FreeSWITCH

The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre8. This latest release is the most stable and secure version of FreeSWITCH to date. All are encouraged to update as soon as possible. The latest files are ... [More] available at files.freeswitch.org.   The newest release of FreeSWITCH adds new features and improvements in stability, security, and performance. For example, support for ZRTP (www.zfone.com) was recently added. ZRTP is an opportunistic encryption protocol that allows for automatic encryption of RTP streams when both endpoints are ZRTP-enabled. Combining ZRTP with TLS, which encrypts signaling information, allows for complete end-to-end security in FreeSWITCH.   Also included with FreeSWITCH 1.0.4pre8 are the latest updates to the OPAL (www.opalvoip.org) integration and Skype. OPAL, the Open Phone Abstraction Library, adds H.323 support to FreeSWITCH. Skype functionality is added with the combination of the Skype's proprietary client and FreeSWITCH's mod_skypiax module. The Skype interface has new speed improvements as well as the ability to run as a service in Windows.   FreeSWITCH (www.freeswitch.org) is an open-source telephony platform that allows for the creation of voice, video, and chat-driven applications. Released under the business-friendly MPL, FreeSWITCH is a cross-platform communications library that runs natively on Linux, Mac OS X, Windows, and a number of Unix variants. FreeSWITCH can efficiently serve as a soft-switch, PBX, IVR, voicemail/auto attendant, and conference server. FreeSWITCH supports a number of protocols both for VoIP (such as SIP and H.323) and traditional TDM telephony with analog or digital interface cards.   [Less]

Ruby fans everywhere will be happy to know that, in the words of Bougyman, "Ruby Loves FreeSWITCH!" How so? Consider the FreeSWITCHeR project, an open source Ruby library that acts as an abstraction layer for the FreeSWITCH event socket. ... [More] FreeSWITCHeR is a FreeSWITCH event socket abstraction library built upon the Ruby event machine. It allows for connecting to - and controlling of - multiple FreeSWITCH instances from a single controller. The following is a sample "get digits" listener program that uses the FreeSWITCH outbound event socket paradigm: require "rubygems"require "fsr"require "fsr/outbound"class GetDigits < FSR::Listener::Outbound def session_initiated exten = session.headers[:caller_caller_id] answer do read("/path/to/smooth.wav") do |read_var| FSR::Log.info("Received #{read_var} from #{exten}") end end endendFSR.start_oes! GetDigits, :port => 8084, :host => "127.0.0.1"Start the program and then send a call to the socket application in the dialplan to see it work. The program will capture DTMFs from the call, play a wav file, and read the digits back to the caller. It will also display the captured digits to standard output.   The fsr library also has an inbound event socket listener module and a command socket module which allows for more traditional sending of api and bgapi commands to a FreeSWITCH machine. The fsr docs page has a lot of interesting information. The fsr developers welcome all Ruby programmers to try out this new abstraction library.   [Less]

The development team is pleased to announce that preliminary support for ZRTP as been added to the Linux, Unix, and Mac OS X versions of FreeSWITCH. Windows support will be added shortly.   ZRTP is a key agreement protocol for establishing SRTP ... [More] streams. ZRTP is not limited to a specific signaling protocol because the key exchange is done within the RTP stream, therefore ZRTP works with SIP and H.323. ZRTP is opportunistic: if ZRTP is available at both ends of a call then an SRTP connection is automatically negotiated, otherwise a standard RTP stream is used.   ZRTP is part of the Zfone project, designed by Philip Zimmerman, the orignal creator of PGP. It was designed with the goal of being a better architecture for protecting VoIP calls. ZRTP offers very good security with low CPU overhead.   To use ZRTP you will need to update to the latest version of FreeSWITCH from the SVN trunk. You will also need to download the Zfone libZRTP Software Development Kit. Complete instructions are available on the FreeSWITCH wiki.   Right now virtually any softphone will can be used with ZRTP. The Zfone project has a plug-in that can be downloaded here. Many softphones have been tested, including x-lite, Gizmo, SJPhone, and others. [Less]

If you haven't heard already there is a nice article about FreeSWITCH. Please follow this link to digg.com and digg the article. Then please go to the vote section of the article and click the thumbs up button to upvote this article. Also, for those ... [More] of you who have other link sharing sites we have links for those: reddit.com - click through and upvote the "Open Source Sofswitch" entry mixx.com - click through and comments, kudos, etc. Please add to your del.icio.us accounts if you have them. Thanks for your support! -Michael   [Less]

FYI, Some of you are familiar with ZDNet's Dave Greenfield. He has a new blog post discussing HD Telephony. Evidently HD Telephony is becoming a big buzzword. We need everyone in the FreeSWITCH community to spread the word that not only is ... [More] FreeSWITCH capable of HD voice, it does it as well or better than anyone on the planet.  If you think that last statement is hyperbole then google around for HD telephony or HD VoIP and see what your options are. Look at the expensive companies - Cisco, Avaya, Nortel, etc. - and see what they offer for HD voice. Compare that to using FreeSWITCH with CELT, Speex WB, or Polycom's Siren codecs, not to mention Skype integration with mod_skypiax. For zero cost software you can get HD voice. Of course, you'll still need some hardware, but who wants to pay for hardware and software both, especially when the paid-for software can't even keep up with FreeSWITCH? If you have used or are using HD voice with FreeSWITCH then please go put a nice comment on Dave's blog. Also, if you know someone going to the HD VoIP Summit in New York on May 21 then definitely put a bug in their ear about FreeSWITCH + HD codecs being a fantastically low barrier-to-entry for HD voice in the enterprise. -Michael [Less]

