Voice Quality Analysis

Real-time Voice Quality Analysis for every Call

Palladion supports the analysis of RTP packet traffic at the tap points for identifying Voice Quality problems first hand and for measuring the Audio Quality for each and every call. It can measure Jitter, Loss, and Late Loss for the Subscriber's data stream and estimate important Voice Quality factors such as Delay impairment and Effective Equipment Impairment according to ITU-T Recommendation G.107 for determining a Voice Quality Rate or the degradation of the Mean Opinion Score. (R factor and MOS)

In addition if the SIP phones support the reporting of call quality parameters in the SIP BYE message, Palladion can also report these.

Palladion collects and presents all that data. The thresholds for all reported Voice Quality parameters can be configured through the web based Operator interface.

In the screen shot shown above the statistics of interest are Jitter Times, Calls with Packet Loss, and Calls with Burst Loss, Ieeff (Equipment, i.e. codec degradation factor), R Factor and Mean Opinion Score The bar chart is green when the Voice Quality is good, yellow with mediocre, and red with bad reported Voice Quality Parameters. Also shown is the source of the statistics for that call. In the case shown above the X-RTP-STAT heading means it comes from a BYE message sent by the User agent involved in the call.

The period over which Call Quality is displayed can be set using the web GUI slider controls.