A SIP phone system lives or dies on the unglamorous middle. Something has to route every call, know who is registered, wake phones that are asleep, and translate between networks that disagree. Flexisip is that middle, built as a modular server suite where you deploy only the pieces your service needs. The proxy routes calls and messages, relays media with ICE, STUN, and TURN support, and authenticates users by digest, TLS certificates, or OpenID Connect. Its push gateway speaks RFC 8599 to Apple and Google, so incoming calls ring phones whose app is closed, and it can lend that gift to an existing Asterisk or FreeSWITCH platform that lacks push of its own. The presence server tracks who is reachable and shares contact lists, the back-to-back user agent bridges SIP networks to PSTN gateways and trunks with codec transcoding and caller identity translation, and a registration event server notifies peer domains when users register. Instances cluster with load balancing for services that outgrow one box, the same suite shrinks onto a Raspberry Pi class intercom, and it has quietly powered the public linphone.org SIP service for over a decade. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. AGPL-3.0 licensed.