Flexisip
A comprehensive, modular, and scalable SIP server suite written in C++17 for building VoIP services.
What is Flexisip?
Flexisip is a comprehensive SIP server suite written in C++17 that provides essential VoIP functionalities like proxy routing, presence, conferencing, and push notifications. It solves the problem of building scalable, real-time communication services by offering a modular and embeddable SIP stack.
Developers and organizations building VoIP services, telephony applications, or embedded communication systems that require SIP protocol support.
Developers choose Flexisip for its proven scalability, modular architecture, and dual licensing model, which allows both open-source and proprietary deployments with features like push notifications and conferencing built-in.
Overview
Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)
Use Cases
Best For
- Building server-based VoIP services like linphone.org
- Embedding SIP functionality into hardware or IoT devices
- Setting up scalable SIP proxy servers for call routing
- Enabling group voice and video conferencing
- Implementing presence and availability status in communication apps
- Adding push notification support for SIP-based mobile apps
Not Ideal For
- Teams needing a simple, out-of-the-box SIP server with minimal configuration and management overhead
- Projects focused exclusively on WebRTC without deep SIP protocol integration requirements
- Environments demanding a web-based graphical interface for server administration and monitoring
- Use cases where proprietary telephony protocols like H.323 are dominant over SIP
Pros & Cons
Pros
Features like presence and B2BUA can be enabled or disabled via CMake flags (e.g., -DENABLE_PRESENCE=OFF), allowing tailored deployments for specific needs.
Has powered the linphone.org service since 2011, demonstrating reliability in handling real-world VoIP traffic for a free public service.
Offers AGPLv3 for open-source use and proprietary licenses for closed-source projects, providing commercial deployment options without copyleft restrictions.
Can be compiled and run on smaller hardware systems, making it suitable for IoT and embedded applications beyond server-based deployments.
Cons
Requires specific compiler versions (GCC >=13 or Clang >=19) and multiple dependencies like OpenSSL and libsrtp2, with optional features adding libraries like XercesC, complicating setup.
No default web UI; configuration is file-based and requires generating a conf file via CLI and deep understanding of SIP settings, as noted in the 'Configuration' section.
Lacks built-in graphical or web-based administration tools compared to alternatives like Asterisk, relying on manual config edits and CLI for management.
Frequently Asked Questions
Related Projects
Sippy B2BUASippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
KamailioOpen source SIP server widely deployed by carriers and providers. Formerly known as OpenSER
OpenSIPSOpen source SIP server, tracing its roots in OpenSER (presently Kamailio)
RoutrLightweight SIP proxy, location server, and registrar written in Node.js
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