Page History

audioTxDuring2833

Use this flag to specify the Audio Transmit state during RFC2833.

• disabled
• enabled (default)

comfortEnergy

-90 to -35 (dBm0)

The initial estimate used for generating comfort noise when the CODEC in the Packet Service Profile is G.711 or G.711SS. For G.711, when no modem has been detected, this represents the level of comfort noise to fill in the audio when packet loss occurs. This plays until the first packet is received. Whenever there is a drop in packets (packet loss or silence periods), the G.711SS represents the level of comfort noise when no SID is received. (default = 56 [which represents -56 dBm0]).

jitterEvalPeriod

10-300000

Time period (in milliseconds) to decide when to periodically evaluate playout occupancy. This parameter determines the rate at which the jitter buffer is adapted.
If the value is too small, then the jitter buffer algorithm may tend to discard samples too aggressively causing small losses of audio. If the value is too large, then the excess delay built up in the jitter buffer will remain for a long time before it can be removed. (default = 1000).

jitterMinOccThsh

2-200

The minimum jitter buffer occupancy threshold (in milliseconds) below which the playout time advances if the occupancy has existed for the jitterEvalPeriod.

This value should be target minimum occupancy of the buffer, assuming the actual network jitter is small enough to reach this number.

The minimum occupancy of the jitter buffer over time represents the delay added before audio is played out to the PSTN. The value is used to prevent excess delay from building up in the jitter buffer. If the network jitter is small enough, occupancy will gradually be brought down to this level or possibly lower. The expected jitter threshold should be set to equal or slightly larger than the jitter to obtain minimum delay. If the actual jitter is higher, then some samples may (infrequent) be discarded, depending on the statistics of the signal. If the actual jitter is somewhat smaller, then some accumulated delay (less than or equal to this value will be formed) in the jitter buffer. This represents the trade-off between maintaining minimum delay and discarding samples. (default = 20). Setting this number to 200 disables jitter buffer adaptation.

playoutTimeseriesPeriod

10000-240000

Specifies the recording interval size (in milliseconds) used by the 

playoutTimeseriesThreshold0

N/A

Specifies the playout loss time series threshold (in milliseconds) used by the 

Loss durations less than or equal to Threshold0 are considered Good. Loss durations greater than Threshold0 and less than Threshold1 are considered Acceptable. The default value is "0", or 0 percent of the playoutTimeseriesPeriod.

This parameter is applicable for all channel instances.

playoutTimeseriesThreshold1

N/A

Specifies the playout loss time series threshold (in milliseconds) used by the 

Loss durations greater than Threshold0 and less than Threshold1 are considered Acceptable. Loss durations greater than Threshold1 and less than Threshold2 are considered Poor. The default value is "200" (0.2 seconds, or 1 percent of the playoutTimeseriesPeriod).

This parameter is applicable for all channel instances.

playoutTimeseriesThreshold2

N/A

Specifies the playout loss time series threshold (in milliseconds) used by the 

Loss durations greater than Threshold1 and less than Threshold2 are considered Poor. Loss durations greater than Threshold2 are considered Unacceptable. The default value is "600" (0.6 seconds, or 3 percent of the playoutTimeseriesPeriod).

This parameter is applicable for all channel instances.

rtpDtmfRelay

0-127

This integer specifies the RTP payload type to use for DTMF Relay during compressed calls. When running RFC 2833 with H.323 or SIP signaling, H.323 disallows 0-95. (default = 100).

sidHangoverTime

80-2560

This integer specifies the minimum time (in milliseconds), after voice is detected inactive before sending a SID packet. (default = 300).

sidMaxNoiseFloor

24-62

This (positive) integer specifies the maximum noise level; above this noise level is considered to be speech (in dBm0s). (default = 48 [which represents -48 dBm0]).

sidMaxTime

50-300000

This integer specifies the maximum time between SID packets (in milliseconds). If SID HEARTBEAT in the Packet Service Profile is enabled, the SID packets will be sent during silence intervals lasting longer than the value specified by this parameter. These packets can be used to keep a minimum level of bearer traffic flowing for RTCP calculation purpose. This value must exceed the sidMinTime (below). (default = 2000, or 2 seconds).

sidMinNoiseFloor

24-62

This (positive) integer specifies the minimum noise level; below this noise level is considered to be silence (in dBm0s). (default = 60 [which represents -60 dBm0]).

sidMinTime

50-300000

This integer specifies the minimum time between SID packets, in milliseconds. This ensures that SID packets are not sent too frequently when the background noise is changing, but instead some minimum amount of compression still occurs. (default = 200).

toneThreshold

1-100

The percentage of threshold crossing value for tone resources. (default = 90).

toneThresholdState

N/A

The state of the tone threshold event.

• disabled
• enabled (default)

universalCompressionThreshold

1-100

This positive integer specifies the percentage of usage the usage threshold for universal compression resources in the node. When this usage level or threshold is reached, an event (and possibly a trap) will be generated. (default = 90).

universalCompressionThresholdState

N/A

When this flag is enabled, a trap is generated when universal compression resources are reduced beyond the configurable universalCompressionThreshold value.

• disabled
• enabled (default)