摘要
现存网络传输系统具有异构和时变的特点,提出了一种基于CCITT的G.726建议的 自适应压缩率的实时语音网络传输系统的实现方法。发送方根据RTCP的接收者报告判断其网 络状况,把用户分为4级,然后为不同级的用户提供相应的压缩率数据,在一定程度上减少 了包丢失率和抖动,从而保证了实时语音传输系统的QoS.
Since now network transport system has heterogenous and time-varying characteristics, this paper presents an adaptive compression ratio real-time t ransport scheme based on recommendation G.726 of CCITT.After determining the net work condition of users by RTCP receiver reports and classifying them as 4 types , it sends corresponding compression ratio audio to the users. This mechanism re duces packet loss and jitter and provides QoS for real-time speech transport sys tem.
出处
《计算机工程》
CAS
CSCD
北大核心
2004年第1期107-109,共3页
Computer Engineering