Welcome To FreeSWITCH
The World's First Cross-Platform Scalable FREE Multi-Protocol Soft Switch
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.
We support 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 builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.
FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.
Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipXecs, The Asterisk Open Source PBX and Call Weaver.
a Spec Sheet is available on our Wiki.
Announcing The First Truly Free RTMP Solution For Open Source VoIP!
The FreeSWITCH team is pleased to announce that we have officially released mod_rtmp - the first truly free RTMP solution for open source VoIP and telephony! A special thanks goes out to Barracuda Networks for sponsoring the professional development of this module and allowing it to be freely released under the MPL along with the rest of FreeSWITCH.
RTMP - the Real Time Messaging Protocol - was originally developed by Macromedia to allow the streaming of audio and video through the ubiquitous Flash player. After Adobe acquired Macromedia they released the RTMP specifications which has allowed third party developers to create server-side applications that work with Flash and other RTMP-enabled clients. mod_rtmp turns FreeSWITCH into an RTMP server, allowing you to bridge RTMP client streams to SIP and TDM connections as well as conferences. Currently, mod_rtmp supports the Speex audio codec.
One of the applications of this technology is to allow Web site visitors to make phone calls right from their browser. A company's Web page can detect Flash and offer the visitor an option to "Click Here" to speak to a customer service representative.
To illustrate the usefulness of mod_rtmp, Anthony Minessale has added a Flash-based connector to the public FreeSWITCH conference. Browse there and try it out. Be sure that Flash is installed for your browser and that you have a headset or other audio device. For security reasons you will need to explicitly allow Flash to access to your audio device. Click "Call FreeSWITCH Conference" and then enter a name in the popup box. Click "Call" and you will be connected to the FreeSWITCH public conference. (Be sure to open the keypad and press zero to unmute!) If others are present you will be able to speak with them. Audio devices that support wide-band audio will produce higher quality audio.
Would you like to learn more about FreeSWITCH and mod_rtmp? Be sure to join the FreeSWITCH developers at ClueCon 2011 this August 9-11 in Chicago. ClueCon is a great opportunity to meet the authors of some of your favorite open source VoIP and telephony software projects.