# 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](how-to-guides/create-sip-trunk.html).

## 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](sip-routing-methods.html).

| 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](../../../phone-numbers/capacity/how-capacity-works.html) 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](#inbound-sip-trunk-type-values) | 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](#inbound-sip-static-endpoint-fields) | DIDWW constructs the destination from the user part, host, and optional port configured on the trunk. |
| [Dynamic Registration](#inbound-sip-dynamic-registration-fields) | 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](call-information/cpc-usage.html#inbound-sip-cpc-usage) 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](call-handling-and-delivery/service-failover-and-load-balancing.html#inbound-sip-dns-srv-failover). |
| Resolve R-URI | Static Endpoint only | Replaces the Request-URI host with its resolved IP address. |
| [Network Protocol](#inbound-sip-network-protocol-values) | Static Endpoint only | Controls the IP protocol used when resolving the host. The default value is **Any**. |
| [Transport](#inbound-sip-transport-values) | Required, Static Endpoint only | Transport protocol used for SIP signaling. The default value is **UDP**. |
| [Preferred Server](#inbound-sip-preferred-server-values) | 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](sip-connectivity-reference.html#inbound-sip-connectivity-reference). |
| 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](how-to-guides/configure-preferred-server-routing.html).

### 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](how-to-guides/view-sip-registration-credentials.html)
and [View Registration history](how-to-guides/view-registration-history.html).

#### 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](sip-connectivity-reference.html#inbound-sip-connectivity-registrars) for the available endpoints and their IP addresses. |

## Number Translations

| Field | Availability | Description |
| --- | --- | --- |
| [CLI Format](#inbound-sip-cli-format-values) | 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](../number-lists/index.html) 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](../trunk-groups/index.html) 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**](https://datatracker.ietf.org/doc/html/rfc2782.html). |
| 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](../analytics/sip-response-codes-reference.html#inbound-technical-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](sip-connectivity-reference.html#inbound-sip-connectivity-codecs). |
| [SRTP Mode](#inbound-sip-srtp-mode-values) | 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](sip-connectivity-reference.html#inbound-sip-connectivity-encryption). |
| [DTMF Send Mode](#inbound-sip-dtmf-send-values) | Editable | Method DIDWW uses to send DTMF events. The default value is **RFC 2833**. |
| [DTMF Receive Mode](#inbound-sip-dtmf-receive-values) | 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](../../../logs-analytics/call-logs/inbound-logs.html#inbound-cdr-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**](https://datatracker.ietf.org/doc/html/rfc4961.html). |
| 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**](https://datatracker.ietf.org/doc/html/rfc4028.html). |
| 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](#inbound-sip-sst-refresh-values) | 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**](https://datatracker.ietf.org/doc/html/rfc3261.html). 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](#inbound-sip-stir-shaken-values) | Editable | STIR/SHAKEN identity and verification information added to or relayed in the SIP `INVITE`. The default value is **Disabled**. |
| [Diversion Relay Policy](#inbound-sip-diversion-relay-values) | Editable | Controls whether and how DIDWW relays PSTN Diversion information. The default value is **Disabled**. |
| [Diversion Inject Mode](#inbound-sip-diversion-inject-values) | 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](call-information/cnam-in.html), including [Trunk Group behavior](call-information/cnam-in.html#inbound-sip-cnam-in-trunk-groups). |

### 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](mailto:sales%40didww.com). |
| 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](how-to-guides/create-sip-trunk.html) - Create a trunk
  and configure the settings described on this page.
- [SIP routing methods](sip-routing-methods.html) - Compare Static Endpoint
  and Dynamic Registration behavior.
- [SIP connectivity reference](sip-connectivity-reference.html) - Review
  signaling, RTP, codec, DTMF, encryption, and registrar specifications.
- [Trunk Groups](../trunk-groups/index.html) - Configure failover and load
  balancing across multiple trunks.

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