Polycom SoundPoint IP 600 SIP Administrator's Manual page 76

Administrator Guide - SoundPoint
Attribute
voice.audioProfile.x.jitterBufferShrink
voice.audiProfile.x.jitterBufferMax
4.6.1.6.2 Volume Persistence <volume/>
The user's selection of the receive volume during a call can be remembered between
calls. This can be configured per termination (handset, headset and handsfree/chassis).
In some countries regulations exist which dictate that receive volume should be reset
to nominal at the start of each call on handset and headset.
Attribute
voice.volume.persist.handset
voice.volume.persist.headset
voice.volume.persist.handsfree
4.6.1.6.3 Gains <gains/>
The default gain settings have been carefully adjusted to comply with the TIA-810-A
digital telephony standard. Changing these is not advised.
Attribute
voice.gain.rx.analog.handset
70
®
IP SIP
Permitted
Values
10, 20, 30, ...
(multiple of
10)
> jitterBuf-
ferMin,
<=500,
multiple of
10
Permitted
Values
0, 1
0, 1
0, 1
Permitted
Values
-30, -27, -24,
..., 0
Copyright © 2004 Polycom, Inc.
Interpretation
The absolute minimum duration time (in
milliseconds) of RTP packet Rx with no
packet loss between jitter buffer size
shrinks. Use smaller values (1000 ms) to
minimize the delay on known good net-
works. Use larger values to minimize
packet loss on networks with large jitter
(3000 ms).
The largest jitter buffer depth to be sup-
ported (in milliseconds). Jitter above
this size will always cause lost packets.
This parameter should be set to the
smallest possible value that will support
the expected network jitter.
Default Interpretation
0 If set to 1, the receive volume will be
remembered between calls.
0
If set to 0, the receive volume will be
reset to nominal at the start of each
1
call.
TIA-810-A
Default Interpretation
0 Rx analog gain in dB when
using handset.
Optimizing SoundPoint