# SIP Trunk 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. --- ## Create a New SIP Trunk To create an inbound SIP trunk, follow these steps: > 1. Navigate to the **Voice** section in the left-hand menu. > 2. Select **Inbound Trunks** from the submenu. > 3. Click the **Create New** button in the top-right corner of the screen. > 4. From the dropdown menu, select **SIP Trunk**. [![Adding a new SIP Trunk](../../_images/fig176.png)](../../_images/fig176.png) **Fig. 1.** Adding a new SIP Trunk --- ## Quick SIP Trunk Configuration Guide To quickly configure a SIP trunk, enter the required information below: 1. Enter a **Name** - Provide a unique name to identify the trunk. 2. Enter your **Host** - Specify the public IP address or domain name of your server. 3. Click the **Create** button to save and activate your configuration. Important Because the preferred server is set to **Auto**, your system must allow traffic from all [DIDWW SIP IPs](technical-data/sip.html#service-did-sip). [![Basic SIP trunk configuration example](../../_images/fig258.png)](../../_images/fig258.png) **Fig. 2.** Basic SIP trunk configuration example --- ## Advanced SIP Trunk Configuration Guide For advanced configurations, you can customize additional SIP trunk settings. **General** Core routing, addressing, and SIP behavior settings. [General](#user-panel-sipin-trunk-general-settings) **Number Translations** Configure caller ID formatting, prefixes, and number lists. [Number Translations](#user-panel-sipin-trunk-number-translations) **Trunk Group** Assign trunks to groups for failover and load balancing. [Trunk Group](#user-panel-sipin-trunk-trunk-group-configuration) **Authorization** Configure additional SIP digest authentication for the trunk. [Authorization](#user-panel-sipin-trunk-authentication) **Media & DTMF** Configure codecs, RTP security, DTMF methods, and media handling. [Media & DTMF](#user-panel-sipin-trunk-media-dtmf) **Signalling** Manage session timers, timeout policies, and call redirection behavior. [Signalling](#user-panel-sipin-trunk-general-advanced-signaling-settings) **Additional Services** Configure STIR/SHAKEN, diversion handling, and CNAM lookup. [Additional Services](#user-panel-sipin-trunk-additional-services) --- ### General The **General** tab defines how inbound SIP calls are routed to your SIP endpoint. You can configure the following settings: #### General properties - **Name** - Enter a unique name to identify the trunk. - **Capacity Limit** - Specify the maximum number of simultaneous calls allowed for this trunk. #### Type The **Type** setting determines **how DIDWW delivers calls to your endpoint**. *Static Endpoint* Use **Static Endpoint** when your SIP endpoint has a fixed, reachable address. DIDWW delivers calls to the configured SIP URI, built from **User Part of R-URI**, **Host**, and **Port**. - **User Part of R-URI** - Define the user part of the `R-URI` in the INVITE request. **Placeholder Variables** - `{DID}` - Inserts the called DID in E.164. - `{CALL_CPC}` - Calling party category. See [CPC usage](technical-data/cpc-usage.html#upcpc). - **Host** - Host part of the `R-URI` in the SIP `INVITE` request. This can be an IP address or a domain. - **Port** - Port part of the `R-URI` in the SIP `INVITE` request. **Port Auto-Resolution Behavior** If **Port** is left empty, DIDWW first attempts an `SRV` record lookup. If no `SRV` record is found, it falls back to resolving the `A` record. - **Resolve R-URI** - Replace the host part of the `R-URI` with the resolved IP address. - **Network Protocol** - Select the IP protocol preference used when resolving the host. | Option | Description | | --- | --- | | IPv4 only | Use IPv4 exclusively. | | IPv6 only | Use IPv6 exclusively. | | Any | Use either IPv4 or IPv6. | | Prefer IPv4 over IPv6 | Prefer IPv4 but fallback to IPv6. | | Prefer IPv6 over IPv4 | Prefer IPv6 but fallback to IPv4. | - **Transport** - Choose the protocol for SIP signaling: **UDP**, **TCP**, or **TLS**. - **Preferred Server** - Choose the DIDWW Point of Presence (POP) for routing: - **Auto (recommended):** Let DIDWW select the optimal SBC dynamically - **US:** LA, MIA, NY - **Germany:** FRA - **Singapore:** SG - **Hong Kong:** HK - **Netherlands:** AMS Important - When using **Auto**, allow all inbound [DIDWW signaling and RTP IPs](technical-data/sip.html#service-did-sip) on your equipment. - The **Auto** preferred server option acts as a failover. If there is any interruption on a single POP, calls are routed through an alternative DIDWW POP to maintain call continuity. - Ensure all POPs are enabled in [settings](settings/index.html#trunks-settings). Disabling any POP may cause unnecessary routing hops. **Routing Examples by Preferred Server** The following figures illustrate how incoming calls are routed based on the selected preferred server: 1. **Preferred Server: FRA** - Calls from the PSTN network reach the DIDWW HK SBC and are routed to the FRA POP. > [![Preferred Server: FRA](../../_images/fig2.svg)](../../_images/fig2.svg) > > > **Fig. 3.** Routing example with FRA as the preferred server. 2. **Preferred Server: Auto** - Calls are dynamically routed from the same DIDWW SBC that received them. > [![Preferred Server: Auto](../../_images/fig3.svg)](../../_images/fig3.svg) > > > **Fig. 4.** Routing example with Auto as the preferred server. [![General tab](../../_images/fig337.png)](../../_images/fig337.png) **Fig. 5.** General tab for static endpoint type *Dynamic Registration* ## Use **Dynamic Registration** when your PBX or SBC has a dynamic IP address, operates behind NAT, or when you prefer registration-based routing. Your SIP peer initiates and maintains registration, and DIDWW sends incoming calls to the **Contact address** learned during the SIP `REGISTER` process. - **Use DID in R-URI** - Replaces the user part of the `R-URI` in the SIP `INVITE` with the DID number, instead of the user part received in the `Contact` header during registration. Important - DIDWW generates the registration credentials and displays them immediately after the trunk is created. - A maximum of **10 simultaneous registrations** are supported per trunk. - View your registration username and password by clicking the **trunk name** on the **Inbound Trunks** page. For detailed steps, see [how to view credentials](#view-inbound-trunk-registration-credentials). [![General tab for dynamic registration type](../../_images/register.png)](../../_images/register.png) **Fig. 6.** General tab for dynamic registration type --- ### Number Translations The **Number Translations** tab defines the caller ID format delivered to your SIP endpoint. - **CLI Format** - Select how the Caller ID (CLI) is formatted for incoming calls: | Option | Description | | --- | --- | | **E.164** | Converts the CLI to E.164 format (Country Code + Area Code + Number). This is the default option. | | **Raw** | Passes the caller ID unchanged. | | **Local** | Converts the CLI to local format (Area Code + Number). | - **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](number-lists.html#user-panel-voice-in-numberlist) documentation page. Warning - CLI format conversion may not work correctly for calls originating from outside the country of the DID. - Number Lists work by **matching** the inbound CLI. If you modify the **CLI format** and **CLI prefix**, you may need to adjust your numbers in the **Number List** accordingly. [![Number translations tab](../../_images/number_translations1.png)](../../_images/number_translations1.png) **Fig. 7.** Number translations tab --- ### Trunk Group 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**. You can configure the following settings: - **Trunk Group** - Select the trunk group to assign the trunk to. A trunk can belong to only one trunk group. If no trunk groups exist, refer to the [Trunk Group Creation Guide](creating-a-new-trunk-group.html#user-panel-trunk-group). - **Priority** - Set the trunk priority within the trunk group. Trunks with lower values are tried first. Trunks with the same priority are ordered by weight. For more details, see [**RFC 2782**](https://datatracker.ietf.org/doc/html/rfc2782.html). - **Weight** - Set the relative weight for trunks with the same priority. A higher value increases the likelihood that the trunk is selected. For more details, see [**RFC 2782**](https://datatracker.ietf.org/doc/html/rfc2782.html). - **Re-routing Disconnect Codes** - Select the disconnect codes that trigger call rerouting to another trunk in the group. For available codes and their meanings, see [SIP Response Codes](technical-data/call-logs-response-codes.html#inbound-technical-sip-response-codes-in-call-logs). [![Trunk Group tab](../../_images/trunk_group1.png)](../../_images/trunk_group1.png) **Fig. 8.