SIP trunk settings reference

This reference describes the fields and values available when creating or editing an inbound SIP trunk. The settings are grouped by the tabs shown in the DIDWW User Panel. For configuration steps, see Create a SIP trunk.

General

The General tab identifies the trunk and determines how DIDWW locates the receiving SIP endpoint. For guidance on choosing a trunk type, see SIP routing methods.

Field	Availability	Description
Name	Required	Unique name used to identify the trunk in the DIDWW User Panel.
Capacity Limit	Optional	Maximum number of simultaneous calls that DIDWW can deliver through the trunk. This can restrict, but cannot increase, the Capacity assigned to a DID number. When the limit is reached, DIDWW attempts another trunk in the same Trunk Group when available. Enter a value from 1 to 32767. The default value is Unlimited.
Type	Required	Determines whether DIDWW uses a configured destination or a Contact address learned through SIP registration. The default value is Static Endpoint.

Type values

Value	Description
Static Endpoint	DIDWW constructs the destination from the user part, host, and optional port configured on the trunk.
Dynamic Registration	DIDWW delivers calls to an active Contact address supplied by the receiving system through SIP REGISTER requests.

Static Endpoint fields

These fields are available when Type is set to Static Endpoint.

Field	Availability	Description
User Part of R-URI	Static Endpoint only	User part placed before @ in the Request-URI. Supports {DID}, which inserts the called DID in E.164 format, and {CALL_CPC}, which inserts the Calling Party Category for Toll-Free numbers only. The default value is {DID}.
Host	Required, Static Endpoint only	Public IP address or DNS hostname used as the Request-URI host.
Port	Optional, Static Endpoint only	Destination signaling port. Enter a value from 1 to 65535. When empty, DIDWW queries DNS SRV. If no SRV record is found, DIDWW falls back to A and AAAA resolution on port 5060. See DNS SRV failover.
Resolve R-URI	Static Endpoint only	Replaces the Request-URI host with its resolved IP address.
Network Protocol	Static Endpoint only	Controls the IP protocol used when resolving the host. The default value is Any.
Transport	Required, Static Endpoint only	Transport protocol used for SIP signaling. The default value is UDP.
Preferred Server	Static Endpoint only	DIDWW Point of Presence used to deliver calls to the endpoint. The default value is Auto.

Network Protocol values

Value	Description
IPv4 only	Uses IPv4 exclusively.
IPv6 only	Uses IPv6 exclusively.
Any	Allows either IPv4 or IPv6.
Prefer IPv4 over IPv6	Uses IPv4 when available and falls back to IPv6.
Prefer IPv6 over IPv4	Uses IPv6 when available and falls back to IPv4.

Transport values

Value	Description
UDP	Sends SIP signaling over User Datagram Protocol.
TCP	Sends SIP signaling over Transmission Control Protocol.
TLS	Sends encrypted SIP signaling over Transport Layer Security.

Preferred Server values

Value	Description
Auto	Delivers calls from the DIDWW Point of Presence that received them. If one Point of Presence is unavailable, DIDWW routes calls through an alternative location to maintain call continuity. This is the default and recommended value. Allow all DIDWW SIP signaling and RTP addresses.
LA, MIA, or NY	Uses the selected United States Point of Presence.
FRA	Uses the Frankfurt, Germany Point of Presence.
SG	Uses the Singapore Point of Presence.
HK	Uses the Hong Kong Point of Presence.
AMS	Uses the Amsterdam, Netherlands Point of Presence.

Disabling Points of Presence can cause additional routing hops. See Configure Preferred Server routing.

Dynamic Registration fields

Field	Availability	Description
Use DID in R-URI	Dynamic Registration only	Replaces the registered Contact user part with the called DID in the Request-URI of the inbound INVITE.

DIDWW generates registration credentials after the trunk is created. One trunk supports up to 10 simultaneous registrations. See View SIP registration credentials and View Registration history.

Post-creation registration fields

The following read-only fields are shown when viewing a Dynamic Registration trunk after it has been created:

Field	Availability	Description
Username	Read-only, Dynamic Registration only	System-generated username that the SIP endpoint uses to register with DIDWW.
Password	Read-only, Dynamic Registration only	System-generated registration password. Use the visibility control to display it or the copy control to copy it.
Registration Endpoints	Read-only, Dynamic Registration only	DIDWW registrar hostnames to which the SIP endpoint can send its authenticated SIP REGISTER requests. sip.didww.com provides automatic DNS-based load balancing between locations. Regional hostnames direct registration to a specific location. See SIP Registrars for the available endpoints and their IP addresses.

