hat LAN. It has a capability provided by a voice or audio codec such as a G.711 or G.728 codec. It will also provide a signaling function for VoIP circuit setup, maintain, and take-down. A VoIP terminal optionally can support video and data streams including compression and decompression of these streams. Media streams are carried on RTP (real-time protocol) or RTCP (real-time control protocol). RTP deals with media content while RTCP works with the signaling functions 15.5 MEDIA GATEWAY CONTROLLER AND ITS PROTOCOLS of status and control. This protocol information is embedded in UDP, which is reliably transported by TCP. Other VoIP entities are gateways and there is a gatekeeper that is optional. The leading issue in VoIP implementation is guaranteed quality of service (QoS). H.323 is based on RTP (real-time protocol) which is comparatively new. RTP-compliant equipment includes control mechanisms for synchronizing different traffic streams. On the other side of the coin, RTP has no mechanisms for ensuring on-time delivery of traffic signals or for recovering lost packets. It does not address the QoS issue related to guaranteed bit rate availability for specific applications. The IEC [19] reports that there is a draft signaling proposal to strengthen the Internet's ability to handle real-time traffic reliably. This would dedicate end-to-end transport paths for specific sessions much like the circuit-switched PSTN does. This is the resource reservation protocol (RSVP). It will be implemented in routers to establish and maintain requested transmission paths and QoS levels. 15.5.2 Session Initiation Protocol (SIP) SIP is based on RFC 2543 [2] and is an application layer signaling protocol. It deals with interactive multimedia communication sessions between end-users, called user agents. It defines their initiation, modification, and termination. SIP calls may be terminal-to terminal, or they may require a server to intercede. If a server is to be involved, it is only required to locate the called party. For interwork