** Trunk Group tab --- ### Authorization Authorization 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. To enable authorization, configure the following settings: > | **Setting** | **Description** | > | --- | --- | > | **Enable Authorization** | Enables authentication for the SIP trunk for incoming INVITE requests from DIDWW. | > | **Auth User** | Defines the username for authentication. | > | **Auth Password** | Defines the password for authentication. | > | **From User** | Specifies a custom user in the **From** field instead of the Caller ID. | > | **From Domain** | Specifies a custom **From** domain in SIP messages. | [![Authorization tab](../../_images/authorization.png)](../../_images/authorization.png) **Fig. 9.** Authorization tab --- ### Media & DTMF The **Media & DTMF** section allows you to configure codec preferences, Dual-Tone Multi-Frequency (DTMF) signaling, and real-time transport protocol (RTP) settings. You can configure the following settings: - **Allowed RTP IP Addresses** - Define the IP addresses that are allowed to send RTP media packets to this SIP trunk. To restrict RTP sources, enter up to 10 IP addresses or subnets in the following format: `IPv4[/mask]` or `IPv6[/mask]`, e.g., `203.0.113.5/32, 2001:db8::1/128`. Note - If left blank, RTP packets are accepted from any source. - If the mask is omitted, `/32` is used for IPv4 and `/128` is used for IPv6. - **Codecs** - Select the codecs to include in the SDP offer. Selected codecs can be removed individually, or you can click **Remove All** to clear the selection. Reorder selected codecs to set their priority; the first codec has the highest priority. For a list of supported codecs, see [Supported Codecs](technical-data/sip.html#service-did-sip-codecs). - **SRTP Mode** - Select the Secure Real-time Transport Protocol (SRTP) mode used for media encryption. DIDWW supports **TLS** for secure SIP signaling and [SRTP](https://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol) for media encryption. | Option | Description | | --- | --- | | **Disabled** | Do not use SRTP media encryption. | | **SRTP SDES** | Negotiate SRTP keys using [Session Description Protocol Security Descriptions](https://en.wikipedia.org/wiki/SDES). | | **SRTP DTLS** | Negotiate SRTP keys using [Datagram Transport Layer Security](https://en.wikipedia.org/wiki/Datagram_Transport_Layer_Security). | | **SRTP ZRTP** | Negotiate SRTP keys using the [ZRTP key agreement protocol](https://en.wikipedia.org/wiki/ZRTP). | - **DTMF Send Mode** - Select how Dual-Tone Multi-Frequency (DTMF) signals are sent from DIDWW to your equipment. | Option | Description | | --- | --- | | **Disable sending** | Do not send DTMF signals to your equipment. | | **RFC 2833** | Send DTMF events using the `telephone-event` RTP payload. This is the default option. | | **SIP INFO application/dtmf-relay** | Send DTMF events in SIP `INFO` requests using the `application/dtmf-relay` content type. | | **SIP INFO application/dtmf** | Send DTMF events in SIP `INFO` requests using the `application/dtmf` content type. | - **DTMF Receive Mode** - Select how Dual-Tone Multi-Frequency (DTMF) signals are received by DIDWW from your equipment. For more details, see [DTMF options](technical-data/sip.html#service-did-sip-dtmf). | Option | Description | | --- | --- | | **RFC 2833** | Receive DTMF events using the `telephone-event` RTP payload. This is the default option. | | **SIP INFO application/dtmf-relay OR application/dtmf** | Receive DTMF events in SIP `INFO` requests using either the `application/dtmf-relay` or `application/dtmf` content type. | | **RFC 2833 OR SIP INFO** | Receive DTMF events using either the `telephone-event` RTP payload or SIP `INFO` requests. | - **RTP Timeout** - Define the maximum time, in seconds, before a call is disconnected if no RTP packets are received. Note RTP Timeout value must be between 5 and 600 seconds. - **Force Symmetric RTP** - Send media to the source IP address and port of received RTP packets, ignoring the remote address negotiated in the SDP. Warning Enabling **Force Symmetric RTP** may expose the call to an RTP Bleed attack, where an attacker can hijack the RTP stream by sending crafted packets from a different source address. Restrict accepted RTP sources using **Allowed RTP IP Addresses** to mitigate this risk. - **Symmetric RTP Ignore RTCP** - When **Force Symmetric RTP** is enabled, use only RTP packets when switching the media path and ignore RTCP packets. - **RTP Ping** - After SDP negotiation is complete, send several empty RTP packets to trigger RTP transmission on the remote side. [![Media & DTMF tab](../../_images/media_and_dtmf.png)](../../_images/media_and_dtmf.png) **Fig. 10.