Introduction
This document provides a structured overview of the Session Initiation Protocol (SIP) call flow, covering the complete lifecycle of a typical VoIP session—from initial registration through call setup, media exchange, and termination.
Key protocols addressed include SIP (signalling), SDP (media negotiation), RTP/SRTP (media transport), and TLS (security). The flow diagrams and tables included serve as a practical reference for network engineers, telecom architects, and system administrators involved in VoIP deployment, troubleshooting, and optimisation.
📊 SIP Call Flow Diagram
The following flowchart provides a visual summary of the complete SIP call session lifecycle, from initial INVITE through to termination. This diagram consolidates the key signalling exchanges between caller, proxy, and callee, including provisional responses, media negotiation, and session tear-down—serving as a quick-reference companion to the detailed sections below.
1. 🚀 Call Initiation (INVITE)
User A’s SIP client transmits an INVITE request to User B.
SIP headers convey:
Caller and callee addresses (phone numbers or SIP URIs)
Routing information (Via, Route headers)
The SDP body specifies:
Media capabilities (e.g., supported codecs: G.711, G.729, etc.)
IP address and UDP port for RTP streams
2. ⏳ Provisional Responses
User B’s SIP proxy returns a 100 Trying response to acknowledge receipt and indicate that request processing is underway (may be generated statelessly).
When User B’s device alerts, a 180 Ringing response is propagated back to User A (stateful, end‑to‑end signalling).
3. ✅ Call Acceptance (200 OK)
Upon answer, User B’s client sends a 200 OK response.
This includes B’s SDP answer—the selected codec, media IP, and port—finalising the negotiation.
Media parameters are now mutually agreed and ready for use.
4. ✉️ Call Confirmation (ACK)
User A’s client issues an ACK to confirm receipt of the 200 OK.
RTP (or SRTP) media streams commence, using the negotiated codecs and ports.
5. 👋 Call Termination (BYE)
Either party initiates a BYE request to end the session.
The recipient replies with 200 OK, terminating the session and ceasing all RTP streams.
📊 Message Flow (Proxy Scenario)
A (Caller) SIP Proxy B (Callee)
| | |
|--- INVITE(SDP)-->| |
|<-- 100 Trying ---| |
| |--- INVITE(SDP)-->|
| |<-- 180 Ringing --|
|<-- 180 Ringing --| |
| |<-- 200 OK(SDP) --|
|<-- 200 OK(SDP) --| |
|------ ACK ------>| |
| |------ ACK ------>|
| |
| [ RTP / SRTP Media Flow ] |
| |
|------ BYE ------->| |
| |------ BYE ------->|
|<----- 200 OK ----| |
| |<----- 200 OK ----|🔑 Key Technical Concepts
Aspect | Description |
|---|---|
SDP Negotiation | Essential pre‑call handshake to align codecs, transport ports, and media formats. |
SIP Response Codes | 1xx (provisional), 2xx (success), 3xx–6xx (final/error responses). |
Stateless vs Stateful | Proxies typically generate 100 Trying statelessly; endpoints handle 180, 200, and ACK with stateful logic. |
Media Protocols | RTP/SRTP carry voice/video payloads; RTCP provides quality feedback (jitter, loss, delay). |
Security & Transport | TLS secures SIP signalling; SRTP encrypts media streams. |
Common Ports | SIP: 5060 (UDP/TCP), 5061 (TLS); RTP: dynamic ports (typically 1024–65535). |
🏗️ Typical Deployment Architecture
Endpoints: IP phones, softphones, WebRTC clients.
Proxy / Registrar: Handles routing, authentication, and registration.
SBC (Session Border Controller): Secures inter‑domain trunks and manages NAT traversal.
Media Servers: Used for conferencing, transcoding, or IVR services.
SIP Trunks: Connect on‑premises systems to PSTN or cloud telephony providers.
📝 SIP Registration (REGISTER) – Summary
The endpoint sends a REGISTER request to a registrar, containing its Address of Record (AOR).
The registrar challenges with 401 Unauthorized or responds with 200 OK upon successful authentication.
Registration binds the user’s SIP URI to their current IP/port, enabling inbound call routing.