FreeSWITCH

Check out NXTVOX http://www.nxtvox.com/ for an inexpensive way to try out our new OpenZAP channel driver for FreeSWITCH. The cards are compatable with both asterisk and FreeSWITCH! 

FreeSWITCH announces fully tested support for TLS sip signalling and SDES secure RTP making it possible to make secure VoIP calls with the standard release of the software! No patches or workarounds are necessary! Now you can communicate with ... [More] someone over the internet using FreeSWITCH and have a completely secure channel of communication. Used in conjunction with the G722 or speex16 HD codecs you can have crystal clear audio as well!             [Less]

Since the advent of the High Definition Television, The HD phenomena has spread far and wide in recent years making the “ordinary” “extraordinary” with the addition of a simple 2 letter prefix. The telecommunications industry is no exception with ... [More] the strengthening concept of HD-Telephony. Is it all hype or is there an actual benefit to better sounding phone calls? How does this affect the hardware and software designed to keep us connected? As a software developer in this field, I hope to shed a little light on the topic and separate the facts from the misconceptions. The term “High Definition” is traditionally applied to a technology when a new innovation allows a noticeable improvement in quality or detail from an existing version. The superior picture quality of an HD TV for example. The improvement in telephone audio quality that HD-Telephony offers is far less profound than that of HD-TV or HD-Radio but it’s an improvement nonetheless. The biggest issue with HD-Telephony is that the majority of phones and phone switches were designed with the mentality that every call they would encounter would be the same audio format and sampling rate. As VoIP grows in popularity we already face this issue with the various encoding formats used to reduce the size of the media stream. Encoded audio cannot be manipulated or analyzed and must first be decoded to alter the volume for instance. Then it must be re-encoded to the proper format when passing the call to the public telephone network. Variances in audio sample rates add another dimension to this problem because the decoded audio may then need to be re-sampled before being re-encoded or processed. If a high-definition audio signal is re-sampled to a lower rate at any point, all benefits of using that format are lost. Therefore, it is important to ensure that any calls using an HD-audio format only pass through devices that use the same sampling rate as the original call. So at this time, using a higher quality audio format for calls destined for the public phone network is pointless. On the other hand, there are many benefits to supporting HD-audio in your application or device with little or no cost to the bandwidth usage on your network as long as you pay attention to when and why you are using it. I have done a great deal of work for the Asterisk open source PBX project. Asterisk supports voice-over-IP but it was really designed to support legacy telephone equipment such as analog telephones and digital circuits where higher quality audio is far from a reality and of little concern. That was among the many reasons I decided to start my own project called FreeSWITCH. Since I had the luxury of knowing about the importance of varying sample rates, I was able to design my application to not only translate audio between various encoding formats but also to mix and resample audio as well. I think this has paid off in the long run because I now see many opportunities to take advantage of HD-audio as newer phones are being developed. One situation where higher quality audio shines is when you have several phones on the same network that support HD-audio. In this case it will be possible to experience a noticeable improvement in quality with every call. When calls are destined for the public telephone network or some other legacy device that will only support lower quality audio, the switch can negotiate the call with the calling phone at the lower quality in anticipation. This puts the burden of re-sampling on the phone and since the phone was designed to operate at either format anyway it has little impact on performance. Once you have the logic in place to determine when to use high definition audio most of the disadvantages begin to melt away. In FreeSWITCH, we can take full advantage of high-definition conferencing, audio playback and speech generation / recognition when applicable. In the case of conferencing we can even allow legacy devices to join a high definition conference by re-sampling the audio to the correct format. As long as the format is chosen wisely or corrected to match the format of the destination there are no drawbacks to the addition of HD-Telephony into an existing architecture. Another concern many have with the idea of HD-Telephony is that it appears to be counter-intuitive with the goal of efficiency. There is a misconception that because the quality of the audio is higher, then the amount of bandwidth necessary to transport that audio must also increase by the same factor. To transmit uncompressed audio, it will indeed require twice bandwidth of an 8khz stream to send a 16khz media stream. However, once encoded, a 16khz audio stream can use the same or less bandwidth than a standard 8khz g711 stream. It may be true that some encoding formats designed for 8khz can greatly reduce the network bandwidth required to send a high volume of calls but again, in those cases, the telephony switch can request the most optimal encoding format from the phone or provide the encoding to the desired format by way of software or hardware encoding. The basic principle is that if the one phone calls another phone on the same network or when the exchange dialed is a known high definition resource, it chooses a high definition format and when the phone calls an exchange that will put the call over a highly encoded trunk using something like g729 it will negotiate g729 with the phone from the beginning to get the optimal results of that media path. Many new SIP phones now support high quality audio using g722 as well as low bandwidth codecs such as g729. The open source speex codec supports wideband 16khz as well as ultra-wide-band 32khz. Many new wireless phones support both narrow and wideband audio formats making it possible to complete a HD-audio call from your mobile phone to a HD-ready conference or SIP phone without giving up any additional resources or functionality. There is simply another factor to consider when negotiating the call and just because it’s more difficult to deal with, we should not ignore this technology that has actually existed for a long time. It may not offer many benefits to Ma Bell but HD-Telephony is emerging in the marketplace and when used properly, has the potential to raise our standards to the next level. [Less]

We promised to stop making new modules or features but.......we lied. We added: * mod_voicemail: a full featured enterprise scale voicemail application tied to the core directory that can be used with a local sqlite database or shared across ... [More] multiple boxes with ODBC. It supports mutiple greetings, fully customizable keys for every function, fully customizable email template, ability to forward messages to your email once you have listened to them, urgent messages with accurate urgent/non urgant message counts in mwi and templatable ivr prompts via the FreeSWITCH phrase macro interface. *mod_limit: a small application to allow restriction of certian simotaneous resources in the dialplan which also works locally with sqlite and at an enterprise scale with ODBC.   [Less]

Here is one of the more interesting examples of ways people are finding to play with FreeSWITCH Check It Out This application actually is integrating FreeSWITCH with a 3D graphics engine.

