게이트키퍼 중재형 호 기반 플레이아웃 알고리즘의 제안 및 성능 비교

Abstract

Recent works for VoIP phones have been focused on both solving the problem of synchronous playout of audio packets and achieving toll-quality audio in the face of variable network delays. In this paper we propose a per-call basis playout algorithm that uses same playout delays during a given call adapted according to the basic network delay value received from the gatekeeper at the initial call stage. Here, the playout delay value consists of the basic network delay and additional waiting delay. The additional waiting delay value is experimentally determined in order to diminish the effects of network-induced jitter to the level of a lateness loss percentage of below 5% even the underlying network has high jitter. We obtained 100 log files of experimentally-measured network delays between several different Internet sites. And we evaluate the playout algorithms using the playout emulator which can calculate the lateness loss rates and average playout delays for different log files. Our results indicate that the per-call basis algorithm can achieve a lower rates of lateness lost packets while at same time keeping playout delays to be quite tolerable in human conversations. With our algorithm, one can achieve better VoIP call quality specially when jitter is high