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.
The FreeSWITCH team would like to announce that mod_ladspa has been added to FreeSWITCH. You may not be familiar with LADSPA, but no doubt you have heard some songs on the radio where the singer's voice is "autotuned" so that it stays on key or has a slightly techno/robotic sound. The new mod_ladspa module allows you to do this with FreeSWITCH.
How Does It Work?
First, as the L in LADSPA implies, you'll need to be on a Linux system. Secondly, you'll need to update to the latest Git HEAD as of late Thursday March 17, 2011. Finally, you'll need to install LADSPA and get at least one LADSPA plugin. The FreeSWITCH wiki has a page with instructions for CentOS 32 and 64 bit installation. There is a sample dialplan file, 00_ladspa.xml, the gets placed into the conf/dialplan/default/ directory and creates a test extension 101. Change that destination from the FreeSWITCH conference to a local extension, or better, to your friends' phone numbers so you can surprise (or scare) them with your Cher-like singing ability.
There are a lot of different things you can do with a voice stream. After following the instructions on the wiki you will have quite a number of LADSPA plugins to try if. Please report back to us with any interesting uses to which you put this module. Better yet, if you find a truly practical application then please let us know so that we can share with others.
Once again, quickly and easily adding a module like mod_ladspa demonstrates just how extensible FreeSWITCH truly is. Let's all thank Anthony Minessale and the rest of the FreeSWITCH development team for all their hard work to make such a wonderful project possible.
This article from Network World just came across my desk. It's a reminder that those of us who use VoIP - and especially those who run public-facing VoIP servers - must keep security in mind.
One thing that stood out is that malicious entities are using cloud-based computing resources to engage in brute-force password attacks. This is a good reminder not to have really simple authorization passwords in your XML directory. If you are using static XML then you have a simple tool to assist: randomize_passwords.pl - a script that will randomize the SIP auth password and/or the VM login password. The script is found in scripts/perl/ under the FreeSWITCH source directory. It is very easy to use and has a number of options. Launch the script with -h to see the various command line arguments.
Also helpful in mitigating attacks is the use of a utility like fail2ban. The fail2ban wiki page has good information about how to set this up on your Linux/Unix based system. In conjunction with iptables, this tool helps you keep malicious attackers from overwhelming your system.
What else are you doing to protect your systems? Please add your comments below.
With the recent announcement about Gizmo5 being discontinued, some have wondered about the future. Dan York and others are reporting that, at least for now, it is possible to call a Google Voice phone number with a SIP URI.
I'm sure more information will become available at some point. In the meantime if you learn anything more about how the SIP-based calling works then please let us know.
Jim Gettys and other members of the Bufferbloat project will be giving a special presentation on the weekly FreeSWITCH conference call, Wednesday March 9. All are welcome to listen in and learn about this interesting topic that affects virtually all network communications.
If you are not already aware of what bufferbloat is, the fastest way to get up to speed is to peruse the slides and listen to 25 minutes of audio before the conference call, at:
It would be great to have an informed audience so we can only touch lightly on the preliminaries and then dive deeper into bufferbloat and VOIP issues. See also Jim Gettys' blog postings, discussions on lwn.net, on slashdot, and elsewhere, as well as two very busy mailing lists.
In short, the bufferbloat problem is that there are really big, bloated network buffers in many (especially new) routers, home (mostly wireless) gateways, hosts, and ADSL/FIOS/cable modems that can dramatically affect VOIP performance.
The bufferbloat project is attempting to identify equipment and software where truly bloated buffers exist, and mitigate or fix the issues with new software algorithms and heightened awareness. Recently they released a debloat-testing Linux kernel that may help in some cases. Bufferbloat is not a Linux-specific problem, it exists in all OSes, and may become more acute as Windows 7 gets rolled out.
Looking forward to talking to you all on Wednesday!
Aberdeen-based internet telephony service provider SureVoIP has been shortlisted for a prestigious industry award.
The company is one of six finalists in the ‘Best Business ITSP (Internet Telephony Service Provider) for SMEs’ category of the 2011 ITSPA (Internet Telephony Service Providers’ Association) Awards.
The winner of the accolade – presented to the company which the judging panel deems provides the best service for small and medium business users – will be announced at an awards ceremony in the iconic BT Tower in London, on Monday, March 7.
In order to make the shortlist, SureVoIP had to undergo a rigorous assessment process, which involved providing comprehensive details on its service offering, as well as a period of technical testing.
