Just keep in mind that the amount of hops or latency to these SIP servers at the data centers should have no effect on the audio of the call (or quality of the call). Unless Voipo has changed the way they handle audio (RTP stream), these SIP servers are just handling the setup/signaling of the call not the audio stream itself. Voipo’s upstream/terminating carriers are handling the audio stream, unless Voipo has specifically flagged your account to proxy the audio.

See Tim’s last 2 responses in this thread a few years ago:
http://www.dslreports.com/forum/r242...Washington-DC-

If this is still the case, your ISP network backbone/routing would be most important between the terminating carriers and your location, not so much to the SIP servers at the data centers. I’m talking about the audio quality of the calls. The amount of hops/latency would come into big play here.

The main concern to the SIP servers is just to have good connectivity (no extreme latency), so as not to have any call setup/signaling problems during the call. In this respect, the backbone/routing just needs to be dependable and reliable more so than bandwidth….