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.
FreeSWITCH Weekly News and Notes
Happy December to everyone!
Last week was painful for many of us as we were dealing with a sustained DDoS attack on most of our infrastructure. Kudos to the guys for working through it. It seems the worst is over and we can get back to the business at hand: doing FreeSWITCH stuff. :)
In spite of the drama last week we did have a conference call and we released 22.214.171.124! We discussed mostly the details of the DDoS we experienced and how the community can assist in the future so that we can mitigate the effects of such an occurrence. With the community's help we will be more resistant to the effects of any future attacks. We appreciate the outpouring of support we received from everyone.
This week we will go back to discussing FreeSWITCH. We are still finalizing future guests so this week we'll do another installment of tips and tricks from the FreeSWITCH community. Among other things I will be showing how Chris Rienzo (IRC: crienzo) and I used the source this weekend to figure out what the XML preprocessor can do and get the wiki updated. I'll then show a simple example of the always-present-but-previously-undocumented command can do. As an added bonus we'll have an update on the ClueCon 2012 videos!
Thanks and have a great week.