dTelecom is an open-source communication infrastructure enabling audio/video conferencing and live streaming using the WebRTC technology. It's crafted to deliver all necessary components to integrate real-time video and audio functionalities into your applications.
dTelecom's server is written in Go, using the awesome Pion WebRTC implementation.
sudo sysctl -w net.inet.udp.recvspace=2500000- Scalable, distributed WebRTC SFU (Selective Forwarding Unit)
- Modern, full-featured client SDKs
- Built for production, supports JWT authentication
- Robust networking and connectivity, UDP/TCP/TURN
- Advanced features including:
- speaker detection
- simulcast
- end-to-end optimizations
- selective subscription
- moderation APIs
- webhooks
- distributed and multi-region
Client SDKs enable your frontend to include interactive, multi-user experiences.
| Language | Repo | Declarative UI | Links |
|---|---|---|---|
| JavaScript (TypeScript) | client-sdk-js | React | docs | JS example | React example |
Server SDKs enable your backend to generate access tokens, call server APIs, and receive webhooks. In addition, the Go SDK includes client capabilities, enabling you to build automations that behave like end-users.
| Language | Repo | Docs |
|---|---|---|
| JavaScript (TypeScript) | server-sdk-js | docs |
dTelecom Cloud is the fastest and most reliable way to run dTelecom. Every project gets free monthly bandwidth credits.
Sign up for dTelecom Cloud.
Pre-requisites:
- Go 1.18+ is installed
- GOPATH/bin is in your PATH
Then run
git clone https://github.com/dTelecom/livekit
cd livekit
./bootstrap.sh
mageWe welcome your contributions toward improving dTelecom! Please join us to discuss your ideas and/or PRs.
dTelecom server is licensed under Apache License v2.0.