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.
I thought this was a nice article over at Tech Crunch. (For those who don't know, TC does lots of articles on web startups and cool new technology stuff.) It discusses a cool project called PBWorks (formerly PBwiki) that uses FreeSWITCH to power the voice component. Check it out!
We frequently get questions about billing solutions for FreeSWITCH. I've personally not had the opportunity to use it, but I've certainly heard of ASTPP. Evidently the ASTPP project has had FreeSWITCH support for some time. Here is a somewhat detailed guide to installing ASTPP on a CentOS system posted on the ASTPP site. We would love to hear some freedback from our users. If ASTPP is a viable option for VoIP billing on a FreeSWITCH server then we'd like to be able to recommend it where appropriate.
The folks over at TMCnet.com have a new blog post up that features some information about FreeSWITCH, CudaTel, and a few quotes from Anthony. It's a nice read. The author has a bit of confusion over the relationship between FreeSWITCH and Barracuda so we'll just mention here that FreeSWITCH is not a "subsidiary" of CudaTel or Barracuda Networks. (The author says that "FreeSWITCH" is a "company" that is a "subsidiary" but, of course, FreeSWITCH is simply a project, not a "company.")
Regardless, this is a nice indication the OSS telephony is indeed gaining momentum and that FreeSWITCH is garnering some serious attention from the telecom industry.
For those of you who may not know, the EFF (Electronic Frontier Foundation) has an endeavor called the Patent Busting Project. The goal is to get the USPTO (Patent and Trademark Office) to review bogus patents that do nothing more than stifle innovation and encourage patent trolls to sue legitimate businesses. The EFF has scored another victory, and this one affects many of us in the VoIP and telephony industries. Please check out this news release from the EFF. It's definitely good news for OSS telephony.
Here's an interesting article over at Fierce VoIP: Within 3 years VoIP will have a 79% penetration rate in businesses. I'm assuming this is worldwide but the article doesn't say specificially. In any case, it's another good sign that FreeSWITCH (along with VoIP technologies, be they OSS or proprietary) will be used a lot in the coming months and years.
We are pleased to announce that the FreeSWITCH development team is going to meet in one place for the release of version 1.0.5. The official release will take place the week of February 8, 2010! To celebrate we invite and encourage the community to buy the development team dinner one night that week. To donate, simply click the PayPal button on the right side-bar of this page.
On behalf of the FreeSWITCH development team: thank you for supporting the project and thank you in advance for giving the guys a well-deserved dinner.
All those interested in learning more about VoIP and SIP should head over to Ars Technica and read An Introduction to the SIP Protocol, Part 1. It is a slightly technical introduction to how SIP works and is a good foundation for those who need to understand more about how SIP works with the rest of the network. I will post links to any followup articles that Ars puts up.
We'd like to extend our thanks to Sangoma, a Friend of FreeSWITCH, for continuing to support our favorite open source telephony project. There is a nice write-up over at TMCnet.com discussing Sangoma's future plans for support not only the FreeSWITCH project itself but also the FreeSWITCH community and the ClueCon open source telephony developer's conference.
This just came in a few days ago. It's a blog post over at chilling silence wherein the author describes his (rather pleasant) experiences dealing with FreeSWITCH and FreePBX v3. It's a nice read that we thought you might enjoy.