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.
After nearly 2 years of non-stop development, FreeSWITCH has evolved from an idea and an empty directory into a fully operational soft-switch capable of withstanding the loads 8-10 times as large as anything else they were able to deploy in the past with other software according to the reports from our early adopters.
I am happy to announce the immediate availability of FreeSWITCH 1.0 “Phoenix“ (Release Candidate 1). The Phoenix symbolizes the rebirth of a new era and I hope this new release will be the first step in a new age of voice-over-IP development we can all benefit from.
We anticipate the official FreeSWITCH 1.0 “Phoenix” release date to be Monday, May 26th 2008. We will have a formal release presentation at this year’s ClueCon 2008 August 5th-7th 2008. http://www.cluecon.com
I would like to thank our community for the testing, bug reporting, code contributions, feedback and financial donations that helped us get this far, especially those who were involved from day 1.
Please take this opportunity to test drive FreeSWITCH as we will be making it available as binary packages for the first time to practice for the official 1.0 release. The code and arrangement of the configuration will not change apart from bug fixes and tweaks from this point forward. As always, the code will still be available in svn at the usual place.
For download instructions please stay tuned to:
The packages will be online soon.
Anthony Minessale II
Check out NXTVOX http://www.nxtvox.com/ for an inexpensive
way to try out our new OpenZAP channel driver for FreeSWITCH.
The cards are compatable with both asterisk and FreeSWITCH!
FreeSWITCH announces fully tested support for TLS sip signalling and SDES secure RTP making it possible to make secure VoIP calls with the standard release of the software! No patches or workarounds are necessary!
Now you can communicate with someone over the internet using FreeSWITCH and have a completely secure channel of communication. Used in conjunction with the G722 or speex16 HD codecs you can have crystal clear audio as well!
Since the advent of the High Definition Television, The HD phenomena has spread far and wide in recent years making the “ordinary” “extraordinary” with the addition of a simple 2 letter prefix. The telecommunications industry is no exception with the strengthening concept of HD-Telephony. Is it all hype or is there an actual benefit to better sounding phone calls? How does this affect the hardware and software designed to keep us connected? As a software developer in this field, I hope to shed a little light on the topic and separate the facts from the misconceptions.
A few new features have been developed in the last few months.
* SRTP support: Make calls with secure RTP
* mod_dialplan_asterisk: A dialplan module similar to the asterisk dialplan syntax.
* mod_cdr_csv: A templated CSV CDR module.
* mod_tone_stream: Play tones anywhere a file can be played.
* USER channel, a way to call a user in the directory as a channel.
* mod_voicemail: a full featured enterprise scale voicemail application tied to the core directory that can be used with a local sqlite database or shared across multiple boxes with ODBC. It supports mutiple greetings, fully customizable keys for every function, fully customizable email template, ability to forward messages to your email once you have listened to them, urgent messages with accurate urgent/non urgant message counts in mwi and templatable ivr prompts via the FreeSWITCH phrase macro interface.
*mod_limit: a small application to allow restriction of certian simotaneous resources in the dialplan which also works locally with sqlite and at an enterprise scale with ODBC.
Here is one of the more interesting examples of ways people are finding to play with FreeSWITCH
This application actually is integrating FreeSWITCH with a 3D graphics engine.
As we move another step closer to our 2nd beta, we added a few more features this week.
1) The channel variable RECORD_STEREO=true will make the record_session
application record a stereo file with the read and write streams in
seperate channels. For the more seasoned developer, this also means the
media bugs have a capability to return audio in mixed or stereo form
for other applications as well.
2) The addition of mod_fifo. This module makes it possible to park
many calls in a fifo queue and unpark them in the order in which they
were received. This will make it possible to make parking and call
distribution applications. The module sends events detailing stages of
callflow as well as an api interface command to generate a full report
of the call state. And is purely dynamic with no configuration necessary
to support an infinite amount of fifo queues.
App_confcall is a conferencing application for Asterisk®. This application provides advanced conferencing functionality, enabling more powerful conferencing applications to be performed.
There are a few different conferencing modules for Asterisk®. This article will discuss app_confcall, a previously unreleased conferencing application now available to the Asterisk® community, those that use Asterisk®. A feature matrix of the 3 most popular conferencing solutions for Asterisk® will also be provided so that you as the user can decide which is best for your needs.
I'll be attending Astricon in Phoenix on Sept. 24th thru the 28th and want to get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Please find me on IRC if you wish to touch base at Astricon.
In addition I would like to point out that Cepstral sponsored work on OpenMRCP so the world could benefit from MRCP v1 and v2 integration for accessing ASR and TTS resources. FreeSWITCH and its team of great people were instrumental in making sure OpenMRCP became a reality. Here is the news release from Cepstral about OpenMRCP.