Voice channels

Back to voice channels

Set up Realtime LLM Voice channel

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)

   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.

   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 (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, 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, for example korevg-np.kore.ai:5060. Select Copy to copy the address.
   SIP Gateway IPs / CIDRs	Provider SIP source addresses. Separate multiple IPv4 addresses or CIDR ranges with commas, for example 203.0.113.50:5080, 3.80.16.0/23.
   SIP Termination URI	Choose whether your SIP provider identifies by IP Address or Domain Name, then enter the SIP termination URI.
   DID / Phone Number	The phone number (DID) your SIP provider routes to this SIP gateway.
   Enable E.164 (leading +)	Prepend a leading + to outbound numbers sent to the carrier. On by default, which international (full-E.164) numbers require. Turn off only for local/national-format numbers — otherwise the leading + misroutes them.

5. Configure Realtime LLM Voice Settings

   Under Realtime LLM Voice settings, set the following fields.

   • Realtime orchestration: Select how runtime coordinates the conversation flow and business tool execution for realtime voice interactions. Realtime orchestration determines whether the realtime model or runtime controls conversation progression and business actions, such as bookings, orders, and payments.

     Mode	Description	Recommended for
     Legacy - model-selected tools	Preserves the existing behavior where the realtime model manages both the conversation and business tool selection.	Existing agents and backward compatibility.
     Runtime Guarded - hybrid freeform (recommended)	Allows the realtime model to conduct the conversation naturally while runtime validates tool execution and context updates against the current FLOW step. Business tools execute only when they belong to the active FLOW step, preventing out-of-order actions while maintaining the authored FLOW.	New realtime voice agents that require natural conversations while maintaining FLOW integrity.
     Runtime Owned - canonical turn	Runtime controls every conversation turn and executes the FLOW exactly as authored. The realtime model handles conversational responses but cannot invoke business tools directly.	Workflows involving payments, bookings, orders, or other irreversible business actions and highly deterministic flows.

     • Changes to the orchestration mode apply only to new realtime sessions. Calls already in progress continue using the mode that was active when the session started.
     • Existing channel configurations without a configured orchestration mode continue to use Legacy behavior.

   • Response wording owner: Select who generates the wording of responses for realtime voice conversations. This setting is available only when the Realtime orchestration mode is set to Runtime Owned — canonical turn.

     • Runtime text (recommended): Runtime generates the response text. This option provides consistent response generation.
     • Provider-grounded voice: Uses the realtime voice provider to generate spoken responses while runtime continues to control conversation flow and business actions. This option is available only for Gemini Live. For unsupported or unsafe conversation turns, it automatically falls back to runtime-generated text responses.

   • Semantic convergence: Semantic Convergence determines how the platform coordinates with the realtime voice provider during a voice interaction. Controls whether realtime voice turns use the coordinated utterance runtime. This setting is available only when the realtime orchestration config is set to Legacy mode.

     Mode	Description	Recommended use
     Off	Uses the legacy realtime voice execution model. The realtime provider executes tool calls directly without semantic coordination.	Use for backward compatibility with existing voice applications.
     Shadow	Uses the legacy execution model while collecting diagnostics to compare legacy behavior with semantic convergence. Doesn’t affect customer interactions.	Use for validation, testing, and migration before enabling Enforce.
     Enforce	Uses semantic convergence to route voice turns through the platform’s coordinator, ensuring tool execution and business logic are processed consistently with the agent runtime.	Recommended for new realtime voice deployments and production environments requiring consistent behavior across voice and text.

   • S2S Provider: The realtime speech-to-speech model the channel uses. Select one of the following: Amazon Nova Sonic (S2S), Azure OpenAI Realtime, Grok Realtime (S2S), Google Gemini Live, or OpenAI Realtime. Each option shows which credentials it uses:

     Provider	Credentials used
     Amazon Nova Sonic (S2S)	Voice Service credentials (Amazon Nova Sonic (S2S) Credentials).
     Azure OpenAI Realtime, Grok Realtime (S2S), Google Gemini Live, OpenAI Realtime	The project-selected model connection. See Realtime model connection below.

   • Realtime model connection: Shown for every S2S provider except Amazon Nova Sonic, which authenticates via Voice Service credentials instead. Selects the project-pinned realtime model this provider uses - shows Use project realtime model until a workspace admin adds one:

     1. Configure the model under Voice Services, then add it to the model catalog as an LLM provider.
     2. In the project, go to Settings -> Model and add the model you configured at the workspace level.

