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Submission Summary: 0 pending, 34 declined, 1 accepted (35 total, 2.86% accepted)

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Submission + - Barracuda Networks Connects FLASH To SIP for FREE (

anthm writes: "Barracuda Networks Inc. Has donated the new mod_rtmp for FreeSWITCH an open source telephony platform. Using the module, you can connect a web page to your server and make phone calls that can gateway to and from SIP or other protocols or just call a conference or other voice app on the FreeSWITCH server. The module is licensed under the MPL and is freely available as part of the FreeSWITCH source tree."

Submission + - ITEXPO – Ooma is all open source FreeSWITCH (

anthm writes: The popular Ooma VoIP device announces at IT expo that they are based on the open source VoIP platform FreeSWITCH

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.

Submission + - FreeSWITCH 1.0.6 Released Supports Skype Codec (

anthm writes: FreeSWITCH, the only open source telephone switch, has released version 1.0.6 that supports the SILK audio codec from Skype.
Also supported is the newly open sourced broadvoice bv-16 and bv-32 audio codecs among a ton of other features making FreeSWITCH capable
of merging computers and telephone networks.

FreeSWITCH is also capable of connecting calls with the Skype applicaton as well.


Submission + - Free Software Connects Skype To Everything Else (

anthm writes: "An Open Source Soft-Switch, FreeSWITCH now has support for the Skype protocol. It's the first step towards bridging the gap between proprietary Voice Over IP and open standards such as SIP, H323 and Jingle. Using the new module it's possible to bridge Skype to SIP, PSTN and Wide-Band Conferencing. The software is first in line for Skype's new SILK codec."

Submission + - Open Source VoIP That Surpasses CD Quality (

anthm writes: "Just a few days after releasing support for the Polycom Siren(tm) codec allowing VoIP at 32khz, FreeSWITCH Announces support for the CELT codec.

The FreeSWITCH implementation of the CELT codec allows any other device that can use CELT to send ultra-high-definition audio in a very small package. The bandwidth rate of 48kps in the FreeSWITCH implementation is actually less data per stream than the audio format that traditional telephones use.

Higher frequencies allow more detail in the audio making things like music and voice sound more rich and true to it's original sound. This offers a new frontier for telephony. The world has grown used to the low quality of the PSTN but many would happily exchange it for crystal clear call-quality to go with this new High-Definition age.

CELT stands for "Constrained Energy Lapped Transform" and is created by the same team responsible for the Speex codec.

FreeSWITCH is an Open Source soft-switch and application server hosted at"


Submission + - Cisco Sued by SFLC (FSF) for Open Source Violation (

anthm writes: "According to this blog post Cisco is being sued by the SFLC

Cisco, the networking giant, should know better than this, but they've worn out the FSF's (Free Software Foundation) patience. So, Cisco is now being sued by the SFLC (Software Freedom Law Center) on behalf of the FSF for Linux and other GPL copyright violations."


Submission + - How does FreeSWITCH compare to Asterisk? (

anthm writes: "How does FreeSWITCH compare to Asterisk? Why did you start over with a new application? These are questions I've been hearing a lot lately so I decided to explain it for all of the telephony professionals and enthusiasts alike who are interested to know how the two applications compare and contrast to each other. I have a vast amount of experience with both applications with about 3 years of doing asterisk development under my belt and well, being the author of FreeSWITCH. First I will provide a little history and my experience with Asterisk, then I will try to explain the motivations and the different approach I took with FreeSWITCH."

Submission + - Will telcos accept an open source switch? (

anthm writes: "This story on ZDNet discusses the viability of an open source telephony switch in the commercial TELCO World.

FreeSWITCH is an Open Source Telephony Switch that can be used to move high volumes of VoIP and TDM phone traffic. The potential for disruption is that the Free Software can hold it's own against commercial Carrier Grade equipment.

"One unasked question is whether telcos will ever embrace this kind of open source and the savings it brings.""


Submission + - An Open Source Class-Five Telephony Switch (

anthm writes: " — Open source projects have revolutionized operating systems, web servers, web browsers, and so why not carrier switches? The FreeSwitch open source project released its Release Candidate 1 (RC1) yesterday providing and by early accounts the software rocks."

Submission + - Open Source Switch Takes Down Telco (

anthm writes: "FreeSWITCH had this report on April 1 (Yes april fools day but no joke It was a backwards April fools joke by telling the truth about something hard to believe)

"Routinely my freeswitch routing servers reach > 400 sessions Per Second (where they are rated limited) and > 4000 concurrent Sessions (where they are also limited) with approx 20% CPU utilization and 30% ram utilization on a Dell 1950 w/ Dual Xeon E5335 2Ghz Quad Core CPUs and 4G of ram" Any SPS and Concurrency limiting is done not to protect my boxes but upstream peers I have had atleast 3 occations where Large Tier 1 / Tier 2 Intl and Domestic US LD Carriers have asked for mercy one called saying "Can you please slow the traffic up, you are melting down my {CENSORED} SBCs" well the vendors in question that I know for a Fact are Veraz and AcmePacket.""


Submission + - FreeSWITCH, a free telephone switch (

anthm writes: "FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, 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, 16 or 32 kilohertz and can bridge channels of different rates.

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.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver."

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