I\'m in startup of designing a client/server audio system which can stream audio arbitrarily over a network. One central server pumps out an audio stream and x number of clients
Depending on the size and shape of the venue, getting everything to be in sync is the easy part, getting everything to sound correct is an art-form in itself, if possible at all. From the technical side, the most difficult part is finding out the delay from your synchronized timeline to actual sound output. Having identical hardware and low latency software framework (ASIO, JACK) certainly helps here, as does calibration. Either ahead of time or active. Otherwise it's just synchronizing the timeline with NTP and using a closed loop feedback to the audio pitch to synchronize the output to the agreed timeline.
The larger problem is that sound takes a considerable amount of time to propagate. 10m of difference in distance is already 30ms of delay - enough to screw up sound localization. Double that and you get into the annoying echo territory. Professional audio setups actually purposefully introduce delays, use a higher number of tweeters and play with reverberations to avoid a cacophony of echoes that wears the listener out.