Polycom SIP 3.0.2 ユーザーズマニュアル

 Administrator’s Guide Addendum for the SoundStation IP 6000

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Configuration changes can performed centrally at the boot server or locally:

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:

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)

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 file: 
sip.cfg

Specify codec priority, preferred payload sizes, and jitter buffer tuning 
parameters.

For more information, refer to 

 

on page 

.