Back to voice channelsCreate a project-owned Genesys Audio Connector connection, choose a Voice mode, configure that mode’s speech settings, wire the connection into Genesys Architect, and validate live audio before enabling traffic.
Voice mode can’t be changed after the connection is created. Choose Pipeline for independent speech-to-text, LLM, and text-to-speech providers, or Realtime for a single speech-to-speech (S2S) model with lower latency. If you need the other architecture later, create a new connection.
Sign in to the regional Genesys Cloud organization that owns the voice deployment. Studio uses this region to select the matching Genesys API host.
2
Identify the voice deployment
In Genesys Cloud Admin, identify the voice deployment associated with the intended Architect flow and copy its Voice Deployment ID.
3
Generate a project API key
Generate a project-specific Audio Connector API key for this connection. Configure the matching value in Genesys when you wire the Architect flow.
4
Keep environments and credentials separate
Use separate API keys, voice deployments, and connections for development and production. Never share one connection’s credentials across environments.
To configure a Genesys Audio Connector channel, go to Deployments -> Channels -> VOICE -> Genesys Audio Connector -> + New and complete the following steps in order.
1
Enter a Display Name
Give the connection a label, for example Genesys Audio Connector Production.
2
Enter the Genesys Organization ID and Allowed Region
Enter the exact Genesys Organization ID and select the region that owns it, for example US East (N. Virginia). A mismatched region can make valid credentials appear unavailable.
3
Enter the Voice Deployment ID
Paste the voice deployment ID associated with the intended Architect flow.
4
Choose the Voice mode
Select Pipeline or Realtime. This choice is permanent for the connection - see the note above.
5
Set the Activation Mode
Start in Diagnostics only so the Testing tab can collect evidence without treating the setup as production-ready. Switch to Enabled only after validation passes.
6
Enter the API Key and optional Webhook Signature Secret
Enter the project-specific Audio Connector API key. Webhook Signature Secret is optional at creation - set it when Genesys is configured to sign outbound requests.
7
Set Max Session Seconds
Set the maximum duration for a single voice session, in seconds (default: 1800).
8
Complete common connection settings
Field
Description
Environment
Select an environment to follow its active deployment, or leave the default to use the working copy.
Provider Verification Strength
Weak is the default. Choose Strong only for channels or providers whose identity assertions you trust for stronger continuity and linking decisions.
Initial Status
Active connections can be used immediately. Inactive connections stay saved but disabled until you activate them.
9
Create the connection
Select Create. The next step - configuring speech settings for your chosen Voice mode - happens after creation, on the connection’s Configuration tab.
The voice deployment associated with the intended Architect flow.
Voice mode
Pipeline (independent STT/LLM/TTS providers) or Realtime (single S2S model). Permanent after creation.
Activation Mode
Diagnostics only (default, safe - setup and validation, no production traffic), Enabled (processes live calls), or Disabled.
API Key
A project-specific Audio Connector API key.
Webhook Signature Secret
Optional at creation. Configure the identical value in Genesys when signed HTTP signaling is enabled. Referred to as HTTP Message Signature Secret later, on the Credentials tab - same value, different label.
Max Session Seconds
Maximum duration for a single voice session, in seconds. Default: 1800.
After creation, open the connection and go to its Configuration tab to set up speech behavior for the Voice mode you chose.
Pipeline mode
Realtime mode
1
Configure Speech Recognition
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
Domain-specific terms to bias recognition toward. Select Add phrase and set a 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.
Multilingual speech recognition
Chooses the recognizer that listens for each configured language. Enable A different recognizer per language to select the configured speech recognizer that should listen after an explicit language change.
2
Configure Speech Synthesis
Optionally enable A different voice per language to give each language its own voice instead of one voice for every caller. Then set:
Field
Description
Provider
The text-to-speech credential set to use, for example ElevenLabs (TTS) Credentials.
Language
The synthesis language, for example Arabic.
Voice
The voice used for synthesized speech.
TTS Model
The synthesis model, for example Eleven Multilingual v2.
If you selected an ElevenLabs voice, an ElevenLabs Voice Settings panel lets you override Speed, Stability, Similarity boost, Style exaggeration, and Speaker boost for this connection, plus a Preview Voice field to hear the configured voice before saving.
3
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.
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
When off, the welcome plays to completion even if barge-in is on. It applies only while barge-in is active for the call, through the channel’s Barge-in setting or the agent’s listening settings.
Connect greeting (Audio)
Plays 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. See Greeting audio.
