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Slide set from my Astricon 2009 Presentation about HDVoice in Asterisk
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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
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?
• 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