A SIP trunk is a virtual connection that delivers inbound voice calls from the public telephone network (PSTN) to an IP-based phone system over the internet.
To quickly configure a SIP trunk, enter the required information below:
Enter a Friendly Name - Provide a unique name to identify the trunk.
Enter Your Host address - Specify the public IP address or domain name of your server.
Click Create button to save and activate your configuration.
Warning
Checking the Assign trunk to all DIDs checkbox will override all previous trunk configurations.
Perform this action only if you intend to assign all DIDs to the new trunk.
Click Use Recommended Values to apply the most widely supported SIP trunk settings instantly.
If the preferred server is set to Auto, ensure your system can accept traffic from all DIDWW SIP IPs.
For advanced configurations, you can customize additional SIP trunk settings. This guide explains the available options based on priority and use case.
The SIP trunk configuration is divided into the following sections:
General settings in SIP Trunk are required to correctly forward inbound calls to your server. These options define how calls are routed, formatted, and processed.
Passes the incoming caller ID unchanged (default).
e164
Converts the CLI to E.164 format (Country Code + Area Code + Number).
local
Converts the CLI to local format (Area Code + Number).
Note
CLI format conversion may not work correctly for calls originating from outside the country of the DID.
CLI Prefix - Prepend a custom prefix to the incoming CLI for identification or routing purposes.
CLI Number List – Assign a Number List to allow or reject incoming calls based on full number matches, prefix matches, or length restrictions. For more information, see the Number List documentation page.
Warning
Number Lists work by matching the inbound CLI. If you moodify the CLI format and CLI prefix, you may need to adjust your numbers in the Number List accordingly.
Username - Define the user part of the R-URI in the INVITE request.
Tip
Use placeholders for default setups:
{DID}: Inserts the called DID number (DNIS) in E.164 format.
{CALL_CPC}: Identifies the type of the calling party. For more details, see here.
Host - Enter the public IP address or domain name of your server.
Transport Protocol - Choose the protocol for SIP signaling: UDP, TCP, or TLS.
Note
TLS is not enabled by default. Contact sales@didww.com to enable it.
Port - Specify the SIP port on your server.
Tip
To enable the DNS SRV failover mechanism, leave this field empty. If empty, the system attempts DNS SRV resolution first, then falls back to the A record with port 5060.
Network Protocol - Specifies the IP protocol version used for communication between the SIP trunk and the endpoint. Options include:
SIP Trunk group configuration is optional and should only be used when multiple trunks need to be assigned to a single trunk group for failover or load balancing.
Authentication is an optional SIP Trunk setting that allows the trunk to be authenticated using digest authentication (credentials). Use this feature only if your SIP server requires authentication.
The Media & DTMF section allows you to configure codec preferences, Dual-Tone Multi-Frequency (DTMF) signaling, and real-time transport protocol (RTP) settings.
The Advanced Signaling Settings section allows you to configure SIP session management, transaction timeouts, failover behavior, and diversion headers settings. These optional settings help maintain call stability, prevent unnecessary call drops, and optimize failover handling in case of network failures.
Session Timers ensure that SIP sessions remain active by periodically exchanging messages. If a session remains idle for too long and the other endpoint is unresponsive, the dialog is automatically terminated. For more details, refer to RFC 4028.
Setting
Description
SST Enabled
Activates SIP Session Timers. Ensures that SIP dialogs remain active by sending periodic keep-alive messages.
This feature requires support from both endpoints and is standardized by the IETF.
SST Accept 501
Prevents call drops when receiving a SIP 501 response for non-critical messages.
SST MIN Timer
Sets the minimum session timer value (default: 600 seconds).
SST MAX Timer
Sets the maximum session timer value (default: 900 seconds).
SST Session Expires
Defines the Session-Expires header value (must be within the MIN and MAX Timer range).
SST Refresh Method
Specifies the SIP method for session updates: INVITE, UPDATE, UPDATE (Fallback to INVITE).
The Diversion Headers Settings section allows you to configure how diversion headers are handled for calls received on the inbound SIP trunk. These headers are used to indicate call forwarding or redirection events prior to reaching the trunk and may be required by downstream systems for routing, billing, or caller identity purposes.
Setting
Description
Diversion Relay Policy
Specifies how DIDWW should handle Diversion headers received from the remote party. Options include:
Do not relay – Remove the Diversion header from outgoing SIP messages.
Relay as SIP URI – Format and relay the Diversion header as a SIP URI (e.g., sip:user@sip.didww.com).
Relay as TEL URI – Format and relay the Diversion header as a TEL URI (e.g., tel:+123456789).
Diversion Inject Mode
Controls whether DIDWW inserts a Diversion header in SIP messages. Options include:
Do not add Diversion – A Diversion header will not be generated or included by DIDWW.
Add Diversion as +DID number – A DIDWW-generated Diversion header will be added using the associated +DID number as the value, formatted in E.164 (e.g., tel:+123456789).
The STIR/SHAKEN framework helps prevent caller ID spoofing by verifying the authenticity of the calling number. This section allows you to configure caller ID attestation headers.
Adds verification status and attestation headers to the SIP INVITE request.
This option is supported by default and does not require enabling Identity Header Transit.
Note
The Transit Identity Header is not enabled by default. To enable it for your DIDWW account or from the originating side, contact our sales team at sales@didww.com.
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