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Voice support layers on top of any Agntix agent. Flip the Voice Enabled toggle, pick TTS/STT providers, and you can talk to the agent from your browser (LiveKit) or attach a phone number for PSTN inbound/outbound calls (Twilio, Telnyx, or BYON).

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Prerequisites

  • A signed-in dashboard account at app.agntix.ai.
  • An existing agent (see Build your first agent).
  • For PSTN calling: at least one phone number imported on the Phone Numbers page. Twilio/Telnyx are supported out of the box; you can also import a custom (BYON) SIP trunk.

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Step 1: Enable voice on the agent

Open your agent at /agents/{id}. In the top-right of the General Settings card, flip Voice Enabled on.
Voice Enabled toggle
Once enabled, a Model Type selector appears with two options:
  • Pipeline (STT → LLM → TTS) — three separate models. Best quality and most flexibility; slightly higher latency.
  • Speech-to-Speech (STS) — a single multimodal model (e.g. OpenAI Realtime, Gemini Live) handles audio in and out. Lowest latency; fewer provider choices.
For most use cases pick Pipeline.
Model Type selector

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Step 2: Configure TTS and STT

Switch to the Calls tab on the right side of the agent page. You’ll see the voice-pipeline configuration:
  • TTS Provider — ElevenLabs, Google, OpenAI, Cartesia, Deepgram Aura, and others (depending on what your org has access to).
  • TTS Model — model variants offered by the provider.
  • Voice — the specific voice (preview each by clicking its play icon).
  • STT Provider — Deepgram, OpenAI, Google, Azure, etc.
  • STT Model — provider-specific models (e.g. nova-3-general for Deepgram).
Calls tab TTS/STT configuration
The voice dropdown shows a searchable list, grouped by provider:
Voice picker
When you’re happy, click Save Changes in the page header.

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Step 3: Test in the browser

You don’t need a phone number to test voice. Click the green Test Agent button in the page header. Agntix mints a short-lived LiveKit token, opens a room, and joins the agent automatically.
LiveKit test room
Allow microphone access, say something, and verify:
  • The agent responds quickly (sub-second time-to-first-audio is normal).
  • TTS sounds right (voice, speed, language match what you configured).
  • STT captures your speech accurately (check the Calls tab’s transcript if available).
If latency feels off, see the Advanced voice tuning section at the bottom.

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Step 4: Attach a phone number

To accept real phone calls (inbound) or place them programmatically (outbound), attach a phone number.
  1. Open Phone Numbers from the left nav.
  2. Click Import in the top right.
  3. Pick a provider (Twilio / Telnyx / BYON), select or paste the E.164 number, and choose which agent handles inbound and outbound traffic.
Import phone number
After importing, the row shows the linked agent’s name in the Inbound Agent / Outbound Agent columns. Inbound calls to that number now route to your agent automatically.
Phone number attached to agent

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Step 5: Place an outbound call

To place a real call out:
  1. On the Phone Numbers list, click the phone icon on the row whose outbound agent is the one you want to call from. Or open the agent’s detail page and click Initiate Outbound Call in the header.
  2. Enter the destination phone number in E.164 format (e.g. +14155550123).
  3. Optionally pass session variables — key/value pairs available to the system prompt as {{variableName}}. Useful for personalizing (e.g. customerName=Alex).
  4. Click Call.
Initiate outbound call modal
The call shows up under Logs / History the moment it connects, with full transcript and call recording once it ends.

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Verify it works

  • The Test Agent browser session connects without errors and the agent speaks back.
  • Inbound calls to the attached phone number ring the agent (not voicemail).
  • Outbound calls from the dashboard reach the destination phone.
  • Each call appears in Logs / History with a transcript and audio.

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Advanced voice tuning

The Calls tab also exposes these knobs (collapsed by default — open the Advanced section):
SettingWhat it does
Allow InterruptionsIf on, the agent stops talking when the user starts speaking (barge-in).
Preemptive SynthesisStart TTS before the LLM finishes — cuts perceived latency at the cost of occasional cut-offs.
Welcome ModeWhether the agent greets the user immediately, after silence, or only when spoken to first.
Background Noise ConfigAdd ambient noise (light office, café) to make the call feel more natural.
End-of-Turn DetectionHeuristic (silence threshold) or SmartTurn (model-based). SmartTurn is better in noisy environments.
Interrupt Speech Duration / Min WordsHow long / how many words the user must speak before the agent yields.
Tweak one at a time and re-test — these settings significantly affect perceived quality.

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Common next steps

Call campaigns

Place outbound calls in bulk against a contact list.

Function tools

Let the agent end calls, transfer to a human, or invoke your APIs.

Streaming events

Subscribe to per-utterance events from your backend.

Webhooks

Receive session.updated and call.ended events.

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API alternative

Every dashboard step above maps to a REST endpoint. Minimal scripted version: 
curl -X POST https://api.agntix.ai/v1/chat/voice/agents/$AGENT_ID \
  -H "x-api-key: $AGNTIX_API_KEY" \
  -H "Content-Type: application/json" \
  -d '{
    "tts": { "provider": "elevenLabs", "voiceId": "..." },
    "stt": { "provider": "deepgram", "model": "nova-3-general" }
  }'

curl -X POST https://api.agntix.ai/v1/chat/voice/token \
  -H "x-api-key: $AGNTIX_API_KEY" \
  -H "Content-Type: application/json" \
  -d '{ "agentId": "agnt_…", "userId": "browser-anon-id" }'
The token in the response is what you hand to the LiveKit Web SDK to join the room. See the Voice API reference for the full schema.