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.
1
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.
2
Choose a Call Routing Option (Kore.ai Voice Gateway)
Once the channel is created, select it, then 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.
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.
3
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.
4
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.
5
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.
Open a channel from Deployments > Channels > VOICE > Realtime LLM Voice shows three tabs: Overview, Configuration, and Deployment.
Overview
Configuration
Deployment
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)
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. 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.
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.
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.
Choose your voice provider before you begin — it determines which additional fields apply later in this setup.
To configure a Pipeline Voice channel, go to Deployments > Channels > VOICE > Pipeline Voice > + New and complete the following steps in order.
1
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 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.
2
Call Routing Option (Kore.ai Voice Gateway)
Once the channel is created, select it, then go to the Configuration tab. If you choose Voice Provider: Kore.ai Voice Gateway, set the following fields under:
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.
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.
3
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.
4
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.
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.
5
Configure Conversation Controls
Under Conversation controls, set the following fields.
Setting
Description
Barge-in
Allows 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.
Welcome Message
The opening prompt the agent speaks, such as Hello! How can I help you today?.
Noise isolation
Applies server-side noise reduction to inbound call audio to filter out background voices and ambient noise.
Open a channel from Deployments > Channels > VOICE > Pipeline Voice shows three tabs: Overview, Configuration, and Deployment.
Overview
Configuration
Deployment
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
Pipeline Voice Channel
Source ID
Unique channel identifier (UUID)
Webhook URL
Use the URL to integrate external SIP or voice pipeline systems.
Contains the same General (Display Name), Provider Verification Strength, call routing, and Conversation Controls described in Set up Pipeline Voice above, editable at any time. Changing the call routing or conversation control settings here updates the live channel immediately after Save Changes.
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.
Configure Genesys Audio Connector for KoreVG-owned AudioHook execution.
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.
To configure a Genesys Audio Connector channel, go to Deployments > Channels > VOICE > Genesys Audio Connector > + New and complete the following steps in order.
1
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.
2
Choose a Call Routing Option (Kore.ai Voice Gateway)
Once the channel is created, select it, then 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.
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.
3
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.
4
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.
5
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.
6
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.
7
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.
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.
Overview
Credentials
Configuration
Deployment
Testing
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.
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.
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 above, editable at any time. Changing any of these settings here updates the live channel immediately after Save Changes.
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.
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.