I'm wondering what the latency difference is between P2P Jacktrip, or Jacktrip Hub + 3 clients, and a SIP call, or conference call with 3 clients, (audio only) RTP would be riding the same circuits.
The two seem similar in that both are riding UDP. On the SIP side, I could use opus at 48k sample rate and stereo for my RTP stream, similar to Jacktrip's default stereo in/out. Which comes out on top in terms of minimal latency and quality? Has this difference, if any, been measured scientifically by the Stanford team?