# Create a SIP trunk

Create an inbound SIP trunk to deliver calls from DIDWW to a SIP endpoint,
such as an IP PBX, session border controller, communications platform, or
application server. Choose **Static Endpoint** when DIDWW should send calls to
a destination configured on the trunk. Choose **Dynamic Registration** when
your SIP system should register with DIDWW and supply the destination.

For a detailed comparison of both options, see [SIP routing methods](../sip-routing-methods.html).

## Before you begin

- An active DIDWW account is required. [Sign in to DIDWW](https://my.didww.com/users/sign_in) or [Create DIDWW account](https://my.didww.com/users/sign_up#/users/sign_up).

## Step 1: Open the creation form

1. In the DIDWW User Panel, open **Voice**.
2. Select **Inbound Trunks**.
3. Click **Create New**.
4. Select **SIP Trunk**.

![Create New menu on the Inbound Trunks page with SIP Trunk available](../../../../_images/fig12.webp)


**Fig. 1.** Select SIP Trunk

## Step 2: Configure the General settings

1. Enter a unique **Name** for the trunk.
2. Optionally set **Capacity Limit** to restrict the number of simultaneous
   calls delivered through this trunk. The default value is **Unlimited**. In
   a Trunk Group, DIDWW tries another trunk when the limit is reached.
3. Select the **Type** that matches how the SIP endpoint receives calls, then
   complete the corresponding settings below.

Static Endpoint

Use **Static Endpoint** when the receiving SIP system has a stable,
publicly reachable IP address or DNS hostname.

1. Select **Static Endpoint**.
2. In **Host**, enter the public IP address or DNS hostname of the SIP
   endpoint.
3. Review **User Part of R-URI**, **Port**, **Resolve R-URI**, **Network
   Protocol**, **Transport**, and **Preferred Server**. Retain their
   default values unless the receiving system requires different SIP
   addressing or delivery behavior. Leave **Port** empty to use DNS SRV.

For how DIDWW builds the Request-URI and resolves the host, see
[Static Endpoint routing](../sip-routing-methods.html#inbound-sip-static-endpoint-routing).
Field definitions, defaults, and available values are listed under
[Static Endpoint fields](../sip-trunk-settings-reference.html#inbound-sip-static-endpoint-fields),
[Network Protocol values](../sip-trunk-settings-reference.html#inbound-sip-network-protocol-values),
[Transport values](../sip-trunk-settings-reference.html#inbound-sip-transport-values), and
[Preferred Server values](../sip-trunk-settings-reference.html#inbound-sip-preferred-server-values).

Important

The default **Preferred Server** value is **Auto**. Allow all
[DIDWW SIP signaling and RTP addresses](../sip-connectivity-reference.html#inbound-sip-connectivity-reference) so DIDWW can deliver calls
through an alternative Point of Presence when required. Disabling
Points of Presence can introduce additional routing hops. See
[Configure Preferred Server routing](configure-preferred-server-routing.html).

![General tab configured for a Static Endpoint SIP trunk](../../../../_images/fig259.png)


**Fig. 2.** Configure Static Endpoint routing


Dynamic Registration

Use **Dynamic Registration** when the SIP system has a changing public
address, operates behind NAT, or normally receives calls through SIP
registration.

1. Select **Dynamic Registration**.
2. Enable **Use DID in R-URI** if DIDWW should replace the registered
   Contact user part with the called DID in the Request-URI. This helps
   the receiving system distinguish calls to different DID numbers that
   use the same trunk.

DIDWW generates the registration username, password, and registrar
endpoints after the trunk is created. One trunk supports up to 10
simultaneous registrations. See [Dynamic Registration
fields](../sip-trunk-settings-reference.html#inbound-sip-dynamic-registration-fields). For how registration-based
routing works, see [Dynamic Registration routing](../sip-routing-methods.html#inbound-sip-dynamic-registration-routing).

The SIP system must refresh its registration before it expires. If no
valid registration is available, DIDWW has no current Contact address
to which it can deliver the call.

![General tab configured for a Dynamic Registration SIP trunk](../../../../_images/register.png)


**Fig. 3.** Configure Dynamic Registration routing

## Step 3: Configure optional settings

Configure only the tabs required by the receiving system and your inbound
call-routing design. You can skip this step and create the trunk with the
**General** settings only.

