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 October! I hope the weather is nice where you are. Here it's still above 100F. :)
Last week was a little bit quieter than the previous few weeks. I had a chance to work on the FreeSWITCH change log and I made a list of some of the APIs, dialplan tools, and channel variables. These have all been added since 1.2.0 was initially released in early August. All of them have wiki entries - thanks to those who took the initiative to do add them. Feel free to add your knowledge and experience to the mix.
Last week's conference call was an object lesson in the challenges of getting a SIP proxy working with TLS and FreeSWITCH. This week we are going to change direction and look at something that has been slowly (and painfully) advancing the past few years: mobile VoIP. We will be having Daniel Pocock share with us some information about Lumicall, an open source mobile VoIP client for Android devices. There is also a service component to Lumicall and we'll be learning about that as well. Come join us to see the state of mobile VoIP.
We are working on some fresh presentations for later this month. We hope to have an update on e164.org and how we can all get involved. We are also preparing a presentation on how to perform some of the data-gathering techniques that are needed for basic and advanced troubleshooting. If you have some input on these or other presentations please let me or Ken Rice know.