>> Yes, that is what we are doing. However, due to calculation speed >> and queuing, this cannot happen immediately but only with a slight delay. >> Also, we need to support a frame size of 2.5 ms instead of 10ms. >> This should be still ok? >> > > I am not quite sure what are the requirements of your applications, > but the earlier email today from Ingemar might be relevant/helpful to > this question. Let's assume a distributed ensemble performance over the Internet using the anticipated IETF audio codec. Distributed ensemble performance require an acoustic one-way delay of 25 ms instead of maybe 150 ms in case of VoIP. Thus, also the frame size must be small, say 5% of the RTT. Also, the coding rate would be 128kbps for high quality music. Then, we send every 2.5ms a RTP packet having a size of 320bytes plus headers. Thus, maybe instead of 10 ms, a TFRC-SP Min Interval of min(10ms,RTT/20) would be reasonable? With best regards, Christian
