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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
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Patent Office Grants EFF Request for Reexamination of Dangerous VOIP Patent
Submitted by
mcollins on Sat, 02/06/2010 - 16:38
::
For those of you who may not know, the EFF (Electronic Frontier Foundation) has an endeavor called the Patent Busting Project. The goal is to get the USPTO (Patent and Trademark Office) to review bogus patents that do nothing more than stifle innovation and encourage patent trolls to sue legitimate businesses. The EFF has scored another victory, and this one affects many of us in the VoIP and telephony industries. Please check out this news release from the EFF. It's definitely good news for OSS telephony.
If you have any prior art that can be used to assist the EFF with busting bogus patents then please contact them. They are doing us all a favor by getting silly patents overturned. Did you know that someone actually got a patent on the oh-so-clever concept of the hyperlink? Enough said.
-Michael
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