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sevanaoy AQuA - Audio Quality Analyzer for IP PBXs 0 Tuesday, 8:28 AM EST by sevanaoy
Thread started: Tuesday, 8:28 AM EST  Watch
AQuA - Audio Quality Analyzer

AQuA provides precise and extensive audio quality testing that can be easily brought to your company's own hardware enabling your monitoring server to graphing MOS/PESQ values for various voice termination locations.

Many companies are using this product to test and monitor voice and audio quality of conference bridges, carriers and VoIP links. Typically companies that provide telecom solutions have automated "test dialers" to test networks of conference bridges, carriers, terminations etc, and the system is able to place test calls and record them to for example a wav file. And the only thing that is missing to obtain full test sequence is an ability to automatically compare wav files to determine if there are any significant audio quality problems.

AQuA voice quality solution is designed exacty to fit these needs. In case of Asterisk for example voice quality monitoring using AQuA is implemented in the following way:

1. Originate a call on the monitoring server using Asterisk manager interface to a server which is running an echo application.

2. Monitor both inbound and outbound legs of the call and save them as wav files.

3. Use AQuA (Linux or Windows version of the software) to compare the wav files.

This approach is quite effective as one does not need to purchase additional hardware thus utilizing existing infrastructure. We sincerely believe that many other companies will find AQuA product useful, competitive and cost effective solution for audio quality testing and monitoring.

Thank you very much for your time and we are looking forward to hearing from you!

Best regards,
Endre Domiczi

Sevana Oy
CEO/Co-founder
Phone: +358 9 23164165
Mobile: +372 53485178
http://www.sevana.fi
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Keyword tags: SIP Software
hankkarl ISDN, R2, RBS to SIP interworking module for IP PBXes 4 Oct 5 2009, 10:14 AM EDT by hankkarl
Thread started: Apr 20 2009, 10:21 AM EDT  Watch
PSTN-SIP module allows IP PBXes to connect to the PSTN via ISDN PRI or BRI, QSIG, CAS R2 or RBS, or analog telephony. Pre-ported version available for Linux (Windows is on the roadmap) that interfaces to DAHDI (was Zaptel) boards so you can use the least expensive hardware solution. DAHDI over ethernet is also supported.

This product allows you to implement an ISDN (or other PSTN) interface with minimal knowledge of ISDN or SIP.

Asterisk and other FOS PBXes benifit from gaining the ability to use the industry-proven TeleSoft ISDN stack that is not performance limited in the way that the free stacks are. Configuration and installation is much easier than configuring and installing the free stacks.
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hankkarl High Performance, High Quality, Fully Supported SIP Source Code 0 Apr 20 2009, 10:13 AM EDT by hankkarl
Thread started: Apr 20 2009, 10:13 AM EDT  Watch
High Performance, High Quality SIP source code written by experts in the Telecom field is fully supported. This stack was written with ease of use and portability in mind. It has both low CPU cycle and low per-call RAM requirements, and scales well so it is suitable for use in the largest SoftSwitches, conference bridges, B2BUAs and other devices as well as being suitable for small devices like TAs and Handsets. Various user interfaces are available, including the native API which is very powerful, and a softphone client for use on mobile devices which allows engineers to implement IMS SIP without a great knowledge of SIP.

Also available is an interface to the industry-standard TeleSoft ISDN and R2/RBS CAS stacks for use in gateways and interworking devices.

http://www.nine-9s.com/prod_embedded_sip_protocol_stack.htm

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