Number Translations

Field	Availability	Description
CLI Format	Editable	Format applied to the caller ID in the SIP From header. The default value is E.164.
CLI Prefix	Optional	Prefix added before the caller ID after format conversion. Enter up to six characters containing digits or #, optionally preceded by +. The optional + is not included in the six-character limit. For example, + alone adds that symbol to every delivered caller ID. The field is empty by default.
CLI Number List	Optional	Number List used to allow or reject calls by full-number match, prefix match, or length.

CLI Format values

Value	Format	Description
E.164	Country code, area code, and number	Converts the caller ID to E.164 format. This is the default and recommended value. The number is delivered without +. For example, 12125551234.
Raw	Unmodified	Sends the caller ID unchanged, as received from the originating network.
Local	Area code and number	Converts the caller ID to the local numbering plan of the DID number’s country of origin. For example, 2125551234.

Note

• DIDWW transfers the caller ID accurately for local calls. International caller ID conversion depends on third-party providers and may not always produce the expected value.

• If CLI Format or CLI Prefix changes the value matched by a CLI Number List, update the Number List accordingly.

Trunk Group

This optional tab places the trunk in one Trunk Group for failover or load balancing.

Field	Availability	Description
Trunk Group	Optional	Group to which the trunk belongs. A trunk can belong to only one group.
Priority	Trunk Group selected	Attempt order. Lower values are attempted first. Trunks with the same priority are selected according to Weight. Enter a value from 0 to 65535. The default value is 1.
Weight	Trunk Group selected	Relative selection weight among trunks with the same priority. Higher values increase selection likelihood. Enter a value from 0 to 65535. The default value is 65535. See RFC 2782.
Re-routing Disconnect Codes	Trunk Group selected	Disconnect codes that cause DIDWW to attempt another trunk. By default, all codes are selected except 401 Unauthorized, 486 Busy Here, 503 Service Unavailable, and 603 Decline. The list also includes Ringing Timeout, which is selected by default and re-routes the call when the trunk’s Ringing Timeout expires. See SIP response codes in Call Logs.

Authorization

Field	Availability	Description
Enable Authorization	Editable	Enables SIP digest authentication for INVITE requests DIDWW sends to the receiving SIP server.
Auth User	Required when authorization is enabled	Username supplied during digest authentication.
Auth Password	Required when authorization is enabled	Password supplied during digest authentication.
From User	Optional when authorization is enabled	Custom user part placed in the SIP From header instead of CLI. Some receiving systems require this value to match Auth User.
From Domain	Optional when authorization is enabled	Custom domain placed in the SIP From header. Some receiving systems require a specific domain.

Media & DTMF

Field	Availability	Description
Allowed RTP IP Addresses	Optional	Up to 10 IPv4 or IPv6 addresses or subnets from which DIDWW accepts RTP, entered as IPv4[/mask] or IPv6[/mask]. Empty accepts any source. An omitted mask defaults to /32 or /128 respectively.
Codecs	Editable	Codecs included in the SDP offer. Their displayed order determines preference. By default, PCMU, PCMA, G729, G723, and telephone-event are selected. You can add, remove, or reorder the selection. See Supported codecs.
SRTP Mode	Editable	Controls RTP media encryption and key negotiation. TLS secures SIP signaling separately and does not enable SRTP automatically. The default value is Disabled. See Encryption.
DTMF Send Mode	Editable	Method DIDWW uses to send DTMF events. The default value is RFC 2833.
DTMF Receive Mode	Editable	Method DIDWW accepts for received DTMF events. The default value is RFC 2833.
RTP Timeout	Editable	Period without received RTP after which DIDWW disconnects the call. The value must be between 5 and 600 seconds. The default value is 30 seconds. In Inbound Call Logs, RTP Timeout (Destination) indicates that DIDWW did not receive media from the endpoint. RTP Timeout (Origination) indicates that DIDWW did not receive media from the caller side.
Force Symmetric RTP	Editable	Sends media to the source address and port of received RTP instead of the destination negotiated in SDP. This COMEDIA behavior can assist with NAT traversal. See RFC 4961.
Symmetric RTP Ignore RTCP	Force Symmetric RTP enabled	Uses RTP packets, but not RTCP packets, to switch the media destination.
RTP Ping	Editable	Sends empty RTP packets after SDP negotiation to trigger remote RTP. This can prevent a media deadlock when both systems use symmetric RTP and wait for the other system to send media first.

Warning

Force Symmetric RTP can expose a call to an RTP Bleed attack. Restrict media sources with Allowed RTP IP Addresses when enabling it.

SRTP Mode values

Value	Description
Disabled	Does not use SRTP media encryption.
SRTP SDES	Encrypts RTP and negotiates keys using SDP Security Descriptions.
SRTP DTLS	Encrypts RTP and negotiates keys using Datagram Transport Layer Security.
SRTP ZRTP	Encrypts RTP and negotiates keys using ZRTP.

