Performance Of The Video In Wired Network by

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Performance of the Video in Wired Network

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Performance of the Video in Wired Network

Part 1

Introduction

Several features are added to IPv6 specification in addition to 128-bits addressing as the IPv6 specification made its way through the IETF committee process. This includes, Levels of assured service, enhance security, and improved reliability(Floyd & Fall, 1999).

Quality of Service (QOS) is an important term and an emerging feature of modern networks. IPv4 networks typically give each and every packet a "best level of effort" service, even if the content of every packet isn't really important or time-sensitive data.

An IPv4-based system has no way to differentiate between data payloads that are time sensitive, such as streaming video or audio, and those that aren't time-sensitive, such as status reports and file transfer. Streaming audio and video application are very sensitive to delay of a few packets - lips move without sound or picture break up - but IPv4 has no way to prevent those problems(Bansal, 2001).

If a packet is lost in transit, TCP recognises the loss and requests a retransmission, but only after an inevitable delay. The single delayed TCP packet is probably part of a much larger packet of audio or video data, so the entire big packet is delayed and probably thrown out because the smallest part didn't arrive on time(Chockalingam, 1999).

IPv6 provides a way for applications to request handling without delay throughout the WAN. The term often used to describe this is low latency. Streaming audio and video requires low latency through high priority. To prevent a break down in the scheme, the various application can share connection via priority level.

Level 0 - No specify priority

Level 1 - Background traffic (news)

Level 2 - Unattended data transfer (email)

Level 3 - Reserved

Level 4 - Attended bulk transfer (FTP)

Level 5 - Reserved

Level 6 - Interactive traffic (Telnet, Windowing)

Level 7 - Control traffic (routing, network management)

Packet fragmentation is a major source of packet delays, or high latency, under IPv4. Each device attached to a network has a payload data limit set inside the Ethernet packet. If program is generating streaming data, such as video or audio, the data stream will be split up into a string of packets, each carrying the maximum payload.

With different devices, these payload sizes are set differently and it is possible that between the originating source and the destination, some transmission path, particularly an asynchronous transfer mode (ATM) link, will have a smaller payload size. (Kohler, 2006)

So the ATM equipment will chop the already fragmented stream of data into even smaller pieces. An ATM switch could divide the data carried in one single Ethernet packet into 20 ATM cells. Somewhere in the dividing and rebuilding or all the data, it is likely that a cell, and there a packet will be dropped or delayed. (Chiu, 1989)

IPv6 uses a more sophisticated approach to handle data from programs requesting priority handling. The originating device will queries the destination in order to determine the maximum size of the payload that call be handled across the ...
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