> ## Documentation Index
> Fetch the complete documentation index at: https://koreai-content-gov.mintlify.site/llms.txt
> Use this file to discover all available pages before exploring further.

# Voice Channels

<Badge icon="arrow-left" color="gray">[Back to Channels](/agent-platform/channels#voice)</Badge>

## Set up realtime LLM voice

Supports live voice conversations using realtime LLM models (OpenAI Realtime, Gemini Live) through the Kore.ai Voice Gateway or a Bring Your Own SIP connection.

To configure a Realtime LLM Voice channel, go to **Deployments** > **Channels** > **VOICE** > **Realtime LLM Voice** > **+ New** and complete the following steps in order.

<Steps>
  <Step title="Set Up the Connection">
    On the **New Realtime LLM Voice Connection** page, provide the following fields.

    | Field                              | Description                                                                                                                                                                                                                                                                               |
    | ---------------------------------- | ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Display Name**                   | A label for the connection.                                                                                                                                                                                                                                                               |
    | **Connection Name**                | A system name for the connection, such as `realtime-voice-prod`. Up to 255 characters.                                                                                                                                                                                                    |
    | **Environment**                    | The deployment the connection binds to. Select an environment to follow its active deployment automatically, or leave the default to use the working copy. When no environment has an active deployment yet, create a deployment first to bind the connection to a published environment. |
    | **Provider Verification Strength** | Controls how strictly the platform trusts the provider's identity assertions for session continuity and linking. **Weak** is the default. Choose **Strong** only for channels or providers whose identity assertions you trust.                                                           |
    | **Initial Status**                 | **Active** connections are usable immediately. **Inactive** connections stay saved but disabled until you activate them.                                                                                                                                                                  |

    Click **Create**.
  </Step>

  <Step title="Choose a Call Routing Option (Kore.ai Voice Gateway)">
    After creating the channel, go to the **Configuration** tab. If you choose **Voice Provider: Kore.ai Voice Gateway**, set the following fields under **Kore.ai Voice Gateway**.

    | Route                  | Description                                                                                                                                              |
    | ---------------------- | -------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Soft Phone** (FREE)  | Dial the agent from your browser with no DID purchase or telephony charges. Use it for evals, QA, and demos. **Launch dialer** opens the browser dialer. |
    | **Phone Number (DID)** | Purchase a real phone number through Twilio. Disable this route to stop DID listing and call routing while keeping the number saved.                     |

    <Note>If you select **Phone Number (DID)**, complete the next step. If you selected **Voice Provider: Bring Your Own SIP** instead, skip to Configure SIP Settings.</Note>
  </Step>

  <Step title="Configure the Phone Number (DID Only)">
    Set the **Provider** to **Twilio**, toggle **Inbound calls** and **Outbound calls**, and select a **Country**, **Number Type**, and optional **Area Code**. Select **Search Available Numbers** to list numbers that match your filters.
  </Step>

  <Step title="Configure SIP Settings (Bring Your Own SIP)">
    Once the channel is created, select it, then go to the **Configuration** tab. If you chose **Voice Provider: Bring Your Own SIP**, set the following fields under **SIP Configuration**.

    | Field                      | Description                                                              |
    | -------------------------- | ------------------------------------------------------------------------ |
    | **Our SIP Server Address** | Configure your SIP provider to send `INVITE` to this address.            |
    | **SIP Gateway IP Address** | Your SIP trunk IP address — optionally include a port (default: `5060`). |
    | **DID / Phone Number**     | The phone number (DID) your SIP provider routes to this SIP gateway.     |
  </Step>

  <Step title="Configure Realtime LLM Voice Settings">
    Under **Realtime LLM Voice settings**, set the following fields.

    | Setting                  | Description                                                                                                                                       |
    | ------------------------ | ------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Semantic convergence** | Controls whether realtime voice turns use the coordinated utterance runtime.                                                                      |
    | **S2S Provider**         | The realtime speech-to-speech model: **Google Gemini Live (S2S)**, **Grok Realtime (S2S)**, or **OpenAI Realtime**. Each uses tenant credentials. |
    | **Noise isolation**      | Applies server-side noise reduction to inbound call audio to filter out background voices and ambient noise.                                      |
  </Step>

  <Step title="Save the Channel">
    Click **Save Changes** to apply the configuration.
  </Step>
</Steps>

### Manage an existing realtime LLM voice channel

Open a channel from **Deployments** > **Channels** > **VOICE** > **Realtime LLM Voice** shows three tabs: **Overview**, **Configuration**, and **Deployment**.

