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.
Today has been a bit busy with tech news that may be of interest to our community so I've summarized a few items and have provided links. They both come via Slashdot:
FCC Ups Penalties For Caller ID Spoofing - This is a good one to keep in mind if you are a service provider. Make sure that your users aren't up to no good - we'd hate to have our community members drawn into litigation because of illegal activities on the part of individual users. Note that modifying caller ID isn't the issue here, but rather "to commit fraud or with other harmful intent."
USPTO Rejects Many of Oracle's Android Claims - More good news from this proverbial clash of the tech titans. If you didn't already know, Oracle is suing Google for infringing on various Java-related patents that were acquired when they bought Sun Microsystems. It seems that most people realize just how silly these patents are - even the USPTO, who in their infinite wisdom issued the patents in the first place, has rejected many of the claims. If you want some deeper insights into this issue I highly recommend reading Groklaw's coverage of the litigation.
Just to let everyone know, Philip Zimmermann, aka the father of PGP, is coming back to ClueCon 2011! (More information available here.) Philip will be headlining the VoIP security roundtable session on Wednesday afternoon.
For more information about ClueCon please visit our website or call 877.742.CLUE.
The FreeSWITCH team is pleased to announce that we have officially released mod_rtmp - the first truly free RTMP solution for open source VoIP and telephony! A special thanks goes out to Barracuda Networks for sponsoring the professional development of this module and allowing it to be freely released under the MPL along with the rest of FreeSWITCH.
RTMP - the Real Time Messaging Protocol - was originally developed by Macromedia to allow the streaming of audio and video through the ubiquitous Flash player. After Adobe acquired Macromedia they released the RTMP specifications which has allowed third party developers to create server-side applications that work with Flash and other RTMP-enabled clients. mod_rtmp turns FreeSWITCH into an RTMP server, allowing you to bridge RTMP client streams to SIP and TDM connections as well as conferences. Currently, mod_rtmp supports the Speex audio codec.
One of the applications of this technology is to allow Web site visitors to make phone calls right from their browser. A company's Web page can detect Flash and offer the visitor an option to "Click Here" to speak to a customer service representative.
To illustrate the usefulness of mod_rtmp, Anthony Minessale has added a Flash-based connector to the public FreeSWITCH conference. Browse there and try it out. Be sure that Flash is installed for your browser and that you have a headset or other audio device. For security reasons you will need to explicitly allow Flash to access to your audio device. Click "Call FreeSWITCH Conference" and then enter a name in the popup box. Click "Call" and you will be connected to the FreeSWITCH public conference. (Be sure to open the keypad and press zero to unmute!) If others are present you will be able to speak with them. Audio devices that support wide-band audio will produce higher quality audio.
Would you like to learn more about FreeSWITCH and mod_rtmp? Be sure to join the FreeSWITCH developers at ClueCon 2011 this August 9-11 in Chicago. ClueCon is a great opportunity to meet the authors of some of your favorite open source VoIP and telephony software projects.
Plans for this year's ClueCon event are proceeding nicely. We have a full slate of speakers as well as giveaways and other special events planned. This year we are at the beautiful Sofitel hotel in downtown Chicago. We are looking forward to seeing everyone again. Be sure to register right away so that you can increase your chances of winning the great laptop and tablet that we are giving away this year!
We would like to thank our longstanding sponsors: FreeSWITCH Solutions, Sangoma, and iCall. We would also like to welcome a first-time sponsor this year: Junction Networks, suppliers of the OnSIP solution. We look forward to Junction Networks CEO John Riordan's talk on Tuesday morning.
If you have any questions at all regarding ClueCon please call us at 877.742.CLUE or email us at firstname.lastname@example.org. We are here to help and we definitely look forward to seeing you this August!
The FreeSWITCH team is pleased to share the news that Plivo has been officially released today. Plivo is a communications framework that employs FreeSWITCH for rapidly building telephony-enabled applications. It is 100% open source, released under the MPL 1.1 license. The Plivo Team is lead by Michael Ricordeau and Venky.
