Posted
almost 16 years
ago
by
mcollins
The FreeSWITCH development team has struck again! FreeSWITCH now supports CELT, a new open source audio codec that allows for CD-quality transmission with VoIP. CELT is described as an "ultra-low delay" audio codec that supports both voice and
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music. 48kHz VoIP can be carried in 48kbps of bandwidth!
Putting this in perspective: CD-quality audio is 44.1kHz. This means CELT supports better than CD-quality audio in a VoIP environment! The only bad news here is that you might need to upgrade your headset so that your microphone is sensitive enough to handle such a wide range. Also, consider bandwidth usage: 48kHz of super high-quality audio requires a mere 48kbps of bandwidth. By comparison, the venerable PCM mu-law codec, G.711u, consumes 64kbps of bandwidth and yields only 8kHz audio quality!
What are some applications of such a high-quality codec with FreeSWITCH? For one, CD-quality conferencing. Another is radio station backhaul. Have you ever heard a radio talk-show? The host is crystal clear and the callers are sometimes very difficult to hear. This could be remedied by using a higher quality codec for guests and callers.
New codecs are creating new opportunities for voice and video applications. Another important consideration is the intellectual property involved in many codecs.
The Evolution of "Free Speech"
Anyone who has dealt with the hassles of licensing a proprietary codec, such as the ubiquitous G.729, knows that such patent-encumbered codecs cost more than just money. Developers certainly know the issues: high initial fees and silly licence enforcement schemes. These schemes require resources that could otherwise be applied to advancing the project that uses the codec.
To get an idea of just how "patent-encumbered" G.729 really is, please see the attached PDF report from the ITU Intellectual Property Rights (IPR) page. By my count, this codec is encumbered by 52 patents from 17 different entities! These entities are part of the G.729 "consortium" and they use Sipro as an agent to handle the business of collecting money from those who use the codec. Beyond that, more patent holders can join the consortium if they demonstrate that their IP is used in the codec. Now look at all the codecs in the ITU's IPR database... That's a lot of encumbered technology.
To combat this trend, a number of technology specialists have devoted themselves to the advancement of "free speech" - that is, unencumbered codec technologies for use in audio and video environments. Many of these codecs are of the highest quality and are as good as, or better, than their proprietary counterparts. Two well-known codecs are Vorbis (music, general audio) and Theora (video). Another such free and open source codec is called Speex. Speex is optimized for speech and like G.729 it is a "lossy" format meaning that some audio quality is sacrificed in order to minimize bandwidth. Speex can operate at 8kHz, 16kHz, and 32kHz.
The CELT codec represents the best of both worlds: audio quality equal to that of Vorbis while acheiving the low latency needed for VoIP transmissions. We anticipate that more equipment manufacturers and service providers will turn to high-quality, low-cost codecs like CELT. To sum it up, CELT is simply an awesome codec and is a welcome addition to the FreeSWITCH project.
The FreeSWITCH community would like to thank Jean-Marc Valin, the Project Lead for the CELT project, whose work made this exciting addition possible.
FreeSWITCH users will see this codec as mod_celt.
For more information on CELT please visit the project home page.
For more information on other free codecs please visit the Xiph page.
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Posted
almost 16 years
ago
by
mcollins
The FreeSWITCH development team is pleased to announce the
availability of two new codecs:
Polycom's G722.1 and G722.1C Codecs (Siren).
The Polycom Siren(tm) family of codecs offers high-quality and low-bandwidth
performance that works for all
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audio, including music. The Siren(tm) 7
G.722.1 (16 kHz) and Siren(tm) 14 G.722.1C (32 kHz) codecs will allow
FreeSWITCH users to have access to HD telephony without expensive
licensing requirements.
This makes FreeSWITCH the first and only open source software that can do transcoding, conferencing and bridging of this exciting new 32khz audio format.
The FreeSWITCH community would like to extend a special thank you to
Steve Underwood for graciously donating his knowledge, experience, and
programming abilities to the creation of these codec libraries as well
as his continued support for the FreeSWITCH project.
