“The fraction of lost packets increases as the traffic intensity increases. Therefore, performance at a node is often measured not only in terms of delay, but also in terms of the probability of packet loss…a lost packet may be retransmitted on an end-to-end basis in order to ensure that all data are eventually transferred from source to destination.” ^[7] The amount of packet loss that is acceptable depends on the type of data being sent. For example, for Voice over IP traffic, the only effect seen due to the occasional dropped packet is jitter, and therefore “[m]issing one or two packets every now and then will not affect the quality of the conversation. Losses between 5% and 10% of the total packet stream will affect the quality significantly.”^[8] On the other hand, when transmitting a text document or web page, a single dropped packet could result in losing part of the file, which is where the aforementioned packet retransmission schemes are used.

When given a situation where the amount of content due to be pushed through a connection is growing at a rate greater than it is possible to push through that connection, also known as a bottleneck, then there is no other solution than to drop packets.^[9] The TCP protocol is designed with a slow-start connection strategy so that excessive packet loss will cause the sender to throttle back and stop flooding the bottleneck point with data (using perceived packet loss as feedback to discover congestion). ^[10] The data packets will be transmitted over a longer duration.

There are many methods used for determining which packets to drop. Most basic networking equipment will use FIFO queuing for packets waiting to go through the bottleneck and they will drop the packet if the queue is full at the time the packet is received. This type of packet dropping is calledtail drop. However, droppi