FreeSWITCH is a scalable open source cross-platform telephony platform
designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.
FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.
FreeSWITCH supports various communication technologies such as Skype, SIP, H.323 and GoogleTalk making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.
FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.
FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz and can bridge channels of different rates. The G.729 codec is also available under a commercial license.
FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.
LYLIX offers hosting services for Freeswitch based distributions. Each distribution provides it's own unique control panel to realize a complete Freeswitch PBX system.
If you are unsure which Freeswitch distribution is best for you, follow the links below to each distribution's page on our site. If you still require further information, visit the distribution's respective website with the links provided in the sidebar.
If you prefer to build your own VOIP system using a bare-bones Linux VPS, check out our Linux VPS services.