IP语音包的自适应编码和封装算法的研究  

AN ADAPTIVE VOICE CODING AND PACKETING SCHEME FOR IP TELEPHONY

作　　者：黄永峰[1] 李星[1] 

机构地区：[1]清华大学电子工程系网络与人机通信研究所,北京100084

出　　处：《电子与信息学报》2002年第12期1829-1834,共6页Journal of Electronics & Information Technology

基　　金：中国博士后基金"自适应语音编码算法研究"

摘　　要：IP电话与传统电话相比语音质量较差,其中最主要的原因是因特网的带宽变化较大,导致丢包率较大。该文根据因特网带宽变化的特点提出了1种应用在IP电话网关中的语音自适应编码与封装策略,采用该策略的编码器能根据网络的带宽变化动态调节语音编码速率和语音包封装大小。据此,本文提出了4种算法:一种基于RTP协议语音包丢失率的计算算法、变速率编码算法、不同长度IP语音包的封装算法和根据丢包率来调整编码速率和封装的自适应算法。QoS of IP telephony is worse than that of circuit telephony because the available bandwidth of Internet varies then results in the loss of voice packets . This paper suggests an adaptive voice codec which can be applied in the IP telephony gateway, it can output various bit rate when Internet's bandwidth varies, the most advantage of the codec is that it can decrease the ratio of packet-losing and improve the QoS of voice. In the implementation of the voice coder, this paper brings forward four algorithms, which include an algorithm for computing the ratio of packet-losing based on real-times transport protocol, an algorithm for implementing a coder that outputs various bit rate, an algorithm for voice packeting, and an adaptive algorithm for encoding and packeting.

关 键 词：IP语音包 自适应编码 封装算法 丢包率 IP电话 分组语音传输技术 

分 类 号：TN916[电子电信—通信与信息系统] TN912.[电子电信—信息与通信工程]