Hello Asterisk Community,
I would like to introduce CloudSIP, an open-source WebRTC SIP client developed for organizations and service providers looking for a modern browser-based telephony experience.
CloudSIP.app enables users to make and receive SIP calls directly from a web browser using WebRTC technology, eliminating the need for traditional desktop softphones while maintaining compatibility with Asterisk deployments.
Key Features
- WebRTC audio calling
- SIP over Secure WebSocket (WSS)
- Incoming and outgoing calls
- Call history
- Hold, Mute, Transfer
- DTMF support
- Responsive user interface
- Easy integration with existing platforms
Available Projects
CloudSIP.app Web Phone
A standalone browser-based SIP phone suitable for customer portals, CRMs, PBX interfaces, and communication platforms.
Repository:
CloudSIP.app Browser Extension
A browser extension that allows users to manage calls directly from Chrome or Edge without keeping a dedicated phone window open.
Repository:
Project Goals
The primary objective of CloudSIP.app is to provide a lightweight, open-source, and easy-to-deploy WebRTC SIP solution that integrates seamlessly with Asterisk-based environments.
The project is designed for:
- VoIP service providers
- Call centers
- Business PBX deployments
- CRM click-to-call integrations
- Remote and hybrid work environments
- Multi-tenant communication platforms
Community Feedback
The project is under active development, and feedback from the Asterisk community is highly appreciated.
I am particularly interested in hearing about:
- Real-world deployment experiences
- Browser compatibility results
- WebRTC interoperability observations
- Feature requests and enhancement suggestions
- Security and scalability recommendations
Links
Web Phone:
Browser Extension:
Thank you for taking the time to review the project. I welcome any feedback, questions, or contributions from the community.