PS Voice/SIP Service

SIP Call Flow

What Is SIP Call Flow?

Generally speaking, the term “call flow” refers to the map or path of how calls are navigated from beginning to end. SIP call flow can be a simple, linear map between two SIP users, or a very complicated series of destinations utilized to reach the final conversation between two users.

Because this is somewhat of an abstract concept, it may be easier to visualize a very simple example of an SIP call flow. For our example, we will demonstrate the SIP call flow between two users (user A and user B). Both users in this example have an SIP proxy server that handles the coding and signaling. We can break SIP call flow down into eight steps.

SIP Call Flow Example:

  1. Invite: User A wants to initiate a phone call with user B. User A sends an invitation to user B. This invitation contains information including both users’ addresses, phone numbers and preferred media settings.
  2. 100 Trying: User B’s proxy receives the invitation and sends back code 100 (trying.) This means user B accepted user A’s invite.
  3. 180 Ringing: Once user B’s phone begins ringing, the code 180 (ringing) is sent back to user A.
  4. 200 OK: Once user B answers the call, code 200 (OK) SIP message is sent to user A. Again, the message contains information on user B’s supported media settings.
  5. ACK: User A relies with code ACK, confirming that 200 (OK) was received.
  6. SIP Call: Finally, the two users communicate directly in a phone conversation.
  7. BYE: Once either user decides to end the call, a BYE message is sent to the other user.
  8. Final 200 OK: The user receiving the BYE message sends back a final 200, (OK), code to confirm the conversation ended.

In this SIP call flow, if user B is unavailable or doesn’t take user A’s call, the navigation is sent to voicemail or another phone number. While this is an example of a simple SIP call flow between two users, SIP call flows can be extremely complex with long navigations to reach the endpoint. This can look particularly complex in large business PBX systems with many extensions and users. These SIP call flows are used in a VoIP environment, utilizing the Internet as the server.

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