Plivo is similar in nature to Twilio. It is easy to port existing Twilio apps to Plivo. However, unlike Twilio, Plivo is 100% open source, and developers are not locked in to any particular cloud, hosting solution, or telecom provider. Plivo enables the developer to use any communications media that FreeSWITCH supports, not just 8kHz calls over the PSTN. Use SIP, H323, Skype, or FreeTDM. Use G.722, SILK, CELT, or G.729. Use the carriers you prefer. Developers are free to use whichever tools they have available.
The Plivo framework is built with open source software: Python, gevent, and Flask. However, developers are free to use any of the following languages for actually building their telephony applications:
Visit the Plivo web site to learn how to get started building telephony applications. We look forward to seeing what our intrepid community members build with this new open source telephony framework.
(See update below.) In a move that is a surprise to pretty much no one with a pulse, the pre-Microsoft-owned Skype has decided not to renew their agreement with Digium, meaning that Skype for Asterisk will no longer be available after July 26, 2011. Existing customers are able to use Skype for Asterisk until at least 2013, after which no one is quite sure what will happen. The cynic in me says that two year is plenty of time for Microsoft to make Skype totally unusable.
In the meantime, we will all be keeping an eye on what Microsoft does with non-Windows Skype clients. Obviously the Windows clients will receive the most attention. From a purely business standpoint, Microsoft has little incentive to make sure that the OS X and/or Linux Skype clients are fully functional. One way to change the dynamic would be if Microsoft found a way to monetize each client, regardless of the underlying operating system. Could we possibly see Bingified Skype clients? Time will tell. Meanwhile it seems that the existing clients will still work with Giovanni Maruzzelli's SkypOpen module. Again, time will tell if Microsoft will try to "fix" that. Rumor has it that Microsoft will be Skype-enabling Lync, Xbox, etc. so there is value in keeping the Skype ecosystem healthy. Microsoft's press release says that it will continue to invest in "non-Microsoft" platforms. Evidently anything by those open-source hippies doesn't count.
UPDATE: This article at PCWorld is actually pretty balanced and has a non-conspiracy theory explanation for the move by Skype. I can't say that I'm holding my breath that MicroSkype will follow through on their grand interoperability plans, but at least there's some sliver of hope.
Good news! I was looking for some information on a ClueCon 2010 talk and it turns out that I had not uploaded any of the day 3 videos! They are on viddler.com, starting with video 94. These talks are all now available:
Notes on the Perception of Imaginary Differences
The blue.box Project
IAX - The Myth, The Reality - And The Status
Telefaks*de Application Server and FreeSWITCH Operator Panel
Creating Phone 2.0 Applications with Adhearsion
What I Learned Supporting Asterisk and AGI Applications For a Year
As you know, ClueCon is fast approaching: August 9 - 11, 2011. That's only three months away! The ClueCon team wants to see as many as possible get registered ASAP, so as an added bonus they are doing an early bird special: extra entries into the drawings for the coolest laptop and iPads ever!
Here's how it works: Get registered before the end of May 2011 and you'll receive a total of four entries into the giveaway drawings. After May 31 the number of entries will decrease:
Register by June 7 and receive three entries
Register by June 14 and receive two entries
All registrations after June 14 will receive a single entry.
Don't wait to sign up for ClueCon - a number of folks have already signed up and have their four entries guaranteed. Get signed up right away so that you maximize your chances of winning the coolest laptop and iPads!
For those of you who were not able to attend last week's conference call (April 27, 2011) I would like to let you know that both audio and video are now available:
* Audio - FreeSWITCH Weekly Conference Main Page
* Video - Stream it at Safi Systems Website or download it here
Thanks again to Zac Wolfe and the gang over at Safi Systems for their hard work. The product is actually very nice and is a great addition to the burgeoning FreeSWITCH ecosystem. If you haven't already seen it then please visit Safi Systems to learn more.