** Media & DTMF tab --- ### Signalling The **Signalling** tab allows you to configure **SIP session management, transaction timeouts, failover behavior, call transfers, and redirects**. These optional settings help maintain **call stability**, prevent **unnecessary call drops**, and optimize **failover handling** in case of network failures. You can configure the following settings: - **SIP Session Timers** - Enable DIDWW to initiate SIP session timer negotiation. Session timers periodically refresh the SIP session and terminate it when the remote endpoint stops responding. For more details, see [RFC 4028](https://datatracker.ietf.org/doc/html/rfc4028). - **SST MIN Timer** - Set the minimum session interval. The default value is 600 seconds. - **SST MAX Timer** - Set the maximum session interval. The default value is 900 seconds. - **SST Session Expires** - Define the `Session-Expires` header value. The value must be within the range defined by **SST MIN Timer** and **SST MAX Timer**. - **SST Refresh Method** - Select the SIP method used to refresh the session. | Option | Description | | --- | --- | | **Invite** | Refresh the session using an `INVITE` request. This is the default option. | | **Update** | Refresh the session using an `UPDATE` request. | | **Update fallback Invite** | Use an `UPDATE` request when supported and fall back to `INVITE` when necessary. | - **SST Accept 501** - Do not terminate the call after receiving a SIP `501` response to a session update request. - **SIP Timer B** - Set the timeout for an `INVITE` transaction, in milliseconds. The default value is 8000 ms. For more details, see [RFC 3261 Section 17.1.1.2](https://www.rfc-editor.org/rfc/rfc3261.html#section-17.1.1.2). - **DNS SRV Failover Timer** - Set how long DIDWW waits for an `INVITE` transaction before rerouting to the next DNS SRV target, in milliseconds. The default value is 2000 ms. - **Ringing Timeout** - Set the maximum time to wait for a `200 OK` response after receiving an `18x` response, in seconds. When exceeded, the routing attempt ends with the **Ringing timeout** disconnect code. - **Max Transfers** - Set the maximum number of SIP `REFER` requests processed for call transfers. The default value is 0. - **Max 30x Redirects** - Set the maximum number of SIP `301` or `302` redirect responses followed during call rerouting. The default value is 0. [![Signalling tab](../../_images/signaling.png)](../../_images/signaling.png) **Fig. 11.** Signalling tab --- ### Additional Services The **Additional Services** tab contains STIR/SHAKEN handling, diversion settings, and CNAM lookup. You can configure the following settings: - **STIR SHAKEN Mode** - Select how STIR/SHAKEN information is delivered in SIP signaling. The STIR/SHAKEN framework helps prevent caller ID spoofing by verifying the authenticity of the calling number. | Option | Description | | --- | --- | | **Disabled** | Do not include additional STIR/SHAKEN headers. | | **Relay Identity header** | Relay the Identity header in SIP `INVITE` messages. | | **Add PAI, P-Attestation-Indicator, P-Origination-ID** | Add the listed attestation and identity headers to the SIP `INVITE` request. | | **Relay Identity header + Add PAI, P-Attestation-Indicator, P-Origination-ID** | Relay the Identity header and add the listed attestation headers to the SIP `INVITE` request. | | **Add P-Stir-Verstat, P-Attestation-Indicator, P-Origination-ID** | Add verification status and attestation headers to the SIP `INVITE` request. | Note The **Relay Identity header** option 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](mailto:sales%40didww.com). - **Diversion Relay Policy** - Control how PSTN Diversion information is delivered in SIP signaling. | Option | Description | | --- | --- | | **Disabled** | Do not relay the Diversion header. | | **Diversion header in SIP URI format** | Format and relay the Diversion header as a SIP URI, for example, `sip:user@sip.didww.com`. | | **Diversion header in TEL URI format** | Format and relay the Diversion header as a TEL URI, for