3c4e3de286
We should be able to capture WebRTC stats in some form for post-processing
so that it helps on debugging support requests (and other use cases, e.g.:
improving field trial analysis on test servers).
Although much of WebRTC stats information can be gathered via server side
components, none have logs as structured for proper post-processing as
the client logs - so we're going the client route for now.
Capture WebRTC stats information for audio and screen sharing via:
- Audio logCodes: new `stats` extraInfo field
- `audio_joined`
- `audio_failure`
- `sfuaudio_error_retry_through_relay`
- `sfuaudio_error_try_to_reconnect`
- Screen share logCodes: new `stats` extraInfo field
- screenshare_presenter_start_success
- screenshare_viewer_start_success
- screenshare_broker_failure
Additionally, add an option to periodically capture WebRTC stats information
for all relevant peers. This is disabled by default since the log can be
verbose (and, consequentially, network taxing when using external
logging targets). It can be enabled via `public.stats.logMediaStats` in
settings.yml. The default interval is 30s. The periodic log format is as
follows:
- logCode: `mediaStats`
- extraInfo.stats: an aggregated stats object of all peers (equivalent
to the `Copy` function in the Connection Status modal).
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| .. | ||
| base.js | ||
| bridge-whitelist.js | ||
| service.js | ||
| sfu-audio-bridge.js | ||
| sip.js | ||