A few questions answered by the Author of FreeSWITCH, Anthonny Minessale II (anthm) Editor's Note: This interview was not made by SineApps

Question 1: What is FreeSWITCH anyway? Is it just another fork of Asterisk or something?

A: That is a good question which I can elaborate on heavily since I have done extensive development on both projects. FreeSWITCH is actually an entirely different application from Asterisk. It shares no code whatsoever with Asterisk but it does share many capabilities. The biggest difference between the two besides the fact that Asterisk is a PBX and FreeSWITCH is a soft-switch is the architecture. Asterisk was originally designed with a purely monolithic architecture meaning that processing, user interface and data all reside within 1 entity (the main asterisk process). Despite the fact that asterisk supports loadable modules the majority of them are not gateways to other technologies but rather inline implementations of whatever technology the module was written for. Also Asterisk is very cautious about the number of concurrent threads they are willing to use and will take any measure to reduce that total. Some work has been done to support Asterisk on a few other UNIX variants but there are no goals set to support anything beyond LINUX and GCC. Since Asterisk feels it only has to concern itself with carrying calls to the PSTN it only supports an 8 kilohertz audio sampling rate, the rate of most telephone circuits. I respect these decisions and because I disagree with some of them at a fundamental level I chose to make a new Application.

FreeSWITCH takes an entirely different approach with its architecture. The initial release of the code already supported multiple sampling rates and even had the ability to bridge calls with unlike sample rates in real time. Portable cross platform code is foremost on the list of requirements which has paid off with support for Windows, MacOSX and many versions of Linux on varying numbers of processors both 32 and 64 bit. External data access and cluster ability has been planned for since day one and the module interfaces are designed to bring existing technology into the core by providing a gateway from existing libraries. So clearly FreeSWITCH aims to create a platform on which to combine technologies where Asterisk, like its namesake the ‘*’ character, aims to implement just about everything.

Question 2:What’s the difference between PBX and soft-switch isn’t this just semantics?

A PBX is an entity that allows a private company to have its own mini phone company providing services like voicemail, extensions and conferencing to phones. The primary focus of a PBX is for multiple phones to find each other and communicate.

A soft-switch is a software application that can connect phone lines from one network to another, often routing calls from one protocol to another or to a termination point such as a PBX. FreeSWITCH has the potential to also implement a PBX but it is not mandated to. Think of it as a lower level application than a PBX. It’s possible to load several modules into FreeSWITCH to make it behave exactly like a cluster of PBX applications. This is entirely easier than trying to make a single monolithic PBX behave as a switch especially when much of the PBX functionality is permanently built into the core of the PBX application.

Question 3: What about all that Asterisk development you did?

It was not in vein, I still use the software, heck I even do consulting on it for people all the time. I spent several years contributing to it and I have developed many third-party modules for Asterisk located at http://www.pbxfreeware.org.

FreeSWITCH simply represents my concept of the future of Telephony.

Question 4: So can you do anything useful with it?

yes, there are endpoints that implement SIP, IAX2, Jingle (GoogleTalk) Woomera, sound cards and Wanpipe TDM cards. You can use it as both a client and a server for VoIP communications, gateway a call from 1 protocol to another and execute IVR scripts written in one of several languages such as JavaScript, Perl or C#. There is an extensive event engine that will inform you of anything that happens on the system as well as an XML RPC interface that allows you to send and receive data from the core via HTTP.

There is abstraction in place to allow data to be pulled from XML, external HTTP requests, flat file (INI format) and via directory lookup, LDAP for example. The dial plan combines the power of XML and Perl Compatible Regular Expressions to allow one to route calls based on most any criteria from the call details.

Question 5: What? Did you say it can talk to GoogleTalk?

Yes in March of 2006 I developed my own XMPP telephony signaling library that is capable of communicating with Google’s GoogleTalk. With a single Jabber account you can receive endless simultaneous calls from GoogleTalk clients and gateway those calls to IVR or another voice protocol like SIP or H.323. When FreeSWITCH is on both ends of the call you can bypass NAT and send extended data such as Caller ID and DNIS.

Question 6: How long have you been working on this project?

The initial release of the code that actually did anything interesting was in early January 2006. I started actual development in private sometime in the beginning of October 2005.

Question 7: When will you have the first tagged release?

ASAP, we are now in the stage of trying to get a first release out in time for ClueCon where we will be presenting FreeSWITCH. We want to finalize any API interfaces and make sure there is ample valid documentation as well before any formal release is made.

Question 8: What is ClueCon?

ClueCon http://www.cluecon.com is an annual telephony developer conference in Chicago where leaders of various popular VoIP projects and other telephony developers adjourn to speak and share ideas as well as give informative presentations. It’s a three-day opportunity to be immersed in this emerging technology by day and spend the evenings in downtown Chicago!

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