In this section:
DSP Pad
This object allows you to specify the characteristics of the DSP packet assembly and disassembly (PAD) resources in the
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. DSP PADs are used for transcoding between different media codecs and/or different packetization times, detecting fax and/or modem tones, interworking DTMF transport modes, and detecting DTMF digits.Audio compression of the following types may be assigned to the above mentioned resources:
- G.711
- G.729A
- G.729A+B (Silence Suppression)
- G.726
A packet outage is the loss of incoming voice (RTP) packets. If a PAD on any server module detects a packet outage that exceeds the PACKET OUTAGE THRESHOLD, a "set" trap is generated after the call is disconnected. The set trap displays a count for the total outage occurrences on the shelf and the slot of the affected module. Ten seconds after the last detected outage, a "clear" trap is generated to indicate that the condition has not occurred for a 10 second interval on the shelf and slot. A counter for the occurrences within the interval is displayed in the clear trap. A total occurrence counter increments with every packet outage that exceeds the threshold on a server. The counter can be reset through PACKET OUTAGE RESET TOTAL COUNTER.
Packet outages cannot be detected if T.38 is used in a call. Calls that use a silence suppression algorithm need to specify a heartbeat of an appropriate interval to detect outages.
Media Performance data
The
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monitors the affects of packet loss and jitter exceeding the jitter buffers capacity using the playout time series.The playout time series consists of 31 quality measurements, with each measurement representing a consecutive time period. Taken as a whole, the measurements represent how the playout buffer viewed the jitter and packet loss over consecutive time periods. Within each time period the quality is classified into four possible values:
- Good
- Acceptable
- Poor
- Unacceptable
Anytime the playout buffer has no data to play due to packet loss or excessive jitter, the
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tracks the duration of this during a time period. The total duration of the missing data during a time period is compared against three programmable thresholds to classify the performance during the period (THRESHOLD0, THRESHOLD1, and THRESHOLD2).The time series provides an approximate indication of the locations (in time) of packet problems for determining call problems due to, for example, a large single- event outage or a continuous series of packet issues distributed throughout the call.
Since the time period is fixed, the duration of the calls affect the number of time period intervals that are used for collecting data. Using a default time period of 20 seconds, a short call of 1-30 seconds only produces data for one or two time periods, whereas a longer call lasting 10 minutes will have data for the last 30 time periods. Calls lasting longer than 31 time periods will have data for the last 31 time periods of the call only (old data is discarded). If you wish to obtain data at a more granular level, you can configure the time period to be shorter, however this precludes you from monitoring longer calls (since only the last 31 time periods are recorded).
Configuring the Playout Time Series Period and Thresholds
To configure the playout time series parameters, you set the thresholds to detect a certain percentage of missing data within a time period.
For example, to configure a 20-second time period where between 1 and 2 percent of missing data is considered Poor quality, and more than 2 percent of missing data is considered Unacceptable:
- Calculate the duration of the percentages of the 20-second period:
- percent of 20 seconds = 0.2 seconds (200msec)
- percent of 20 seconds = 0.4 seconds (400msec)
- Assign these values (in milliseconds)
To View and Edit DSP Pad
On SBC main screen, go to Configuration > Profile Management > Category: DSP Pad > DSP Pad.
The DSP Pad window is displayed.