The testing element of the assessment stage entailed over 1000 calls being made via SureVoIP’s internet telephony lines over a 3 month period to measure a number of metrics including voice quality, delays and background noise.
VoIP (Voice over Internet Protocol) offers significant improvements to the flexibility, speed and cost of global communication through transmitting voice and multimedia communication through the Internet, as opposed to the fast-disappearing Public Switched Telephone Network (PSTN).
Gavin Henry, Managing Director of SureVoIP – which became the first company in Scotland to achieve ITSPA Quality Mark accreditation in December 2010 – commented on the company making the shortlist:
“We are thrilled to be in the running for such a highly respected industry award.
“Being named as a finalist is a major, exciting achievement for SureVoIP, particularly considering the company has only been operating for a matter of months, having been launched in November 2010, but with all services being quality tested over the previous year before launch.
“Making the shortlist – which involved SureVoIP being assessed by top industry professionals – highlights the exceptional quality of service we already provide customers with and bodes well for the future success of the company”.
“We couldn't have done this without FreeSWITCH and the FreeSWITCH community”
SureVoIP, which is a division of IT and telecoms support specialist, the Suretec Group, was established to harness and utilise the vast technical expertise and experience with VoIP suppliers, which the group had acquired through its initial services relating to VoIP consultancy.
Established in Scotland in 2003, the Suretec Group specialises in open source IT support and consultancy, open source telecoms and training in open source software. SureVoIP is the fourth division within the group, which also includes Suretec Systems, Suretec Telecoms and Suretec Training.
For further information about SureVoIP please call 01224 900 123 or visit www.surevoip.co.uk
Issued by Mackenzie PR on behalf of SureVoIP. For further information please contact Paul Beaton or Andrew Reid on (01224) 580 188 or email firstname.lastname@example.org
Imagine this scenario: you're a VoIP engineer working on a soft-switch. You walk in the door at 8AM and your boss says, "Here's the specs on a new codec you've never heard of. Please implement this in our soft-switch before the end of the day." (Queue the theme from "Mission Impossible".) Sounds crazy, doesn't it?
Well, at 8:21AM CST this morning Anthony Minessale (chief architect and lead developer of FreeSWITCH) learned of the existence of the Opus codec. At 3:45PM Anthony committed mod_opus to the FreeSWITCH git repo. Less than 10 minutes later I had done a "git pull", compiled, and loaded mod_opus on a test server. It wasn't even necessary to reboot FreeSWITCH.
If you want to give mod_opus a test drive then update to latest git and add "codecs/mod_opus" to your modules.conf file and do a "make mod_opus-install" in your FreeSWITCH source directory. You'll need to go into fs_cli and issue a "load mod_opus" command as well. Look for a new codec named "OPUS" in your codecs list. Opus is a hybrid of SILK and CELT. Lower frequencies use a CELP-style encoding and higher frequencies (like music) use an MDCT form of encoding. Wikipedia has entries on all these terms if you'd like to learn more.
Enjoy the new codec!
I have been splitting my time developing not only FreeSWITCH but the CudaTel Phone system by Barracuda Networks that uses FreeSWITCH as the telephony engine. I'm proud to announce that not only has the product reached a 2.0 status, it's now 100% deployed within Barracuda Networks running the entire company's phone services across multiple locations. The CudaTel is using FreeSWITCH right from GIT with no special modifications at all. The best part is it works great as a PBX you can then feed out to your existing FreeSWITCH setup to route the traffic to custom applications or whatever you can think of.
The next step is to develop more exciting and impressive PBX features that can be added for free with an energize update package available at the time of purchase. CudaTel comes with the same great support you receive today on FreeSWITCH.
This is also a great opportunity for those of you out there looking to get into the reseller market. Contact me at the consulting link at the top of this page or call 408-588-3633 to learn how you can become a reseller of CudaTel. Use your existing VoIP expertise to support and deploy CudaTel to others.
Check it out at http://www.cudatel.com
Our friends over at GMVoices conducted a number of interviews at ClueCon MMX last August. They've finished editing them and they turned out very well. Here are the links to each interview:
Dr. Moshe Yudkowsky
We hope you enjoy these videos. Also, start gearing up for ClueCon 2011! It'll be in Chicago, August 9-11. More information will be announced in the coming weeks.
This is a nice interview with Anthony, Mike, and Brian conducted by VoIPon. It has a Youtube "video" embedded as well as a transcript.