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.
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
| Parameter | Speaker Role | Listener Role | Protocol / Invariant |
|---|---|---|---|
| Audio Codec | Opus Fullband (48 kHz) | Opus Adaptive (16 - 48 kHz) | RFC 6716 WebRTC Standard |
| Bitrate Range | 32 kbps - 128 kbps | 16 kbps - 64 kbps | Dynamic Bandwidth Adaptation |
| Metering Unit | Per 1,000 speaker minutes | Per 1,000 listener minutes | Independent RatePlanVersion |
| Telemetry Freshness | ≤ 10 seconds | ≤ 10 seconds | NATS JetStream Deduplicated Ingest |