Two protocols

• SIP (Session Initiation Protocol, UDP/TCP 5060, TLS 5061) sets up, changes and ends calls.
• RTP carries the audio, on a range of UDP ports (often 10000–20000), described by SDP inside the SIP messages.

A call, step by step

1. Phone A → PBX: INVITE (with SDP: codecs, IP, RTP port)
2. PBX → A: 100 Trying, then 180 Ringing or 183 Session Progress
3. B answers → 200 OK with its SDP
4. A → ACK; audio flows as RTP
5. Either side → BYE → 200 OK

Authentication challenges appear as 401/407 followed by a repeated request with credentials — normal, not an error.

Codecs