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Network 320 Week 4 Wireless Network Traffic

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Wireless Network Traffic

DeVry University

Wireless Network Traffic More and more traffic is being moved to wireless networks of various types. Voice traffic in particular can provide substantial savings on a company’s telephone service. It allows companies to use an IP network to make phone calls instead of the traditional telephone companies' public switched telephone network (PSTN). Although cost savings is the biggest advantage to using wireless networks for voice traffic, there are still a few issues and limitations. When voice traffic travels over a wireless network, the packets must compete with occurrences that may affect the overall voice quality. The primary factors that determine voice quality are packet loss, latency and jitter. Packet loss occurs when packets sent over the network are not properly received by end stations, causing them to be discarded by the receiver. There are several reasons why packet loss can occur. Overloaded links, overload in the receiving device, excessive collisions in the wireless link, physical media errors due to interference and low link quality are among a few of the reasons why packet loss happens. Packet loss can and will degrade voice quality. Some protocols such as TCP will ask the receiver for retransmission or the sender automatically resend any segments that have not been acknowledged when packet loss occurs. Others solutions to eliminate packet loss are: reduce the number of packets transmitted and enable the end stations to react more quickly to packet loss. Excessive jitter can result from congestion on low bandwidth WAN links. High levels of jitter can cause large numbers of packets to be discarded by the jitter buffer in the receiving IP phone or gateway. This may result in severe degradation in call quality or large increases in delay. Solutions to eliminate jitter are increasing the jitter size buffer in the end station, enabling the adaptive jitter buffer operation. In the event there is large amounts of jitter, steps should be taken to eliminate the jitter at the source. The jitter buffer deliberately delays incoming packets and will also fix any out-of-order errors by looking at the sequence number in the RTP
Frame.
Latency is the time it takes the data packets to get from one designated point to another. In voice traffic it is described as the “mouth-to-ear” overall delay. Two way phone conversations can be quite sensitive to latency. Sometimes latency is measured by a sending packet’s that are returned, making it a round trip. Factors contributing to network latency are: * Propagation: the time it takes for a packet to travel between one place and another at the speed of light * Transmission: the medium itself introduces some delay. * Router and other processing: each gateway node takes time to examine and possibly change the header in a packet.
Changing the router settings can sometimes decrease latency. In conclusion, when voice traffic travels over a wireless network, collisions in the wireless link can reduce capacity and increase latency and jitter. This should be taken into consideration when subscribing users to each sector. Proper planning should include traffic planning that reduces the number of packets per second generated, thus reducing the overhead affects.

References
Online diagnostic tools for network managers problems:jitter. (n.d.). Retrieved from http://www.voiptroubleshooter.com/problems/jitter.html
Gottlieb, A. (2012, November 26). What can be done about wan packet loss and its impact on wan application performance. Retrieved from http://www.networkworld.com/community/blog/what-can-be-done-about-wan-packet-loss-and-its-impact-wan-application-performance

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