webrtc-stats
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WebRTC Statistics
as described in RFC 5245 15.1 If a candidate is server *or* peer reflexive, and are equal to the base for that server or peer reflexive candidate *** Preview |...
This issue is _not_ arguing to change when `inbound-rtp` is created, because in that case the SSRC is in theory not known until the first packet is received, and we...
Sender-side [packetsReceived](https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceived) is unsigned, yet is a result of a subtraction of two RR values that could be garbage: Having stats reflect garbage input seems helpful for debugging, but it's...
RTCVideoSourceStats contains [frame counters](https://w3c.github.io/webrtc-stats/#videosourcestats-dict*) as "frames originating from this source". But the spec is not clear about which point in the pipeline this is measured. If you measure this as...
In the [W3C Identifiers for WebRTC's Statistics API](https://www.w3.org/TR/webrtc-stats/) spec, the logic for marking a video freeze, is defined as follows(see [freezeCount](https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-freezecount)): 1. Calculate the linear average of the past 30...
We already have video-related metrics like [freezeCount](https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-freezecount) and [totalFreezesDuration](https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalfreezesduration), but audio-related metrics are missing, although they have been [implemented in libwebrtc](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/stats/rtcstats_objects.h;l=359;drc=1ab61886a9b1bb6c2307653d4e7d66e593715ed1) please consider exposing the audio-related metrics to Javascript Developers.
https://w3c.github.io/webrtc-stats/#example-of-a-stats-application uses variable names that refer to both RTCStats (`report`) and RTCStatsReport (`currentReport`) as reports. Clearly one is a report and one is a stats object. The former is clearly...
[§ 2. Conformance](https://w3c.github.io/webrtc-stats/#conformance) says: _"This specification does not define what objects a conforming implementation should generate. Specifications that refer to this specification have the need to [SIC] specify conformance."_ Typo...
Related: #673 `RTCOutboundRtpStreamStats.scalabilityMode` represents the `scalabilityMode` value set for the RTP stream, and it provides useful information for the SVC stream. However, the SVC stream may have a different temporal/spatial...
For example tests like "this counter should increase over time" or "the two sides seem to roughly agree on the state of things"