FYI, The FreeSWITCH dev team is continuing to make good progress on the road to officially releasing FreeSWITCH version 1.0.4. Please download the latest version here and put it into production as soon as possible. Remember, these prereleases are ... [More] actually the most stable version of FreeSWITCH that you can get. However, we we've noticed that some have been using mod_loopback in creative ways and possibly have been relying upon loopback channels behaving in what might be called unnatural ways. We especially need those using mod_loopback to get updated and offer feedback. As always, if you are already on SVN trunk then please just do a "make current" to get your system updated. Thanks for helping to make FreeSWITCH so awesome! -Michael [Less]

Congratulations to FreeSWITCH for making this top-11 list: http://digg.com/linux_unix/Best_VoIP_Software_LinuxLinks_News Please read and digg. -Michael

We have good news to report to the FreeSWITCH community: OPAL integration is moving forward. OPAL integration, via mod_opal, is officially in beta status. Here is some background information. OPAL - the Open Phone Abstraction Library - is a ... [More] continuation of the open source openh323 project. OPAL expands upon its predecessor by including protocols other than H.323, including IAX2 and SIP. Furthermore, OPAL is not limited strictly to voice applications as it can handle fax and video. It comes with a number of audio and video codecs included and has a run-time loadable codec interface to allow for loading of custom or proprietary codecs. The mod_opal FreeSWITCH module is authored by one of OPAL's developers, Robert Jongbloed. I asked him about the history of OPAL and he offered the brief version as follows: This is a VERY long story. The highly abridged version begins in 1998 with a desire for a firewall proxy program Craig and I had developed to be able to proxy Netmeeting calls. So, a product called PhonePATCH was created, which then became the open source OpenH323 library, which then became OPAL. As to the use of open source, Robert mentions that this really is a matter of being pragmatic rather than philosophical. Rather than going down the closed source route the OPAL developers chose a model that many open source users can relate to and what Robert calls the "give the software away and try and make money on the support" model. In addition to getting help from the community this also allows for interesting scenarios like integrating OPAL into FreeSWITCH. While technically it can be construed that OPAL and FreeSWITCH are "VoIP competitors," Robert points out that each project has its own distinct emphases, and therefore being complementary is much more valuable than been competetive. FreeSWITCH benefits from OPAL because it brings, among other things, H.323 support and OPAL gains exposure by being included in FreeSWITCH. We encourage anyone who needs H.323 support in FreeSWITCH to try out mod_opal and to visit the OPAL website to learn more about what a great project it is. We are also happy to report that Robert Jongbloed will be speaking at ClueCon 2009! If you are a fan of OPAL or H.323 then be sure to get registered for this year's ClueCon event. If you would like to try mod_opal be sure to use the 'buildopal.sh' script in the build subdirectory of the FreeSWITCH source directory. This script will install OPAL and the required PTLIB. Also, be sure to update your FreeSWITCH installation to the latest SVN. -Michael S Collins [Less]

SipFoundry has announced that their new 4.0.0 version of sipXecs has been released. This new version of their flagship IP PBX product includes FreeSWITCH as an underlying component of the automated attendant and conferencing systems. I asked Andy ... [More] Spitzer about the decision to include FreeSWITCH in the open source sipXecs product. He reports: "We started looking at FreeSWITCH in order to provide a conferencing solution for the open source sipXecs.  The commercialized version sold by Nortel included a bolted on proprietary conferencing module.  We could not use that in the open source side, so we turned to FreeSWITCH." SipFoundry needed an OSS-friendly conferencing solution that is flexible and robust. The powerful media processing of FreeSWITCH provides that solution, and the XML-RPC interface allows for designing an elegant user interface.  Spitzer continues: "We were quite impressed by the audio quality of the mixing, and the ease with which it fit into the sipXecs architecture, being pure SIP and having simple XML based configuration.  The performance has shown to be outstanding as well, performing as well as (and sometimes better than) the proprietary solution on the same hardware. Features like being able to poll the conference bridge with XML-RPC queries made the Web User Interface for conferencing quite lively, with updating displays of participants and status and such.  As a bonus, the HD codecs like G.722 that FreeSWITCH supports give us a nice edge over many other solutions, including the ability to mix different codecs within a single conference." In addition to the conferencing needs, FreeSWITCH also lends itself to traditional telephony applications like automated attendants. Interestingly, SipFoundry wanted to replace the existing VoiceXML-based automated attendant. The event socket in FreeSWITCH gave SipFoundry the power and flexibility to create the necessary interface for the automated attendant module. Spitzer notes: "As for the Auto Attendant, we want to replace the existing VoiceXML based AA and Voice Mail with something that can handle more simultaneous calls, and be much easier to improve and add new features.  As a proof-of-concept of using FreeSWITCH as a media server, we started with the AA, writing the code in Java using FreeSWITCH's "event socket" interface.  The project went very well, and we couldn't be more happy with the performance and scalability of the new Auto Attendant. Programming in Java, with the ability to write unit tests and use nice IDE's like Eclipse, combined with the simplicity of the event socket interface, has made for an outstanding IVR programming model." With the success of its automated attendant, SipFoundry plans to use FreeSWITCH as the underlying voicemail engine in the next sipXecs release. For more information on SipFoundry please visit: http://www.sipfoundry.org/about.html [Less]

The FreeSWITCH development team would like encourage everyone to update to the latest pre-release version of FreeSWITCH. The tarball can be downloaded here. FreeSWITCH development is moving forward rapidly, so please check back often. The official ... [More] 1.0.4 release is very close. Having many people download the pre-release versions has helped tremendously. Please keep the feedback coming. -Michael [Less]