139
edits
Media/WebRTC/WebRTC Debugging: Difference between revisions
Fix wrapping
(Temp adding peerconnection expand image) |
(Fix wrapping) |
||
| Line 28: | Line 28: | ||
The key metrics in [about:webrtc] are RTT (round-trip-time) and jitter. They can be found in the RTP stats section of the PeerConnection. The PeerConnection informational blocks start out in a minimized state, and one will need to expand a block to find the RTP stats section: | The key metrics in [about:webrtc] are RTT (round-trip-time) and jitter. They can be found in the RTP stats section of the PeerConnection. The PeerConnection informational blocks start out in a minimized state, and one will need to expand a block to find the RTP stats section: | ||
[[File:About webrtc reveal PeerConnection stats.png|800px|border|How to reveal the full statistics of a PeerConnection in about:webrtc]] | [[File:About webrtc reveal PeerConnection stats.png|800px|border|How to reveal the full statistics of a PeerConnection in about:webrtc]] | ||
Once expanded one can locate the RTP Stats section and find the key diagnostic stats: | Once expanded one can locate the RTP Stats section and find the key diagnostic stats: | ||
[[File:About webrtc Jitter and RTT.png|800px|border|Location in about:webrtc of jitter and RTT stats]] | [[File:About webrtc Jitter and RTT.png|800px|border|Location in about:webrtc of jitter and RTT stats]] | ||