AI-native telephony infrastructure

SIP Trunking

One trunk handles inbound DIDs and outbound termination to 190+ countries at ~80ms average latency, with automatic failover and support for G.711, G.729, and Opus. Scales from 10 to thousands of concurrent calls without re-provisioning.

DIDs & Special Series

Provision local, national-rate, and toll-free numbers in 130+ countries via a self-serve API with eKYC verification — no sales call required. Any number can be assigned to a SIP trunk, Voice API flow, or WebRTC session and reassigned later.

Voice API

Make, receive, and control calls across 190+ countries at ~80ms latency. A JSON callback fires the moment a call arrives, and real-time controls let you stop, hold, transfer, or bridge any live call from your server.

WebRTC

Browser-based calling that connects directly into the same ~80ms, bidirectional-audio infrastructure used for PSTN calls — no separate SDK stack, no hardware, no PBX.

Streaming & WebSocket

Audio streams in and out over secure WebSockets in 20ms frames with barge-in detection, so your STT, LLM, and TTS pipeline gets the speed it needs for natural turn-taking. Streaming-first, not bolted on.

Deep Call Controls

DTMF capture for PINs and OTPs, dual-channel call recording stored encrypted and retrievable via API, and VoiceXML for scripting call flows — play audio, gather input, branch, transfer — without building call-control logic from scratch.