Hello everyone,
I’d like to share an open-source project I’ve been working on that may be useful for FreePBX, Asterisk, and SIP users.
CloudSIP.app is a WebRTC SIP Phone available both as a standalone web application and as a browser extension. It is designed to work with FreePBX, Asterisk, and other SIP-compatible platforms supporting WebRTC/WSS.
Features
- SIP over WebSocket (WSS)
- WebRTC audio calling
- Browser Extension for Chrome/Edge
- Standalone Web Phone
- Call history
- Mute, Hold, Transfer
- DTMF support
- Presence and registration status
- Lightweight and easy deployment
- Self-hosted or cloud-hosted
Browser Extension Repository
GitHub:
The extension allows users to make and receive calls directly from their browser without requiring a separate desktop softphone.
Web Phone Repository
GitHub:
The web version can be integrated into existing portals, CRMs, PBX systems, and customer platforms.
Typical Use Cases
- FreePBX deployments
- Asterisk-based PBX systems
- Call centers
- CRM click-to-call integrations
- Remote and hybrid teams
- Multi-tenant VoIP providers
Feedback Welcome
The project is actively being developed and I’d appreciate any feedback, testing results, feature requests, or contributions from the community.
AI Disclosure: AI was used only to help structure and proofread this post.