### Number Translations

Change how the caller ID is formatted, add a prefix to it, or filter calls by
caller ID with a Number List.

Field definitions, available values, and constraints are listed in
[Number Translations](../sip-trunk-settings-reference.html#inbound-sip-trunks-settings-number-translations)
and [CLI Format values](../sip-trunk-settings-reference.html#inbound-sip-cli-format-values). For how Number
Lists allow or reject calls, see [Number Lists](../../number-lists/index.html).

Warning

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. A **CLI Number List** matches
the caller ID after **CLI Format** and **CLI Prefix** are applied. Update
the Number List if either setting changes the value it must match.

![Number Translations tab of the Create Inbound SIP Trunk form with CLI Format, CLI Prefix, and CLI Number List
fields](../../../../_images/number_translations1.png)


**Fig. 4.** Configure Number Translations

### Trunk Group

Add the trunk to a Trunk Group and set how it is selected and re-routed
within the group. Use this tab only when the trunk will participate in
failover or load balancing.

Field definitions, available values, and constraints are listed in
[Trunk Group settings](../sip-trunk-settings-reference.html#inbound-sip-trunks-settings-trunk-group).

For how Trunk Groups work, see [Trunk Groups](../../trunk-groups/index.html).
For how priority, weight, and re-routing work, see [Service failover
and load
balancing](../call-handling-and-delivery/service-failover-and-load-balancing.html#inbound-sip-failover-load-balancing).

![Trunk Group tab of the Create Inbound SIP Trunk form with Trunk Group, Priority, Weight, and Re-routing Disconnect
Codes fields](../../../../_images/trunk_group1.png)


**Fig. 5.** Configure Trunk Group membership

### Authorization

Enable SIP digest authentication when the receiving SIP server requires
DIDWW to authenticate its `INVITE` requests.

Field definitions, available values, and constraints are listed in
[Authorization settings](../sip-trunk-settings-reference.html#inbound-sip-trunks-settings-authorization).
For how authentication applies in each direction, see [Authentication
direction](../sip-connectivity.html#inbound-sip-connectivity-authentication-direction). For the
supported methods, see [Authentication](../sip-connectivity-reference.html#inbound-sip-connectivity-authentication) in
the SIP connectivity reference.

Note

**Authorization** credentials are supplied by the receiving SIP system and
used by DIDWW when sending calls. They are different from the
system-generated credentials that a customer endpoint uses to register a
**Dynamic Registration** trunk with DIDWW.

![Authorization tab of the Create Inbound SIP Trunk form with Enable Authorization turned on and Auth User, Auth
Password, From User, and From Domain fields](../../../../_images/authorization.png)


**Fig. 6.** Configure Authorization

### Media & DTMF

Configure accepted RTP sources, codecs, media encryption, DTMF handling, and
RTP behavior.

Field definitions, available values, and constraints are listed in
[Media and DTMF settings](../sip-trunk-settings-reference.html#inbound-sip-trunks-settings-media-dtmf).
For supported [codecs](../sip-connectivity-reference.html#inbound-sip-connectivity-codecs), [DTMF
methods](../sip-connectivity-reference.html#inbound-sip-connectivity-dtmf), [media
encryption](../sip-connectivity-reference.html#inbound-sip-connectivity-encryption), and [RTP
addresses](../sip-connectivity-reference.html#inbound-sip-connectivity-rtp), see the SIP connectivity reference. For how
signaling and media paths, codec negotiation, and DTMF interoperability work,
see [Signaling and media paths](../sip-connectivity.html#inbound-sip-connectivity-signaling-media-paths) and [Codec and
DTMF
interoperability](../sip-connectivity.html#inbound-sip-connectivity-codec-dtmf).

Retain the default codecs and DTMF modes unless the receiving system requires
different values.

Enable **Force Symmetric RTP** when the receiving system requires symmetric
RTP for NAT traversal. Enable **RTP Ping** when both systems use symmetric RTP
and each waits for the other system to send media first.

Warning

**Force Symmetric RTP** can expose a call to an RTP Bleed attack. When
enabling it, restrict accepted media sources with **Allowed RTP IP
Addresses**. See [Force Symmetric RTP](../sip-trunk-settings-reference.html#inbound-sip-force-symmetric-rtp-warning).

