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Michael Graves Astricon 2009 Hd Voice Demo Rev2
1. Asterisk v1.6 & HDVoice Demonstrating HDVoice In Asterisk Using Polycom’s Siren Codecs
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14. Preparing Audio Samples PC with soft phone & Vemotion software plays the uncompressed wav files into a call Polycom VVX-1500 Records G.711,G.722 & G.722.1 calls to USB stick Line out from VVX to Zoom H2 recorder for G.722.1C SIP call analog
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Editor's Notes
15 years technology sales & marketing experience, primarily in broadcast television
“What we have here is a failure to communicate” - credit
“Life’s Too Short” – credit Jerry Jeff Walker “What we have here is a failure to communicate” - credit
The sense that a device is cheap lives on long after the memory of the deal has faded.
* Had to add a fourth $ to indicate the VVX-1500 Business Media Phone
Encoding technique uses Modulated Lapped Transform (MLT) vs G.722 using simpler sub-band ADPCM, G.722.2 (AMR-WB) uses ACELP with is patented 16 kbps mode is an extension of the codec and not part of the ITU specification for G.722.1 ITU G.722.1 approved back in 1999 after several years of testing
Similar properties to Siren7 but a little higher bitrate Twice the audio bandwidth ITU standardization approved in 5/2005
MeetMe relies on DAHDI which does not support sample rates over 16 KHz ConfBridge supports higher sample rates
higher sample rates in future (Q2/2010?)
MeetMe relies on DAHDI which does not support sample rates over 16 KHz ConfBridge supports higher sample rates
Jeff Rodman, interview by TMC said something to this effect
Specific advantages to using the Siren codecs
Also MS OCS when in conference situations (Siren7), it uses RTAudio for point-to-point calling
General advantages of HDVoice 30,000 ft perspective