VOICE INFRASTRUCTURE● OPUS 48KHZ HD

Carrier-Grade Voice Calls & Global Audio Lounges

Engineered for crystal-clear interactive voice streaming. Independent metered pricing for speakers vs. listeners per 1,000 minutes, dynamic jitter buffering, noise cancellation, and automated media lease enforcement.

Audio Quickstart Guide →View Audio Pricing TiersFirst-Success Journey

Speaker vs. Listener Separation

Fair billing based on actual role classification. Passive listeners consume fraction of bandwidth and are billed at separate listener rates.

Sub-100ms Edge Latency

Global WebRTC routing with dynamic edge relays ensures natural, conversational voice interaction across continents without awkward overlaps.

Autonomous Lease Suspension

Platform issues finite operational time leases. If a customer balance reaches overdraft, streams teardown gracefully upon lease expiry.

Client-Side Audio Connection Example

Standard WebRTC connection token must be generated on your application server.

import { VoxeraAudioClient } from "@voxera/client-sdk";

// Initialize low-latency voice room
const client = new VoxeraAudioClient({
  appId: "app_live_prod_01",
  token: "eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9...", // generated on your secure server
});

// Join room as publisher (speaker) or subscriber (listener)
await client.joinRoom({
  roomId: "voice_lounge_402",
  role: "speaker", // or "listener"
  audioConfig: {
    codec: "opus",
    sampleRate: 48000,
    bitrate: 64000,
    echoCancellation: true,
    noiseSuppression: true,
  },
});

console.log("Joined voice room. Connected to global edge.");

Audio Technical Specifications

ParameterSpeaker RoleListener RoleProtocol / Invariant
Audio CodecOpus Fullband (48 kHz)Opus Adaptive (16 - 48 kHz)RFC 6716 WebRTC Standard
Bitrate Range32 kbps - 128 kbps16 kbps - 64 kbpsDynamic Bandwidth Adaptation
Metering UnitPer 1,000 speaker minutesPer 1,000 listener minutesIndependent RatePlanVersion
Telemetry Freshness≤ 10 seconds≤ 10 secondsNATS JetStream Deduplicated Ingest