The latency of the network will be the same from any input to output and vice versa. that’s based on pinging for worst case scenario. that’s post adc and pre dacs, you plug a laptop in the switch with dante controller application, and it will tell you what this latency is, along with many infosNot sure about the accuracy of this. May be I misunderstood but last time I looked at Dante specs, the latency "handshake" was designed to manage transceiver processing latency (not network latency). If a transceiver at the other end cannot process packets faster than every X ms (including buffering), there is no reason to send packets faster than that. So that is the minimum latency requirement as far as that transceiver is concerned. This is hardwired into the transceiver (which knows what latency happens inside its processing) and is the same on any network (topology). The sender could decide to send packets at a slower speed than that.
If there is a peer-peer direct connection between them then that is the latency for transmission.
However, if it is on a shared IP network, then I am not sure how the network latency is automatically figured out. PTP is just for clock synchronization as far as I understand it.
Then there is the issue of variable latency in a shared IP network unless the network is purpose-built with QoS guarantees for latency.
I don't see this protocol as a simple plug-and-play for a consumer application except for point to pont (or you have network engineering expertise) in which case an USB will do just as well without the transceiver complexity.
In studio settings and local or wide area distribution, absolutely.
Having said that, I don't think the latency is an issue with most simple topologies on private networks but it could make a difference in the ability to do live recording/playback from multiple sources (unless the latencies are as low as the typical audio interfaces).