FreeSWITCH

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Software-defined telecom stack for VoIP, SIP, and real-time media

5.1k stars 1.8k forks last commit first released

Actively maintained

Last commit 26 Aug 2026.

FreeSWITCH screenshot

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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