   • Provider-specific settings: After you select an S2S provider, a configuration section for that provider appears. Expand the provider below to see its settings.

     Amazon Nova Sonic

     Under Amazon Nova Sonic Configuration, set the following fields.

     Field	Description
     Voice	The voice Nova Sonic uses to speak to callers. Default: Tiffany.
     Model	The Amazon Bedrock model ARN, for example arn:aws:bedrock:us-east-1::foundation-model/amazon.nova-2-sonic-v1:0.
     Maximum tokens	The maximum number of tokens in a model response. Default: 1024.
     Temperature	Controls response randomness. Lower values give more predictable responses; higher values give more varied responses. Default: 0.70.
     Top P	Limits the model to the smallest set of likely tokens whose combined probability reaches this value. Default: 0.90.
     Endpointing sensitivity	How quickly Nova decides the caller has finished speaking. Medium (default) suits general conversation. High responds sooner but may interrupt callers. Low allows more pause before responding.

     Pending tool fillers

     Turn on Pending tool fillers to let Nova speak a brief hold phrase when a project tool is still running. Fast tools finish without a filler.

     Field	Description
     Filler delay (milliseconds)	Nova speaks only if the tool is still pending after this delay. Leave empty to use the default of 900 ms.
     Tool filler instruction	The instruction Nova follows to generate the hold phrase. Use {{toolName}} to insert the name of the running tool. Default: The tool {{toolName}} is still in progress. Briefly tell the caller that you are working on it.

     Coordinator turn filler

     Turn on Coordinator turn filler to let Nova speak a short hold phrase during convergence-enabled coordinator turns while Runtime decides and executes the turn.

     Field	Description
     Coordinator filler instruction	The instruction Nova follows to generate the hold phrase. Default: Briefly tell the caller that you are still working on their request. Runtime sends this as native cross-modal input and keeps its transcript out of the canonical conversation history.

     Advanced voice behaviour (Nova-specific)

     Field	Description
     Failed tool call fallback > Tell the caller and end the call	When a Nova tool call fails, Nova says a short apology and ends the call. Off by default.

     Also configure the Advanced voice behaviour settings that apply to all providers, described after the provider list.

     Azure OpenAI Realtime

     Under Azure OpenAI Realtime Configuration, set the following fields.

     Field	Description
     Voice	The voice used to speak to callers. Default: Marin (recommended).
     Temperature	Controls response randomness. Range: Focused (0.6) to Creative (1.2). Default: 0.8.

     Turn detection

     Field	Description
     Speech detection sensitivity	Adjusts how readily the model detects speech. Default: 0.5.
     Silence Duration (ms)	How long the caller must be silent before the turn is treated as complete. Default: 700.
     Prefix Padding (ms)	Audio kept before detected speech so the start of the caller’s utterance is preserved. Default: 300.

     Also configure the Advanced voice behaviour settings that apply to all providers, described after the provider list.

     Grok Realtime

     Under Grok Realtime Configuration, set the following fields.

     Field	Description
     Voice	The voice used to speak to callers. Default: Ara.
     Temperature	Controls response randomness. Range: Focused (0.0) to Creative (2.0). Default: 0.8.
     Threshold	Controls speech detection. Range: Sensitive (0.0) to Conservative (1.0). Default: 0.5.
     Silence Duration (ms)	How long the caller must be silent before the turn is treated as complete. Default: 500.
     Prefix Padding (ms)	Audio kept before detected speech so the start of the caller’s utterance is preserved. Default: 300.

     Also configure the Advanced voice behaviour settings that apply to all providers, described after the provider list.

     Google Gemini Live

     Under Gemini Live Configuration, set the following fields.

     Field	Description
     Voice	The voice used to speak to callers, for example Puck — Upbeat.
     Enable context window compression	Lets Gemini compress older session context so long-running Live conversations can continue beyond the model’s context window. Off by default.
     Temperature	Controls response randomness. Range: Deterministic (0.0) to Creative (2.0). Default: 0.8.

     Tool response scheduling

     Field	Description
     Tool response scheduling	Controls how Gemini resumes after a final tool result. Interrupt responds immediately to the final result. When idle — finish the current response first (default) waits for the current response to finish. Silent retains the final result without speaking right away. Progress and filler updates always wait until Gemini is idle.
     Pending tool fillers	Lets Gemini Live acknowledge long-running project tools while the original tool call remains pending. Off by default.

     Realtime turn filler

     Separate from project Voice Fillers. Applies only to Google or Vertex realtime S2S calls while semantic convergence is set to Enforce.

