Administrator’s Guide for SIP-T5 Series Smart Media Phones
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Enables or disables the IP phone to periodically (every 5 seconds) send RTCP-XR packets to
another participating phone during a call for call quality monitoring and diagnosing.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter “voice.rtcp_xr.enable” is set to 1 (Enabled).
If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
None
Phone User Interface:
None
VQ-RTCPXR
The VQ-RTCPXR mechanism, compliant with RFC 6035, sends the service quality metric reports
contained in SIP PUBLISH messages to the central report collector. Three types of quality reports
can be enabled:
Session: Generated at the end of a call.
Interval: Generated during a call at a configurable period.
Alert: Generated when the call quality degrades below a configurable threshold.
A wide range of performance metrics are generated in the following three ways:
Based on current values, such as jitter, jitter buffer max and round trip delay.
Covers the time period from the beginning of the call until the report is sent, such as
network packet loss.
Computed using other metrics as input, such as listening Mean Opinion Score (MOS-LQ)
and conversational Mean Opinion Score (MOS-CQ).
To operate with central report collector, IP phones must be configured to forward their voice
quality reports to the specified report collector. You can specify the report collector on a
per-line basis.
Users can check the voice quality data of the last call via web user interface or phone user
interface. Users can also specify the options of the RTP status to be displayed on the phone user
interface. Options of the RTP status to be displayed on the web user interface cannot be
specified.
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