Voice Platform

Programmable Voice for Every Stack

Terminate calls via REST API, SIP trunk, or WebRTC - whichever fits your architecture. AHOI's voice platform gives you global numbering in 100+ countries, HD audio quality, intelligent routing, and full programmatic call control from a single account.

Build contact centers, embed browser-based calling, connect your PBX to the PSTN, or power millions of automated outbound calls. One platform, three termination paths, unlimited flexibility.

100+

Countries

99.99%

Uptime SLA

HD

Audio Quality

<150ms

Avg Latency

What Programmable Voice Means for Your Business

Programmable voice turns phone calls into software events. Every inbound ring, outbound dial, DTMF press, and voicemail drop becomes an API event you can capture, route, record, transcribe, and act on — in real time, from your own application code.

Traditional telephony locks you into rigid PBX configurations and carrier contracts. Programmable voice breaks that model. You control call flow with code, swap routing logic instantly, spin up new numbers in any country through an API call, and pay only for the minutes and numbers you actually use.

AHOI supports three termination methods — REST API for full programmatic control, SIP trunking for existing PBX and softswitch infrastructure, and WebRTC for browser and mobile calling with no plugins. All three share the same global carrier network, the same number inventory, and the same real-time analytics dashboard, so you can mix and match based on each use case without managing separate providers.

1

Provision Numbers

Search and activate local, toll-free, or mobile numbers in 100+ countries via API or dashboard.

2

Choose Termination

Connect via REST API, register a SIP trunk, or embed WebRTC calling in your app.

3

Define Call Logic

Build IVR trees, routing rules, queues, and failover paths with code or visual builder.

4

Go Live

Calls flow through AHOI's global carrier network with HD audio and real-time event webhooks.

5

Monitor & Optimize

Live dashboards, call recordings, transcriptions, and analytics to continuously improve quality.

REST API

API-Driven Call Control

Full programmatic control over every aspect of a call. Initiate outbound calls, accept inbound calls, and manipulate live calls — all through RESTful API requests and webhook callbacks. The most flexible option for custom applications and automation.

  • Outbound call initiation with a single POST request
  • Inbound call handling via configurable webhook URLs
  • Mid-call control: transfer, mute, hold, record, bridge
  • Call flow defined in code — IVR, queues, routing logic
  • Real-time event streaming and status callbacks
SIP Trunking

SIP Trunk Termination

Connect your existing PBX, softswitch, or SBC to AHOI's global carrier network via standards-based SIP trunking. Keep your current infrastructure while gaining access to elastic capacity, global numbers, and per-second billing.

  • Standards-compliant SIP (UDP, TCP, TLS, SRTP)
  • IP authentication or credential-based registration
  • Elastic trunking — unlimited concurrent call capacity
  • Compatible with FreeSWITCH, Asterisk, 3CX, Broadsoft, Metaswitch
  • Failover and load balancing across multiple endpoints
WebRTC

Browser & Mobile Calling

Embed voice calling directly into web and mobile applications with no plugins, downloads, or phone hardware. Users make and receive PSTN-quality calls from any browser or native app using AHOI's JavaScript and mobile SDKs.

  • JavaScript SDK for browser-based calling
  • iOS and Android SDKs for native mobile apps
  • PSTN-to-browser and browser-to-PSTN bridging
  • Opus codec support for HD audio in-browser
  • Token-based authentication with short-lived credentials
Global Network

Global Numbering & Carrier-Grade Infrastructure

Provision local, national, mobile, and toll-free numbers in over 100 countries from a single API. Every number is backed by direct carrier interconnects, geo-redundant media servers, and a private backbone that keeps latency under 150ms worldwide.

Local DID Numbers Toll-Free Numbers Mobile Numbers National Numbers Short Codes 100+ Countries STIR/SHAKEN
Global Number Inventory
United States Local, Toll-Free, Short Code
Live
United Kingdom Local, National, Mobile
Live
Australia Local, Toll-Free, Mobile
Ready
Canada Local, Toll-Free, DID
Live
Germany Local, National, Toll-Free
Provisioning

Call Termination & Origination

Inbound and outbound calling across the global PSTN with carrier-grade reliability and competitive per-minute rates.

  • Outbound termination to 100+ countries
  • Inbound origination with local presence worldwide
  • Elastic capacity — no trunk provisioning limits
  • Automatic codec negotiation (G.711, G.729, Opus)
  • Per-second billing with no minimum commitments

Routing & Call Flow

Intelligent call routing, IVR, queuing, and failover logic that adapts in real time to deliver the best customer experience.

