How audio packages travel

I have a dumb question about SIP. During the call do audio packages go from first phone to server then second phone or directly first phone to second phone?

Which one is correct for default asterisk configuration?

-Audio packages transfer-

  1. first phone<–>server<–>second phone
  2. first phone<---->second phone

If second one is correct it doesnt really matter number of concurrent calls. Server load wont increase dramatically and I dont need higher bandwidth or powerful server.

Yes. They do one of those two things.

It depends on the details of the configuration, and potentially on the phones. A lot of features cannot be implemented when using directmedia, and it is only possible if the codecs are compatible, so it may be ignored, even if you try to enable it.