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Exists on Condition | /system/sbcPersonality/role !='ssbc' |
Parameter | Presence | Type | Default | Description | |
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jitterEvalPeriod | O | uint32 | 1000 | Time period to decide when to periodically evaluate playout occupancy (in milliseconds). | |
jitterMinOccThsh | O | uint32 | 20 | If occupancy falls below this threshold for some time period we advance the playout time (in milliseconds). | |
rtpDtmfRelay | O | uint32 | 100 | RTP Payload type to be received as DTMF relay during compressed calls. | |
sidMinTime | O | uint32 | 200 | Minimum time (in milliseconds) between silence packets. | |
sidMaxTime | O | uint32 | 2000 | Maximum time (in milliseconds) between silence packets. Must be larger than minimum time between silence packets. | |
sidHangoverTime | O | uint32 | 300 | Minimum time (in milliseconds) after DSP detects speaker going inactive before sending silence packets. | |
sidMinNoiseFloor | O | uint32 | 60 | Minimum noise floor (in -dBm). If energy level is below this it will be considered as silence. Configuration range is between -62dBm -24dBm. | |
sidMaxNoiseFloor | O | uint32 | 48 | Maximum noise floor (in -dBm). If the energy level is above this it will be considered as speech. Configuration range is between -62dBm -24dBm. | |
comfortEnergy | O | uint32 | 56 | Initial estimate (in -dBm) to be used for generating comfort noise. It is used when no silence packets are received to generate comfort noise. Configuration range is between -90dBm and -35dBm. | |
universalCompressionThreshold | O | uint32 | 90 | Percentage Threshold crossing value for Universal Compression resources. When this threshold is reached an event will be generated if universalCompressionThresholdState is enabled. | |
universalCompressionThresholdState | O | enumeration | enabled | State of Universal Compression Threshold Event. An event will be generated only if state is enabled. | |
playoutTimeseriesPeriod | O | uint32 | 20000 | Duration of one period of the RTP playout buffer loss timeseries statistic. This statistic is used only when an RTP stream is terminated (it does not operate in a Packet Monitor channel type). Units: milliseconds (will be converted to 10 msec increments) | |
playoutTimeseriesThreshold0 | O | uint32 | 0 | Threshold #0 for generation of the playout loss timeseries statistic. The first quantization level is <= this threshold. The set of three thresholds specified here are used for all channel instances. Units: milliseconds (will be converted to 10 msec increments) Default: 0 (0.0 sec, 0% of playoutTimeseriesPerdiod) | |
playoutTimeseriesThreshold1 | O | uint32 | 200 | Threshold #1 for generation of the playout loss timeseries statistic. The second quantization level is: Threshold #0 < #Interpolations <= Threshold #1 Units: milliseconds (will be converted to 10 msec increments) Default: 200 (0.2 sec, 1% of playoutTimeseriesPerdiod) | |
playoutTimeseriesThreshold2 | O | uint32 | 600 | Threshold #2 for generation of the playout loss timeseries statistic. The third quantization level is: Threshold #1 < #Interpolations <= Threshold #2 Units: milliseconds (will be converted to 10 msec increments) Default: 600 (0.6 sec, 3% of playoutTimeseriesPerdiod) | |
toneThreshold | O | uint32 | 90 | Percentage Threshold crossing value for tone resources. When this threshold is reached an event will be generated if toneThresholdState is enabled. | |
toneThresholdState | O | enumeration | enabled | State of Tone Threshold Event. An event will be generated only if state is enabled. | |
audioTxDuring2833 | O | enumeration | enabled | State of Audio Transmit During 2833. Parameter that allows the user to inhibit the transmission of audio packets during the period lasting from the start of a transmitted RFC4733 event to the end of the transmitted RFC4733 event. |
REST API: GET Example |
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curl -kisu 'admin:secret' -X GET https://{SBX-SERVER}/api/config/system/dspPad |