webrtc-stats
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w3c
WebRTC Statistics
when using chrome to make a call on https://webrtc.github.io/samples/src/content/peerconnection/constraints/ the `inbound-rtp` codecId stat is pointing to RTCCodec_1_Inbound_96 which resolves to VP8. However, Chrome actually wraps this in RED (historically, see...
Currently, `framesPerSecond` ([RTCInboundRtpStreamStats](https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framespersecond), [RTCOutboundRtpStreamStats](https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framespersecond) , [RTCVideoSourceStats](https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats-framespersecond)) is defined as the number of encoded frames during the last second. The problem with the current definition is that it is too restrictive...
webrtc-stats [sdpFmtpLine](https://w3c.github.io/webrtc-stats/#dom-rtccodecstats-sdpfmtpline) says: _"The a=fmtp line in the SDP corresponding to the codec, i.e., after the colon following the PT. This defined by [JSEP] in Section 5.7."_ This isn't very...
Some statistics, such as `freezeCount`, `totalFreezeDuration`, `pauseCount` and `totalPauseDuration` are only available from a native level or through *webrtc-internals*. Those are documented as [webrtc-provisional-stats](https://github.com/w3c/webrtc-provisional-stats) and seem to be implemented some...
The stats collected by this API enable two new privacy harms / risks. This spec should enable the main uses of WebRTC, without automatically exposing these additional risks. a) Leaking...
In the webrtc-pc spec, it is clear that the timestamp on a stats record is based on Unix time (https://w3c.github.io/webrtc-pc/#dom-rtcstats-timestamp). It is also mentioned elsewhere (in CSRC) that the value...
The definition says that it is "calculated as defined in [RFC3550] section 6.4.1", though I think it means "... the cumulative number of RTP payload bytes received, calculated in the...
i'm currently adding an alias for chrome/webrtc's `ip` field on candidate statistics. However, one of the WPT failed because for prflx addresses the field is an empty string instead of...
[sdpFmtpLine is defined](https://w3c.github.io/webrtc-stats/#dom-rtccodecstats-sdpfmtpline) as > The a=fmtp line in the SDP corresponding to the codec, i.e., after the colon following the PT. This defined by [JSEP] in Section 5.7. at...
End-to-end delay refers to the time it takes between the capture of a video frame or audio sample and the playout of that frame or sample at another endpoint. This...
