> Perhaps the better thing to focus on is the frequency of underruns (or "xruns" as they say on Linux, which also covers overruns) for a specific sample rate and buffer size setting.

Right, that's what I meant. There will be some jitter depending on the scheduler, system load, network performance and traffic, and the quality of hardware/drivers; so more latency gives you headroom to absorb the jitter without underruns. The "ideal configurations" I was referring to were the low-traffic network and high-quality NIC. On a setup with more jitter, you might have to increase the buffer size (and thus latency) for reliable operation.

Mostly, I was just trying to contextualize the numbers for readers who aren't super familiar with low-latency audio networking. Sure, this project may not achieve the sub-1-ms roundtrip latencies that you can get with dedicated Dante hardware (like a Yamaha mixer and stagebox); but Dante can't do better than 8ms when one of the ends is a PC (although they were probably aiming for reliability on setups not aggressively tuned for minimum jitter.)