ed media streams. Protocols such as RTP (which attaches a header to UDP packets that can be used to detect packet loss), RTCP (which guarantees quality of service) and RSVP (which establishes service classes) are bringing streaming data to TCP/IP, as supporting router infrastructure is being deployed. Overview of RTP In January 1996, the IETF's Audio-Video Transport Working Group published RFC 1889, entitled, "RTP: A Transport Protocol for Real-Time Applications" and with it, a companion standard, RFC 1890, "RTP Profile for Audio and Video Conferences with Minimal Control." RTP and RTCP are part of the IETF's IIS (Internet Integrated Services) model, along with RSVP (described below). A second version of RTP and RTCP became available shortly thereafter. RTP bridges the gap between the lower- level UDP transport and a Real-Time multimedia application. The RTP is a streams-oriented protocol. It provides end-to-end network transport services for Real-Time multimedia applications over IP based networks. It can be used in multicast (multipoint) or unicast (point-to-point) applications. How RTP Works RTP applications can compensate, to a degree, for the instability of UDP/IP networks by appending a 10-byte header onto each UDP packet. The header is designed with Real-Time transport in mind. It contains a time stamp and a sequence number to keep multimedia packets in the proper order during transmission. The time stamp is used to pace frame display and synchronize multiple streams (e.g., voice, video and Video Communications: The Whole Picture Heade Heade RTP Payload Figure 10-4. IP Packet Containing RTP Data data) while the sequence number is used to reassemble the data stream at the receiving end. The header also carries information about the information payload (in other words, the types of compression used to generate the media stream.) Unlike the point-to-point TCP, only end-points (and not network devices) are required to support RTP. An interactive audiovisual application in an RTP-based conference must buffer da