Due to packet loss and random delay problems in the Internet, the transmission of real-time voice over such an environment is a complex and challenging issue. Many researchers have been devoted to this active research area. In the present paper, we develop a voice-delivery scheme to tackle the above problems. To overcome jitter and packet loss, an adaptive voice synchronization scheme is constructed in a feedback configuration. This scheme consists of the two following mechanisms, a delay and loss measurement mechanism and a QoS adjustment mechanism. Clearly, voice is a crucial medium for networked multimedia systems. The proposed voice tool is extremely valuable in supporting applications such as computer supported cooperative work (CSCW), Internet telephony, or multimedia instruction on demand (MID). Therefore, the results obtained in the present work are of benefit to many networked multimedia systems.
關聯:
Communication Technology Proceedings, 2000. WCC - ICCT 2000. International Conference on Vol.1. pp.628-632