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 just saw this news item from The Register and I thought you all might be interested. It's a story about how Republic Wireless is going to be offering unlimited voice and data for $19 per month. Naturally, there is a catch: the phone will "seamlessly" piggyback on any open WiFi network to which it has access, including open WiFi hotspots. Supposedly this all happens in the background without any user intervention. I'm very curious to see if this actually works or not. Furthermore, how long before AT&T WiFi starts blocking these calls? We'll have to wait and see.
We've all heard of CenturyLink by now, and I'm sure some of you may not exactly be big fans. This story from Fierce Telecom will do nothing to change that perception. DSL users can start expecting data caps, either 150GB or 250GB per month, depending on the speed of the connection. If you have CenturyLink DSL and you use it for VoIP or other data-hungry apps then be on the lookout for their new "EUP" - Excessive Use Policy. On a brighter note, at least they aren't being like AT&T and charging extra right off the bat. Instead, those who go over will receive a phone call or email notification, most likely trying to upsell them to a new plan.
This item just came across my desk. It's a TMCNet article written by a guest contributor, Fred Goldstein. In it he discusses the chaotic mess that is the state of VoIP regulation here in the USA. This article is a good read for anyone doing any kind of VoIP business in the United States. I highly recommend it.
I must confess that I had never heard of Genesys Labs until I read this blog post today. In any case, the author of the blog post has some nice things to say about FreeSWITCH, although I was more interested in his thoughts on setting up secure VoIP. His comments on setting up TLS and dealing with the challenges of certificates and CA's was pretty insightful. If you are thinking about doing TLS and/or SRTP then definitely check out this blog post - it is a good read.
Over at Astricon this story came out: someone got NAILED for $400K in fraudulent toll charges. Ouch, indeed! Remember the mantra: lock down your VoIP systems!
Note: OnSIP was a silver sponsor at ClueCon 2011. They are great folks and we're glad to have them in the FreeSWITCH community!
This story comes from David Rowe, VoIP hacker extraordinaire. This is a reminder that even the best of us can make mistakes and otherwise forget to plug holes that we already know about. I recommend you read up on his experience. It shows that a combination of factors - in this case a characteristically evil SIP ALG and a config file miscue - can conspire to make your system quite hackable.
Let's all be careful out there!
Our buddy Dan York has written a piece about Microsoft's assimilation acquisition of Skype. For better or for worse it is a done deal. I don't know about you but I must confess that I'm keeping my eye on this one. It could be great or it could be a train wreck. In either case I've got the popcorn ready...
We wanted to keep everyone informed of the schedule for upcoming FreeSWITCH training. Darren Schreiber, co-author of the FreeSWITCH book and CEO of the 2600hz project, is the lead instructor. There are beginner and advanced classes as well as a 5-day complete training. Visit VoIP.com for all the details. Also, don't forget that Darren and Company can also offer customized training suited to your specific business and technical needs. Contact them at firstname.lastname@example.org to obtain more information.
Thanks to Darren and all the great folks at 2600hz for helping to expand the reach of the FreeSWITCH community all over the world!