Jitter - Is the accuracy of the packets showing up in the right order at the destination. Latency (Ping) ms. IP Address: For Attendees: Recommended Minimum Download Speed is > 5Mbps For Presenters/Organizers: Recommended Minimum Download AND Upload Speed is > 5Mbps Latency: 0-150ms = Good, 150-400ms = OK, >400ms = Poor . If using wifi, try moving closer to your wifi router, Make sure that nobody is downloading or uploading large files, or watching movies using the internet connection, Try disabling and enabling wifi on your computer, For users in China or UAE we recommend using a VPN for best performance. The public IP address of the browser conducting the test. The minimum throughput measured throughout the test conducted. Then I can just see which ones are missing. The time it takes to create an initial full connection to the TURN server using UDP. tests the reachability and connectivity to a list of HTTPS or WSS addresses. Low minimum throughput, as well as the high variance between minimum, average and maximum, may indicate a connection that is unstable and jittery. Specifically, since this tests an actual WebRTC session towards Talkdesk. In the case of a VPN or a proxy, what we see in our service is the public IP address of the proxy server itself. The lower the value, the higher the media quality. The test server is located at Digitalocean host in the Frankfurt datacenter. Talkdesk Network Test Tool provides the user with a series of widgets displaying valuable information regarding location and connection details, namely: To proceed with the test, please insert your email and a reason for doing it. Now, we conducts similar measurements with an RTMP player via the Wowza server and a simultaneous test with a WebRTC player using Web Call Server. The Call Quality Widget tests for the actual session quality when connecting a WebRTC session with Talkdesk. To proceed with the test, please insert your email and a reason for doing it. for a few minutes for the most accurate results.
Testing latencies RTMP vs WebRTC. See, everywhere else, your browser uses TCP, which, when a packet fails, it will keep resending it until it works or gives up. By using an external geoIP service, we convert the IP address to a country.
WebRTC sessions prefer sending media over UDP and need low latency to establish real-time sessions. The higher the uplink and downlink values and the lower the jitter values, the better. Mean Opinion Score.
It is performed by using the SCTP protocol relayed via TURN.