example, `tel:+123456789`. | - **Diversion Inject Mode** - Control whether DIDWW adds its own Diversion information to outgoing SIP messages. | Option | Description | | --- | --- | | **Disabled** | Do not add a DIDWW-generated Diversion header. | | **Add Diversion header with DID number** | Add a DIDWW-generated Diversion header containing the associated DID number. | Note If both **Diversion Relay Policy** and **Diversion Inject Mode** are enabled at the same time, two Diversion headers may be included in the SIP message, which may cause unexpected behavior on your system. - **Enable CNAM Lookup** - Retrieve and display the caller name associated with a US CLI. The caller name is delivered as the display name in the SIP `From` header. - The system queries a remote CNAM database when a call originates from the US. - Billing applies only if the lookup is successful. Warning If a **Trunk Group** is used and any of its trunks have CNAM lookup enabled: - A CNAM lookup will be performed for all incoming calls. - A successful lookup will be billed additionally per call. - The retrieved CNAM value will be displayed only for trunks with CNAM lookup enabled. **CNAM in the SIP From Header** The following examples show how CNAM appears in the **From** header: > - **CNAM lookup not enabled:** `"12899230448" ;tag=24-6917EC3D-6453984E000C1C1E-5EA5B700` > - **CNAM lookup enabled and lookup successful:** `"KRIS TOTAL" ;tag=24-7EB42886-645381060002B85A-5EB5C700` > - **CNAM lookup enabled but lookup failed:** `"Unavailable" ;tag=24-7EB42886-645381060002B85A-5EB5C700` The CNAM value is also displayed in the **Source Name** field in [inbound call logs](../../logs-analytics/call-logs/inbound-logs.html#inbound-cdr-logs). [![Additional Services tab with STIR/SHAKEN, diversion, and CNAM lookup settings](../../_images/additional_services.png)](../../_images/additional_services.png) **Fig. 12.** Additional Services tab --- ## View Inbound SIP Registration Credentials If your inbound SIP trunk uses [Dynamic Registration](#user-panel-sip-trunk-registration), the system automatically generates a unique set of credentials. You can view these credentials by clicking on the **trunk name** on the **Inbound Trunks** page. [![Accessing SIP Registration credentials on the Inbound Trunks page](../../_images/credentials11.png)](../../_images/credentials11.png) **Fig. 13.** Accessing SIP Registration credentials. A **SIP Credentials** pop-up window will appear, showing the necessary details for registering your SIP endpoint: The pop-up displays the following information: | **Field** | **Description** | | --- | --- | | **Username** | The unique, system-generated username required for your SIP endpoint to register with DIDWW. | | **Password** | The password required for registration. Select the **eye** icon to view the password in clear text. | | **Host** | The DIDWW SIP server hostname your SIP endpoint must register to (e.g., `sip.didww.com`). For more details, see [General SIP Information](technical-data/sip.html#service-did-sip). | [![My SIP Trunk SIP Credentials pop-up window](../../_images/credentials2.png)](../../_images/credentials2.png) **Fig. 14.** SIP Registration Credentials pop-up. --- ## Edit SIP Trunk To edit an existing SIP Trunk, follow these steps: > 1. Navigate to the **Voice** section in the left-hand menu. > 2. Select **Inbound Trunks** from the submenu. > 3. Locate the SIP trunk that you want to edit and click the [![actions](../../_images/three_dots_action_button3.png)](../../_images/three_dots_action_button3.png) > button. > 4. Click **Edit**. > > [![Actions Button.](../../_images/actions_edit3.png)](../../_images/actions_edit3.png) > > > **Fig. 15.** Actions Button. > > 5. In **Edit Inbound SIP Trunk**, make the required changes and click **Submit** to save them. > > [![Edit Inbound SIP Trunk.](../../_images/edit_screen.png)](../../_images/edit_screen.png) > > > **Fig. 16.** Edit Inbound SIP Trunk. --- ## View Registration History Note The **Registration History** is available only for trunks that use [Dynamic Registration](#user-panel-sip-trunk-registration). It displays recent successful registration intervals for the selected trunk. Follow these steps to view the Registration History: > 1. Go to the **Voice** section in the menu. > 2. Select **Inbound Trunks**. > 3. Find the trunk you want to inspect and open the **Actions** menu. > 4. Click **View Registration History**. > > [![Opening the Registration History window from the Actions menu.](../../_images/registration_statistics1.png)](../../_images/registration_statistics1.png) > > > **Fig. 17.