SIP is the protocol VoIP systems use to start, change and end calls. It carries the signaling only: who is calling whom, which codecs both sides support and when the call ends. The voice itself travels separately, as RTP packets. SIP is an IETF standard (RFC 3261) used by IP desk phones, softphones, PBXs and operators alike.
A basic call is a short exchange of text messages:
- INVITE: the caller asks to open a call and lists the codecs it can use.
- 100 Trying, then 180 Ringing: the network is routing the call and the destination is ringing.
- 200 OK: the person answered; the caller confirms with ACK and the audio starts.
- BYE: either side hangs up.
Failures come back as numbered codes, much like on the web: 486 Busy Here, 404 Not Found when the number does not exist, 503 Service Unavailable when a route is down. SIP usually runs on port 5060 over UDP or TCP, and on 5061 when the signaling is encrypted with TLS.
Businesses meet SIP in two places: phones register to a PBX with a SIP account, and the PBX reaches the public phone network through a SIP trunk. OVOKY's SIP trunk connects a PBX by IP authentication or by registration with a SIP username and password, with the G.711, G.729 and Opus codecs.