FreeSWITCH
Software-defined telecom stack for VoIP, SIP, and real-time media
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Actively maintained
Last commit 26 Aug 2026.

FreeSWITCH is an open-source, software-defined telecom stack used to build and run voice, video, and real-time communications infrastructure on commodity hardware. It is commonly used as a softswitch and media server for SIP-based telephony and can also provide browser-based calling via WebRTC.
Key Features
- SIP signaling and media handling suitable for softswitch and PBX-style deployments
- Native WebRTC capabilities for browser-based real-time communications
- Modular architecture with loadable modules for telephony features and integrations
- Built-in conferencing and real-time media services (including mixing and related functions)
- Common voice features such as call routing, IVR building blocks, call recording, and voicemail support
Use Cases
- Build and operate a SIP softswitch or VoIP application server for carriers and enterprises
- Implement IVR, call routing, and contact-center style call flows
- Provide WebRTC calling to web applications without requiring a separate gateway
Limitations and Considerations
- Configuration and operation can be complex for newcomers due to the breadth of telecom concepts and module options
- Certain advanced capabilities may be available only via commercial modules or enterprise distributions
FreeSWITCH is a mature foundation for telecom and real-time media systems, from embedded devices to large-scale deployments. Its modular design and protocol support make it a flexible core for custom telephony platforms and communication products.
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