The transmission of voice packets over the Internet is an active field of research. In the present paper, we develop an adaptive scheme in dealing with this issue. As it is well known, the present Internet environment may present packet delay, jitter and packet loss. These phenomena are hostile to continuous medium transmission. To overcome the above difficulties, in our design, an adaptive voice synchronization scheme is constructed in a feedback manner. This scheme consists of the following mechanisms: (1) a packet QoS probing mechanism, (2) a redundant packet sending mechanism, and (3 ) an adaptive playback mechanism. With the integration of these mechanisms, the quality of voice transmission is improved. Supporting of voice transmission is a crucial issue to networked multimedia systems. Therefore, the results obtained in the present work are of benefits to applications such as computer supported cooperative work(CSCW) and multimedia on demand (MOD).
一九九八年第十六屆國際電信研討會論文集第二冊=16th (1998) International Telecommunication Symposium=Proceedings Volume (II)，頁195-202