SIP Basic Call Flow Explained

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.

🔃 Call Flow from Carrier to end User

A VoIP call originates with a carrier or telecommunications provider over the PSTN or a SIP trunk. The carrier is responsible for the availability, redundancy, and security of its own telecommunications network and for delivering the call to our Session Border Controller (SBC)/Switch. Our responsibility begins at the Switch level, where the call is secured, normalized, inspected, governed by routing and security policies, and passed to the appropriate cloud service.

The cloud service applies the customer’s call-routing logic and features, such as extensions, IVR, voicemail, and call control. The call is then delivered over the customer’s own network—including its Internet connection, firewall, router, LAN/WAN, and switching infrastructure—to the customer’s endpoint, such as an IP phone or softphone. The customer is responsible for the availability, configuration, security, and redundancy of that network and endpoint environment.

In summary, the carrier manages the upstream carrier network; our managed service responsibility begins at the Switch/SBC and continues through the cloud service; and the customer manages the network and device environment on their side of the connection.

📊 SIP Call Flow Diagram in Details

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