Polycom SIP 3.0.2 Administrator's Manual page 283

Note
Central Configuration file:
sip.cfg
(boot server)
Voice Quality Monitoring
Configuration File Changes
Sampled Audio for Sound Effects <saf/>
The network bandwidth necessary to send the encoded voice is typically 5-10%
higher than the encoded bit rate due to packetization overhead. For example, a
G.722.1C call at 48kbps consumes 5xkbps of network bandwidth (one-way audio).
Two-way audio would take over 100kbps.
Configuration changes can performed centrally at the boot server or locally:
Specify codec priority, preferred payload sizes, and jitter buffer tuning
parameters.
• For more information, refer to
on page 1-4.
Voice Quality Monitoring is not supported on the SoundStation IP 6000
conference phone at this time.
The following sip.cfg configuration file changes were made to support the
SoundStation IP 6000 conference phone:
• Sampled Audio for Sound Effects <saf/>
• Voice Coding Algorithms <codecs/>
• Gains <gain/>
• Receive Equalization <rxEq/>
• Transmit Equalization <txEq/>
• Feature <feature/>
The following new sampled audio WAVE file (.wav) formats are supported:
• L16/32000 (16-bit, 32 kHz sampling rate, mono)
• L16/48000 (16-bit, 48 kHz sampling rate, mono)
Administrator's Guide Addendum for the SoundStation IP 6000
Codec Preferences <codecPref/>
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