     Field	Description
     Native turn filler	Lets Gemini speak a short hold phrase while Runtime settles an enforced turn. It never applies to transfers, handoffs, or protected playback.
     Generative native filler	Lets Gemini choose a short, contextual hold phrase in the current language instead of using the deterministic phrase pool.
     Generative filler prompt override	Overrides the prompt Gemini uses to generate the hold phrase. Default: Reply with a short, natural hold phrase in the current language.

     Filler delivery safeguards

     Shared by pending-tool and realtime turn fillers.

     Field	Description
     Settlement watchdog	Time allowed for filler generation or a held final response to settle before Runtime releases the safe fallback. Accepted range: 1000–60000 ms. Leave empty to use the platform default of 10000 ms.

     Voice activity detection

     Controls when Gemini starts listening to a caller turn and how long it waits before closing that turn.

     Field	Description
     Start Sensitivity	Higher detects speech sooner but may trigger on noise. Lower waits for a stronger speech signal. Default: Default.
     End Sensitivity	Higher ends turns sooner for lower latency. Lower allows longer pauses before Gemini responds. Default: Default.
     Silence Duration (ms)	Milliseconds of non-speech before Gemini closes the caller turn. Increase to allow pauses; decrease for faster responses. Default: 100.
     Prefix Padding (ms)	Milliseconds of detected speech required before Gemini commits a speech start. Lower values catch shorter utterances but can increase false starts. Default: 20.

     Caller transcription

     Field	Description
     Caller language (Optional)	The language callers speak, such as en-US. If left empty, Gemini guesses the language on every turn, which can turn English into another script. The call’s language setting, when there is one, takes precedence.

     Silent-turn recovery

     Field	Description
     Recover silent turns	Re-prompts, then restarts, a Gemini session that stays silent when it owes the caller an answer.
     Recover unanswered caller turns only	Limits recovery to caller turns that received no answer. Included automatically when Recover silent turns is on. If Gemini hears the caller but does not respond within 8 seconds, the Gemini session restarts with the caller’s words, without re-prompting tool results.

     Advanced voice behaviour (Gemini-specific)

     Field	Description
     Silent resume recovery > Recover a silent resumed session	Starts a fresh Gemini session when a resumed one connects but never speaks. Off by default.

     Also configure the Advanced voice behaviour settings that apply to all providers, described after the provider list.

     OpenAI Realtime

     Under OpenAI Realtime Configuration, set the following fields.

     Field	Description
     Voice	The voice used to speak to callers. Default: Marin (recommended).

     Turn detection

     Tune how OpenAI detects when speech starts and when the caller has finished speaking.

     Field	Description
     Turn detection mode	How OpenAI detects turns, for example Server VAD — audio activity.
     Speech detection sensitivity	Range: More sensitive (0.0) to More conservative (1.0). Lower values pick up quieter speech; higher values reduce background-noise triggers. Default: 0.5.
     Silence Duration (ms)	Quiet time before OpenAI treats the caller turn as complete. Default: 700.
     Prefix Padding (ms)	Audio kept before detected speech so the beginning of the caller’s utterance is preserved. Default: 300.

     Caller transcription

     How the caller’s speech is written down. This text is stored as the caller’s turn and is what the agent checks answers against.

     Field	Description
     Caller transcriber	The transcription model, for example Whisper (default).
     Caller language (Optional)	The language callers speak, such as en-US. If left empty, the transcriber guesses the language on every turn, which can turn English into another script.
     Words to expect	Brand names and terms callers use, for example Acme, Rewards, email addresses spelled out letter by letter. Used by Whisper and GPT-4o Transcribe.

     Never put a caller’s own details in Words to expect. The transcriber would then hear them from anyone.

     Also configure the Advanced voice behaviour settings that apply to all providers, described after the provider list.

   • Advanced voice behaviour (all providers): Optional behaviours for this channel. Each one is off, empty, or set to the earlier platform behaviour until you change it.

     Field	Description
     Caller wording > Wording a caller turn runs on	Select the wording the agent runs a caller turn on: Model’s wording (default), Auto: prefer the caller’s transcript, or Caller transcript.
     Default transfer announcement > Line spoken before a transfer (Optional)	The line the agent speaks before a transfer, for example One moment while I connect you.

   • Noise isolation: Applies server-side noise reduction to inbound call audio to filter out background voices and ambient noise.

   • Voice transfer and survey capabilities: A read-only panel that shows which transfer and survey capabilities Runtime supports for the selected provider.

