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The variation of packet arrival time is one of the problems to be solved in realizing the real-time voice communication on asynchronous networks such as Internet. Although the variation can be absorbed by means of a receiving buffer, the end-to-end delay increases in proportion to the buffer size. Therefore, the guideline for the receiving buffer's design which considers the tradeoff between voice quality and delay is needed. In this paper, the design of the receiving buffer by using the voice speed conversion technique is discussed. The buffer size should be equivalent to the time length of about 95-99% of the probability distribution function of the packet arrival intervals. However, the synchronized voice reproducing can be achieved with the preset delay, which is shorter than the corresponding time to the buffer size, by controlling the rate of voice speed conversion within the range of 50%-150% depending on the variation of the intervals. The proposed buffer design was implemented in the conventional non-real-time UNIX workstations (WSs) connected to Ethernet and its performance was measured. The average voice delay between WSs for the authors' scenario is 150^J180msec and the synchronization is achieved with the standard deviation of 13^-25msec. The proposed method is applied to VSAT satellite channel routers to realize the real-time voice communication between the WSs via the VSAT channel.

Synchronization of Voice Data on Asynchronous Networks and its Implementation for a VSAT Channel Router