Keypad tone
Plays an audible keypad tone to the caller on each DTMF keypress. Off by default.
Voice gateway turn detection
Lets the voice gateway decide when the caller has finished speaking, using voice and wording signals instead of the speech provider’s silence timer. Off by default. Requires a voice gateway that supports turn detection.
Noise isolation
Applies server-side noise reduction to inbound call audio to filter out background voices and ambient noise.
Trust inbound SIP headers
Off by default. Turn it on only if your SBC sets or overwrites these headers. If the headers can arrive from the caller, the value isn’t trustworthy. When on, transfer headers configured to pass through from the inbound call (for example, User-to-User) are sent on the transfer. When off, they are dropped.
4
Configure the Transfer Queue
Field
Description
Transfer Queue Header Key
SIP transfer header name that carries the queue, for example X-Queue. Must match what Genesys expects. Required for escalate-to-human.
Transfer Queue Name
The Genesys queue used when the agent transfers a caller to a human, for example Kore Inbound. Must match a queue configured in Genesys.
5
Save your changes
Select Save Changes.
Under Realtime LLM Voice Settings, select an S2S Provider. Only the panel for the selected provider applies to this connection.
Azure OpenAI Realtime, Grok Realtime, Google Gemini Live, and OpenAI Realtime each require a Realtime model connection: a project-level model pinned to this channel. Amazon Nova Sonic does not use this field; it authenticates via Voice Service credentials instead.To make a model available here:
Go to Settings -> Model and add the model you configured at the workspace level.
Once added, the model becomes selectable in the Realtime model connection dropdown below.
Field
Description
Realtime model connection
The project-pinned realtime model this provider uses. Shows No project model selected until a workspace admin adds one following the steps above; once selected, shows the model name and provider (for example GPT Realtime 2 · openai).
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 fillersTurn 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 fillerTurn 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).
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 fillerSeparate 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 safeguardsShared 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 detectionControls 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 detectionTune 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 transcriptionHow 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.
Realtime turn fillerApplies only to OpenAI Realtime calls while semantic convergence is set to Enforce. Off by default.
Field
Description
Native turn filler
While Runtime works on a caller turn, OpenAI says a short hold phrase in the caller’s language. Hold phrases stop when the answer is ready, never play during a transfer or handoff, and aren’t added to the conversation.
Filler prompt override (Optional)
Replaces the default instruction OpenAI words each hold phrase from. Default: Reply with a short, natural hold phrase in the current language. Runtime appends fixed safety rules that prevent claims about results, actions, promises, questions, or tool calls.
Advanced voice behaviour (OpenAI-specific)
Field
Description
Caller interruption > Let callers interrupt the agent
When off, the caller speaking doesn’t cut off what the agent is saying.
Uses model playback for protected outputs and protects the welcome unless interruption is allowed. Off by default. With server VAD, turning this on lets the model play protected outputs and protects the welcome unless ON_START allows interruption. An ON_START greeting authored with barge-in disallowed is protected automatically with server or semantic VAD, regardless of this switch. Input that repeats a protected output word for word is treated as echo. Menu answers, numbers, spelled names, and other caller input still run.
Silent caller reprompt > Reprompt a silent caller
Uses the project’s voice no-input settings (timeout, reprompt messages, retries, and final action) on this realtime channel. Turn it off to keep the channel silent.
Caller transcript cross-check > Check the model’s wording against the transcript
Waits briefly for the caller’s transcript and gives it to the agent beside what the model reported.
Longest wait for the transcript (ms) (Optional)
Sets how long the turn waits for the caller’s transcript, up to 5000 ms. Leave empty to size the wait to the caller transcription model of the selected realtime model: 1500 ms for whisper-1 and 400 ms for gpt-4o transcription. The turn runs without the transcript once the wait ends.
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.
Runtime makes the voice model say its answers word for word, once per caller turn, and records what was said.
Noise isolation
Applies server-side noise reduction to inbound call audio to filter out background voices and ambient noise.
Trust inbound SIP headers
The same setting as on the Settings tab. Off by default. Turn it on only if your SBC sets or overwrites these headers.
Transfer Queue Header Key
SIP transfer header name that carries the queue, for example X-Queue. Must match what Genesys expects. Required for escalate-to-human.
Transfer Queue Name
The Genesys queue used when the agent transfers a caller to a human, for example Kore Inbound. Must match a queue configured in Genesys.
Select Save Changes.
