Information about Packet Loss

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

Causes

Packet loss can be caused by a number of factors, including signal degradation over the network medium, oversaturated network links, corrupted packets rejected in-transit, faulty networking hardware, maligned system drivers or network applications, or normal routing routines.

Effects

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. [1]

Packet recovery

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 designed to handle this type of packet loss.

Acceptable Packet Loss

Perhaps the way to garner this number and still maintain a semblance of objectivity is to offer target numbers for different classes of service or types of architectures

For example, certain routers allow you to "prioritize" your traffic according to content [putting the different types into different queues]. For such an architecture, you might specify that the highest priority for a critical service, packet types should pass with less than 1% packet loss. Lower priority packet types might pass with less than 5% and then 10% for the lowest of priority of services.

See also

References and notes

Footnotes

  1. Packet loss or latency at intermediate hops. (HTTP). Retrieved on 2007-02-25.
  2. Packet loss practically represented (German) (HTTP). Retrieved on 2007-03-12.
  3. Excerpts from ietf.ippm: 6-Sep-96 Re: Thoughts on metrics Jamshid Mahdavi@psc.edu (1628) (HTTP). Retrieved on 2007-09-28.

References

1. ^ Packet loss or latency at intermediate hops. (HTTP). Retrieved on 2007-02-25.

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as a college campus, industrial complex, or a military base. A CAN, may be considered a type of MAN (metropolitan area network), but is generally limited to an area that is smaller than a typical MAN.
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Jitter is an unwanted variation of one or more signal characteristics in electronics and telecommunications. Jitter may be seen in characteristics such as the interval between successive pulses, or the amplitude, frequency, or phase of successive cycles.
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This article relies largely or entirely upon a .
Please help [ improve this article] by introducing appropriate of additional sources. ()
This article has been tagged since October 2007.
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Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other IP-based network.
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A multiplayer game is a video game in which more than one person can play the same game at the same time. Unlike most other games, computer and video games are often single-player activities because the computing power exists to create artificial opponents.
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A videoconference (also known as a videoteleconference) is a set of interactive telecommunication technologies which allow two or more locations to interact via two-way video and audio transmissions simultaneously.
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The Transmission Control Protocol (TCP) is one of the core protocols of the Internet protocol suite. TCP provides reliable, in-order delivery of a stream of bytes, making it suitable for applications like file transfer and e-mail.
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throughput is the amount of digital data per time unit that is delivered over a physical or logical link, or that is passing through a certain network node. For example, it may be the amount of data that is delivered to a certain network terminal or host computer, or between two
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Sliding Window Protocol is a bi-directional data transmission protocol in the data link layer (OSI model). It is used to keep a record of the packet sequences sent and their respective acknowledgements received by both the users.
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User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages sometimes known as datagrams (using Datagram Sockets) to one another.
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throughput is the amount of digital data per time unit that is delivered over a physical or logical link, or that is passing through a certain network node. For example, it may be the amount of data that is delivered to a certain network terminal or host computer, or between two
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In computer networks (including wireless networks), goodput is the application level throughput, i.e. the number of useful bits per unit of time forwarded by the network from a certain source address to a certain destination, excluding protocol overhead, and excluding retransmitted
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