DTMF Send Mode values

Value	Description
Disable sending	Does not send DTMF events.
RFC 2833	Sends the telephone-event RTP payload. This is the default.
SIP INFO application/dtmf-relay	Sends SIP INFO with application/dtmf-relay.
SIP INFO application/dtmf	Sends SIP INFO with application/dtmf.

DTMF Receive Mode values

Value	Description
RFC 2833	Receives the telephone-event RTP payload. This is the default.
SIP INFO application/dtmf-relay OR application/dtmf	Receives SIP INFO with either listed content type.
RFC 2833 OR SIP INFO	Accepts either RFC 2833 or SIP INFO events.

Signalling

Field	Availability	Description
SIP Session Timers	Editable	Enables session-timer negotiation. Session refreshes verify that the remote endpoint remains responsive and allow an inactive session to be terminated. Both endpoints must support the negotiation. See RFC 4028.
SST MIN Timer	Required when SIP Session Timers is enabled	Minimum session-refresh interval. Enter a value greater than 0 seconds. The default value is 600 seconds.
SST MAX Timer	Required when SIP Session Timers is enabled	Maximum session-refresh interval. The value must be greater than or equal to SST MIN Timer. The default value is 900 seconds.
SST Session Expires	SIP Session Timers enabled	Session-Expires value. It must be between the configured minimum and maximum.
SST Refresh Method	SIP Session Timers enabled	SIP method used to refresh the session. The default value is Invite.
SST Accept 501	SIP Session Timers enabled	Keeps the call connected after SIP 501 is returned to a refresh.
SIP Timer B	Editable	Timeout for an INVITE transaction, as defined by RFC 3261. The value must be between 1000 and 64000 milliseconds. The default value is 8000 milliseconds. If DIDWW receives no final response before the timer expires, the transaction fails.
DNS SRV Failover Timer	Editable	Time DIDWW waits before attempting the next DNS SRV target. Enter a value greater than 0 milliseconds. The default value is 2000 milliseconds.
Ringing Timeout	Optional	Time DIDWW waits for 200 OK after an 18x response. When the timeout is exceeded, the routing attempt ends with the Ringing timeout disconnect code. Enter a value from 1 to 32 seconds.
Max Transfers	Editable	Maximum SIP REFER requests processed. Enter a value from 0 to 8. The default value is 0, which disables transfers.
Max 30x Redirects	Editable	Maximum SIP 301 and 302 responses followed. Enter a value from 0 to 8. The default value is 0, which disables redirects.

SST Refresh Method values

Value	Description
Invite	Uses a re-INVITE. This is the default.
Update	Uses UPDATE.
Update fallback Invite	Uses UPDATE when supported and falls back to a re-INVITE.

Additional Services

Field	Availability	Description
STIR SHAKEN Mode	Editable	STIR/SHAKEN identity and verification information added to or relayed in the SIP INVITE. The default value is Disabled.
Diversion Relay Policy	Editable	Controls whether and how DIDWW relays PSTN Diversion information. The default value is Disabled.
Diversion Inject Mode	Editable	Controls whether DIDWW adds a Diversion header containing the DID. The default value is Disabled.
Enable CNAM Lookup	Optional	Looks up the caller name associated with a U.S. caller ID and delivers it as the display name in the SIP From header. An additional charge applies to each call from a U.S. source number. The default value is Disabled. See CNAM IN for SIP trunks, including Trunk Group behavior.

STIR SHAKEN Mode values

Value	Description
Disabled	Does not include additional STIR/SHAKEN headers.
Relay Identity header	Relays the received Identity header. Account enablement is required. Contact DIDWW Sales.
Add PAI, P-Attestation-Indicator, P-Origination-ID	Adds the listed identity and attestation headers.
Relay Identity header + Add PAI, P-Attestation-Indicator, P-Origination-ID	Relays Identity and adds the listed headers.
Add P-Stir-Verstat, P-Attestation-Indicator, P-Origination-ID	Adds the listed verification and attestation headers.

Diversion Relay Policy values

Value	Description
Disabled	Does not relay the received Diversion header.
Diversion header in SIP URI format	Relays a SIP URI, for example, sip:user@sip.didww.com.
Diversion header in TEL URI format	Relays a TEL URI, for example, tel:+123456789.

Diversion Inject Mode values

Value	Description
Disabled	Does not add a DIDWW-generated Diversion header.
Add Diversion header with DID number	Adds a Diversion header containing the DID number.

Enabling both Diversion settings can produce two Diversion headers. Confirm that the receiving system supports this behavior.

Related resources

• Create a SIP trunk — Create a trunk and configure the settings described on this page.

• SIP routing methods — Compare Static Endpoint and Dynamic Registration behavior.

• SIP connectivity reference — Review signaling, RTP, codec, DTMF, encryption, and registrar specifications.

• Trunk Groups — Configure failover and load balancing across multiple trunks.