An adaptive voice coding and packeting scheme for ip telephony
IP语音包的自适应编码和封装算法的研究

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.