<Tabs>
  <Tab title="Overview">
    The **Connection Summary** shows these fields.

    | Field           | Description                       |
    | --------------- | --------------------------------- |
    | **Status**      | for example: *Active*             |
    | **Created**     | Timestamp the channel was created |
    | **Updated**     | Timestamp of the last change      |
    | **Environment** | for example: `dev`                |
    | **Source Type** | `Realtime LLM Voice Channel`      |
    | **Source ID**   | Unique channel identifier (UUID)  |
  </Tab>

  <Tab title="Configuration">
    The channel includes the same **General** (Display Name), **Provider Verification Strength**, call routing, and **Realtime LLM Voice Settings** options described in [Set up Realtime LLM Voice](#set-up-realtime-llm-voice). These settings can be edited at any time. Any changes to call routing or Realtime LLM Voice settings take effect immediately after you click **Save Changes**.
  </Tab>

  <Tab title="Deployment">
    The **Deployment** tab shows these fields.

    | Field                 | Description                                                                                                             |
    | --------------------- | ----------------------------------------------------------------------------------------------------------------------- |
    | **Agent Version**     | An **Environment** dropdown (for example: `Development`) that selects which deployed version handles incoming messages. |
    | **Active Deployment** | Shows the bound deployment version and when it was deployed.                                                            |

    Click **Save Changes** after switching the environment/version.
  </Tab>
</Tabs>

## Set up pipeline voice

It's a voice channel that handles calls through separate speech-to-text, LLM, and text-to-speech stages, connected through Kore.ai Voice Gateway. Unlike Realtime LLM Voice, which uses a single speech-to-speech model, this pipeline lets you choose and configure the ASR (speech-to-text) and TTS (text-to-speech) providers independently.

<Note>Choose your voice provider before you begin — it determines which additional fields apply later in this setup.</Note>

To configure a Pipeline Voice channel, go to **Deployments** > **Channels** > **VOICE** > **Pipeline Voice** > **+ New** and complete the following steps in order.

<Steps>
  <Step title="Set Up the Connection">
    On the **New Pipeline Voice Connection** page, provide the following fields.

    | Field                              | Description                                                                                                                                                                                                                                                                               |
    | ---------------------------------- | ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Display Name**                   | A label for the connection.                                                                                                                                                                                                                                                               |
    | **Connection Name**                | A system name for the connection, such as `pipeline-voice-prod`. Up to 255 characters.                                                                                                                                                                                                    |
    | **Environment**                    | The deployment the connection binds to. Select an environment to follow its active deployment automatically, or leave the default to use the working copy. When no environment has an active deployment yet, create a deployment first to bind the connection to a published environment. |
    | **Provider Verification Strength** | Controls how strictly the platform trusts the provider's identity assertions for session continuity and linking. **Weak** is set by default. Choose **Strong** only for channels/providers whose identity assertions you trust for stronger continuity and linking decisions.             |
    | **Initial Status**                 | **Active** connections are usable immediately. **Inactive** connections stay saved but disabled until you activate them.                                                                                                                                                                  |

    Click **Create**.
  </Step>

  <Step title="Choose a Voice Provider">
    After creating the channel, go to the **Configuration** tab. Under **Pipeline Voice Settings**, set **Voice Provider** to either **Kore.ai Voice Gateway** or **Bring Your Own SIP** — this determines which call routing fields appear next.
  </Step>

  <Step title="Configure Call Routing (Kore.ai Voice Gateway)">
    If you chose **Kore.ai Voice Gateway**, set the following fields under **Call routing**. Choose how calls reach this agent — enable Soft Phone for browser testing, or add a DID when this channel needs a real phone number.

