PulseRTC
LIVEKIT SFU // COTURN TURN // TYPE-SAFE WEB SDK

Real-Time Voice & Video Calling Infrastructure.

Launch your own developer-first RTC platform with multi-tenant App IDs, prepaid minute billing, ephemeral STUN/TURN traversal, and a clean TypeScript Web SDK.

< 100ms
Regional SFU hop
48 kHz
Full-band Opus
1,000 Mins
Free signup grant
SFU TELEMETRY SIMULATOR // LIVEKIT + COTURN
Median Latency
128 ms
720p/1080p Simulcast
Audio Quality (MOS)
4.53 / 5.0
Noise Suppression + Redundant Opus FEC
Transport Security
DTLS-SRTP
Ephemeral HMAC TURN + Zero Retention
Platform Capabilities

Everything required to run an RTC SaaS

Built cleanly across a TypeScript SDK, a Fastify control plane with Postgres/SQLite persistence, and Docker-ready LiveKit + Coturn media nodes.

48kHz Full-Band Voice SDK

Opus stereo audio with adaptive jitter buffering, echo cancellation, and active speaker detection.

Adaptive Simulcast HD Video

Multi-layer WebRTC video with dynamic bitrate scaling, camera switching, and screen sharing.

1-on-1 & Group SFU Rooms

Powered by LiveKit SFU so participants send one upstream stream and subscribe selectively.

Coturn STUN / TURN Relay

Ephemeral HMAC-SHA1 TURN credentials bypass strict corporate firewalls and symmetric NATs.

Multi-Tenant App Provisioning

Issue isolated App IDs (prtc_...) and App Secrets with per-tenant room JWT minting.

Prepaid Minute Metering

Webhook-driven CDR billing and automated HTTP 403 ZERO_BALANCE cutoff when quotas reach zero.

@pulsertc/sdk

Integrate voice & video calls in 10 lines

Your customers never touch raw LiveKit secrets. Their backend authenticates with x-app-id and x-app-secret against your control plane, which enforces minute balances and mints scoped room tokens.

import { PulseClient } from '@pulsertc/sdk';

const client = new PulseClient({
  appId: 'prtc_live_8f92a1c4...',
  tokenEndpoint: 'https://api.yourdomain.com/v1/rtc/token',
});

await client.joinRoom({
  roomName: 'support-call-42',
  identity: 'agent_riya',
  audio: true,
  video: true,
});