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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
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Spit Will Be Worse Than Spam?? NOT!!!
Submitted by
admin on Thu, 06/12/2008 - 15:44
::
I was reading this story over on Slashdot about how SPIT (spam over internet telephony) will be worse than SPAM. I find it hard to believe that will be the case. If you use software like FreeSWITCH to frontend all your phone calls it can give you the ability to knock out* the people you don't want to talk to in a heart beat.
What is your take on this?
* No you can't literally knock people out via telephone... yet!
I'm sure they will come up
I'm sure they will come up with something to stop it.
Think of the filter mechanisms
Spam has been relatively easy to assess, because it is text based. Even when the spam is little more than image files, there are still a lot of clues in the little text that is present.
Spit is audio, or maybe video. It is hard to analyse, and can be easily made different on each call, even if they are pushing recordings at you. It is hard to assess where it came from. Unless you want to blot out calls from anyone you haven't explicitly authorised, you have problems.
Unless tricky techniques come to light I thing the potential for streaming media based spam is huge........ Hey, huge potential. VCs, its time for Web 3.0!