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 have bugs in the core now, media bugs that is! A mechanism to tap into a session and get a mixed stream of the inbound and outbound audio.
FreeSWITCH now supports the G.726 8 Kilohertz Audio codec!
Most implementors only support the easy 32kps version but we wanted to have it all.
Now that the the dust has settled, we now support the G.726 codec at all 4 bitrates:
16kps, 24kps, 32kps, 40kps
We also support the G.726 codec at both the proper RFC3551 mode and the improper mirror image mode (AAL2) which is implemented on many non-compliant software and hardware.
If you need to pack some bits, we have an interface for that now too in the core.
The code was updated this week to allow multiple destinations to be called from 1 dial string. You also have the option to bridge the call to the first of the targets to dial a paticular DTMF while listening to a file loop or the first target to survive any application you choose without being hung up.
Please Digg it!
Coding was finished today to change mod_dingaling to use the new variant of the GoogleTalk protocol. It now supports 16khz so you can use FreeSWITCH in conjunction with GoogleTalk to run apps like mod_conference in high definition!
FreeSWITCH's mod_cdr attaches to the global hangup event handler, and can log to 3 logging methods at the current time: Perl Data Dumper records (1 per file), CSV format, and MySQL using prepared statements (requires 4.1.x or greater). Both sides of every call are logged, adding additional possibilities for data analysis. Arbitrary channel variables can be set and logged as well, and are key-value pairs. Additionally, mod_cdr is written in C++ making use of object orientation, and proves how easy it is to integrate C++ modules in FreeSWITCH, giving developers even more leverage in the languages of choice. For more information, get a current copy of FreeSWITCH today!
RocketSource is a meta-project created to bring together a community with common interest in developing open source telephony applications using VoiceXML, CCXML, and related standards. Voxeo sponsors RocketSource to encourage visibility and collaboration across open source VoiceXML and CCXML application projects. One of our goals is to bring other sponsors and code contributors to the RocketSource project.
More information on this project can be found on their website at http://www.rocketsource.org/.
Cool! We have a Story On DIGG. Now go vote!
We are pleased to announce a Getting Started guide to help you get in the game fast!
Looks like we were slashdotted.
Can you Digg it?