Abstract

The statistical nature of data traffic and the dynamic routing techniques employed in IP networks results in a varying network delay (jitter) experienced by the individual IP packets which form a VoIP flow. As a result voice packets generated at successive and periodic intervals at a source will typically be buffered at the receiver prior to playback in order to smooth out the jitter. However, the additional delay introduced by the playout buffer degrades the quality of service. Thus, the ability to forecast the jitter is an integral part of selecting an appropriate buffer size. The use of playout delay adaptation algorithms in order to determine suitable delays is introduced during the speech playout process in order to mask this delay variation.

We consider time series prediction methodologies for packet jitter and techniques to enhance the performance of the overall playout process. We also consider the application of non-intrusive speech quality estimation methodologies (such as ITU -T standard P563 are based) in the process of playout delay adaptation.