Abstract

The quality of the audio in IP telephony is significantly influenced by various factors, such as type of encoder, distance, delay variation, rate and distribution of packet loss, type of error concealment, and others. Hence, the performance of any IP telephony system is highly dependent on understanding the contribution of these factors to audio quality, and their impact on adaptive transport mechanisms such as error and buffer control. We conducted a large-scale audio transmission experiment over the Internet in a 12-month period in order to evaluate the effects and the correlation of such parameters on audio transmission over IP. As a part of studying and analyzing the collected data, we have made a number of new observations on the correlation of loss and RTT (round trip time) variation, and various RTT measurement mechanisms that are significant for adaptive audio transmission over IP networks.

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