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.
As you may know the FreeSWITCH team is continuing to update the project's infrastructure. Among other things this includes getting ready for IPv6. Last week Brian West finished getting several of our servers all set up to handle IPv6 traffic. These includes www.freeswitch.org and conference.freeswitch.org. Thank you to all those who did testing and gave us valuable feedback.
On last week's conference call we enjoyed our very own Ken Rice giving us some great reminders on how to gather data for troubleshooting as well as tips on opening bug reports in Jira. We had a number of users comment on how useful it was to see examples of how to do this. The audio is up in the usual location and we have a community member who is preparing a video which will be posted as soon as it is ready.
This week we have Chad Engler from Patlive coming to discuss with us his node-esl library. Chad has made the code available here on Github. He has included an interesting channel monitor example to give you an idea of what can be done by combining node.js with ESL. We look forward to hearing more about it on this week's conference call.
Have a great week!
Welcome to October! I hope the weather is nice where you are. Here it's still above 100F. :)
Last week was a little bit quieter than the previous few weeks. I had a chance to work on the FreeSWITCH change log and I made a list of some of the APIs, dialplan tools, and channel variables. These have all been added since 1.2.0 was initially released in early August. All of them have wiki entries - thanks to those who took the initiative to do add them. Feel free to add your knowledge and experience to the mix.
Last week's conference call was an object lesson in the challenges of getting a SIP proxy working with TLS and FreeSWITCH. This week we are going to change direction and look at something that has been slowly (and painfully) advancing the past few years: mobile VoIP. We will be having Daniel Pocock share with us some information about Lumicall, an open source mobile VoIP client for Android devices. There is also a service component to Lumicall and we'll be learning about that as well. Come join us to see the state of mobile VoIP.
We are working on some fresh presentations for later this month. We hope to have an update on e164.org and how we can all get involved. We are also preparing a presentation on how to perform some of the data-gathering techniques that are needed for basic and advanced troubleshooting. If you have some input on these or other presentations please let me or Ken Rice know.
Welcome to the last Monday of September 2012!
We've had quite the interesting week. Perhaps the most interesting item the team dealt with was a vulnerability in the Sofia SIP stack that would cause a segmentation fault while processing a specially crafted SIP message. Just to show you how nimble the FreeSWITCH developers are, from the time the vulnerability was reported it took less than a day to fix, test, and roll a new version of FreeSWITCH. We encourage everyone on 1.2 to get updated to version 1.2.3 as soon as possible. (The fix is also in the 1.3 development branch as of last Wednesday, September 19.) We tip our hats to Anthony and the rest of the dev team for their hard work on our behalf.
Last week's conference call was also very informative. We received an introduction to the repro SIP proxy software. We look forward to this coming Wednesday where Scott Godin and Daniel Pocock will continue the discussion and will get deeper into how to set up the proxy and use it with FreeSWITCH. If you haven't already tried to install repro please do so. Daniel has a nice tutorial over at OpenTelecoms.org - be sure to check it out and bring your questions on Wednesday.
Finally, we'd like to draw your attention to this blog post by long time FreeSWITCH and open source telephony supporter Kristian Kielhofner. Kristian reports that his company, Star2Star Communications, is sponsoring the FreeSWITCH stable branch by giving direct financial support to the project. This allows for a full-time team member to work on things like the stable branch and packaging as well as community interaction and documentation. We appreciate those who support FreeSWITCH and open source telephony!
Have a good week and we'll see you again in October.
Packt Publishing, the company who produces the two FreeSWITCH books, has announced a celebration of their 1000th title! This quite a milestone and we are happy that FreeSWITCH has two (soon to be three!) titles in the Packt catalog. As part of the celebration Packt is inviting everyone to sign up for a free account by September 30, 2012. Included in the celebration is a "surprise gift" - but you must be signed up in order to receive it.
Packt Publishing supports the FreeSWITCH project so we try our best to support them. Please sign up and if possible purchase the FreeSWITCH books.
Thanks to everyone who supports FreeSWITCH and open source!
