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The Video P2P Widget simulates a real-time WebRTC peer-to-peer session to measure the accuracy and quality of both incoming and outgoing video traffic.

By connecting the user's machine to a managed cloud browser, this test provides a highly accurate estimation of how a device handles live video calls, even in scenarios where the user only receives video (e.g., no webcam present).

Data Collected and Shared

Metric Path Description
Estimated Out only The bandwidth estimation WebRTC has for the outgoing direction.
Bitrate In + Out The actual bitrate of video data currently traveling over the connection.
Downlink In only The round trip time (RTT) reported on incoming traffic. Lower is better.
Jitter In + Out The percentage of packet loss observed in each direction. Lower is better.

Key Observations for Troubleshooting

  • Latency and Stability: Focus on keeping Round Trip Time and Packet Loss as low as possible to prevent "stuttering."
  • Volume: High values for Bitrate and Estimated bandwidth are ideal.
  • Camera Factors: If outgoing bitrate is low, check if the video content is static or low-resolution (e.g., a physical privacy slider is covering the lens).

Pro-Tip: High packet loss (Jitter) often correlates with Wi-Fi interference. If Jitter is red, advise the user to move closer to the router or switch to a wired Ethernet connection.