Creating a New SIP Trunk

Step 1. Under the “Trunks” section, select “Voice IN” and click the “Create New/ SIP Trunk” button.(Fig. 1).


Fig. 1. Adding a new SIP Trunk

Step 2. You will be prompted to enter your SIP Trunk details (Fig. 2).

Basic configuration options:

Friendly name - a desired trunk name.

Username - the user part of R-URI in the INVITE request. You may also use “{DID}” pattern which will be replaced by the called DID number (DNIS) in E164 format.

Host - your server’s public IP address or domain.

Port - SIP port on your server. In order to use DNS SRV failover mechanism leave port empty. In case of empty value DIDWW system will try to resolve DNS SRV records, then A record will be tried (with default SIP signaling port 5060)

Preferred server - the preferred DIDWW POP server, with the options being US: LA, MIA, NY, Germany: FRA, SG (Singapore) or Local.

Protocol - the underlying transport layer protocol (UDP or TCP) that will be responsible for SIP requests and responses.

Capacity Limit - the maximum number of simultaneous calls per trunk.

Resolve rURI - if checked, host part or R-URI will be replaced by the resolved IP address.

Resolve DNS SRV record - if checked, the system will attempt to perform a DNS SRV lookup for the host part of the R-URI to override the port settings.

CLI Format - the following format options are available:

  • “raw” - DIDWW will pass the incoming caller ID unchanged (default).

  • “e164” - DIDWW will attempt to convert the CLI to E.164 format (Country Code + Area Code + Number)

  • “local” - DIDWW will attempt to convert the CLI to local format (Area Code + Number)


CLI format conversion may not work correctly for phone calls originating from outside the country of the specific DID

CLI Prefix - a field that allows prepending an additional prefix to the incoming CLI (maximum of 7 characters)

Map all DID(s) - if checked, then all DIDs in your account will be assigned to this inbound trunk.

In addition to the basic configuration options, there are advanced options which may be left unchanged, or recommended values can be set by pressing “Use recommended values” button. These options include:

  • Auth enabled - if checked, then authorization for the SIP server is enabled

  • Auth user - optional authorization user for the SIP server

  • Auth password - optional authorization password for the SIP server

  • Auth “From” User - Specifies user in a “from:” field instead of the Caller ID

  • Auth “From” Domain - Specifies the default “From” domain in SIP messages

  • Media & DTMF - Optional signalling configuration with parameters:

  • Available codecs/Selected codecs - codecs that will be sent with an SDP offer may be selected from the list of supported codecs

  • Customer To DIDWW DTMF format - the method of receiving DTMF signals from the CPE (options are RFC 2833, SIP INFO application/dtmf-relay OR application/dtmf, or RFC 2833 OR SIP INFO)

  • DIDWW to Customer DTMF format - the method of sending DTMF signals to the CPE CPE (options are disable sending, RFC 2833, SIP INFO application/dtmf-relay, or SIP INFO application/dtmf)

  • RTP Timeout - timeout (seconds) for disconnecting the call if RTP packets are not received

  • RTP Ping - if checked, an RTP ping should be used for connecting the call

  • Force Symmetric RTP - if checked, the trunk will operate in the Symmetric RTP/COMEDIA mode

  • Symmetric RTP Ignore RTCP - if checked, then only RTP packets will be considered while operating in the Symmetric RTP/COMEDIA mode, and switching of the RTP session based on RTCP packets will be avoided

  • Advanced Signalling Settings - Optional SIP configuration parameters as follows:

  • SST enabled - (RFC 4028) if checked, then the customization of SIP Session Timers is enabled in order to ensure that a session remains alive

  • SST Accept 501 - if checked, then the call will not be dropped after receiving SIP 501 response for non-critical messages

  • SST MIN Timer - (RFC 4028) minimum SIP session timer value (default 600 seconds)

  • SST MAX Timer - (RFC 4028) maximum SIP session timer value (default 900 seconds)

  • SST Session expires - (RFC 4028) Optional ‘Session-Expires’ header values (should be in range SST MIN Timer to SST MAX Timer)

  • SST Refresh method - (RFC 4028) SIP method that is used for session update (options are Invite, Update and Update Fallback Invite)

  • SIP Timer-B - (RFC 3261, Section INVITE transaction timeout (default 8000 ms)

  • DNS SRV failover timer - Invite transaction timeout for each of the gateways with DNS SRV rerouting (default 2000 ms)

  • Max Transfers - maximum REFER Request count

  • Max 30x Redirects - maximum 301/302 SIP Redirect count

  • STIR/SHAKEN - Optional caller-id attestation headers based on the new STIR/SHAKEN policy. See Voice IN STIR/SHAKEN page for details.


Fig. 2. SIP Trunk configuration details window

Step 3. Click “Create” to complete the trunk configuration.

SIP Technical information is available here.