In this section:

DSP Pad

Use this screen to specify the characteristics of the DSP packet assembly and disassembly (PAD) resources in the SBC. 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 SBC 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

Whenever the playout buffer has no data to play due to packet loss or excessive jitter, the SBC 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:

  1. Calculate the duration of the percentages of the 20-second period:
    1. percent of 20 seconds = 0.2 seconds (200msec)
    2. percent of 20 seconds = 0.4 seconds (400msec)
  2. Assign these values (in milliseconds)

To View and Edit DSP Pad

On SBC main screen, choose a path:

  • Configuration > Profile Management > Category: DSP Pad > DSP Pad
  • All > System > DSP Pad > DSP Pad

The DSP Pad window is displayed.


DSP Pad Parameters:

Parameter

Length/Range

Description

Jitter Eval
Period

10-300000

The Jitter evaluation period during which to decide when to periodically evaluate playout occupancy (in milliseconds).

This parameter determines the rate at which the jitter buffer is adapted. Set this value to a range that covers somewhere between 0.5 seconds to 2 seconds; although, you can set it to numbers outside this range. Using a setting of 1 second is a reasonable compromise.

  • 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 value is 1000 ms.

Jitter Min Occ
Thsh

2-200

The Jitter buffer occupancy threshold (in milliseconds) below which playout time is advanced if this occupancy has existed for the Jitter Eval Period.

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, 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) get discarded, depending on the statistics of the signal.
  • If the actual jitter is somewhat smaller, then you may experience 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.

The default value is 20 ms. Setting this number to 200 will disable jitter buffer adaptation.

RTP DTMF Relay

96-127

This integer specifies the RTP payload type to use for DTMF Relay during compressed calls.

The default value is 100. When running RFC 2833 with H.323 or SIP signaling, H.323 disallows payload types 0-95.

Sid Min Time

50-300000

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

The default value is 200 ms.

Sid Max Time

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 level of bearer traffic flowing for RTCP calculation purposes. This value must exceed sid MinT i me (below). The default value is 2000 ms (i.e., 2 seconds).

Sid Hangover Time

80-2560

This integer specifies the time after voice is detected inactive before sending a SID packet, in milliseconds. The default value is 100 ms

Sid Min Noise
Floor

24-62

This (positive) integer specifies the noise level below which level any noise is considered to be silence (in dBm0s). The default value is 60 (or-60 dBm0). The configuration range is between -62dBm0 and -24dBm0.

Sid Max Noise
Floor

24-62

This (positive) integer specifies the maximum noise level above which level any noise is considered to be speech (in dBm0s). The default is 48 (or -48 dBm0). The configuration range is between -62dBm0 to -24dBm0.

Comfort Energy

35-90

This (positive) integer specifies the initial estimate to use for generating comfort noise when the CODEC in the Packet Service Profile is G711 or G711SS.

  • For G711, when no modem is 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).

The configuration range is 90dBm0 to -35dBm0. The default value is 56 (or -56 dBm0).

Universal Compression Threshold

1-100

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) is generated. The default value is 90.

Universal Compression Threshold State

N/A

Enable this flag to generate a trap when universal compression resources are reduced beyond the Universal Compression Threshold value.

  • Disabled
  • Enabled (default)

Playout Timeseries Period

10000-240000

(10-240 seconds)

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

The default value is 20000 (20 seconds).

Playout Timeseries Threshold0

N/A

This  parameter is applicable for all channel instances.

Specify the playout loss time series threshold (in milliseconds) used by the SBC when quantifying packet loss as applied to the Playout Timeseries Period.

  • 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 (0.0 seconds, or 0 percent of the Playout Timeseries Period). 

Playout Timeseries Thershold1

N/A

This parameter is applicable for all channel instances.

Specify the playout loss time series threshold (in milliseconds) used by the SBC when quantifying packet loss as applied to the Playout Timeseries Period.

  • 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 Playout Timeseries Period). 

Playout Timeseries Threshold2

N/A

This parameter is applicable for all channel instances.

Specify the playout loss time series threshold (in milliseconds) used by the SBC when quantifying packet loss as applied to the Playout Timeseries Period.

  • 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 Playout Timeseries Period). 

Tone Threshold

1-100

The Percentage Threshold crossing value for tone resources. When this threshold is reached, an event is generated if Tone Threshold State is enabled.

The default value is 90.

Tone Threshold
State

N/A

The Tone Threshold event state. An event is only generated when this flag is enabled.

  • Disabled
  • Enabled (default)

Audio Tx
During2833

N/A

The state of Audio Transmit During 2833. This flag allows you to inhibit the transmission of audio packets during the period lasting from the start to the end of a transmitting an RFC4733 event.

  • Disabled
  • Enabled (default)
High Priority Compression Reservation0-100

The percentage of compression resources reserved for high priority calls.

The default value is 0.

High Priority Tone Reservation0-100

The percentage of tone resources reserved for high priority calls.

The default value is 0.

Make the required changes and click Save.