Voice over IP (VoIP) carries phone calls as packets over IP networks instead of over dedicated circuits. For the history of how voice moved from copper to packets, see VoIP and the retirement of copper.
How a VoIP call works¶
- SIP (Session Initiation Protocol) handles signalling: registering a phone with a server, setting up a call, ringing, answering, and hanging up.
- RTP (Real-time Transport Protocol) carries the actual audio, usually directly between the endpoints or through a media server.
- Codecs encode the audio. G.711 is uncompressed 64 kbps PCM, the same as the traditional phone network. G.722 is wideband ("HD voice"), and Opus is a modern adaptive codec.
The parts of a VoIP setup¶
| Component | Role |
|---|---|
| Endpoint | A SIP desk phone, a softphone app, or an analog phone behind an ATA |
| PBX (optional) | Switches calls between internal extensions and outside lines, and provides voicemail, IVR menus, ring groups, and call recording |
| SIP trunk / provider | Connects to the public phone network and supplies phone numbers (DIDs) |
The simplest setup skips the PBX: an ATA or SIP phone registers directly with a SIP provider. Adding a self-hosted PBX between them gives full control over extensions, dial plans, and voicemail.
Sections¶
- SIP providers: pay-as-you-go SIP trunks for an ATA or a PBX.
- PBX software: Asterisk, FreePBX, the wider open-source ecosystem, and legacy projects like Trixbox.