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Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination.

Packet loss is distinguished as one of the three main error types encountered in digital communications; the other two being bit error and spurious packets caused due to noise.

Packet loss can be caused by a number of factors including signal degradation over the network medium due to multi-path fading, packet drop because of channel congestion, corrupted packets rejected in-transit, faulty networking hardware, faulty network drivers or normal routing routines (such as DSR in ad-hoc networks).

When caused by network problems, lost or dropped packets can result in highly noticeable performance issues or jitter with streaming technologies, voice over IP, online gaming and videoconferencing, and will affect all other network applications to a degree. However, it is important to note that packet loss does not always indicate a problem. If the latency and the packet loss at the destination hop are acceptable then the hops prior to that one don't matter.

Some network transport protocols such as TCP provide for reliable delivery of packets. In the event of packet loss, the receiver asks for retransmission or the sender automatically resends any segments that have not been acknowledged. Although TCP can recover from packet loss, retransmitting missing packets causes the throughput of the connection to decrease. This drop in throughput is due to the sliding window protocols used for acknowledgment of received packets. In certain variants of TCP, if a transmitted packet is lost, it will be re-sent along with every packet that had been sent after it. This retransmission causes the overall throughput of the connection to drop.

Protocols such as UDP provide no recovery for lost packets. Applications that use UDP are expected to define their own mechanisms for handling packet loss.

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