Understanding the Core Mechanism Driving SIP Trunking Communications


SIP Trunking relies on SIP (Session Initiation Protocol) to facilitate voice and multimedia communication over IP networks. To gain a comprehensive understanding of SIP Trunking, it’s essential to explore the underlying signalling mechanisms that drive these communications. In this sixth instalment of our 10-part blog series on SIP Trunking, we will delve into the details of SIP signalling, its components, and how it operates within a SIP Trunking infrastructure.

SIP Signalling Overview

SIP is an application-layer protocol that establishes, maintains, and terminates multimedia communication sessions over IP networks. SIP signalling is the process of exchanging messages and information between SIP-enabled devices, such as IP phones, PBX systems, and SBCs, to initiate and control communication sessions.

Key Components of SIP Signalling

  1. SIP Messages: SIP messages are textual, human-readable messages exchanged between SIP-enabled devices. They consist of two types: requests and responses. Requests are sent by a client to initiate or modify a communication session, while responses are sent by the server to acknowledge or provide information about the session.

  2. SIP Methods: SIP methods, also known as methods or verbs, are used in request messages to indicate the desired action. Some common SIP methods include INVITE (initiate a session), ACK (acknowledge a session), BYE (terminate a session), and REGISTER (register with a SIP server).

  3. SIP Response Codes: SIP response codes are three-digit numeric codes used in response messages to indicate the outcome of a request. They are grouped into six classes, based on the first digit, with each class representing a specific category of response (e.g., 2xx for successful responses, 4xx for client errors, and 5xx for server errors).

  4. SIP URIs: SIP Uniform Resource Identifiers (URIs) are used to identify SIP-enabled devices and resources. They are similar in format to email addresses and typically include a username and domain, such as [email protected].

  5. SIP Headers: SIP headers carry additional information about a message, such as the sender and recipient, call duration, and authentication data. They are included in both request and response messages and provide important context for SIP signalling.

SIP Signalling in a SIP Trunking Infrastructure

In a SIP Trunking environment, SIP signalling is used to initiate and manage communication sessions between a business’s PBX system and the SIP trunk service provider. The signalling process typically follows these steps:

  1. Call Initiation: When a user initiates a call, the PBX sends an INVITE message to the SBC, which then forwards it to the provider’s SIP server.

  2. Call Setup: The provider’s SIP server processes the INVITE message and sends a response back to the SBC. If the call is accepted, a series of messages (including ACK, PRACK, and UPDATE) is exchanged between the devices to establish the session.

  3. Media Exchange: Once the call is set up, media (voice, video, or data) is exchanged between the caller and recipient over a separate channel called the Real-time Transport Protocol (RTP). SIP signalling only handles the session control, not the actual media transmission.

  4. Call Termination: To end the call, either party sends a BYE message, which is propagated through the SIP signalling chain. This terminates the session and releases the associated resources.


SIP signalling is the foundation of SIP Trunking communications, enabling the initiation, control, and termination of multimedia sessions over IP networks. By understanding the key components and process of SIP signalling, businesses can gain valuable insight into the operation of their SIP Trunk