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
We are all back to a full week after many of us enjoyed some well-deserved time off last week. However, even though there was a holiday here in the US, the intrepid FreeSWITCH development team was working hard on your behalf. As Ken Rice previously mentioned, Anthony spotted a potential issue in the recently released 1.2.5 version. Therefore, this past Saturday they made 126.96.36.199 available for us. Many thanks to those who work so hard to make sure that FreeSWITCH is running smoothly for us all.
On last week's conference call we spent some time getting everyone up to speed on how to edit the FreeSWITCH wiki, specifically focusing on channel variables pages. Updating documentation is one of the least glamorous aspects of maintaining an open source project. Many thanks to those who've stepped up over the past weeks and months to help us out. With the end of the year upon us we are slowing down a bit in our speaking schedule for the weekly community conference call. We have a few things in the works but nothing yet scheduled. On this week's call we will be doing a community scrum. Be sure to bring your questions and topics for discussion. If you have a tip or trick that you'd like to share with the group that would be most welcomed. If time permits we will crowdsource a few selected questions from the mailing list.
Have a great week and we'll talk to you on Wednesday.