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John, I think that you are mixing the latency/jitter requirements with the synchronization requirements. In case a) the 15mS latency requirement presented in the tutorial seems reasonable - although the use of headphones in this case makes the task an order of magnitude easier than the tutorial case. In case b) there is no definitive latency requirement - certainly 1/2 second would be reasonable. Therefore a >100mS jitter buffer could be expected at the receivers making the network latency/jitter constraint very easy indeed. However, there is a synchronization requirement (as there is in case a). The synchronization of multi-channel audio requires that all receivers have a frequency lock mechanism with a precision of <10uS (video is much less stringent). The use of Ethernet (as opposed to ATM) makes this job more difficult but a proper implementation of a heartbeat mechanism with hysteresis and a settling time of several seconds should be able to achieve <1uS precision. Hugh. John Grant wrote: At 08:50 12/04/2005 +0100, Arthur Marris wrote: [snip]iii) You need to come up with a quantitative requirement for jitter and relative latency and justify it. I saw the figure of 10us mentioned on the Yahoo mailing list. This is ridiculously tight. An 802.3 voter who is experienced in VOIP pointed out to me that even 1ms is too tight when you consider that sound only travels one foot in a millisecond. Once you have this requirement nailed a lot else will fall into place.However, the requirements for (a) performers listening over headphones to a mix that includes their own voice and (b) forming a stereo or surround image from speakers that are individually (and independently) connected to the network are more stringent than those for voice telephony. John Grant |