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.
We had a chance to play around on an Amazon Elastic Compute Cloud so we checked out a copy of FreeSWITCH then compiled and ran a conference on it. And it WORKED!
Amazon's EC2 is a web platform the can expand it's computational power on-the-fly to meet the load required using a clustering technology.
It's good to know that our "bend, but don't break" philosophy is paying off!
We checked in mod_lumenvox this week an interface to the LumenVox ASR engine.
More information in the Commit Log.
Due to popular demand. We downloaded and installed a copy of Media Wiki, a much revered wiki platform. A huge thanks to Mike Murdock who has done a great job adding a huge amount of content in a very short period of time.
Here is a small example script.
You should be able to do some really cool stuff with this module.
We added mod_enum today allowing ENUM lookups from the dialplan, as a dialplan or via the console.
We have just finished up the interface for our new speech recognition abstraction layer. Using the new API, it is now possible to attach a speech recognizer in the background that can react to particular speech and turn the text into events that are passed to the channel the same way as DTMF and text messages. We wrapped this up in a nice high level interface and made an example IVR that illustrates a pizza ordering system written in pure java-script.
We've been hard at work coding so there has been a lack of news lately but the bright side is we have a whole bunch of new functionality.
We now support outbound and inbound SIP registration, transfer functions and SIMPLE messaging.
We also have new support for doing SDP pass-through to allow a call passing through the switch to negotiate media with the other end of the call directly.
The biggest addition is probably the new functionality to gateway SIP/SIMPLE to Jingle/GoogleTalk. So now we can not only bridge audio but we can bridge text messages, presence and signalling too!
There is a bunch more work to do to get a release out the door, so stop by and pitch in!
We have had a tester report passing 1.1 million SIP calls today with a peak of 3000 simultaneous channels(!) and the audio was still playing clearly. This was with our new partially finished SIP module. We have a long way to go but this is a good start. You can help! Press the PayPal link on the right!
Embed With Mono
James Martelletti (author of mod_syslog), has committed the first working trunk edition of mod_mono (an open source .NET engine). Now you can write modules in C#, VBscript and many more .NET languages.
In fact Brian Fertig has contributed 3 modules this week based on a head start from Anthony Minessale last Friday.
mod_freehp (a php module)
mod_ruby (a ruby module)
mod_python (a python module)
Add that to the existing language modules:
Now we have 6 Language modules and mono has around 14 of it's own.