An improved speech playout buffering algorithm based on a new version of E-model in VoIP

In Voice over IP (VoIP) applications, playout buffering algorithms based on a tradeoff between delay and loss can be used to alleviate the effect of jitter. In the past, the aim of most buffer algorithms was not to improve the perceived speech quality directly, but to reduce the buffer delay and the packet loss rate. Then a quality-driven approach was proposed, which uses a quality model to control the playout buffer in order to maximize the Mean Opinion Score (MOS) in terms of delay and loss. However, this method can only be used in random loss condition. Thus an improved quality-driven approach is proposed in this paper to deal with bursty loss condition. For this purpose, we make use of the latest version of ITU-T E-Model to incorporate the effects of loss burstiness in the perceived quality. The experimental results show that the proposed method can achieve an "optimum" perceived speech quality, and reduce the bursty loss simultaneously.