Posted
almost 16 years
ago
by
mcollins
The FreeSWITCH team is pleased to announce the immediately availability of FreeSWITCH version 1.0.4. (Source tarball available here.) This new version contains many improvements in stability and security as well as some notable additions.
Skype -
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VoIP For The Masses
FreeSWITCH 1.0.4 now supports Skype integration. If you have a Skype account you can send and receive Skype calls from FreeSWITCH. The Skype module (mod_skypiax) uses the native Skype client for the target operating system. Currently, Windows and Linux are supported. Skype calls can be made or received, and all of FreeSWITCH's transcoding and resampling capabilities are available with Skype calls. The Skype audio codec, SILK, yields higher quality calls than the traditional PSTN, and FreeSWITCH's ability to transcode and resample other high quality codecs allows for Skype calls to connect to SIP endpoints with excellent quality when those endpoints are using HD codecs such as CELT or Polycom Siren.
OPAL - Adding H.323, IAX2
The OPAL project is an open-source library of VoIP functionality. Robert Jongbloed, co-founder of the OPAL project, has been working with the FreeSWITCH developers to create mod_opal, which adds H.323 functionality to FreeSWITCH as well as IAX2 support. The developers are also working on adding more functionality in future releases, such as IAX registrar support.
ZRTP - Secure media streams
ZRTP is a means by which telephone sets and phone systems can easily encrypt media streams with SRTP. ZRTP is co-authored by Philip Zimmermann of PGP fame. ZRTP works with virtually any software that supports SRTP. ZRTP allows for automatic, opportunistic encryption of the media streams. If two endpoints have ZRTP then calls betwen them are encrypted automatically.
MRCP - Controlling Media With External Servers
MRCP, or Media Resource Control Protocol, is now available with a new module: mod_unimrcp. This new module builds takes advantage of the well-written UniMRCP library, authored by Arsen Chaloyen. The module is compliant with MRCP version 2 (SIP). MRCP allows for media servers to be separated from FreeSWITCH. For example, it is possible to have an MRCP server on a seperate machine running software such as Voxeo Prophecy which can perform text-to-speech (TTS) functionality. The MRCP server generates the media stream and feeds it to the MRCP client. MRCP allows for distribution of resource-intensive operations such as ASR and TTS.
NAT Traversal Made Easier With UPnP, NAT-PMP
The FreeSWITCH developers have included additional NAT-handling functionality in version 1.0.4. NAT-traversal is important for SIP calls and is made easier with the automatic NAT handling feature. NAT devices (like routers and firewalls) that support either UPnP or NAT-PMP can be polled by devices inside the network, which allows them to determine the external IP address without resorting to more clumsy methods such as STUN.
Real-Time Billing
FreeSWITCH now supports real-time pre-paid billing applications with mod_nibblebill. The name of the module is telling: it monitors calls in progress and "nibbles" away at the available credit of a pre-paid user's account. It also allows for dyanmically sending audio to the pre-paid caller, alerting him that his account is about to run out and that that call will be disconnected. When the user's credit runs out, the call disconnects and there is no need to handle awkward situations like negative balances.
In addition to the aforementioned additions, FreeSWITCH 1.0.4 contains numerous updates, improvements, and bug fixes. The latest version is the most stable version available and all users are strongly encouraged to upgrade as soon as possible.
The FreeSWITCH development team would like to thank the many community members who have dontated their time and efforts to making FreeSWITCH such a successful open source project. [Less]
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Posted
about 16 years
ago
by
admin
Just a quick note for those coming to ClueCon this year: iCall is sponsoring a meet and greet on Tuesday night. More information is available at the ClueCon website.
Only five more days until ClueCon 2009! Sign up now if you haven't already done so.
-Michael
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Posted
about 16 years
ago
by
admin
FYI,
The three-part interview with Anthony has been posted. Click here to see part three, which contains links to parts one and two. Also, please be sure to Digg and comment on the interview.
-Michael
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Posted
about 16 years
ago
by
mcollins
Okay, everyone needs to check out this link to a great interview with Anthony. Among other things you'll find out more about how Anthony got into telephony. Check it out!
-Michael
|
Posted
about 16 years
ago
by
mcollins
The inexorable march to the official release of FreeSWITCH 1.0.4 continues. Version 1.0.4pre9 is now availble for immediate download. All those not using the latest SVN are encouraged to upgrade as soon as possible. The tarball is available here.
