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.
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.
I just saw this interesting series of posts from "patris1" over at persianadmins.com. It discusses the "marriage" of FreeSWITCH and Speech Server 2007. The posts are all written in English, but it looks like the author is multilingual so if you know of any who speaks Farsi and is looking for other FreeSWITCHers to network in that language then this author would probably be a good resource.
We would like to let everyone know that the development team is now using git for management of the FreeSWITCH project. The SVN repo is now read-only. This primarily affects only those who have commit access. For more information on using git we recommend this informative site: http://progit.org/book/
The folks over at Ars Technica have posted part two of a nice introduction to the SIP protocol. This series of articles is a pretty nice and gentle introduction to the protocol with which all seem to have a love/hate relationship. It's definitely worth a read, especially for those who are still a little new to VoIP and SIP.
Here's a provocative post up over at FierceVoIP suggesting that possibly Polycom and Siemens Enterprise will merge. Evidently Polycom is feeling the heat from "recent consolidation of its competitors." Curious, no?
This post is up over at FierceVoIP regarding "Skype on Verizon's network." Too bad it's Skype on the voice network and not on the data network. Oh well, maybe next time...
Congratulations to Brian West for having the honor of the 17000th SVN commit for the FreeSWITCH project! Development on the project continues to gain steam as the core developers (Anthony Minessale, Michael Jerris, and Brian K West) have been joined by a small but talented group of programmers like Mathieu Rene and Rupa Schomaker. The pace of commits has sped up over the past few years as these and others have joined the project.
We would like to say thanks to all those in the FreeSWITCH community who continue to support FreeSWITCH and open source.
This interesting blog post came across the wire today. It's from a tech professional who needed a scalable solution to handle thousands of inbound and outbound calls with IVR capabilities. Those of us who have taken on challenges like his can appreciate the magnitude of the project and how difficult it can be to implement a viable solution. It sure sounds like he's happy with the solution he built around FreeSWITCH.
This press release from Sangoma just came in today. Sangoma has been a great supporter of the FreeSWITCH project and they are now doing even more! Sangoma is supporting the new TDM abstraction layer, FreeTDM, that will be replacing OpenZAP. More details will be forthcoming.
The FreeSWITCH team would like to thank all those corporations, organizations, and individuals who support the FreeSWITCH project.
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!