基于RTP协议的音频流同步控制策略及实现

The Policy and Implementation for Adaptive Synchronization in Audio-terminal Based on RTP

提出一种利用RTP协议所提供的时间戳字段在接收端计算出传输延迟和延迟抖动,根据延迟抖动值自适应地调整一个语音段的播放时刻从而提高语音播放的连续性的算法。最后,通过在局域网环境下进行模拟实验,得出实验数据。

This paper proposes an adaptively synchronized arithmetic applied to audio data receiver in Internet for purpose of achieving high playing-continuity.h is based on the timestamp field of RTP to calculate the transmit delay value and delayjitter value which determines the time at which the audio decoder writes the RTP packet from audio buffer，At last,to get experiment data through doing an experiment in LAN environment。