Sazrika Foundation / Learn / SIP Call Flow
Interactive diagram

How a SIP call works

Step through a complete SIP call from the moment you dial to the moment you hang up — with real SIP messages at each stage.

Step 0 of 0
// Current step
Press Play to begin
This diagram shows a complete SIP call flow between two endpoints, passing through a SIP proxy, Session Border Controller, SIP trunk provider, and the PSTN.
// SIP message reference
INVITE
Session Initiation
Initiates a call. Contains SDP with codec and media info.
100 Trying
Provisional
Server received INVITE and is processing. Stops retransmissions.
180 Ringing
Provisional
Destination phone is ringing. Ringback tone plays to caller.
200 OK
Success
Call answered. Contains SDP answer with agreed codec.
ACK
Acknowledgement
Confirms receipt of 200 OK. Completes the 3-way handshake.
BYE
Session Termination
Ends the call. Either party can send. Media stops immediately.
CANCEL
Cancel Pending
Cancels a pending INVITE before it's answered. Like hanging up while ringing.
RTP/SRTP
Media Stream
Actual audio flows over RTP, separate from SIP signalling. Usually direct between endpoints.