Open Bug 868626 (webrtc-telemetry) Opened 7 years ago Updated 6 years ago
[meta] Collect Telemetry data for Web
We'd like to collect a number of bits of data via telemetry about Webrtc: Bandwidth available and variability in it Video resolution used Frame rate Losses/re-transmissions Delay A/V sync quality AEC quality (ERLE/ERL/etc) ICE information CPU use during call # of calls Duration of calls User-supplied call quality metrics ala Skype others?
also good to capture is jitter buffer depth statistics. E.g. how many people are getting 500ms of jitter buffer added delay. Likewise, tracking jitter buffer drop and insert events (how much degradation do we have in just modulating the buffer depth) .
This bug appears to call for the collection of these statistics in the aggregate. It could be much more interesting (from a scientific perspective) if these metrics could be associated with some idea of where the endpoints of the call are on the network. For example, the endpoint IP addresses of the call, aggregated to at least /24 granularity. Obviously, there are privacy concerns here, but I wanted to raise the question to see if there were some level of aggregation that might make this acceptable. In a related vein, you might want to expand on "ICE information" a little, since there's likely some privacy concerns lurking in there as well.
greg: The NetEQ jitter buffer is a fully adaptive jitter buffer, and as such doesn't have discrete grow-by-a-packet/drop-a-packet-to-shrink events. You can measure underflows and overflows, and depth sampled at a moment and average or histogram of depth.
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