Consider one of the UDP segments generated by our Internet phone applicatio.  The UDP segment is encapsulated in an IP datagram.  As the datagram wanders through the network, it passes through buffers in the routers in order to access outbound links.  It is possible that one or more of the buffers in the route from sender to receiver is full and cannot admit the IP datagram.  The IP datagram is discarded, never to arrive at the receiving application.

 

Loss could be eliminated by sending packets the packets oer TCP rather than over UDP.  Recall that TCP retransmits packets that do not arrive at the destination.  Retransmission mechanisms are often considered unacceptable for interacitve real-time audio applications such as Internet phone, because they increase end-to-end delay.

 

End-to-end delay is the accumulation of transmission, porcessing, and queuing delays in routers; propagation delays in the links; and end-system processing delays.

 

A crucial component of end-to-end delay is the random queuing delays in the routers.  Boecause of these varying delays within the network, the time from when a packet is generated at the source until it is received at the receiver can fluctuate from packet to packet.  This phenomenon is called jitter.

 

Jitter can often be removed by using sequence numbers, timestamps, and a playout delay.