ce of a number of VBR traffic sources at a network element (such as a router) can soon overload the buffer capabilities. Packets can also be discarded if they have been delayed. An IP packet has a Time-to-Live (TTL) stamp, which when decre- mented to 0 results in the packet being discarded (Peterson & Davie, 2000; Tanenbaum, 2003). This can actually be desirable because for real-time traffic the playout time at the receiving codec is likely to have passed. Packet loss can result from the protocols employed by the transmitting sources them- selves (Baldi & Ofek, 2000). For example, in the TCP protocol, if packets are lost because of congestion, the TCP control mechanism will supply a negative acknowledgment and a retransmission will follow. Acknowledgment and retransmission packets interact with current data packets, causing delay, loss, and increased network congestion. If the video is sent using UDP, it still has to interact with other traffic, which is likely to be TCP, and there- fore the same argument applies (Baldi & Ofek, 2000). The end-to-end behavior of the Internet has been extensively studied, and losses can be summarized as bursty and typically between 2 and 10 percent (Sanghi et al., 1993). Bolot, Crepin, and Vega-Garcia (1995) showed that the number of consecutively lost packets is low provided the network load is not high. Packet loss causes severe effects on video traffic, affecting overall Quality of Experience (QoE) (Woods & Siller, 2006). The type of distortion is dependent on the type of information that has been lost (i.e., does it belong to an inter- or intracoded frame?). Degradations include parts of the image missing, the image becoming blocky, the reconstruction not Chapter 3 Digital Video Transmission resembling the actual, and abrupt/jerky motion. Some works (e.g., Woods & Siller, 2006) have suggested the selective discarding of packets during times of congestion: B frames first followed by P and then finally I frames. Delay and Jitter Packet delay is the time required to transmit the u