    | Route                  | Description                                                                                                                                                     |
    | ---------------------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Soft Phone** (FREE)  | Dial the agent from your browser. No DID purchase or telephony charges. Use this for evals, QA, and demos. Select **Launch dialer** to open the browser dialer. |
    | **Phone Number (DID)** | Purchase a real phone number through Twilio. Disable this route to stop DID listing and call routing while keeping the number saved.                            |

    <Note>If you enable **Phone Number (DID)**, complete the next step. If you chose **Voice Provider: Bring Your Own SIP** instead, skip to Configure SIP Settings.</Note>
  </Step>

  <Step title="Configure the Phone Number (DID Only)">
    Set the **Provider** to **Twilio**, toggle **Inbound calls** (route calls from this number into the voice agent) and **Outbound calls** (allow this number to be used for campaign dialout), and select a **Country** and **Number Type**. Optionally enter an **Area Code** to filter available local numbers. Select **Search Available Numbers** to list matching numbers.
  </Step>

  <Step title="Configure SIP Settings (Bring Your Own SIP)">
    If you chose **Voice Provider: Bring Your Own SIP**, set the following fields under **SIP Configuration**.

    | Field                      | Description                                                                                                                              |
    | -------------------------- | ---------------------------------------------------------------------------------------------------------------------------------------- |
    | **Our SIP Server Address** | Configure your SIP provider to send `INVITE` to this address, for example `korevg-np.kore.ai:5060`. Select **Copy** to copy the address. |
    | **SIP Gateway IP Address** | Your SIP trunk IP address — optionally include a port (default: `5060`), for example `203.0.113.50` or `203.0.113.50:5080`.              |
    | **DID / Phone Number**     | The phone number (DID) your SIP provider routes to this SIP gateway, for example `+14155551234` or `98492`.                              |
  </Step>

  <Step title="Configure Speech Recognition">
    Under **Speech Recognition**, set the following fields.

    | Field                 | Description                                                                                                                                                               |
    | --------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Provider**          | The speech-to-text credential set to use, for example `Deepgram Speech Credentials`.                                                                                      |
    | **Language**          | The recognition language, for example `Multilingual`.                                                                                                                     |
    | **STT Model**         | The recognition model, for example `Nova 3 (latest, most accurate)`.                                                                                                      |
    | **Custom vocabulary** | Domain-specific terms to bias recognition toward. Select **Add phrase** to add a row, then set the **Phrase** (for example `ACME, policy number`) and optional **Boost**. |
  </Step>

  <Step title="Configure Speech Synthesis">
    Under **Speech Synthesis**, set the following fields.

    | Field                                | Description                                                                                                              |
    | ------------------------------------ | ------------------------------------------------------------------------------------------------------------------------ |
    | **Provider**                         | The text-to-speech credential set to use, for example `ElevenLabs (TTS) Credentials` or `Deepgram Speech Credentials`.   |
    | **Language**                         | The synthesis language, for example `Arabic` or `English (PH)`.                                                          |
    | **Voice**                            | The voice used for synthesized speech, for example `Sarah - Mature, Reassuring, Confident - (american, young, female)`.  |
    | **Switch TTS for detected language** | When enabled, calls can change the TTS language, voice, and model after speech recognition detects a different language. |

    <Note>Available fields vary by provider. ElevenLabs exposes additional voice tuning and a preview option (see next step); other providers may only show **Provider**, **Language**, **Voice**, and **Switch TTS for detected language**.</Note>
  </Step>

  <Step title="Tune ElevenLabs Voice Settings (ElevenLabs Only)">
    If you selected an ElevenLabs voice, override playback and prosody for this channel connection under **ElevenLabs Voice Settings**.

    | Field                  | Description                                                                                                                         |
    | ---------------------- | ----------------------------------------------------------------------------------------------------------------------------------- |
    | **TTS Model**          | The synthesis model, for example `Eleven Multilingual v2`.                                                                          |
    | **Speed**              | Adjusts speaking pace. `1.0` is normal; lower slows the voice down, higher makes it faster. Range: `0.7`–`1.2`.                     |
    | **Stability**          | Controls consistency across generations. Higher is more stable and less expressive; lower allows more variation. Range: `0`–`1`.    |
    | **Similarity boost**   | Keeps the output closer to the selected voice. Higher values preserve more of the original voice character. Range: `0`–`1`.         |
    | **Style exaggeration** | Amplifies voice style and emotion. Higher values can sound more expressive but may increase latency or instability. Range: `0`–`1`. |
    | **Speaker boost**      | Enhances speaker similarity and clarity. Turning it off may reduce latency for some voices.                                         |

