MCU or fully meshed interconnection instead of multicast. SIP works in conjunction with RSVP (Resource Reservation Protocol), RTP (Real-Time Transport Protocol), RTSP (Real-Time Streaming Protocol), SAP (Session Announcement Protocol) and SDP (Session Description Protocol). Unlike H.323, SIP has little overhead, as this protocol reuses most of the header fields, encoding rules, error codes and authentication mechanisms of HTTP.

Media Gateway Control Protocol

MGCP is a standard proposed by the IETF (RFC 2705) for the conversion of audio signals carried on PSTN to data packets that travel over the Internet. MGCP enables MGCs (media gateway controllers) and media gateways to communicate. It combines IPDC (IP Device Control) and SGCP (Simple Gateway Control Protocol). It assumes an architecture in which the call-control "intelligence" is outside the gateways and handled by external call-control elements, or call agents. It is a master/slave protocol, wherein gateways execute commands sent to them by the call agent. MGCP is losing momentum as a true VoIP standard with the emergence of the Megaco/H.248 standard.

Megaco/H.248

This new protocol, jointly proposed by the ITU-T Study Group 16 and Megaco work group of IETF, lets an MGC control media gateways. As a successor to MGCP, it adds peer-to-peer interoperability capabilities and provides a means of control appropriate for IP telephone devices operating in a master/slave relationship, similar to MGCP. Megaco/H.248 decomposes the H.323 gateway function into subcomponents and specifies the protocols used by each component for communication. Megaco/H.248 will allow low-cost gateway devices to interface with signaling systems found in circuit-switched networks. H.248 leverages on the existing PSTN, making its implementation cheaper and faster for network operators.

--Rajeev Vijayan