Adaptation for the retransmission in VoIP applications

Abstract

An adaptive retransmission scheme can be used for voice over Internet protocol services to provide a certain level of quality of service. However, parameters used for adaptive retransmission must be selected carefully due to unintended frame loss in the channel and in the playout buffer. In this paper, we evaluate the frame loss rate (FLR) in the output of playout buffer for the adaptive retransmission. This paper also proposes a parameter selection algorithm based on the evaluation. The proposed algorithm controls the probability of starting the retransmission process and the maximum number of retransmissions considering a bursty channel model and the loss in the playout buffer. The evaluation results show that the proposed scheme can select the proper parameters to provide a desired FLR in the output of the playout buffer. In addition, the measured mean opinion score-listening quality objective, which is an expected objective quality score, also shows that the proposed algorithm has a potential to provide a desired level of quality score.