High-quality audio calls with Google Voice require a properly configured network. Verify that your network is ready for voice traffic:
- Make sure your network has enough bandwidth to handle all concurrent calls.
- Open the outbound ports to allow voice traffic to flow to and from your network.
- Whitelist the URIs used by Voice for web traffic, APIs, feedback reports, logs upload, and connectivity setup.
Outbound ports need to allow voice traffic
Set up your network so the following ports allow media traffic to flow to and from your organization:
- Outbound UDP ports range: 19302–19309
- Outbound TCP port 443
Voice port range 19302–19309 uses the Chrome WebRTC UDP Ports Setting.
Bandwidth requirements per site
A site’s network should have enough bandwidth for all concurrent voice calls, plus extra bandwidth for other needs.
To calculate the optimal bandwidth requirement range, multiply the amount of bandwidth required for each voice call by the peak number of concurrent participants.
Average bandwidth per participant * peak number of concurrent participants = Minimum site bandwidth requirements
Bandwidth recommendation per participant
- Minimum requirement: 32 kbps (to account for actual media and RTCP signaling overhead)
- (Recommended) Good quality: Above 50 kbps (to account for actual media and RTCP signaling overhead)
- Calls between two Google Voice clients will attempt to adapt to lower bitrates where possible
Make sure there's full network access to core Google services. If your network has restrictions or filtering policies for users, make sure there’s network access to the following URI patterns.
These patterns are used by Voice for web traffic, APIs, feedback reports, logs upload, and connectivity setup.
Polycom desk phones require this network access:
|Trusted IP Address||UDP||TCP||TLS|