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Michael Graves Astricon 2009 Hd Voice Demo Rev2
 

Michael Graves Astricon 2009 Hd Voice Demo Rev2

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Slide set from my Astricon 2009 Presentation about HDVoice in Asterisk

Slide set from my Astricon 2009 Presentation about HDVoice in Asterisk

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  • 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

Michael Graves Astricon 2009 Hd Voice Demo Rev2 Michael Graves Astricon 2009 Hd Voice Demo Rev2 Presentation Transcript

  • Asterisk v1.6 & HDVoice Demonstrating HDVoice In Asterisk Using Polycom’s Siren Codecs
  • Who Am I?
    • Michael Graves, Pixel Power Inc
      • System Integration Manager
      • Oversee telecom for North America
    • VoIP Blogger
      • Focused on SOHO/SMB space
      • Asterisk user since 2003
    • VoIP Users Conference
      • Guest host & frequent contributor
  • Statement Of Principles
    • Anything that helps people to communicate better is desirable
  • Statement Of Principles
    • Asterisk is an engine for disruption of the telecom industry
  • Statement Of Principles
    • Life’s just too short to suffer though using a lousy phone
  • A Few Simple Facts
    • Polycom hardware is extremely popular with Asterisk installations
    • Polycom offer high-quality SIP hard phones at prices from $ to $$$$*
    • The SoundPoint IP series desktop models support G.722
    • VVX-1500 & SoundStation conference phones support G.722, Siren7 & Siren14
  • Polycom Siren7 Basics
    • HDVoice call quality at reduced bitrates
      • Modulated Lapped Transform (MLT)
      • Suitable for music & speech
      • 16 KHz sample rate
      • 50 – 7,000 Hz audio response
      • Three bitrates: 16 * , 24 or 32 kbps
      • 40 ms algorithmic delay (20ms frames)
      • ITU-T Standard G.722.1 (9/1999)
  • Polycom Siren14 Basics
    • Siren 14 – “Super-Wideband” calling
      • 32 KHz sample rate
      • Suitable for music & speech
      • 50 – 14,000 Hz audio response
      • Three bitrates: 24, 32 or 48 kbps
      • 40 ms algorithmic delay (20ms frames)
      • ITU-T Standard G.722.1 Annex C
  • Asterisk Development
    • Access to Siren is in very recent code
      • Asterisk v1.6.20 trunk
      • Progressed significantly over the past two weeks
    • Polycom offers royalty free license
      • Siren 7&14 are not released under GPL
      • The Siren codecs are not part of the normal Asterisk distribution
      • Distribution is as a binary module
  • Using Siren In Asterisk
    • Canonical names used in sip.conf & iax.conf
      • Allow=siren7
      • Allow=siren14
      • Allow=g.722.1
      • Allow=g.722.1c
  • Using Siren In Asterisk
    • Current implementation supports:
      • Full rate only
        • Siren7 @ 32 kbps
        • Siren14 @ 48 kbps
      • Asterisk is limited to 16 KHz sampling
        • Siren14 streams down-sampled (32 > 16/8 KHz)
        • Support higher sample rates in future
          • CELT, Siren14, SILK
  • Using Siren In Asterisk
    • Current implementation supports:
      • Recording (ex voicemail)
      • Playback (ex IVR prompts)
      • Conferencing
        • MeetMe application relies upon DAHDI for timing and does not support wideband conferencing
        • Using ConfBridge app @ 16 KHz sample rate
          • Considered “a bit experimental”
  • Performance Considerations
    • CPU requirement is asymmetrical
      • Encoding uses more CPU than decoding
    • Siren playback/decoding is very cheap
      • Similar in complexity to GSM-FR
      • Down-sampling is similarly easy
        • 16 > 8 KHz, 32 > 16 KHz, or 32 > 8 KHz
      • Less processor intensive than the aged G.722!
    • Encoding not especially processor intensive
      • Estimated <20% of the CPU load of a G.729a stream
  • 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
  • Sample #1: The Female Voice
    • I asked Mrs Evelyne Resnick
    • the following question:
    • “ What’s it like being married
    • to a VoIP geek?”
  • Sample #2: The Male Voice
    • Michael Iedema
    • Lead Developer
    • Askozia Project
    • http://www.askozia.com
  • Sample #3: The Male Voice
    • Ruben Olsen
    • VoIP blogger
    • http://www.open-voip.com
    • Norwegian
  • Name That Codec!
    • Languages :
    • Chinese
    • French
    • German
    • Russian
    • Spanish
    • Sample encodings :
    • Uncompressed
    • Siren14
    • Siren 7
    • G.711
    • Comparative
  • Name That Codec! Round #1
    • Languages :
    • Chinese
    • French
    • German
    • Russian
    • Spanish
    • Sample encodings :
    • Siren14
    • Siren 7
    • G.711
  • Name That Codec! Round #2
    • Languages :
    • Chinese
    • French
    • German
    • Russian
    • Spanish
    • Sample encodings :
    • Siren14
    • Siren 7
    • G.711
  • Name That Codec! Round #3
    • Languages :
    • Chinese
    • French
    • German
    • Russian
    • Spanish
    • Sample encodings :
    • Siren14
    • Siren 7
    • G.711
  • Name That Codec! Round #4
    • Languages :
    • Chinese
    • French
    • German
    • Russian
    • Spanish
    • Sample encodings :
    • Siren14
    • Siren 7
    • G.711
  • Name That Codec! Round #5
    • Languages :
    • Chinese
    • French
    • German
    • Russian
    • Spanish
    • Sample encodings :
    • Siren14
    • Siren 7
    • G.711
  • What Are The Advantages?
    • Optimal wideband interop with the broad range of Polycom HDVoice hardware
  • What Are The Advantages?
    • Superior voice quality at reduced bitrates for on-net calling
      • Internal calling
      • Inter-office calling
      • SIP trunking
      • IP Peering
  • What Are The Advantages?
    • High-quality interop with larger conference systems
  • What Are The Advantages?
    • A more enjoyable call experience
    • Fewer misunderstandings
    • Reduced call fatigue & frustration
    • Happier customers
    • Better quality of life
    • World peace
  • Special Thanks
    • Darrick Hartman, DJH Solutions
      • Help with Asterisk v1.6.20 trunk in Astlinux
    • Plantronics
      • Savi Go: a truly wideband capable Bluetooth headset
    • Michael Iedema, Askozia PBX
      • Chinese, English, German, Russian & Spanish
    • Ruben Olsen
      • English & Norwegian
    • Randy & Evelyne Resnick, Resmo
      • French & English