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.
FreeSWITCH Weekly News and Notes
As you may know the FreeSWITCH team is continuing to update the project's infrastructure. Among other things this includes getting ready for IPv6. Last week Brian West finished getting several of our servers all set up to handle IPv6 traffic. These includes www.freeswitch.org and conference.freeswitch.org. Thank you to all those who did testing and gave us valuable feedback.
On last week's conference call we enjoyed our very own Ken Rice giving us some great reminders on how to gather data for troubleshooting as well as tips on opening bug reports in Jira. We had a number of users comment on how useful it was to see examples of how to do this. The audio is up in the usual location and we have a community member who is preparing a video which will be posted as soon as it is ready.
This week we have Chad Engler from Patlive coming to discuss with us his node-esl library. Chad has made the code available here on Github. He has included an interesting channel monitor example to give you an idea of what can be done by combining node.js with ESL. We look forward to hearing more about it on this week's conference call.
Have a great week!