nostr:npub1fy0nvfj5gpn5wqcnmfjnurx3wpnaqdah0j75dhmnrv3m5qvf805skpmmfu between the phone and VoIP.ms, there's no decompression and recompression occurring most likely, probably not even on the pbx since as far as it's concerned it's SIP->SIP. It's after VoIP.ms when that diciness of sound quality kicks in for compression nonsense at least. But definitely a solid test as well of that latency stability!