    Under **Preview Voice**, enter sample text (up to 500 characters) and select **Generate** to hear the configured voice before saving.
  </Step>

  <Step title="Configure Conversation Controls">
    Set the following fields.

    | Field                   | Description                                                                                                |
    | ----------------------- | ---------------------------------------------------------------------------------------------------------- |
    | **Barge-in**            | Allow the caller to interrupt the agent while it is speaking.                                              |
    | **Speech Timeout (ms)** | The time the agent waits for caller speech before it times out, for example `3000`.                        |
    | **Welcome Message**     | The opening prompt the agent speaks, such as `Hello! How can I help you today?`.                           |
    | **Noise isolation**     | Apply server-side noise reduction to inbound call audio to filter out background voices and ambient noise. |
  </Step>

  <Step title="Save the Channel">
    Click **Save Changes** to apply the configuration.
  </Step>
</Steps>

### Manage an existing Pipeline Voice channel

Open a channel from **Deployments** > **Channels** > **VOICE** > **Pipeline Voice** shows three tabs: **Overview**, **Configuration**, and **Deployment**.

<Tabs>
  <Tab title="Overview">
    The **Connection Summary** shows these fields.

    | Field           | Description                       |
    | --------------- | --------------------------------- |
    | **Status**      | for example: *Active*             |
    | **Created**     | Timestamp the channel was created |
    | **Updated**     | Timestamp of the last change      |
    | **Environment** | for example: `dev`                |
    | **Source Type** | `Channel Connection`              |
    | **Source ID**   | Unique channel identifier (UUID)  |
  </Tab>

  <Tab title="Configuration">
    Contains the same **General** (Display Name, Provider Verification Strength), voice provider and call routing, **Speech Recognition**, **Speech Synthesis** (including ElevenLabs voice tuning and preview, where applicable), and **Conversation Controls** described in [Set up Pipeline Voice](#set-up-pipeline-voice) above, editable at any time. Changing any of these settings here updates the live channel immediately after **Save Changes**.
  </Tab>

  <Tab title="Deployment">
    The **Deployment** tab shows these fields.

    | Field                 | Description                                                                                                             |
    | --------------------- | ----------------------------------------------------------------------------------------------------------------------- |
    | **Agent Version**     | An **Environment** dropdown (for example: `Development`) that selects which deployed version handles incoming messages. |
    | **Active Deployment** | Shows the bound deployment version and when it was deployed.                                                            |

    Click **Save Changes** after switching the environment/version.
  </Tab>
</Tabs>

## Set up Genesys audio connector

Configure Genesys Audio Connector for KoreVG-owned AudioHook execution.

<Note>Complete the Studio and Genesys Architect setup before you begin — it produces the AudioHook callback URL this connection needs, and determines how the connection is validated before it goes live. Also choose your voice provider before you begin — it determines which additional fields apply later in this setup.</Note>

To configure a Genesys Audio Connector channel, go to **Deployments** > **Channels** > **VOICE** > **Genesys Audio Connector** > **+ New** and complete the following steps in order.

<Steps>
  <Step title="Set Up the Connection">
    On the **New Genesys Audio Connector Connection** page, provide a **Display Name** (up to 255 characters), then set up **Audio Connector setup**, which creates a project-owned Audio Connector setup that targets KoreVG AudioHook execution by default.

