Abstract

The quality of audio in IP telephony is significantly influenced by various factors, including type of encoder, delay, delay variation, rate and distribution of packet loss, and type of error concealment. Hence, the performance of IP telephony systems 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. We have noticed that the correlation of loss and delay is not linear, but stronger correlation is observed as the delay approaches certain thresholds. We have made a number of new observations on various delay thresholds that are significant for loss prediction for adaptive audio transmission over IP networks. We also have made new observations to assess the audio quality of PCM μ-law and G.728 codecs under different loss and delay conditions. The paper provides a number of recommendations for implementing efficient adaptive FEC mechanisms based on our measurement observations and analysis.

Full Text
Published version (Free)

Talk to us

Join us for a 30 min session where you can share your feedback and ask us any queries you have

Schedule a call