Basically, I want to implement a VoIP system with sip in a vps server. But it seems that it would not be able to handle more than ~20 simultaneous calls(just bare sip). What are the workarounds to this problem? Can the sip server be just used as a database to tell the clients where to find their intended targets..? Like p2p? I am quite new to sip. Additional info is appreciated.
SIP scalability comes from delegating as much work to the endpoints and doing as little on the servers as possible. What you describe is a "redirect server": it accepts and stores registrations from the endpoints (softphones, hardphones, etc), and responds with "3xx redirect" to incoming calls and forgets about them immediately.
This is probably the most extreme example of server minimization. SIP is a very versatile protocol, it lets you set up your server infrastructure in many different ways with varying degree of control over calls. It lets you trade off features for performance.
Even the flimsiest VPS should be able to handle the signalling for way more than 20 parallel calls even in full "stateful proxy" mode.
Just make sure media (the RTP streams) is not routed through your server. Set up STUN to help firewalled endpoints send media to each other directly.