These codecs will appear in FreeSWITCH as mod_siren.
More information on these codecs can be found at Polycom's Web site:
G.722.1 - http://www.polycom.com/usa/en/company/about_us/technology/siren_g7221/siren_g7221.html
G.722.1C - http://www.polycom.com/usa/en/company/about_us/technology/siren14_g7221c/siren14_g7221c.html [Less]
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Posted
almost 16 years
ago
by
mcollins
We would like to call attention to some new modules that have been added by enterprising members of our community:
mod_vmd - Voice message beep detection (by Eric Des Courtis)
mod_http - API for fast HTTP operations (by Eric Des Courtis)
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mod_limit - Improved limit functionality using hashes (by Mathieu René)
munin plugin - A plugin to graph channel usage (by William King)
I am working with the authors to make sure that the wiki gets updated properly.
The FreeSWITCH developers very much appreciate these voluntary contributions that help further the success of the project. Well done!
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Posted
about 16 years
ago
by
ranahimal
Dear All,
I couldnt find out where to put my note so m puting it here.
First of all i was wondering how much the support for the h323 channel is there in FS.
I read in couple of threads that there is mod_opal being written in place of the
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mod_wommera.
I was having some doubt regaridng it.
As h323 librady is written as per connection one thread it would be very heavy on the system when there are more concurrent calls ( with tunneling off and fs off).
In that case they have implemented the SOCKET AGREGATION in h323 library.
But it is very absolute for now as saw the code.
Was thinking somone is working on that end so there can be very good h323 library.
Himal
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Posted
about 16 years
ago
by
anthm
Celliax has recently announced they will be adding support for FreeSWITCH.
http://www.celliax.org/node/503
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Posted
about 16 years
ago
by
anthm
We've come in at 8th place in this impressive list of VoIP platforms.
Not too bad considering 1st,2nd,4th and 7th are all derivitaives of Asterisk in one way or another.
http://blog.voipsupply.com/uncategorized/need-an-ip-pbx-101-alternatives...
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Posted
about 16 years
ago
by
brian
Providers need an inexpensive solution for hosting multi-tenant, secure, scalable VoIP services. Such a solution is now available in the marketplace. When VoIP technology first moved into the mainstream, solutions were rare and expensive. Solutions introduced included Session Border Controllers (SBCs), softswitches, and Gateways.
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Posted
about 16 years
ago
by
anthm
We are mentioned in this article on TMCNET.
http://digg.com/software/VoIP_Providers_Needs_are_Changing_over_Time
Have a look and be sure to Digg it!
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Posted
about 16 years
ago
by
brian
For those who were not able to make it to Cluecon this year can now watch the videos on your iPhone and iPod touch or your PC/Mac.
Here are the videos!
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Posted
about 16 years
ago
by
anthm
I have been so busy coding for the last few years I haden't even noticed. Before I started FreeSWITCH, I was an avid Asterisk developer putting all my spare time into the betterment of the project. When the book was released Asterisk: The Future
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of Telephony by O'Reilly Media I thought it was awesome that they mentioned one time Asterisk bug marshall and now FreeSWITCH QA Manager Brian West as well as myself in the book:
• Brian K. West, for your commitment to the community, Asterisk, our book, and
open-source telephony
Anthony Minessale (a.k.a. anthm) is one of the unsung heroes of Asterisk develop-
ment. The number of people who have contributed to Asterisk development are many; the number who can claim to have matched Anthony’s efforts are few.
WOW! guess what? They cut both mentions out of the book. Yet they certianly didn't cut all the contributions we gave them from the code base or the book. I would not have beleived it but they actually seem to have purposely removed our names from the book. I guess my efforts were not quite as memorable as they first thought. Yet I believe most of the things on my list of contribtions at http://www.cluecon.com/anthm.html still remain.
I have to say, This is the biggest dissapointment to date from Asterisk since the time Mark Spencer told me I was going to receive a gift from the Digium for all my hard work on the Asterisk project back in 2005 and then never actually gave it to me...........Sigh......
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