VoIP SIP (Session Initiation Protocol) Gateway: An Overview

  Need Assistance? Contact us
Logo Mobile   Empowering

Please select a VOIP SDK of your interest
VoIP SIP (Session Initiation Protocol) Gateway: An Overview


A network gateway is a point of connection between two networks. It acts as a bridge, facilitating communication between the two networks and enabling the flow of data between them. The gateway serves as a crucial link, connecting both networks to ensure seamless data exchange.


SIP (Session Initiation Protocol) is a key protocol in VoIP (Voice over Internet Protocol) technology.

A SIP gateway is a device that links SIP-based VoIP phone networks to various other phone networks, including landline PSTN (Public Switched Telephone Network), mobile GSM (Global System for Mobile Communication), landline ISDN (Integrated Services Digital Network), and other phone system networks.


VoIP (Voice over Internet Protocol) is a technology that encompasses various protocols, with SIP being one of them. While a SIP gateway can also be referred to as a VoIP gateway, it's important to note that only gateways based on the SIP protocol are specifically termed SIP gateways.


An organization operates a VoIP phone system with a SIP protocol-based IP PBX installed and running, SIP-based phone extensions are online, users utilizing these extensions can only dial and receive calls internally, and they face limitations when attempting to make or receive calls to and from mobile numbers or other landline numbers.

To address this limitation and extend the system's capabilities, the organization incorporates SIP to PSTN or SIP to GSM gateways within the VoIP phone system.


In a setup where SIP-based IP PBX and SIP-based phones are interconnected and operate over the same Local Area Network (LAN), the SIP gateway plays a crucial role.

The gateway is connected to the same LAN and becomes an integral part of the network by obtaining an IP address. The IP PBX initiates communication with the gateway, enabling users to dial and receive PSTN and GSM calls through this integrated system.


Once the gateways are connected to the same Local Area Network (LAN) using either a wired or Wi-Fi connection, they seamlessly become part of the network, ready for communication with the SIP PBX.

If the SIP gateway functions as a SIP to PSTN gateway, local landline phone connections are necessary to establish a link between the gateway and the local landline phone company. The number of PSTN landline connections that can be accommodated by the SIP to PSTN gateway depends on the total number of channels it possesses.

If the SIP gateway operates as a SIP to GSM gateway, it requires the addition of regular GSM connections with SIM cards. If the GSM gateway supports 8 channels, then a maximum of 8 SIM cards can be added to the GSM gateway.

When configuring the SIP-based IP PBX to route calls to a SIP gateway, you'll need to specify the IP address of the SIP gateway and the number prefix.

Open administrative interface of your SIP-based IP PBX system. Enter the IP address of the SIP gateway in the relevant configuration field. This is the address where your IP PBX will forward outbound calls.

Specify the number prefix at which the IP PBX should route calls to the SIP gateway. As mentioned, many systems default to using '9', but this may vary based on your SIP PBX brand and its specific configuration.

With the setup in place, users using SIP phones can dial external numbers by prefixing the desired number with the configured prefix (e.g., '9'). The SIP PBX will then communicate with the SIP gateway to complete the call.


In the contemporary era of VoIP, integrating SIP gateways is no longer obligatory for connecting an organization's SIP phone system to the PSTN or GSM network.

IP-telephony service providers (ITSP) offer solutions for these requirements, providing services that enable SIP phone systems to initiate and receive calls to and from the PSTN and GSM networks.

The ITSP configures gateways within their premises and network infrastructure. The ITSP's SIP PBX communicates with these gateways, facilitating seamless integration. SIP accounts are provided by the ITSP for utilization within the organization's SIP PBX phone system.

The configured organization's SIP PBX system can both initiate and receive calls through the ITSP's SIP PBX.

In essence, the ITSP (Internet Telephony Service Provider) provides VoIP SIP-based connections, facilitating communication between the organization's SIP PBX and external PSTN and GSM phone networks.


DID (Direct Inward Dialing) numbers are actual phone numbers, similar to mobile or PSTN numbers, associated with a specific city or area in a country.

ITSPs also offer DID numbers for various countries, acquire Direct Inward Dialing (DID) numbers, connect DID numbers to the organization's SIP PBX by using SIP account authentication credentials from your ITSP.

DID Routing: When an inbound call reaches a designated DID number, the ITSP’s SIP PBX establishes a connection to the organization's SIP PBX via the internet.

SIP PBX Handling: The organization's SIP PBX takes control and routes the call to the appropriate SIP phone within the system.