Design VoIP System

Problem framing and VoIP scope

A VoIP system carries voice calls over IP networks, letting applications and devices place and receive calls programmatically, including to and from the regular phone network (the PSTN). The fact that defines the architecture is the split between signaling and media. Signaling, done with SIP (Session Initiation Protocol), sets up, modifies, and tears down a call, the INVITE, ringing, answer, and BYE; media, the actual audio, flows separately as RTP packets. The platform sets up calls via SIP, bridges the two legs of a call, routes the RTP audio, and interconnects with carriers over SIP trunks through session border controllers (SBCs).

The spine is therefore: a control surface (an API and SIP signaling) that originates and receives calls idempotently, a call state machine driven by SIP responses (trying, ringing, answered, completed), a media plane that routes RTP audio between legs and to the PSTN, SIP-trunk interconnection to carriers via SBCs, webhooks that surface call events to the customer, per-tenant concurrent-channel governance, and call detail records (CDRs) for billing. State the framing up front: VoIP is signaling (SIP) plus media (RTP) with PSTN interconnection, and the hard parts are the call state machine, media routing, and carrier-grade reliability.