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2025-05-23 17:50:31
- Asterisk - Easy to use but advanced IP PBX system, VoIP gateway and conference server. (Source Code) GPL-2.0 C/deb
- Eqivo - Eqivo implements an API layer on top of FreeSWITCH facilitating integration between web applications and voice/video-enabled endpoints such as traditional phone lines (PSTN), VoIP phones, webRTC clients etc. (Source Code) MIT Docker/PHP
- Flexisip - Complete, modular and scalable SIP server, includes a push gateway, to deliver SIP incoming calls or text messages on mobile device platforms where push notifications are required to receive information when the app is not active in the foreground. (Source Code) AGPL-3.0 C/Docker
- Freepbx - Web-based open source GUI that controls and manages Asterisk. (Source Code) GPL-2.0 PHP
- FreeSWITCH - Scalable open source cross-platform telephony platform. (Source Code) MPL-2.0 C
- FusionPBX - Web interface for multi-platform voice switch called FreeSWITCH. (Source Code) MPL-1.1 PHP
- Kamailio - Modular SIP server (registrar/proxy/router/etc). (Source Code) GPL-2.0 C/deb
- openSIPS - SIP proxy/server for voice, video, IM, presence and any other SIP extensions. (Source Code) GPL-2.0 C
- Routr - A lightweight sip proxy, location server, and registrar for a reliable and scalable SIP infrastructure. (Source Code) MIT Docker/K8S
- SIP3 - VoIP troubleshooting and monitoring platform. (Demo, Source Code) Apache-2.0 Java
- SIPCAPTURE Homer - Troubleshooting and monitoring VoIP calls. (Source Code) AGPL-3.0 Nodejs/Go/Docker
- Wazo - Full-featured IPBX solution built atop Asterisk with integrated Web administration interface and REST-ful API. (Source Code) GPL-3.0 Python
- Yeti-Switch - Transit class4 softswitch(SBC) with integrated billing and routing engine and REST API. (Demo, Source Code) GPL-2.0 C++/Ruby
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