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.
Mike Jerris already posted this on the mailing list but for posterity's sake I am putting it here as well. Please fill out Digium's survey about OSS PBX software. We are a bit curious to see if the 5% reported last year is at all realistic. You can take the survey here.
Thank you for supporting FreeSWITCH and OSS telephony.
Here is an item some of you might find interesting. An intrepid user has created a nice script for installing FreeSWITCH and FreePBX V3 on a Debian Linode using StackScripts. Check out his blog post on the subject. If you use something like this for packaging and/or distribution please send us some feedback on how well it works for you.
Our friends at Sangoma have released a new piece of hardware, the D100, which does on-board transcoding of calls. Check out the writeups over at TMCNet and DigitalJournal.
If you plan on purchasing one of these for your system then please let us know how it works out. We would love to have some feedback on how well this card performs in real-world scenarios.
Here is an interesting heads up from Slashdot: On May 5, at 17:00 UTC, DNSSEC will be rolled out across all 13 root servers. Chances of this going smoothly and not affecting anyone is pretty small. So be prepared! Oh, and if your phones stop working don't call us! Not that you could anyway...
Be sure to check the test site if you have any questions about DNS and your networks.
Many of us have been around FreeSWITCH for a number of years. Like you, I've lost count of how many times someone has asked, "Is there a GUI for FreeSWITCH?" There have been several projects started, all of which are in various stages of development. One project of particular note is FreePBX Version 3. Now the guys over at Chillingsilence have put up a nice little step-by-step tutorial on getting FreePBX V3 up and running with a LiveCD. It's a good read, especially for anyone who is interested in learning more about the current state of FreePBX V3 with FreeSWITCH as the telephony engine.
We are pleased to announce that Vestec, makers of high-quality and low-cost speech recognition engines, is now a certified FreeSWITCH partner. The Vestec Speech Recognition Engine now has a FreeSWITCH connector that allows for direct interaction with the engine right from the dialplan. The connector is free and available for download from Vestec.
I thought this story would be of interest to those who work in the VoIP field. It's about a small company who wanted to save money by going VoIP. They chose Asterisk because "it's cheap" and they wanted to implement it themselves to save even more. Their in-house IT staffers had trouble keeping the system stable. Many FreeSWITCH users are familiar with Asterisk instability issues - indeed they are using FreeSWITCH because of such issues. However there are other things to be gleaned from this example.
The FreeSWITCH team is proud to announce the official release of version 1.0.6, the latest release of the popular open source soft-switch library. FreeSWITCH 1.0.6 builds upon previous FreeSWITCH releases and brings dozens of new features and scores of enhancements in codecs, SIP processing, CPU utilization, TDM hardware support, and more. In the eight months since the release of FreeSWITCH 1.0.4, the core developers and key contributors have made improvements in almost all areas of the project.
New Audio Codecs
The FreeSWITCH team is pleased to announce that users may now purchase licenses for the G.729 audio codec. Licenses are immediately available for FreeSWITCH systems running on Linux. Mac OSX and Windows systems will be supported soon. A single license includes one encoder and one decoder, which is suitable to handle one phone call.
G.729 is a toll-quality, low-bandwidth audio codec supported by many VoIP providers. The voice quality of a G.729-encoded call is about the same as that of G.711, the codec used for standard PSTN calls. However, the amount of bandwidth needed to carry a G.729 call is much lower than that for a G.711 call. Enterprises using G.729 will make more efficient use of available network bandwidth.