The medium (wireless or ethernet) does not matter. What matters is that there is little network (IP packet) latency and small amounts of packet retries. The retry/error-correction features of the TCP/IP protocol are fine for data, because re-tries or resending data has no adverse effect. Not true for audio which is real-time. If packets are lost or delayed the audio will have dropouts. I suppose this can be alleviated to some extent, if not effectively, with packet buffering/management at the application level (above the IP protocol level). This is probably what sets apart products which work well or do not... how effective they are at buffering, if they do it at all.