Access the Call Quality Monitoring Report

The Call Quality Monitoring Report gives you a single quality rating for a call along with the underlying audio measurements that produced it. You open the report straight from your call history in the Cloud Voice App, which makes it a quick way to find out why a particular conversation sounded rough (choppy audio, dropouts, or delay) instead of guessing.

Note

“Personal Logs” shows only the calls placed and received by the extension you are signed in as. If you need another user’s calls, sign in as that user or use an admin-level report. The report is generated per call, so you pick one call log entry at a time.

Open the report

1. Sign in to the Cloud Voice App and go to Call Logs > Personal Logs.

2. Find the call you want to inspect, click the arrow next to it, and choose Call Quality Monitoring Report from the menu.

   

The report opens in a pop-up window.

Tip

Read the Overall Score and the MOS value first. Together they tell you at a glance whether the call was healthy. If the score is low, work down through the packet loss, jitter, and latency figures to see which one caused it.

Understand the metrics

The report is grouped into three parts: an overall assessment, endpoint details for each party (Caller and Callee), and separate audio measurements for each direction of the call.

PBX stands for Private Branch Exchange, the phone system that switches your calls. Because the two audio directions are measured separately, you can tell which leg of the call was degraded:

• Audio from PBX is the audio your client received (the download direction).
• Audio to PBX is the audio your client sent (the upload direction).

Metric	What it tells you
Overall Score	The combined quality score for the call, with its rating.
Summary	A short, plain-language assessment of how the call performed.
Caller & Callee
Name	The party’s display name and phone number.
Monitoring Duration	How long call quality was measured.
Codec	The audio codec (coder/decoder) negotiated for the call, which sets how the audio is compressed.
Client	The client application that handled the call.
Round Trip Time (ms)	The time, in milliseconds (ms), for an audio packet to travel to the far end and back. Lower is better.
Local Address	The private IP (Internet Protocol) address the client used inside its own network.
Public IP Address	The public IP address the client presented to the internet. When it differs from the Local Address, the client is behind NAT (Network Address Translation, where a router maps private addresses to one public address).
Network Type	The kind of network connection the client was on, such as Wi-Fi or wired.
Audio from PBX & Audio to PBX
MOS	Mean Opinion Score, a 1-to-5 rating of audio quality for that direction. Higher is better: below 3.5 is poor, 3.5 to under 4.0 is moderate, and 4.0 to 5.0 is good.
Packet Loss Rate (%)	The share of audio packets that never arrived. Even a few percent can cause audible gaps.
Jitter Buffer (ms)	The buffer, in milliseconds, used to smooth out variation in packet arrival timing.
Latency (ms)	The one-way network transmission delay, in milliseconds.

Note

These numbers reflect the network path as seen by the client that ran the report. High packet loss or latency usually points to the network (Wi-Fi signal, congestion, or the internet link) rather than the phone system itself, so it is a good first place to look when a user complains about call quality.