ry or control capabilities. Internet Fig. 3.19 Simplistic view of end-to-end communication. Video Transport over the Internet While TCP offers a reliable delivery service, UDP is more suitable and mostly used for transmission of real-time media such as video and audio because there are no retrans- missions to increase the round-trip delay in the packets. The Real-Time Transport Proto- col (RTP) (Schulzrinne et al., 1996) can be used on top of UDP to provide a more robust end-to-end delivery service. RTP services include payload identification, sequence num- bering, and timestamping. The payload identifier specifies payload type as well as the encoding/compression scheme used, enabling the receiving application to interpret the payload. The sequence number allows the ordering and detection of lost packets. The timestamp helps the receiver to play the video at the correct rate. The Internet layer manages transfer across multiple networks. Information moves from one network to another through routers and gateways, where networks are identified by their unique IP (Internet Protocol) addresses (Schulzrinne et al., 1996; Peterson & Davie, 2000). The behavior of packets on the Internet is widely documented (Baldi & Ofek, 2000), and the general conclusion is that the Internet is not an ideal medium for the transport of real- time traffic. The current Internet offers a single-best-effort level of service, where all packets are independently routed to the next node without guarantee of delivery and might be subject to delay or jitter. Packet Loss Packet loss is when packets go missing in a network. The principal cause of packet loss is when sources overload network elements. As a network becomes congested, forwarding nodes cannot process the data at the rate at which it arrives, and it is therefore discarded by the overflowing buffers. Other loss due to equipment failure or bit error caused by inter- ference can occur but is in general rare. Bursty traffic is particularly adept at causing buffer overflow, and the convergen