Voice · Streaming

The real-time audio pipe for production voice AI.

Fork live call audio over a bidirectional WebSocket to your speech-to-text, language model and text-to-speech with sub-second round trip. Works from PSTN, SIP and WebRTC calls, in the cloud or on-premise, with any agent framework you already use, and with the EnableX AI Voice Agent when you want managed.

Sub-second · WebSocket · Bidirectional · Any STT, LLM or TTS · Open-source agent frameworks · On-prem ready

Stream your first call to your AI

On a connected call, send one request with your voice_id and WSS host, and EnableX opens a bidirectional audio pipe to your STT, LLM, and TTS. Copy, paste, run.

curl -X PUT https://api.enablex.io/voice/v1/call/$voice_id/stream \
  -u APP_ID:APP_KEY \
  -H "Content-Type: application/json" \
  -d '{ "wss_host": "wss://your-host:port" }'

Read the quickstart →  ·  Media Streaming API reference →

sub-second round-trip
Bidirectional WebSocket
30+ languages ASR
On-prem or cloud

Everything a production voice AI agent needs

Stream audio and plug your STT, LLM, TTS, or hand off to the managed EnableX AI Voice Agent. Both paths first-class.

Pipe

Bidirectional WebSocket

Receive caller audio and send synthesised audio back, both directions, base64-encoded mu-law 8 kHz mono, standard telephony audio.

Latency

Sub-second round-trip

Tier-1 routes deliver end-to-end round-trip sub-second: the difference between an agent that feels human and one that feels like a bot.

Stack

Bring your own stack

Works with open-source agent frameworks and any STT, LLM or TTS provider. Plug your stack, we move the audio.

ASR

Speech recognition built in

30+ languages including 13 Indian languages, with code-switching, or bring your own engine.

Sources

PSTN, SIP, WebRTC sources

Stream from inbound PSTN calls, SIP trunks, or WebRTC sessions. Same WebSocket endpoint, same payload format.

Deployment

Cloud, hybrid, on-prem

Run inside your datacentre for sovereign AI and voice biometrics, for regulated voice workloads that must stay in-country.

What teams build with Voice Streaming

From a phone number to a production voice AI agent on one stack.

Voice AI

AI voice agents

STT → LLM → TTS pipelines that handle inbound and outbound calls end-to-end. Pipecat, LiveKit, or your own orchestrator.

AI voice agents
Banking

Voice biometrics & auth

Verify caller identity in the first seconds of a call. On-prem deployment keeps voiceprints inside your data centre.

Voice AI for banking
Insurance

Real-time call analytics

Live transcription, sentiment, intent, and compliance scoring as the call happens, not hours later.

Voice AI for insurance
Healthcare

Tele-triage with AI

Stream patient calls to clinical NLP for symptom triage, appointment routing, and structured EHR capture. HIPAA-ready controls.

Voice AI for healthcare
Contact centre

Supervisor whisper & coaching

Live listen-in, real-time keyword alerts, AI-suggested next-best-utterance, designed to surface risk in the moment.

Talk to sales
Compliance

Real-time recording & archive

Stream both legs to your archive in real time, with encryption in transit and at rest and retention rules per region.

Talk to sales
Spotlight · Real-time WebSocket pipe

Sub-second. Bidirectional. Production-grade.

We built the pipe for what voice AI needs: bidirectional, low-jitter, reconnect-safe, with hooks into the EnableX AI Voice Agent if you want managed instead of DIY.

  • Base64 mu-law 8 kHz mono: the same telephony format your ASR engine already speaks.
  • Bidirectional: receive caller audio, send synthesised audio back. Open-source agent frameworks work out of the box.
  • Sub-second round-trip on tier-1 routes: the threshold where conversation feels natural.
  • EnableX handles WebSocket reconnect, sequencing, clock-sync, and DTMF eventing; you focus on the agent.
  • On-prem deployment for sovereign AI, voice biometrics, and regulated healthcare workloads.

Why teams build voice AI on EnableX

What sets the EnableX Voice Streaming API apart.

01

On-prem & sovereign-AI ready

Our cloud, your cloud, private cloud or fully on-premise, including the speech models, for voice biometrics, sovereign AI mandates and regulated workloads.

02

Speech recognition built in

30+ languages including 13 Indian languages, with code-switching, or bring your own engine on the same pipe.

03

DIY pipe + managed AI Agent

Use the WebSocket directly with your stack, or hand off to the managed EnableX AI Voice Agent: same platform and billing. Both paths supported.

Pricing

Pay-as-you-stream. No platform fee.

Per-minute streaming on top of voice minutes. Bidirectional priced once. Volume pricing is handled by our sales team. Free trial credits, no credit card required.

Voice Streaming API FAQs

A voice streaming API lets you fork the raw audio of a live phone call and stream it in real time over a WebSocket to a speech-to-text engine, an AI voice agent, a sentiment-analysis service, or a supervisor dashboard. Unlike call recording, which gives you a file after the call ends, voice streaming delivers audio as it happens, with latency in hundreds of milliseconds.

The EnableX Voice Streaming API forks the raw audio of any voice call on the platform and streams it over a WebSocket to a destination you specify: your transcription engine, AI voice agent, analytics pipeline, or your own application. It integrates directly with the EnableX AI Voice Agent and the Dialogs Conversational AI Platform.

When a call is placed, EnableX opens a WebSocket to your URL, streams base64-encoded audio frames as the call progresses, and closes the socket when it ends. Control messages (call started, ended, DTMF) travel on the same socket, and EnableX handles reconnection, sequencing, and clock sync.

Base64-encoded mu-law (ulaw) at 8000 Hz mono, the standard telephony-grade format, compatible with Google Speech-to-Text, Deepgram, AWS Transcribe, Azure Speech, OpenAI Whisper. Both directions use the same format.

Yes. Your WebSocket application can receive caller audio and send audio back to play on the call, enabling AI voice agents, conversational IVR, and real-time voice assistants.

Yes. Stream the audio to Google Speech-to-Text, Deepgram, AWS Transcribe, Azure Speech, OpenAI Whisper, or any ASR engine, or use EnableX's built-in speech-to-text, which covers 30+ languages including 13 Indian languages with code-switching.

Yes. Raw streamed audio can be fed to a voice-biometrics engine to verify caller identity in the first seconds of a call, useful for banking IVR, high-value support, and fraud prevention in financial services. On-premise deployment keeps voiceprints inside your data centre.

Yes. Fork an agent call into a supervisor dashboard for live listen-in, transcription with keyword alerts, or AI-driven whisper coaching. Because EnableX streams both sides of the call, you can surface sentiment, escalation risk, and compliance violations in real time.

The EnableX AI Voice Agent is built on the Voice Streaming API. For a managed option, configure the AI Voice Agent and EnableX runs the WebSocket, ASR, LLM and TTS stack for you. Use the Voice Streaming API directly when you want your own LLM, ASR, or TTS.

Sign up for an EnableX account, enable Voice Streaming, generate an API key, and set your WebSocket endpoint URL in the call configuration. Working samples are in the Developer Docs, and you can pilot it on your own call flows during the free trial.

Build with Voice Streaming API: Developer Docs

See EnableX in action.

Talk to sales or start a free trial. No credit card required.

Free trial credits · No credit card · API keys in 2 minutes

Related guide: How to build a voice bot on your own telephony (WebSocket)