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
It's been another productive week on the FreeSWITCH team. We are pleased to let you know that we have officially tagged FreeSWITCH version 1.2.2 in the git repo. Source tarballs are available in the usual location. Thanks to all those whose efforts make more frequent releases a reality. It is much appreciated.
On last Wednesday's conference call we enjoyed a nice Adhearsion presentation by Ben Langfeld and Ben Klang. Adhearsion is a Ruby-based framework for developing telephony applications. Ben and Ben discuss how Adhearsion works, why Ruby is cool for building telephony apps, and why the Adhearsion guys love FreeSWITCH. FreeSWITCH community members are invited to join the Adhearsion team at AdhearsionConf in Palo Alto, CA on October 20-21, 2012. Community members receive a special rate by using discount code AHNLOVESFREESWITCH. Thanks to Ben and Ben for a great presentation with cool slides.
For the next few weeks we look forward to hearing from Daniel Pocock and Scott Godin who will be telling us more about the Repro SIP proxy and the ReSIProcate SIP stack. For many of us it will be our first look at a SIP proxy that does not have its roots in the OpenSER project. We look forward to learning more on this Wednesday's conference call.
Have a great week!