    | Field                              | Description                                                                                                                                                                                                                     |
    | ---------------------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Genesys Organization ID**        | Your Genesys Cloud organization identifier.                                                                                                                                                                                     |
    | **Allowed Region**                 | The Genesys region this connection is scoped to, for example `US East (N. Virginia)`.                                                                                                                                           |
    | **Voice Deployment ID**            | The Genesys voice deployment this connector serves.                                                                                                                                                                             |
    | **AudioHook Target**               | The audio execution target, for example `KoreVG AudioHook`.                                                                                                                                                                     |
    | **KoreVG AudioHook URL**           | The AudioHook callback URL, for example `wss://voice.example.com/...`.                                                                                                                                                          |
    | **Activation Mode**                | The connection's rollout stage, for example `Diagnostics only`. Promote to live traffic only after Architect validation passes.                                                                                                 |
    | **API Key**                        | The API key used to authenticate AudioHook requests.                                                                                                                                                                            |
    | **Webhook Signature Secret**       | A secret used to verify inbound webhook signatures.                                                                                                                                                                             |
    | **Max Session Seconds**            | The maximum duration of a single voice session, in seconds. Defaults to `1800`.                                                                                                                                                 |
    | **Environment**                    | The deployment the connection binds to. Select an environment to follow its active deployment automatically, or leave the default to use the working copy.                                                                      |
    | **Provider Verification Strength** | Controls how strictly the platform trusts the provider's identity assertions for session continuity and linking. **Weak** is the default. Choose **Strong** only for channels or providers whose identity assertions you trust. |
    | **Initial Status**                 | **Active** connections can be used immediately. **Inactive** connections stay saved but disabled until you activate them.                                                                                                       |

    Click **Create**.
  </Step>

  <Step title="Choose a Call Routing Option (Kore.ai Voice Gateway)">
    After creating the channel, go to the **Configuration** tab. If you choose **Voice Provider: Kore.ai Voice Gateway**, set the following fields under **Genesys Audio Connector Settings** > **Call routing**, which controls how calls reach the agent.

    | Route                  | Description                                                                                                                                              |
    | ---------------------- | -------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Soft Phone** (FREE)  | Dial the agent from your browser with no DID purchase or telephony charges. Use it for evals, QA, and demos. **Launch dialer** opens the browser dialer. |
    | **Phone Number (DID)** | Purchase a real phone number through Twilio. Disable this route to stop DID listing and call routing while keeping the number saved.                     |

    <Note>If you select **Phone Number (DID)**, complete the next step. If you selected **Voice Provider: Bring Your Own SIP** instead, skip to Configure SIP Settings.</Note>
  </Step>

  <Step title="Configure the Phone Number (DID Only)">
    Set the **Provider** to **Twilio**, toggle **Inbound calls** (routes calls from this number into the voice agent) and **Outbound calls** (allows this number to be used for campaign dialout), and select a **Country**, **Number Type**, and optional **Area Code** to filter available local numbers. Select **Search Available Numbers** to list matching numbers.
  </Step>

  <Step title="Configure SIP Settings (Bring Your Own SIP Only)">
    If you selected **Voice Provider: Bring Your Own SIP**, set the following fields under **SIP Configuration**.

    | Field                      | Description                                                                                                                  |
    | -------------------------- | ---------------------------------------------------------------------------------------------------------------------------- |
    | **Our SIP Server Address** | Configure your SIP provider to send `INVITE` to this address. Shows `Not configured — contact admin` until an admin sets it. |
    | **SIP Gateway IP Address** | Your SIP trunk IP address — optionally include a port (default: `5060`).                                                     |
    | **DID / Phone Number**     | The phone number (DID) your SIP provider routes to this SIP gateway.                                                         |
  </Step>

  <Step title="Configure Speech Recognition">
    Under **Speech Recognition**, set the following fields.

    | Field                 | Description                                                                                                                                                                                                                                                                                |
    | --------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ |
    | **Provider**          | The speech-to-text credential set to use, for example `Deepgram Speech Credentials`.                                                                                                                                                                                                       |
    | **Language**          | The recognition language, for example `Multilingual`.                                                                                                                                                                                                                                      |
    | **STT Model**         | The recognition model, for example `Nova 3 (latest, most accurate)`.                                                                                                                                                                                                                       |
    | **Custom vocabulary** | Domain-specific terms to bias recognition toward. Select **Add phrase** to add a row, then set the **Phrase** (for example `ACME, policy number`) and optional **Boost**. Deepgram Nova-3 and Flux use keyterm prompting and ignore boost; older Deepgram models use keyword intensifiers. |
  </Step>

