FreeSWITCH, a brand new telephony application that came on the scene at the beginning of 2006 and made news last month for its ability to communicate with GoogleTalk, is coming close to its first “official” public release with a release candidate in the near future hoping to be adopted in time for the ClueCon Telephony conference this August cluecon.com
FreeSWITCH is a application known to individuals in
telephony circles as a soft-switch, an application designed to accept input
from various Voice over IP protocols or from analog or digital equipment and
connect them to each other allowing soft phones, IP phones and analog
telephones (the ones most of us have) to communicate with one another.
This particular soft-switch also makes it possible to answer the call with a
script written a simple language and do many interesting things such as
transmit TTS (Text to Speech see http://www.cepstral.com/).
The creator and lead developer of FreeSWITCH, Anthony
Minessale II had previously spent his time doing development for the Asterisk
PBX. Asterisk is another telephony application geared towards home
enthusiasts on the Linux platform. After adding many invaluable features
to the application (http://www.cluecon.com/anthm.html)
Anthony finally concluded that the application had limitations that could not
be repaired without a complete rewrite. Since the other developers of
Asterisk did not share this vision and firmly rejected the notion, Anthony set
out on his own armed with nothing but his prior experience and a text editor.
Now, after countless hours of frenzied development,
FreeSWITCH has emerged from the dust able to handle several VoIP protocols as
well as power PRI ISDN circuits with a little help from Sangoma’s innovative
line of telephony hardware. (http://www.sangoma.com)
best of all it runs on several platforms including Win32, MacOSX, Linux
and most other UNIX variants.
FreeSWITCH has a modular design that allows you to
load what you need and save the rest for a rainy day. Some of the more
interesting features include:
VAD
– Voice Activity Detection
Secure
RTP
– Encrypted Audio
Crash
Protection – Ability to
withstand otherwise fatal errors
XML-RPC
– Control the application with HTTP and XML instructions.
TTS/ASR
– Add modules to allow the application to comprehend and generate speech.
Event
Engine
– Internal event bus can allow external applications to know about everything that takes place in the application.
Pluggable
Codecs – API anticipates hardware-driven
codecs and timing devices.
The FreeSWITCH core is suitable to develop both server and
client applications. The source code is freely available at http://www.freeswitch.org“