As we move another step closer to our 2nd beta, we added a few more features this week. 1) The channel variable RECORD_STEREO=true will make the record_session application record a stereo file with the read and write streams in seperate channels. ... [More] For the more seasoned developer, this also means the media bugs have a capability to return audio in mixed or stereo form for other applications as well. 2) The addition of mod_fifo. This module makes it possible to park many calls in a fifo queue and unpark them in the order in which they were received. This will make it possible to make parking and call distribution applications. The module sends events detailing stages of callflow as well as an api interface command to generate a full report of the call state. And is purely dynamic with no configuration necessary to support an infinite amount of fifo queues. http://www.freeswitch.org/eg/xml_cdr/fifo_log.html       [Less]

App_confcall is a conferencing application for Asterisk®. This application provides advanced conferencing functionality, enabling more powerful conferencing applications to be performed. There are a few different conferencing modules for ... [More] Asterisk®. This article will discuss app_confcall, a previously unreleased conferencing application now available to the Asterisk® community, those that use Asterisk®. A feature matrix of the 3 most popular conferencing solutions for Asterisk® will also be provided so that you as the user can decide which is best for your needs. Before going further, let me define what a conference application is. It is an application that bridges voice (or potentially video) of 1 or more endpoints with each other. Generally it is dynamic in that users may join or leave during a conference, when one speaks the others hear them. All of the modules that I will talk about currently only support voice, and utilizing Asterisk® can use any channel type that Asterisk® supports. This includes VoIP and TDM, or a mix of the two. MEETME Meetme is the default Asterisk® conferencing module, which provides the basic conferencing functionality as well as a few additional features. Generally this is built by default, and available for immediate use. CONFERENCE This is a seperate module, which expands upon MeetMe's functionality, and adds a few other features. This is available at http://sourceforge.net/projects/appconference/. CONFCALL ConfCall is a module written to provide even more flexibility and functionality. As you can see from the feature matrix below, it has many capabilities that are not present in other conferencing solutions. App_confcall may be downloaded at the FreeSWITCH.org's Asterisk Stuff page. FEATURE MATRIX   meetme conference confcall audio x x x mute x x x Music when alone x   x menus x   x deaf   x x join/leave sounds x   x AGI in background     x Music when muted     x record name before enter     x record conference x   x pins x x x dynamic conference   x x close conf when moderators leave x   x admin ability x   x background noise reduction     x selectable context on exit     x lock conference x   x adjustable volume per member     x send dtmf to conference     x kick users x   x play sounds     x manager interface   x x talker detection   x x place outbound calls     x operate without timer   x   VAD/DTX support   x     Some of the features are quite handy, for example the manager interface allows you to know when things occur in the conference and allow you to programatically respond to events. They can even able more advanced web based solutions to be created. The background noise reduction is also a very nice feature, this will attempt to detect constant background noise levels, and only bridge media to other participants when this level is exceeded. In this way if one participant is in a somewhat noisy environment it does not impair the other participants to hear each other. By having the ability to select the context that someone goes to when they exit the conference, you get the ability to use the conference app as part of a larger system, perhaps a party line with multiple conference rooms letting you control better callflow when they leave a conference. ABOUT THE AUTHOR app_confcall was written by Anthony Minessale II, who has contributed to the Asterisk® community by way of modules as well as many core contributions to Asterisk®. Currently he spends most of his time working on FreeSWITCH™, an advanced communications platform enabling embedding into other applications as well as being a standalone softphone, pbx or softswitch. Anthony often helps people with Asterisk® as well as FreeSWITCH™ and participates with OSTAG, the Open Source Telephony Advancement Group. He is committed to further research and development in enabling people to communicate more effectively with less effort. [Less]

I'll be attending Astricon in Phoenix on Sept. 24th thru the 28th and want to get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Please find me on IRC if you wish to touch base at Astricon. Thanks, Brian West  

Its been a long and hard road but we are finally at Beta1. Please download, test and report any bugs on Jira. In addition I would like to point out that Cepstral sponsored work on OpenMRCP so the world could benefit from MRCP v1 and v2 ... [More] integration for accessing ASR and TTS resources. FreeSWITCH and its team of great people were instrumental in making sure OpenMRCP became a reality.  Here is the news release from Cepstral about OpenMRCP. I would like to thank the Community for all their support. Feel free to join in. Thanks, Brian West FreeSWITCH.org [Less]

I will be attending SpeechTek in NYC next week(August 20-23). If you're in the area please feel free to find me (Brian West). I'll be hanging around in the LumenVox/Cepstral area. Also be on the look out for some existing announcements coming from these guys. If your all about speech then this is the event to be at. Hope to see you there!