  • Time-of-day, geo, and skills-based routing
  • Multi-level IVR with speech recognition (ASR)
  • Call queues with estimated wait time and callbacks
  • Automatic failover and disaster recovery routing
  • Whisper, barge, and monitor for agent supervision

Voice Intelligence

AI-powered transcription, sentiment analysis, and call insights that turn raw audio into actionable business data.

  • Real-time and post-call transcription
  • Speaker diarization (who said what)
  • Sentiment and keyword detection
  • Call summarization and action item extraction
  • Voice biometric authentication

Security & Compliance

Enterprise-grade security, encryption, and regulatory compliance built into every layer of the voice platform.

  • TLS/SRTP encryption for all signaling and media
  • STIR/SHAKEN caller attestation
  • PCI DSS compliant call recording with auto-redaction
  • HIPAA-eligible voice infrastructure
  • GDPR-compliant data handling and storage

Outbound Calling

Initiate calls programmatically via API with custom caller ID, call progress detection, answering machine detection (AMD), and configurable ring timeout and retry logic.

Inbound Call Handling

Route inbound calls to webhooks, SIP endpoints, WebRTC clients, or PSTN numbers. Define complex routing trees with conditional logic, time rules, and geographic overrides.

Interactive Voice Response

Build multi-level IVR menus with DTMF and speech input, text-to-speech in 30+ languages, dynamic prompts from your database, and seamless handoff to live agents.

Call Recording

Record full calls or specific segments with dual-channel stereo recording, pause/resume controls, automatic storage, and secure download via API. PCI-compliant redaction available.

Real-Time Transcription

Stream live transcripts via WebSocket as the call happens. Power real-time agent assist, keyword alerts, compliance monitoring, and automatic call notes without post-call delays.

Conferencing

Create ad-hoc or scheduled conferences for up to 250 participants with moderator controls, mute/unmute, hold music, recording, and real-time participant management via API.

Voicemail & AMD

Answering machine detection identifies live humans vs. voicemail within seconds. Leave pre-recorded messages automatically, or route live pickups to agents without delay.

Text-to-Speech & SSML

Generate natural-sounding speech in 30+ languages and 60+ voices using neural TTS. Fine-tune pronunciation, speed, pitch, and pauses with full SSML markup support.

Global Number Management

Search, provision, and manage phone numbers in 100+ countries through a single API. Local, national, mobile, and toll-free number types with instant activation.

Number Porting

Port existing numbers to AHOI with managed LOA submission, carrier coordination, and zero-downtime cutover. Bulk porting supported for large number inventories.

Call Analytics & CDRs

Detailed call detail records for every call with duration, cost, quality metrics (MOS, jitter, packet loss), and custom metadata. Export via API, webhook, or scheduled reports.

SIP Domain Routing

Register custom SIP domains and route calls between SIP endpoints, PSTN, and WebRTC clients. Support for SIP REFER, NOTIFY, and in-dialog transfers across all endpoints.

Build Voice Apps in Minutes

Our programmable voice APIs, SDKs in six languages, and WebRTC client libraries let you embed calling into any application. Full documentation, a sandbox environment, and webhook debugging tools are included with every account.

REST API Python Node.js PHP Java C# WebRTC SDK
make-call.py
import ahoi

# Initialize client
client = ahoi.Client(api_key="your_key")

# Make an outbound call
call = client.calls.create(
  to="+1234567890",
  from_="+1987654321",
  url="https://app.co/voice.xml",
  record=True,
  transcribe=True
)

print(call.sid, call.status)

Any Termination, One Platform

REST API, SIP trunking, and WebRTC all connect to the same global network and number inventory. Choose the right method for each use case without managing separate providers or contracts.

True Global Numbering

Provision local, national, mobile, and toll-free numbers in 100+ countries through a single API call. Give your business a local presence anywhere without local infrastructure or carrier relationships.

Sub-150ms Global Latency

Geo-distributed media servers and a private carrier backbone keep round-trip latency under 150ms for calls anywhere in the world. No jitter, no dropped audio, no awkward silences.

Elastic Scale

From 10 concurrent calls to 100,000. No trunk planning, no capacity reservations. The platform auto-scales to your traffic patterns and you pay only for what you use, billed per second.

Built-In Voice AI

Real-time transcription, sentiment analysis, voiceprint authentication, and answering machine detection come native to the platform. No third-party integrations or additional billing required.

Enterprise Security

TLS/SRTP encryption on every call, STIR/SHAKEN attestation, PCI DSS-compliant recording, HIPAA-eligible infrastructure, and GDPR-ready data controls. Security is not an add-on.

Ready to Build With Voice?

Start building with AHOI's programmable voice platform. Whether you need API call control, SIP trunking, or WebRTC — our team will help you architect the right solution.

Get Started