Run the Genesys Cloud WebRTC Diagnostics app
The Genesys Cloud WebRTC Diagnostics app provides you with a set of diagnostics that verifies your WebRTC configuration is properly configured and identifies potential problems.
- You must log in to Genesys Cloud to run the WebRTC Diagnostics app.
- You must have your voicemail set up for the WebRTC Phone Test to work properly.
- If you have recently used your phone, you’ll need to disconnect the persistent connection before you run the diagnostic. For more information, see Terminate a persistent connection on a Genesys Cloud WebRTC phone.
- Access the Genesys Cloud WebRTC Diagnostics app for your region:
|Region||Region-specific Genesys Cloud WebRTC Diagnostics app|
|US East (N. Virginia)||https://apps.mypurecloud.com/webrtc-troubleshooter/|
|US East (Ohio)||https://apps.us-gov-pure.cloud/webrtc-troubleshooter/|
|US West (Oregon)||https://apps.usw2.pure.cloud/webrtc-troubleshooter/|
|Canada (Canada Central)||https://apps.cac1.pure.cloud/webrtc-troubleshooter/|
|South America (Sao Paulo)||https://apps.sae1.pure.cloud/webrtc-troubleshooter/|
|Asia Pacific (Seoul)||https://apps.apne2.pure.cloud/webrtc-troubleshooter/|
|Asia Pacific (Sydney)||https://apps.mypurecloud.com.au/webrtc-troubleshooter/|
|Asia Pacific (Tokyo)||https://apps.mypurecloud.jp/webrtc-troubleshooter/|
|Asia Pacific (Mumbai)||https://apps.aps1.pure.cloud/webrtc-troubleshooter/|
- Select one of the available tests:
WebRTC Phone Test
This WebRTC Phone Test verifies that your network connection is properly set up for WebRTC and also verifies that audio is functional.
When you click Start Tests, the connection is verified and a series of tests are run.
- Streaming Connection: Verify that a streaming connection is established.
- WebRTC Station: Verify that the WebRTC station is identified.
- Call Connected: Verify that the call is connected.
- Call Quality: Provides an estimation of the quality of the call based on various network conditions.
To see the test results, click Test Results.
- Mean Opinion Score: MOS (Mean Opinion Score) is a measurement of the voice quality of an interaction. The calculation of MOS uses an industry standard measurement methodology to rank audio quality from 1 (unacceptable) to 5 (excellent).
- Packet Loss: Packet loss has a serious effect on call quality. The packet loss statistic should be less than 1 percent.
- Average RTT: The average RTT (round-trip time) measures the latency between your Genesys Cloud AWS region and your location. The round-trip average time metric should be below 150ms.
- Jitter: Jitter is a variation between the time when packets are sent and when they are received. The jitter stat should be less than 30ms.
The Network Test verifies that your network can access the Genesys media servers. More specifically, these tests confirm that DNS resolution is working and that you are able to connect to the public AWS pool and to the new Genesys Cloud cloud media services /20 CIDR IP range.
If you encounter errors when running the test, check your firewall settings against the recommended settings found in the About ports and services for your firewall article. If you still have problems, contact Customer Care.