Voice API

Programmable Voice API

The EnableX Voice API is a REST API and SDK set for making and receiving phone calls over PSTN, SIP and WebRTC. It includes programmable call flows, IVR, recording, conferencing, number provisioning and a WebSocket stream of live call audio for AI agents, in 50+ countries, with a 99.5% uptime SLA.

Make and receive calls with one REST API. Build IVRs, record calls, bridge agents and stream live audio to your own AI models at sub-second round trip. 100K+ calls a day already run on this stack.

50+ countries · PSTN, SIP, WebRTC · Real-time streaming · 99.5% SLA

Your first call in one request

Authenticate with your App ID and Key, post one JSON body, and the platform places the call and speaks your text to the answerer. Copy, paste, run.

curl -X POST https://api.enablex.io/voice/v1/call \
  -u APP_ID:APP_KEY \
  -H "Content-Type: application/json" \
  -d '{
    "from": "+14155550100",
    "to":   "+442079460000",
    "action_on_connect": {
      "play": { "text": "Hello, and welcome to EnableX.", "language": "en-US", "voice": "female" }
    },
    "event_url": "https://yourserver.com/webhook"
  }'

Read the quickstart →  ·  Full Voice API reference →

50+ countries
sub-second round-trip
3 channels PSTN · SIP · WebRTC
99.5% uptime SLA
Features

Everything a production voice app needs

Place and receive calls, build IVRs, record, conference and stream audio to AI, all pay as you use with no platform fee.

Inbound and outbound

Programmable call flows

Place and answer calls with one REST API. Branch on DTMF, speech or a webhook response. No markup dialect to learn.

IVR and routing

Menu or speech-driven IVR

Build IVRs with built-in speech recognition for natural-language intent capture and skill-based routing to agents.

Real-time AI

Voice Streaming for AI

Bidirectional WebSocket audio for any speech-to-text, language model and text-to-speech, with sub-second round trip end to end.

Recording

Server-side recording

Mono, dual-channel or composite recording, delivered to your storage with signed URLs and encrypted at rest, with retention you control.

Conferencing

Conference and bridging

Multi-party rooms with mute, hold, transfer, whisper and barge. Mix human agents and bot participants on the same media servers.

Numbers and SIP

Numbers, SIP trunks, masking

Local, mobile and toll-free numbers in 50+ countries, SIP trunking and bring-your-own-carrier, plus number masking and click-to-call.

Spotlight · Voice Streaming

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

The Voice API handles the carrier side: interconnects, DTMF, recording and call records. Voice Streaming forks the live audio both ways over WebSocket, so you can plug in any speech-to-text, language model and text-to-speech, including open-source agent frameworks.

  • Bidirectional WebSocket audio, raw PCM or Opus.
  • Works with any STT, LLM or TTS provider and with open-source agent frameworks.
  • Sub-second round trip on direct routes.
  • Carrier interconnect, DTMF, recording and call records handled by EnableX. You focus on the agent.
  • On-premise deployment for regulated voice workloads.

Why teams build voice on EnableX

What sets the EnableX Voice API apart.

01

Full-stack

The same account adds video and messaging, so a voice flow can send a WhatsApp link or escalate to a video call without a second vendor.

02

Deployment choice

Our cloud, your cloud, private cloud or fully on-premise, with controls aligned with ISO 27001 and SOC 2 Type II, for teams whose regulators require call data to stay in-country.

03

White-label

Run EnableX voice under your own brand and bill on your own invoice, as an annual licence.

Pricing

Pay per minute. No platform fee.

Per-minute rates by destination and call type. Recording, streaming and number rental are priced as add-ons. Free trial, no credit card. Volume pricing through sales.

Voice API FAQs

A programmable Voice API lets developers make, receive, route, record and stream phone calls from their own application through REST calls, without running telephony infrastructure or managing carrier compliance. The EnableX Voice API covers PSTN, SIP and WebRTC calls in 50+ countries, with IVR, recording, conferencing, number provisioning and a WebSocket stream of live call audio for AI agents.

Inbound and outbound calls on PSTN, SIP trunks and WebRTC, app-to-app calls between your own users, and bring-your-own-carrier through SIP trunking. Calls can be bridged, conferenced, transferred, recorded and streamed.

Voice Streaming forks the live call audio in both directions over a WebSocket. You connect any speech-to-text, language model and text-to-speech, or an open-source agent framework, and EnableX handles the carrier interconnect, DTMF, recording and call records. Round trip is sub-second on direct routes.

Yes. Recording is server-side, in mono, dual-channel or composite form, delivered to your own storage through signed URLs and encrypted at rest. You control retention.

Yes. The full voice stack runs on our cloud, in your cloud, in a private cloud or fully on-premise, for banks, insurers and healthcare providers whose regulators require call data to stay in-country. On-premise deployments are licensed annually.

Per minute, with rates that vary by destination and call type. Recording, streaming and number rental are priced as add-ons. There is no platform fee, the free trial needs no credit card, and volume pricing is handled by our sales team.

Server-side SDKs and samples for Node.js, Python, Java, PHP and Ruby call the REST API, and WebRTC client SDKs for web, iOS and Android handle in-app calling. The developer portal lists the current set and versions.

See EnableX in action.

Start a free trial, or talk to sales about volume pricing and on-premise. No credit card required.

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