For each NAM Probe, you can set global thresholds determining the call quality metrics values triggering the reporting of bad calls
Open NAM Console ► Deployment ► Manage devices, NAM Probe Configuration ► Open configuration, and navigate to Global ► Other Protocols Monitoring ► VoIP.
You can fine tune the VoIP monitoring in cases when the NAM Probe is not able to provide the full scope of data due to call inactivity or missing data in the monitored traffic.
Call Inactivity Timeout
The timeout to terminate the call due to inactivity.
Default Delay to Subscriber
Default Delay to Remote
The default delay value is used when the NAM Probe is not able to retrieve the actual delay from the RTCP stream.
The VoIP monitoring requires that the NAM Probe can "see" the two data streams:
- RTP/RTCP carrying the audio call details. NAM Probe understands the following audio codecs: G.711 (PCMU), G.726, GSM, G.723.1 (ACELP), G.723.1 (MP-MLQ), LPC, G.711 (PCMA), G.722-64, G.729.
- VoIP Signaling (SIP or H.323) carrying the control/signaling details.
You can set the thresholds for the following metrics:
VoIP average Mean Opinion Score (MOS) rating of the call quality, for both downstream and upstream traffic.
VoIP average R-factor value, for both downstream and upstream traffic. It is a transmission quality rating, with a typical range of 50-100. An R-Factor score is derived from multiple VoIP metrics, including latency, jitter, and loss.
VoIP loss rate
The percentage of VoIP packets lost or discarded that needed to be retransmitted, measured for both upstream and downstream traffic.
VoIP average jitter measured by the probe, for both downstream and upstream traffic. Jitter is a variation in voice data transit delay, in milliseconds. In general, higher levels of jitter are more likely to occur on either slow or heavily congested links.
VoIP average networking delay, as reported by Real Time Transport Protocol (RTCP), measured for both downstream and upstream traffic.