FreeSWITCH is the world's first cross-platform multi-protocol softswitch which is scalable and free. It is a VoIP open source platform created for routing and connecting the most famous communication protocols using audio, video, text or other mediums.
The founders of FreeSWITCH are former developers of the popular Asterisk open source PBX. Although originally FreeSWITCH was created solely for design purposes such as modularity, cross-platform, support, scalability and stability, now it is developed to do even much more.
FreeSWITCH is designed to fill the space of proprietary commercial solutions. It is also a perfect platform for creating and managing many telephony applications with a rich set of free tools. Thanks to its flexible design, it is a great platform for PBX to transmit switch, TTS (text-to-switch) conversion, audio/video conferencing and even VoIP. Moreover, it is not limited with this.
Every day a lot of developers and users help to improve the project. FreeSWITCH now supports Skype, SIP, H.323 and WebRTC, making their interface user-friendly to operate easily with other open sources like sipXecs, Call Weaver, Bayonne, YATE or Asterisk.
Conferencing module for video transcoding and MCU functionality
Transparent proxy for performing as SBC and proxy T.38 and others
Many advanced SIP features such as presence/BLF/SLA; TCP TLS and sRTP
Wide and narrow band codecs to connect legacy devices to new technologies
Thanks to developers, exceptional support is available within FreeSWITCH. They put all their efforts in being involved in open source; they are donating code and even other resources to telephony projects such as openSER, sipXecs, the Asterisk Open Source PBX and Call Weaver. You can find A Spec Sheet on the FreeSWITCH Confluence.
The latest version of FreeSWITCH™ 1.6 is released in 2015, which later has come with some additional improvements. FreeSWITCH is based on MPL 1.1 license.
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