NETW320 – Converged Networks with Lab
Lab #7 Title: CODEC Selection for a WAN
A codec is a device capable of performing encoding and decoding on a digital signal. Each codec provides a different level of speech quality. The reason for this is that codecs use different types of compression techniques in order to require less bandwidth. The more the compression, the less bandwidth you will require. However, this will ultimately be at the cost of sound quality, as high-compression/low-bandwidth algorithms will not have the same voice quality as low-compression/high-bandwidth algorithms. The following table shows three standard codec types along with their corresponding Coder Type, Bit Rate, Frame Size Delay, and Look Ahead Delay. These particular three were chosen because these are the ones we will use in this lab, but there are others. ITU Recommended Delay Values/G.114
Frame Size Delay (milliseconds)
Look Ahead Delay (milliseconds)
A common scale used to determine the quality of sound produced by various codecs is the mean opinion score (MOS). This standard is based on a scale from 1 to 5, where 1 is very poor and 5 is excellent. Here is a common table used to rate values of MOS.
No Meaning Understood
In this lab, you will build a wide-area network that will consist of several LANs spread across the country and connected via the Internet with two types of traffic: data and voice. You will configure three scenarios where the data and voice traffic generated will be held constant. The only parameter that you will alter will be the type of codec used. In the first scenario, the voice traffic will use G.711 (PCM); in the second scenario, the voice traffic will use G.729 (CS-ACELP); and the third scenario, the voice traffic will use G.723.1 (ACELP). By holding the traffic generated for the data and voice users constant, and only changing the codec values, we will be able to see how the varying bit rate of the specific codecs used will affect the end-to-end delay for the data traffic, the end-to-end delay for the voice traffic, and the packet delay variation for the voice traffic for a wide-area environment. Motivation
Deciding on what codec to use is a tradeoff between bandwidth and speech quality; the more bandwidth the codec uses, the better the speech quality will be. Here are the codecs we will use, along with their corresponding MOS values. Codes
However, this is not the only factor that must be considered. Delay will greatly affect the quality of voice traffic because it is real-time traffic. Words and responses to words must be received in a coherent and timely manner for the voice conversation to be tolerated by the users. Here is a table with ITU-T recommendations for the overall one-way delay times for voice traffic. ITU-T Recommend Acceptable One-Way Overall Delay Times/G.114 Range in Seconds
Range in Milliseconds
0 to 0.150
0 to 150
Acceptable for most user applications.
0.150 to 0.400
150 to 400
Acceptable, provided that administrators are aware of the transmission time and the impact it has on the transmission quality of user applications. Above 0.400
Unacceptable for general network planning purposes: it is recognized that in some exceptional cases this limit is exceeded.
The table shown in the introduction lists a Frame Size Delay and a Look Ahead Delay. The frame size refers to the time it takes the sender to transmit a frame. For complex compression algorithms that reduce bandwidth, this delay can...
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