Hello everyone,
I'd like to share an open-source project I've been working on for the Asterisk community.
CloudSIP is a WebRTC SIP Phone that allows users to make and receive SIP calls directly from a web browser without installing a traditional softphone. It is available both as a standalone web application and as a browser extension.
Web Application
https://github.com/CONNXTA/cloudsip.app
Browser Extension
https://github.com/codebg/cloudsip.app-browser-extension
The goal was to provide a lightweight and modern SIP client that works directly in the browser while remaining simple to deploy and integrate with existing Asterisk environments.
Many organizations are moving away from desktop softphones and prefer browser-based communication tools. This project aims to simplify that transition while keeping the deployment process straightforward.
The project is actively being developed and I would appreciate feedback from other Asterisk users regarding:
If anyone is interested in testing, contributing, or sharing ideas, I'd be happy to discuss them.
I'd like to share an open-source project I've been working on for the Asterisk community.
CloudSIP is a WebRTC SIP Phone that allows users to make and receive SIP calls directly from a web browser without installing a traditional softphone. It is available both as a standalone web application and as a browser extension.
Main Features
- SIP over WebSocket (WSS)
- WebRTC calling
- Call history
- Hold, Mute, Transfer
- DTMF support
- Incoming call notifications
- Browser-based softphone experience
- Easy integration into existing portals and CRM systems
Repositories
Web Application
https://github.com/CONNXTA/cloudsip.app
Browser Extension
https://github.com/codebg/cloudsip.app-browser-extension
Why I Built It
The goal was to provide a lightweight and modern SIP client that works directly in the browser while remaining simple to deploy and integrate with existing Asterisk environments.
Many organizations are moving away from desktop softphones and prefer browser-based communication tools. This project aims to simplify that transition while keeping the deployment process straightforward.
Looking for Feedback
The project is actively being developed and I would appreciate feedback from other Asterisk users regarding:
- SIP registration reliability
- WebRTC interoperability
- Audio quality
- Browser compatibility
- Feature suggestions
If anyone is interested in testing, contributing, or sharing ideas, I'd be happy to discuss them.
Phone System 