An Adaptive Jitter Buffering Algorithm for Voice over IP Networks

The continuous playout of voice packets in the presence of variable network delays is often achieved by buffering the received voice packets for sufficient time. Basic jitter buffering algorithms can work well only when the delay does not spike in the IP networks. In this work, an adaptive jitter buffering algorithm based on the detecting and studying the spike status of the networks, is presented to promote the quality of voice communication. It timely adjusts the minimal and maximal depth of buffer queue according to the control target of end-to-end delay and packet loss rate. The algorithm can much more easily achieve the continuous playout because it plays voice packet at a fixed inter-play time in the most time of a talk-spurt. The control target of packet loss rate can be extended to 20%. However, the basic algorithms can only bear 5%～10% of the packet loss rate. Perceptual evaluation of speech quality (PESQ) is applied to assess the speech quality in the simulation. It is shown that the algorithm can obviously promote the quality of voice communication in IP networks with spike delay. The practical application in voice gateway can also prove the effects of voice quality promotion.