Manage an existing realtime LLM voice channel

Open a channel from Deployments > Channels > Voice > Realtime LLM Voice. It contains three tabs: Overview, Configuration, and Deployment.

Overview

Connection Summary displays the following 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 identifier (UUID) of channel

Configuration

The channel includes the same General (Display Name), Provider Verification Strength, call routing, Realtime LLM Voice Settings, and provider-specific settings (which vary by the selected S2S provider) described in Set up Realtime LLM Voice. You can edit these settings any time and with immediate effect.

Deployment

The Deployment tab displays these fields.

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

Click Save Changes after switching the environment/version.

Set up Pipeline Voice channel

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 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.

2. 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.

3. 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.

   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.
4. 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.

5. 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 IPs / CIDRs	Provider SIP source addresses. Separate multiple IPv4 addresses or CIDR ranges with commas, for example 203.0.113.50:5080, 3.80.16.0/23.
   SIP Termination URI	The termination address your SIP provider expects outbound calls to reach, as either an IP Address or Domain Name. Enter the URI in the field below the selected type.
   DID / Phone Number	The phone number (DID) your SIP provider routes to this SIP gateway, for example [phone] or 98492.

   Toggle Enable E.164 (leading +) to prepend a leading + to outbound numbers sent to the carrier - on by default, which international (full-E.164) numbers require. Turn it off only for local/national-format numbers, since the leading + otherwise misroutes them.

6. 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).
   Minimum STT confidence	Reject transcripts below this confidence (0-1) and treat them as no-input. 0 disables the gate. Provider-relative - calibrate per channel.
   Custom vocabulary	Add domain-specific terms to improve speech recognition. You can set the Phrase (for example ACME, policy number) and optional Boost. For Emotech, add hotwords (domain-specific words or phrases you want the speech recognition system to recognize more reliably) and set the Hotword Weight. A value from 5 to 10 is recommended. Clear the custom vocabulary list to disable hotwords.
   Multilingual speech recognition	Choose the recognizer for each configured language. Enable A different recognizer per language to select the speech recognizer for each language. Add a language profile before configuring language-specific recognition.

7. Configure Speech Synthesis

   Under Speech Synthesis, set the following fields.

   Field	Description
   Voices by language	Enable this option to configure separate text-to-speech settings for each language supported by the agent. When disabled, the channel uses a single TTS configuration for all languages.
   Language	Select the language for which you want to create a TTS profile, then click Add language.
   Provider	The text-to-speech credential set to use for the selected language, for example ElevenLabs (TTS) Credentials or Deepgram Speech Credentials.
   Voice	The voice used for synthesized speech for the selected language, 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.

   • After you click Add language, the platform creates a dedicated TTS configuration section for that language. Configure the provider, voice, model, and any provider-specific settings independently for each language.
   • Available fields vary by provider. Some providers expose additional settings such as TTS model, output format, speed, temperature, and voice tuning controls. Each language profile maintains its own provider configuration, allowing multilingual agents to use different voices and models for different languages.
8. 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.

9. 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)	Controls how long the speech provider waits through silence before finalizing the current speech segment. This does not combine separate final results. For example, 3000.
   Transcript Grouping Timeout (ms)	After the speech provider returns a final result, waits this long for another final result and combines the parts into one user input. Leave blank to process each final result immediately. For example, 1500.
   Welcome Message	The opening prompt the agent speaks, such as Hello! How can I help you today?.
   Allow caller to interrupt the welcome	Allows callers to interrupt the welcome message when barge-in is active.
   Noise isolation	Apply server-side noise reduction to inbound call audio to filter out background voices and ambient noise.

   Under Connect greeting (audio), optionally play a clip as soon as the call connects, before the agent’s first turn. If the project’s greeting-audio library has no clips yet, the panel shows a note and an Open Greeting Audio link to add one.

10. Save the Channel

    Click Save Changes to apply the configuration.

Manage an existing Pipeline Voice channel

Open a channel from Deployments > Channels > Voice > Pipeline Voice. It displays the following three tabs.

Overview

The Connection Summary displays 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 identifier (UUID) of the channel

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 above, editable at any time. Changing any of these settings here updates the live channel immediately after Save Changes.

Deployment

The Deployment tab displays these fields.

Field	Description
Agent Version	An Environment menu (for example: Development) that selects which deployed version handles incoming messages.
Active Deployment	Displays the bound deployment version and when it was deployed.

Click Save Changes after switching the environment or the version.