Troubleshoot Google Voice call quality

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 

1. Sign in with an administrator account to the Google Admin console.

   If you aren’t using an administrator account, you can’t access the Admin console.

2. On the left of the Admin console Home page, click Security Security center Investigation tool.
3. Click  Data sourceand select Voice log events.
4. Click  Add Condition and select Eventfrom Attributedrop-down list.
5. Select Isfrom the list of operators.
6. 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 Destinationor Call Source.
7. 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.

Related topics

• About the security investigation tool
• Voice log events