Signaling System 7 (SS7)

Configuring an SS7 signaling gateway

Element

Duration

Description

Duration of the tone in milliseconds. This value can range from 0 to 2631. If the duration is 0, a tone will be played continuously until it is stopped by a second STN command.

For example, the following string defines a 1004hz tone at 22748 amplitude for 1 second:

"1004, 0, 22748, 1000"

The following string defines a dual tone with frequency 697Hz and 1477Hz at 14567 amplitude for 2 seconds:

"697, 1477, 14567, 2000"

The following string defines a 2050, -3dBm0 tone played continuously until stopped by a second STN message:

"2050, 0, -3, 0"

Reporting VoIP call statistics

A TAOS unit operating as a network access server (NAS) with a signaling gateway can report VoIP call statistics in the output of the NAS messaging interface. IPDC VoIP call statistics are reported once a call is cleared. The source that originates call clearing can be either the signaling gateway or the TAOS unit.

When the unit reports VoIP statistics

IPDC 0.12 statistics tags are reported when the signaling gateway or the TAOS unit clears calls under the following conditions:

When the access server initiates a call teardown using an RCR message.

For packet-based calls when the access server acknowledges a call teardown using an ACR message

The TAOS unit reports the following VoIP statistics, as defined by IPDC 0.12:

Number of Real-Time Protocol (RTP) audio packets sent and received by the TAOS unit.

Number of RTP audio packets that failed to reach the TAOS unit as determined by missed sequence numbers.

Number of audio bytes in the RTP payload sent by the TAOS unit.

Number of audio bytes received in the RTP payload that failed to reach the TAOS unit. Because the number of bytes per packet is variable, this value can only be estimated, based upon an average packet size multiplied by the number of nonreceived packets. This value can also be estimated by the control server with the information supplied.

Number of RTP audio packets received.

Number of audio bytes received in the RTP payload.

Estimated interarrival jitter (in milliseconds) Interarrival jitter is an estimate of the statistical variance among the arrival times of RTP packets, which is equivalent to the difference in their relative transit times. Relative transit time is the difference between a packet’s RTP timestamp at the sender and the receiver’s clock at the time of arrival.

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APX 8000/MAX TNT/DSLTNT Physical Interface Configuration Guide

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Lucent Technologies 7820-0802-003 manual Reporting VoIP call statistics, When the unit reports VoIP statistics