Profile Management - DSP Pad
The following fields are displayed:
DSP Pad Parameters
Parameter | Description |
|---|---|
| Jitter evaluation period. Time period in which to decide when to periodically evaluate playout occupancy in milliseconds. This parameter determines the rate at which the jitter buffer is adapted. This value should be set in a range that covers somewhere between 0.5 seconds to 2 seconds although you can set it to numbers outside this range. If this number is too small the jitter buffer algorithm may tend to discard samples too aggressively causing small losses of audio. If the number is too large the excess delay built up in the jitter buffer will remain for a long time before it can be removed. The default setting of 1 second is a reasonable compromise. Must be 10-300000, default is 1000. |
| jitter buffer occupancy threshold (in milliseconds). The occupancy below which playout time is advanced if this occupancy has existed for the jitterEvalPeriod. This value is the target occupancy of the buffer assuming the actual network jitter is small enough to reach this number. The occupancy of the jitter buffer over time represents the delay added before audio is played out to the PSTN. This value is used to prevent excess delay from building up in the jitter buffer if the delay is not needed. If the network jitter is small enough the occupancy will gradually be brought down to this level or possibly lower. If you know the expected jitter in your network then you should set this threshold equal to or slightly larger than this jitter in order to have delay. If the actual jitter is higher then some samples may (infrequently) be discarded, depending on the statistics of the signal. If the actual jitter is somewhat smaller then you may have some accumulated delay (less than or equal to this value) in the jitter buffer. This represents the trade-off between maintaining delay and discarding samples. Must be 2- 200 (covering a delay of 2 to 200 milliseconds), default is 20 (milliseconds). Setting this number to 200 will disable jitter buffer adaptation. |
| This integer specifies the RTP payload type to use for DTMF Relay during compressed calls. Must be 96-127, default is 100. When running RFC 2833 with H.323 or SIP signaling, H.323 disallows 0-95. |
| time between silence packets. This integer specifies the time between SID packets, in milliseconds. This ensures that SID packets will not be sent too frequently when the background noise is changing, but instead some amount of compression will still occur. Must be 50-300000, default is 200 (milliseconds). |
| 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 level of bearer traffic flowing for RTCP calculation purposes. This value must exceed sid MinT i me (below). Must be 50-300000, default is 2000 (or 2 seconds). |
| This integer specifies the time after voice is detected inactive before sending a SID packet, in milliseconds. Must be 80-2560, default is 100 (milliseconds). |
| This (positive) integer specifies the noise level below which level any noise is considered to be silence (in dBm0s). Must be -62dBm0 to -24dBm0, default is 60 (or-60 dBm0). |
| This (positive) integer specifies the maximum noise level above which level any noise is considered to be speech (in dBm0s). Must be -62dBm0 to -24dBm0, default is 48 (or -48 dBm0). |
| This (positive) integer specifies the initial estimate to be used for generating comfort noise when the CODEC in the Packet Service Profile is G711 or G711SS. For G711, when no modem has been detected, it represents the level of comfort noise to generate to fill in the audio if packet losses occur; it is played until the first packet is received. For G711SS, it represents the level of comfort noise to generate if no SID is received, whenever there are gaps without packets (due to either packet losses or silence periods). Must be - 90dBm0 to -35dBm0, default is 56 (or -56 dBm0). |
| This positive integer is a percentage that indicates 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. Must be 1-100, default is 90. |
| Specifies whether a trap will be generated when universal compression resources are reduced beyond a threshold value:
|
| Specifies the recording interval size (in milliseconds) used by the Unable to show "metadata-from": No such page "_space_variables" when monitoring RTP playout buffer quality; used only when an RTP stream is terminated. This parameter applies only to the RTP playout buffer in the DSP. It does not apply to the RTP monitoring function in the network processor. This value must be greater than or equal to 10000 (10 seconds) and less than or equal to 240000 (240 seconds). The default is 20000 (20 seconds). |
| Specifies the playout loss time series threshold (in milliseconds) used by the Unable to show "metadata-from": No such page "_space_variables" when quantifying packet loss as applied to the playoutTimeseriesPeriod. 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 is 0 (0.0 seconds, or 0 percent of the playoutTimeseriesPeriod). This parameter is applicable for all channel instances. |
| Specifies the playout loss time series threshold (in milliseconds) used by the Unable to show "metadata-from": No such page "_space_variables" when quantifying packet loss as applied to the playoutTimeseriesPeriod. 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 is 200 (0.2 seconds, or 1 percent of the playoutTimeseriesPeriod). This parameter is applicable for all channel instances. |
| Specifies the playout loss time series threshold (in milliseconds) used by the Unable to show "metadata-from": No such page "_space_variables" when quantifying packet loss as applied to the playoutTimeseriesPeriod. Loss durations greater than Threshold1 and less than Threshold2 are considered Poor. Loss durations greater than Threshold2 are considered Unacceptable. The default is 600 (0.6 seconds, or 3 percent of the playoutTimeseriesPeriod). This parameter is applicable for all channel instances. |
| Percentage Threshold crossing value for tone resources. When this threshold is reached an event will be generated if toneThresholdState is enabled. |
| State of Tone Threshold Event. An event will be generated only if state is enabled. |
| Specifies the 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. The options are:
|
Make the required changes and click Save at the right hand bottom of the panel to save the changes made.
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