![Media & DTMF tab of the Create Inbound SIP Trunk form with codecs, SRTP Mode, DTMF modes, RTP Timeout, Force Symmetric
RTP and its RTP Bleed warning, and RTP Ping](../../../../_images/media_and_dtmf.png)


**Fig. 7.** Configure Media & DTMF

### Signalling

Configure SIP session timers, call timeouts, failover timing, and limits for
SIP `REFER` transfers and SIP `301` and `302` redirects.

Field definitions, available values, and constraints are listed in
[Signalling settings](../sip-trunk-settings-reference.html#inbound-sip-trunks-settings-signalling).

Retain the default timer values unless the receiving system or routing design
requires different behavior. For how the DNS SRV failover timer and Ringing
Timeout affect failover, see [Service failover and load
balancing](../call-handling-and-delivery/service-failover-and-load-balancing.html#inbound-sip-failover-load-balancing).
For SIP `REFER` transfers and SIP
`301` and `302` redirects, see [Inbound call redirection](../call-handling-and-delivery/inbound-call-redirection.html).

Note

**Max Transfers** and **Max 30x Redirects** default to `0`, which disables
the corresponding method. Increase the limit only when the receiving system
is expected to request that behavior.

![Signalling tab of the Create Inbound SIP Trunk form with SIP Session Timers, SIP Timer B, DNS SRV Failover Timer,
Ringing Timeout, Max Transfers, and Max 30x Redirects](../../../../_images/signaling.png)


**Fig. 8.** Configure Signalling

### Additional Services

Configure [STIR/SHAKEN](../call-information/stir-shaken-headers.html)
information, Diversion headers, and [CNAM lookup](../call-information/cnam-in.html).

Field definitions, available values, and constraints are listed in
[Additional Services settings](../sip-trunk-settings-reference.html#inbound-sip-trunks-settings-additional-services).

Relaying the received `Identity` header requires account enablement, and the
customer is responsible for validating it. Contact [DIDWW Sales](mailto:sales%40didww.com) for enablement.

Warning

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

CNAM lookup adds a charge for each call from a U.S. number. If the trunk
belongs to a Trunk Group, see [Trunk Group behavior](../call-information/cnam-in.html#inbound-sip-cnam-in-trunk-groups).

![Additional Services tab of the Create Inbound SIP Trunk form with STIR SHAKEN Mode, Diversion Relay Policy, Diversion
Inject Mode, and Enable CNAM Lookup](../../../../_images/additional_services.png)


**Fig. 9.** Configure Additional Services

## Step 4: Create the trunk

1. Review the trunk configuration.
2. Click **Create**.

The new trunk appears on the **Inbound Trunks** page. For **Dynamic
Registration**, DIDWW displays the generated credentials and registrar
endpoints immediately after the trunk is created.

## Next steps

- [Assign the trunk to one or more DID numbers](../../../../phone-numbers/my-numbers/how-to-guides/assign-voice-trunk.html#user-panel-assign-trunk) before receiving calls. Each DID number must have
  sufficient [Capacity](../../../../phone-numbers/capacity/how-capacity-works.html).
- For **Dynamic Registration**, [view the registration credentials](view-sip-registration-credentials.html) and configure them on the SIP endpoint.
  Use [View registration history](view-registration-history.html) to confirm
  that registration is active.
- Place a test call, then review [Inbound Call Logs](../../../../logs-analytics/call-logs/inbound-logs.html#inbound-cdr-logs)
  if the call does not reach the SIP endpoint.

## Related resources

- [SIP trunk settings reference](../sip-trunk-settings-reference.html) -
  Look up field definitions, defaults, dependencies, available values, and
  limits.
- [SIP routing methods](../sip-routing-methods.html) - Compare Static
  Endpoint and Dynamic Registration behavior.
- [SIP connectivity](../sip-connectivity.html) - Understand signaling,
  media, authentication, security, and firewall requirements.
- [Assign a voice trunk](../../../../phone-numbers/my-numbers/how-to-guides/assign-voice-trunk.html#user-panel-assign-trunk) - Assign the trunk to
  one or more DID numbers.
- [Manage SIP trunks](manage-sip-trunks.html) - View, edit, or delete an
  existing SIP trunk.

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