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This latest pre-release is now the most stable version of FreeSWITCH available. There have been numerous improvements and bug fixes. The community has been providing a lot of feedback and the developers are doing a good job of addressing any issues that are found. There are new features as well. One in particular that you may appreciate is a new FS API command: show distinct_channels.
Please keep those bug reports and feedback flowing. The developers are working very hard to make sure that every bug is squashed and every user concern is addressed. Keep up the good work.
-Michael [Less]
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Posted
about 16 years
ago
by
mcollins
A special thanks goes out to Kristian Kielhofner of Astlinux fame - and Yahoo! message board infamy - for creating bootable ISO's with FreeSWITCH. Kristian has graciously added some automation to his setup and each night a new ISO is created. The
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link is here:
http://mirror.astlinux.org/freeswitch/daily/
Many thanks to all of the folks out there who are working so hard to make the FreeSWITCH community the best in the OSS world!
-MC [Less]
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Posted
about 16 years
ago
by
mcollins
The FreeSWITCH development team is pleased to announce that FreeSWITCH now supports MRCP with the addition of mod_unimrcp. A very special thanks goes out to Arsen Chaloyan, author of the fantastic UniMRCP library.
What is MRCP and why is it valuable?
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Media Resource Control Protocol (MRCP) is a protocol that allows a server to offer voice services to its clients. These services include automated speech recognition (ASR) and text-to-speech (TTS). MRCP allows these services to be server-based and therefore decentralized. This in turn allows for better scalability and reduncy of these services as they do not need to reside on the same hardware that runs FreeSWITCH. There are two different versions of MRCP. Version 1 supports RTSP and version 2 supports SIP.
A number of ASR and speech synthesis vendors have products that are MRCP-compliant. These vendors include Cepstral, Voxeo, LumenVox, and Nuance. FreeSWITCH uses configuration files called MRCP profiles to specify connection parameters to each MRCP server. Several predefined profiles have been created. Additionally, the user can create or modify MRCP profiles to fit his specific configuration. For example, I was testing mod_unimrcp on my Macbook Pro and I did not have access to the Voxeo 8.0 Prophecy software, so I installed the beta of Prophecy 9.0 and created a custom MRCP profile.
How does MRCP work in practice? In my example, the client (FreeSWITCH) makes a connection to the server (Voxeo) and requests some sort of media service. I set up my dialplan to call the speak dialplan application like this:
<action application="speak" data="unimrcp||This is a text to speech test."/>
When the server accepts the request it will then negotiate a media connection and send the audio back to the client. The net effect is that this line in the dialplan will use the MRCP server to generate TTS to the connected channel. In this test the Voxeo and FreeSWITCH were on the same machine, but in production the MRCP server(s) would be on different machines.
The addition of mod_unimrcp to FreeSWITCH was relatively quick. It demonstrates the wise design philosophy of the FreeSWITCH architecture. With the help of a few enterprising community members, and the author of the UniMRCP library, the bulk of the work was accomplished in just a few days. Features are being added to FreeSWITCH very rapidly. The community can help out by downloading the latest trunk from the SVN repository and testing these new features in your various environments. [Less]
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Posted
about 16 years
ago
by
mcollins
FYI,
IBM's Rob Smart has been busy with FreeSWITCH! For those of you living in the U.K. I recommend this nice blog post explaining how to set up FreeSWITCH as your home PBX/voice communications server. Now if we can just get Rob and IBM to sponsor ClueCon... :)
-MC
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Posted
about 16 years
ago
by
mcollins
Some of you may be aware of the fact that the OpenSimulator project has some enterprising community members who've used FreeSWITCH as the voice engine. We just wanted to report that they are also getting ready to have a VoIP-enabled image on Amazon EC2. Check it out and spread the word.
-MC
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Posted
about 16 years
ago
by
mcollins
We are pleased to let FreeSWITCHers everwhere know that long-time FreeSWITCH contributor Raymond Chandler and his wife Samantha are expecting their fifth child next month. We all want to give young Kaiden Anthony Chandler a warm welcome to the
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FreeSWITCH family. To that end, we are happy to let you all know that Samantha is registered on WalMart's baby registry here. Raymond (IRC: intralanman) spends a lot of time and energy working on FreeSWITCH and his family is very supportive. Samantha (IRC: lanwifie) spends a fair amount of time in the channel when she's not being a very busy mom. Let's all show our appreciation to the Chandler family by getting something for Kaiden! [Less]
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