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