** Opening the Registration History window. ### Registration History Chart The Registration History chart shows **successful registration intervals**, indicating when a valid `Contact` header was active for call routing. Note - The chart shows the times when the trunk was successfully registered and online. Individual SIP `REGISTER` requests or failed attempts are not displayed. - Registration data is available for the **last 7 days**. If no successful registrations occurred in this period, no data will appear. - Registration activity is shown in **one-minute intervals**. Hover over any point in the chart to view details about that registration interval, including: | **Field** | **Description** | | --- | --- | | **Contact** | The Contact address used by your system during registration. | | **IP/Port** | The source IP address and port used for the registration. | | **Transport Protocol** | The transport protocol used for the registration (UDP, TCP, or TLS). | | **User Agent** | The User-Agent string reported by the registering system. | | **Expires** | The registration lifetime (Expires value) provided by your endpoint. | [![SIP Registration History chart.](../../_images/registration_statistics2.png)](../../_images/registration_statistics2.png) **Fig. 18.** SIP Registration History chart. You can zoom in to focus on a specific time range. Click and drag across the chart to select the period you want to view. [![Zooming and narrowing the Registration History chart.](../../_images/registration_statistics_gif.gif)](../../_images/registration_statistics_gif.gif) **Fig. 19.** Zooming into a selected part of the chart. --- ## Delete SIP Trunk(s) You can either delete a single SIP trunk or delete multiple SIP trunks by using batch actions. > Note > > Trunks assigned to DID numbers or in use cannot be deleted. ### Delete a Single SIP Trunk To delete an existing SIP Trunk, follow these steps: > 1. Navigate to the **Voice** section in the left-hand menu. > 2. Select **Inbound Trunks** from the submenu. > 3. Locate the SIP trunk that you want to delete and click the actions button. > 4. Click **Delete**. > > [![Actions Button.](../../_images/actions_delete4.png)](../../_images/actions_delete4.png) > > > **Fig. 20.** Actions Button. > > 5. In the pop-up Delete Trunk(s) window click **Delete** to confirm the Trunk deletion. > > [![Delete a Single Trunk.](../../_images/delete_one.png)](../../_images/delete_one.png) > > > **Fig. 21.** Delete a Single Trunk. ### Delete Multiple SIP Trunks To delete **multiple** existing SIP Trunks, follow these steps: > 1. Navigate to the **Voice** section in the left-hand menu. > 2. Select **Inbound Trunks** from the submenu. > 3. Select the SIP trunks that you want to delete. > 4. Click on the **Batch Actions** button and click **Delete**. > > [![Batch Actions Button.](../../_images/delete_batch_actions3.png)](../../_images/delete_batch_actions3.png) > > > **Fig. 22.** Batch Actions Button. > > 5. In the pop-up Delete Trunk(s) window click **Delete** to confirm the Trunk deletion. > > [![Delete Multiple Trunks](../../_images/delete_batch.png)](../../_images/delete_batch.png) > > > **Fig. 23.** Delete Multiple Trunks. --- ## Additional Information **General SIP Information** Learn more about DIDWW inbound signaling, RTP IPs, supported codecs, DTMF transport methods, and encryption. [General SIP Information](technical-data/sip.html#service-did-sip) **Assign Trunks to DID Numbers** Learn how to configure DID numbers with trunks and trunk groups for routing inbound calls. [Assign a voice trunk](../../phone-numbers/my-numbers/how-to-guides/assign-voice-trunk.html#user-panel-assign-trunk) **Create a Trunk Group** Discover how to create and configure a trunk group for failover and load balancing. [Trunk Groups](creating-a-new-trunk-group.html#user-panel-trunk-group) **STIR/SHAKEN** Understand STIR/SHAKEN, how it works, and the available handling modes for the Voice IN service. [STIR/SHAKEN](technical-data/stir-shaken.html#service-did-stir-shaken) **CPC Usage** Explore the Calling Party Category (CPC), its functionality, and usage examples in SIP URIs. [CPC Usage](technical-data/cpc-usage.html#upcpc) On this page