If your organization is experiencing call quality problems with Google Voice, it might be caused by disruptions on your network that result in latency, jitter or packet loss. These issues indicate network quality issues that impact voice calls over the internet.
To determine if call quality is being impacted by network disruptions, use the security investigation tool to search Voice log events for latency (RTT), jitter, and packet loss events.
View latency, jitter and packet loss for Voice calls
-
Sign in to your Google Admin console.
Sign in using your administrator account (does not end in @gmail.com).
- On the left of the Admin console Home page, click Security
Security center
- Click Data source and select Voice log events.
- Click Add Condition and select Event from Attribute drop-down list.
- Select Is from the list of operators.
- Under Event, select Network Statistics (client).
(Optional) To add more search conditions, repeat steps 3–5. For example, to view specific calls, filter by attributes such as Actors (users), Call Destination or Call Source. - Click Search.
To understand the values, see Recommendations for Voice call quality.
About Network Statistics events
Network Statistics events are shown for applicable calls when the information is available and are not the only indicators of potential call quality problems.
For example, “Network Statistics (client)” events are available only for the call leg between the Voice client and Google media servers. The values are measured on the link from Voice client to Google media servers. Events for the call leg between Google media servers and Public Switched Telephone networks (PSTN) are not available. Calls using a forwarding number on a mobile network also don’t produce Network Statistics events. For calls due to Ring Group, the values are available in the “Network Statistics (client)” event, which is separate from Call Received (Ring Group) and Call Transferred (Ring Group) events.
Review Fix problems with Voice calls if call quality problems continue.
Recommendations for Voice call quality
| Network Statistics Event | RTT Mean (ms) | Receive Jitter Mean (ms) | Receive Jitter Max (ms) | Receive Packet loss per 1000 | ANQ (Audio network quality) |
|---|---|---|---|---|---|
| Description | Average time it takes to ping the nearest data center and get a response back | Average variation in the delay of received packets during a call | Max variation in the delay of received packets during a call | Packets lost while traveling on the network |
Percentage of call duration that has good network quality. (No audible artifacts are heard by the participants in the call) |
| Recommended value | Below 100 ms | Below 30 ms | Should not differ greatly from average Jitter | As close to zero as possible | As close to 100% as possible |
| Cause | Speed and quality of internet connection | Network congestion | Network congestion | Poor WiFi and network congestion |
Poor WiFi and/or network congestion |
| Solution | Avoid using networks, proxies, VPNs, or VDI. Find faster or more stable internet providers if possible. | Disable deep packet inspection firewalls if applicable. Increase network bandwidth to allow more traffic during peak activity. | Disable deep packet inspection firewalls if applicable. Increase network bandwidth to allow more traffic during peak activity. | Increase network bandwidth to allow more traffic during peak activity. |
See suggestion solutions for Receive Jitter Mean and Receive Packet loss per 1000. |