  <Step title="Configure Speech Synthesis">
    Under **Speech Synthesis**, set the following fields.

    | Field                                | Description                                                                                                              |
    | ------------------------------------ | ------------------------------------------------------------------------------------------------------------------------ |
    | **Provider**                         | The text-to-speech credential set to use, for example `Deepgram Speech Credentials`.                                     |
    | **Language**                         | The synthesis language, for example `English (PH)`.                                                                      |
    | **Voice**                            | The voice used for synthesized speech.                                                                                   |
    | **Switch TTS for detected language** | When enabled, calls can change the TTS language, voice, and model after speech recognition detects a different language. |
  </Step>

  <Step title="Configure Conversation Controls">
    | Field                   | Description                                                                                                |
    | ----------------------- | ---------------------------------------------------------------------------------------------------------- |
    | **Barge-in**            | Allow the caller to interrupt the agent while it's speaking.                                               |
    | **Speech Timeout (ms)** | The time the agent waits for caller speech before it times out, for example `3000`.                        |
    | **Welcome Message**     | The opening prompt the agent speaks, such as `Hello! How can I help you today?`.                           |
    | **Noise isolation**     | Apply server-side noise reduction to inbound call audio to filter out background voices and ambient noise. |
  </Step>

  <Step title="Save the Channel">
    Click **Save Changes** to apply the configuration.
  </Step>
</Steps>

### Manage an existing Genesys audio connector channel

Opening a channel from **Deployments** > **Channels** > **VOICE** > **Genesys Audio Connector** shows six tabs: **Overview**, **Credentials**, **Configuration**, **Deployment**, **Testing**, and **Activity**.

<Tabs>
  <Tab title="Overview">
    The **Connection Summary** shows these fields.

    | Field                       | Description                       |
    | --------------------------- | --------------------------------- |
    | **Status**                  | for example: *Active*             |
    | **Created**                 | Timestamp the channel was created |
    | **Updated**                 | Timestamp of the last change      |
    | **Environment**             | for example: `dev`                |
    | **Source Type**             | `Genesys Audio Connector`         |
    | **Source ID**               | Unique channel identifier (UUID)  |
    | **Genesys Organization ID** | Shown for reference.              |
  </Tab>

  <Tab title="Credentials">
    Credentials are saved and encrypted after creation; the fields show placeholder text and accept new values only if you want to update them.

    | Field                             | Description                                                                                                                                                                                      |
    | --------------------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ |
    | **Use Auth Profile**              | Toggle on to authenticate through a saved Auth Profile instead of entering credentials manually. When enabled, select the profile from the **Auth Profile** picker in place of the fields below. |
    | **API Key**                       | The API key used to authenticate AudioHook requests. Shown as *(saved — enter to update)*; leave blank to keep the current value.                                                                |
    | **HTTP Message Signature Secret** | The secret used to verify inbound webhook signatures. Shown as *(saved — enter to update)*; leave blank to keep the current value.                                                               |

    Click **Save** to apply any changes.
  </Tab>

  <Tab title="Configuration">
    Contains the same **General** (Display Name, Provider Verification Strength), voice provider and call routing, **Speech Recognition**, **Speech Synthesis**, and **Conversation Controls** described in [Set up Genesys Audio Connector](#set-up-genesys-audio-connector) above, editable at any time. Changing any of these settings here updates the live channel immediately after **Save Changes**.
  </Tab>

  <Tab title="Deployment">
    The **Deployment** tab shows these fields.

    | Field                 | Description                                                                                                                                                                                                                      |
    | --------------------- | -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
    | **Environment**       | An environment selector that chooses which deployed version handles incoming calls. Selecting the working copy shows a **Working Copy** warning; an environment with no active deployment shows a **No active deployment** hint. |
    | **Active Deployment** | Shows the bound deployment's version and deploy date, or indicates that none is bound.                                                                                                                                           |

    Click **Save Changes** after switching the environment/version.
  </Tab>

  <Tab title="Testing">
    Use this tab to review diagnostics and validation evidence for the connection, and to test it end to end before promoting it beyond diagnostics mode.
  </Tab>
</Tabs>