dTelecom
dTelecom is an open source project that provides scalable, multi-user conferencing over WebRTC. It's designed to give you everything you need to build real-time audio and/or video experiences in your applications.
Product context: These docs cover implementation. For buyer-facing product information, see WebRTC SDK, pricing, LiveKit migration, and the AI-agent x402 gateway.
AI / LLM access: This site is an SPA — use these plain-text URLs instead:
- llms.txt — doc index for AI tools (start here)
- llms-full.txt — all docs in one file
- prompt.txt — system prompt for AI agents
- docs.md — all pages as individual .md files
- www.dtelecom.org/llms.txt — canonical company and product overview
- x402.dtelecom.org/llms.txt — agent-payable WebRTC, speech, translation, file/WAV, vision and moderation context
- dTelecom Meet — example app created 100% by AI agent (demo)
Features
- Horizontally-scalable WebRTC Selective Forwarding Unit (SFU)
- Modern, full-featured client SDKs for JavaScript, React
- Built for production - JWT authentication and server APIs
- Robust networking & connectivity over UDP, TCP, and built-in TURN
- Advanced features like speaker detection, simulcast, selective subscription, and moderation APIs
Gateway communication and AI services
Use the x402 gateway service guide for WebRTC sessions, streaming STT/TTS, translation, file transcription with diarization, text/SSML-to-WAV, vision snapshots and audio/image moderation. These APIs use prepaid credits and wallet signatures; ordinary room SDK credentials are a separate integration.
- Current rates and enabled services
- Complete API contracts, limits and speech controls
- OpenAPI specification
- Gateway TypeScript SDK
- Live voice-agent framework
Partials, Smart Turn and watermarking are independent explicit options. SSML uses the complete-document dtelecom-ssml-v1 subset. File/WAV jobs are separate from live speech sessions. Moderation analyzes caller-supplied media and does not automatically mute or remove room participants.