Polycom SIP 3.0.2 manual Two-way audio would take over 100kbps

Administrator’s Guide Addendum for the SoundStation IP 6000

	Note		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:
Central		Configuration file:		Specify codec priority, preferred payload sizes, and jitter buffer tuning
(boot server)		sip.cfg		parameters.
				• For more information, refer to Codec Preferences <codecPref/>
				on page 1-4.

Voice Quality Monitoring

Voice Quality Monitoring is not supported on the SoundStation IP 6000 conference phone at this time.

Configuration File Changes

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/>

Sampled Audio for Sound Effects <saf/>

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)

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