An algorithm for playout of packet voice based on adaptive adjustment of talkspurt silence periods

In a typical real-time voice application, voice packets are produced at deterministically-spaced time intervals. In the network they encounter a variable amount of delay that changes the deterministic time intervals. A receiving host can employ a buffer to delay the playout of the voice packets in order to reconstruct the original timing. Adaptive techniques can perform continuous estimations of the network delays and dynamically adjust the buffering delay at the beginning of each talkspurt. Such adjustments are usually undetectable by the human listener. This research develops a new, adaptive "gap-based" algorithm that can be tuned for both end-to-end delay and packet loss to satisfy a user-desired tolerance. This new gap based algorithm adapts the buffering delay based on historical information of arrival and playout times of received voice packers in the previous talkspurt. A simulation study shows that the new gap based algorithm can reduce delay by 10% when compared with existing methods.