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Media/WebRTC/WebRTC Debugging: Difference between revisions
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Adding about:webrtc tour TODO
(Delay remedy) |
(Adding about:webrtc tour TODO) |
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# Add this file as an attachment to your bug.<br /> | # Add this file as an attachment to your bug.<br /> | ||
This file contains statistics about your call, the signalling that was used to setup your call, and information about the network transports. | This file contains statistics about your call, the signalling that was used to setup your call, and information about the network transports. | ||
===about:webrtc=== | |||
====Tour of About Webrtc==== | |||
This section will contain screen shots of about:webrtc and description of each of the sections. This information needs to be referenced from multiple places later in the wikipage. | |||
<syntaxhighlight lang="sh">TODO</syntaxhighlight> | |||
===Common Call Quality Issues=== | ===Common Call Quality Issues=== | ||
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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 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 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]] | ||