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.
We are looking forward to a busy but good week of activity. Last week was busy as well. On last week's conference call we had an engineer from JeraSoft join us to answer some questions about the newly released VCS 3.4.2. We are happy to have this new version as part of the FreeSWITCH ecosystem.
One other bit of good news: All of the content for the new FreeSWITCH book has been submitted to the publishers. We are doing a bit of editing on one of the chapters but otherwise we are basically done. We anticipate a
Take care and have another fantastic week!
The FreeSWITCH team is pleased to announce that we have another product in our ecosystem: JeraSoft VCS version 3.4.2. The JeraSoft official announcement and official release notes can be found here.
For those who may not know, JeraSoft VCS is a routing and billing solution that works with a number of VoIP platforms. Since FreeSWITCH continues to grow as a carrier/provider solution it is good for those using it as such to have such an option. Some highlights of the new release are:
- RADIUS authentication of customers
- RADIUS authorization of calls (including balance limits)
- RADIUS start/stop accounting
- Dynamic routing via SIP redirect server
Feel free to visit JeraSoft to learn more about how to use this software with your FreeSWITCH implementation.
Happy tax day to those in the USA - we hope all is well with your business. Speaking of business, I thought I would relay the interesting news about Dish Network making a bid for Sprint. Many here in North America will be keeping a close eye on this one. Whether or not this is just a big mess or an opportunity for Sprint to become a "real" competitor to AT&T and Verizon remains to be seen. Regardless of the outcome, most of us here are hoping for a healthier Sprint so that we can avoid another duopoly.
On last week's conference call we decided to have a preliminary discussion so that we can prepare for this week. Dave Kompel will be showing us how to build rapidly FreeSWITCH applications in MS Visual Studio 2012 and have those run under mod_managed. Be sure to consult this document so that you can get all the rerequisites installed in time for our call on Wednesday.
In other news I would like to let everyone know that I spoke with Kashif Kahn over at Vestec. We are gearing up for the automatic speech recognition application building contest. The winners will be announced at ClueCon 2013. The official contest page will be posted on the ClueCon website shortly. Stay tuned for more information and be ready to start building your ASR applications!
The ClueCon 2013 call for speakers recently went out and we've had a number of submissions already. We look forwarding to hearing more talk ideas, so please send those in right away. In the meantime ClueCon registration is now open so be sure to get signed up, and don't forget to book your room at the Hyatt Chicago Magnificent Mile hotel for only $169 per night.
Have a great week!
ClueCon - the open source IP communications conference by developers, for developers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea for a technical presentation for ClueCon 2013 then we would like to hear about it.
What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon loves technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points:
- ClueCon talks are 30 minutes in length, including Q&A time with the audience
- ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio
- Attendees enjoy hearing about projects built with open source tools, especially those in a production environment
- Highly technical discussions that show the nuts and bolts are especially well-liked
- The audience appreciates seeing and participating in live demonstrations
- We are especially interested in WebRTC-related talks and demonstrations
Please send your proposals to email@example.com.
Be sure to include the following items:
- Working title
- Brief description of the talk (abstract)
- Name of the presenter
Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time.
ClueCon 2013 registration is now open!. Visit the registration page for details. Be sure to book your room at the Hyatt Chicago Magnificent Mile and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference call each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events!
Since its spring, we decided to mess up up our tree by adding a bunch of new code overnight. We added all kinds of new standards and media processing code because, hey why not!
We also addeed new lossless audio codec. mod_b64. It actually preserves 100% of the original audio signal by seamlessly transmiting the stream as rot-13-encrypted-base-64-encoded-audio in regular and high def! Add that to the list of many FS firsts! At this rate we probably even support webICEE or Big gulp or whatever they call that fancy new stuff.
What'll they think of next? !!!
Welcome to FeeSWITCH, the worlds first fee-driven open source project!
In order to use FeeSWITCH you simply download the code and build it and use paypal to send donations to paypal via the button on the right of the page. The fee can be of any size though we prefer hefty fees over light fees. We prefer that you pay the fee as often as possible. The more often you pay the fee, the better your calls will sound especially if you purchase the new FeeSWITCH ultra gold plated VoIP-Ready voice-enabled ethernet cables from the same guys who brought you Monster TV cables and Beats headphones.
Via Slashdot comes this encouraging story about MPEG-LA giving a royalty-free license for all patents that may apply to the VP8 video codec. The official announcement can be found here.
This is an important step for the VP8 codec. MPEG-LA handles the licensing of patents for many patent holders. By acquiring the rights to these patents - and on a royalty-free basis, Google can be much more confident that VP8 can be put into production without concerns about patent litigation. Of course, there may be individual patent holders (or patent trolls) out there who may feel that their patents are infringed by VP8. Time will tell if those with other patents will come forward, but this is good news for Google none-the-less.
We have a few interesting news items that have come along, both found on Slashdot:
* Jitsi 2.0 Released - Includes an overhauled interface, support for new codecs like VP8 and OPUS, and ZRTP encryption. Check it out and let us know how it goes.
* Do Kiosks and IVRs Threaten Human Interaction - This is an interesting article about how many of us prefer not to interact with a human under many circumstances.
Enjoy the articles and be sure to send along any VoIP/Telecom news items you'd like to share.
Ken Rice has just announced on the weekly FreeSWITCH conference call that FreeSWITCH version 1.2.6 has officially been released!
This version has numerous bug fixes and lots of little memory leaks have been plugged. The v1.2 git branch has been tagged and is ready for you to update. Please use this version in production as soon as possible.
As always, give us your feedback and thanks for using FreeSWITCH!