Voice transfer and survey capabilitiesA read-only panel that shows how many of the transfer and survey capabilities Runtime evaluates as available for the concrete voice execution boundary (the execution surface is KoreVG realtime S2S), for example 11 of 11 capabilities available. It lists the following capabilities, each marked Available or not.
Capability
Description
Human transfer
Transfer the caller to a human agent.
Pre-transfer announcement
Speak an announcement before a transfer.
Transfer outcome recovery
Recover when a transfer doesn’t complete.
SIP transfer
Transfer the call over SIP.
PSTN transfer
Transfer the call to a phone number on the public telephone network.
Bridged agent transcript
Carry the transcript to the agent the call is bridged to.
Transfer CSAT
Collect a satisfaction rating after a transfer (csat, nps, likeDislike).
Template survey
Run a template survey with a rating type of star, thumbs, or scale.
Survey comments
Collect free-text comments in a survey.
Survey follow-ups
Ask follow-up questions in a survey.
Return-leg survey
Run a survey on the return leg of a transferred call.
The Genesys Organization ID and the SIP URI are displayed for reference. Select the copy icon next to a value to copy it.
Open the connection’s Testing tab and copy the AudioHook Callback URL - a wss:// URL with an embedded connection ID, for example wss://korevg-np.audiosocket.kore.ai/audiosocket/genesysconnect/<connection-id>. Preserve the full URL and connection ID; do not substitute a generic Runtime WebSocket.
2
Configure an Audio Connector action
In the intended Architect flow, add or update the Audio Connector action and paste the callback and API key into the corresponding integration fields.
3
Use the supported audio framing
Keep the configured PCMU framing, sample rate, channel count, and media direction aligned with the platform contract. Transcoding or buffering at an intermediate proxy adds latency and can corrupt audio.
4
Map completion and disconnect variables
Capture normal completion, caller hangup, gateway error, and fallback outcomes so Architect can continue or exit the flow deterministically.
Use a dedicated Architect test flow and test number. Confirm session creation, greeting playback, caller speech, agent response, barge-in, and clean hangup.
2
Validate media and backpressure
Confirm PCMU framing, bidirectional audio, queue behavior, packet loss, and disconnect variables under normal speech, silence, interruption, and rapid turn-taking.
3
Capture the voice latency budget
Record audio jitter p95, first audio p95, STT first partial p95, TTS first audio p95, and representative end-to-end turn latency. Investigate regressions before promotion.
4
Save trace and operational evidence
On the Testing tab, record AudioHook and KoreVG session IDs, Architect flow name and version, Runtime trace IDs, sandbox evidence, dashboard and alert references, concurrency, and fallback results, then select Save Evidence.
5
Promote only after validation passes
Mark validation Passed, retain the evidence, then change Activation Mode to Enabled. Keep a tested rollback or fallback path available.
Opening a connection from Deployments -> Channels -> VOICE -> Genesys Audio Connector shows four tabs: Credentials, Configuration, Testing, and Activity.
Credentials
Configuration
Testing
Credentials are saved and encrypted after creation; 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 an API key.
API Key
Shown as (saved - enter to update); leave blank to keep the current value.
HTTP Message Signature Secret
Shown as (saved - enter to update); leave blank to keep the current value. This is the same secret set as Webhook Signature Secret at creation.
Select Save Credentials to apply any changes.
Shows General (Display Name, Provider Verification Strength) plus the Pipeline or Realtime speech settings described in Configure the connection above, editable at any time. Select Save Changes to apply.
The Audio Connector Health panel shows:
Field
Description
AudioHook Target
The execution target for this connection, for example KoreVG (SAVG audiosocket).
Last Success / Last Failure
Timestamps of the most recent diagnostic outcomes, or a prompt such as Live Architect validation required.
Provider URL
The underlying provider endpoint the connection routes to.
AudioHook Callback URL
The full wss:// URL to configure in the Architect Audio Connector action.
Keep the API key and HTTP Message Signature Secret in Genesys and encrypted Studio credentials. Use distinct values per environment and rotate exposed credentials.
Audio - Don’t log raw audio or sensitive transcript content. Use scrubbed trace identifiers and aggregate voice phase metrics for operational debugging.Capacity - Validate the expected concurrent-session ceiling and queue pressure. KoreVG has bounded per-session queues and environment-level capacity controls.Fallback - Define what Architect does when the AudioHook can’t connect, media stalls, STT or TTS fails, or the agent exceeds its response budget.Observability - Monitor connection failures, audio jitter, packet loss, STT, LLM, and TTS phase latency, session termination reasons, and fallback activation.