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
Welcome to the last Monday of September 2012!
We've had quite the interesting week. Perhaps the most interesting item the team dealt with was a vulnerability in the Sofia SIP stack that would cause a segmentation fault while processing a specially crafted SIP message. Just to show you how nimble the FreeSWITCH developers are, from the time the vulnerability was reported it took less than a day to fix, test, and roll a new version of FreeSWITCH. We encourage everyone on 1.2 to get updated to version 1.2.3 as soon as possible. (The fix is also in the 1.3 development branch as of last Wednesday, September 19.) We tip our hats to Anthony and the rest of the dev team for their hard work on our behalf.
Last week's conference call was also very informative. We received an introduction to the repro SIP proxy software. We look forward to this coming Wednesday where Scott Godin and Daniel Pocock will continue the discussion and will get deeper into how to set up the proxy and use it with FreeSWITCH. If you haven't already tried to install repro please do so. Daniel has a nice tutorial over at OpenTelecoms.org - be sure to check it out and bring your questions on Wednesday.
Finally, we'd like to draw your attention to this blog post by long time FreeSWITCH and open source telephony supporter Kristian Kielhofner. Kristian reports that his company, Star2Star Communications, is sponsoring the FreeSWITCH stable branch by giving direct financial support to the project. This allows for a full-time team member to work on things like the stable branch and packaging as well as community interaction and documentation. We appreciate those who support FreeSWITCH and open source telephony!
Have a good week and we'll see you again in October.