It's been another productive week on the FreeSWITCH team. We are pleased to let you know that we have officially tagged FreeSWITCH version 1.2.2 in the git repo. Source tarballs are available in the usual location. Thanks to all those whose efforts make more frequent releases a reality. It is much appreciated.
On last Wednesday's conference call we enjoyed a nice Adhearsion presentation by Ben Langfeld and Ben Klang. Adhearsion is a Ruby-based framework for developing telephony applications. Ben and Ben discuss how Adhearsion works, why Ruby is cool for building telephony apps, and why the Adhearsion guys love FreeSWITCH. FreeSWITCH community members are invited to join the Adhearsion team at AdhearsionConf in Palo Alto, CA on October 20-21, 2012. Community members receive a special rate by using discount code AHNLOVESFREESWITCH. Thanks to Ben and Ben for a great presentation with cool slides.
For the next few weeks we look forward to hearing from Daniel Pocock and Scott Godin who will be telling us more about the Repro SIP proxy and the ReSIProcate SIP stack. For many of us it will be our first look at a SIP proxy that does not have its roots in the OpenSER project. We look forward to learning more on this Wednesday's conference call.
Have a great week!
From the gang over at Slashdot comes this interesting story about security researchers find new and creative ways to crash phone systems and the back-end systems to which they are connected. I thought you might find this interesting. Also, if you use IVR or voice to interact with callers then this is a good reminder to make sure that your systems are locked down and that they don't do silly things like buffer overflows and SQL injections.
Just a heads up - Ken Rice has officially tagged FreeSWITCH version 1.2.2 in the git repo. The source tarballs are available here. If you are already on git and have checked out the v1.2-stable branch then you can simply "git pull" and rebuild.
Thanks to Anthony, Mike, Brian, and Ken for all their hard work in keeping FreeSWITCH fresh and released frequently. Please help us out by testing those release candidates!
A nice article over at TMCNet discusses the value of HD Voice. It mentions our friends over at Sangoma as well as a few other experts. The value of HD Voice is readily apparent to many FreeSWITCH users. The FreeSWITCH dev team is on a conference call all day. The weekly FreeSWITCH conference calls also have many people who call in with wideband audio. The question is: why has adoption been all but non-existent? Why hasn't the demand for HD Voice caused carriers to work on supplying it? Let us know what you think.
We are pleased to report that our friends at Xiph.org have successfully gotten the Opus codec through a major milestone - IETF standardization! It's official: RFC 6716. Congrats to Jean-Marc Valin (IRC: jmspeex) and crew for getting this done. Feel free to check out the official announcement as well as the one from Mozilla and the Slashdot story.
We hope you all had a great week. On our Wednesday conference call we discussed several items as a community. One item of note was how to handle mailing list posts that are overly broad and reflect a lack of research on the part of the individual doing the posting. After much discussion we decided that we would create some online documentation that helps new ones get their bearings when considering the big picture in FreeSWITCH. (For example, what are modules and why do we have them?) Thanks to Dave Kompel for helping to get that started.
We are also pleased to announce that we have started up the Adopt-a-module project. The idea is simple but powerful: community members who are knowledgeable and enthusiastic about a specific module will volunteer to "adopt" that module. Adopting a module means doing several things: watching the mailing list and IRC channel for questions, monitoring the Git repository for new commits, keeping the module's wiki page up-to-date, and acting as a bug marshal for any Jira tickets that are opened. We've had several people step up already. Please visit the list of modules needing adoption to see if you there is one that fits your area of expertise. We give a special note of thanks to Anshel Blum for helping to get this one going.
This week we welcome Ben Langfield and Ben Klang who will be discussing the Adhearsion framework for FreeSWITCH. Adhearsion is a Ruby-based framework for building telephony applications. You may recall that Ben Klang joined us at ClueCon 2012 to make the announcement about Adhearsion being available for FreeSWITCH. We look forward to learning more about how Adhearsion works with FreeSWITCH.
Let's all have a great week!