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以提高音频分组在 IP网络接收端的回放质量为目标 ,根据音频分组的回放原理 ,研究音频回放自适应算法 ,针对处理话音期内出现的延迟尖峰 ,提出了一种改进算法。根据实时传输 /控制协议(RTP/RTCP)框架 ,实现了基于 RTP的 Internet音频传输。仿真实验表明 ,改进算法可适时调整回放延迟 ,在网络负载较重时 ,能有效地减少分组丢失 /平均回放延迟开销
In order to improve the playback quality of the audio packets at the receiving end of the IP network, an adaptive audio replay algorithm is studied according to the playback principle of the audio packets. An improved algorithm is proposed to deal with the delay peaks appearing in the speech period. According to the Real-time Transport / Control Protocol (RTP / RTCP) framework, Internet-based RTP-based audio transmission is achieved. The simulation results show that the improved algorithm can adjust the playback delay timely and effectively reduce the packet loss / average playback delay overhead when the network load is heavy