In BBB 3.0, a change was made to collect full WebRTC stats continuously.
This method gathers stats from *all* peers and *all* senders and receivers
every 2 seconds. Originally, it was intended to run only when the user opened
the connection status dialog, providing in-depth info in the UI and making it
available for copying.
This new behavior is not ideal. Running full stats collection every 2 seconds
in meetings with 20+ peers/transceivers wastes client resources since the
collected data is unused 99% of the time.
This commit reverts to the pre-3.0 behavior (≤2.7), where full stats collection
(`startNetworkMonitoring`) runs only when the connection status modal is open.
As a bonus, it fixes the packet loss status transition log to use the packet
loss percentage, which is the actual trigger metric.
Adjust an inline comment in connection status' service about packet loss metric
usage.
Now it correctly states that the absolute counter SHOULD NOT be used for
alert triggers.
In 3.0, the packet loss metric used to trigger connection status alerts was
changed to the one generated by the `startMonitoringNetwork` method used by the
connection status modal. Since packet loss thresholds were not adjusted (0.5,
0.1, 0.2), a single lost packet causes the status alert to be permanently
stuck on "critical". This is explained by how different those metrics
are:
- **Before (2.7):** A 5-probe wide calculation of inbound packet loss
fraction based on `packetsLost` and `packetsReceived` metrics.
- **Now (3.0):** An absolute counter of inbound lost packets.
This commit restores the previous packet loss metric used to trigger
connection status alerts, reverting to the original collection method via
`/utils/stats.js`. This resolves the issue, but further work is needed in
subsequent PRs:
- Unify the collection done in `/utils/stats.js` with the
`startMonitoringNetwork` method.
- Incorporate the remote-inbound `fractionsLost` metric to account for packet
loss on both legs of the network (in/out).
- Update the packet loss metric displayed in the connection status modal to
show a more meaningful value (e.g., packet loss percentage over a specific
probe interval). An absolute counter of lost packets isn't useful for end
users.
- Update the alert log to use the fraction or percentage above
* Refactor: Make bundle using webpack
* Fix: restore after install codes and a few settings
* Fix: build script folder permission
* Refactor: Remove support to async import on audio bridges
* Upgrade npm using nvm
* Avoid questions on npm ci execution
* Let npm ci install dev dependencies (as we need the build tools here)
* Fix: enconding
* Fix: old lock files
* Remove: bbb-config dependency to bbb-html5 service, bbb-html5 isn't a service anymore
* Fix: TS errors
* Fix: eslint
* Fix: chat styles
* npm install with "lockfileVersion": 3 (newer npm)
* build: allow nodejs 22
* node 22; drop meteor from CI and bbb-conf
* TEMP: use bbb-install without mongo but with node 22 and newer image
* build: relax nodejs condition to not trip 22.6
* build: ensure dir /usr/share/bigbluebutton/nginx exists
* init sites-available/bbb; drop disable-transparent-
* nginx complaining of missing file and ;
* TMP: print status of services
* WIP: tweak nginx location to debug
* Fix: webcam widgets alignments
* akka-apps -- update location of settings.yml
* build: add locales path for nginx
* docs and config changes for removal of meteor
* Fix: build encoding and locales enpoint folder path
* build: set wss url for media
* Add: Enable minimizer and modify to Terser
* Fix: TS errors
---------
Co-authored-by: Tiago Jacobs <tiago.jacobs@gmail.com>
Co-authored-by: Anton Georgiev <anto.georgiev@gmail.com>
Co-authored-by: Anton Georgiev <antobinary@users.noreply.github.com>
Jitter evaluation, as an alert trigger, was changed in 3.0 to get the internal
average jitter used in the conn-status component data (which is total jitter
delay divided by jitterbuffer emit events). This was done accidentally and that
metric is _very_ different from the one used in 2.7 (point-in-time jitter from
remote-inbound-rtp/inbound-rtp, highest on the interval, gathered on
/utils/stats.js). The alert thresholds were preserved, which makes it overly
sensitive in regards to jitter (and thus causes it to be critical whenever the
user is in audio).
Remove jitter as a connection status alert trigger, which fixes the
false positive. The implementation on <= 2.7 is also not ideal - if
anything, it generates false negatives. That's why I'm removing jitter for
the time being since it's ill-suited (at least in the way it's used)
for what we want to achieve.
* refactor(storage): replace Tracker.Dependency with observer hook
* fix(storage): set initial value
* refactor(storage): stop using Meteor's Session singleton
For browsers that don't support headerBytesSent in RTCOutboundRtpStreamStats
neither headerBytesReceived in RTCInboundRtpStreamStats, we are now able
to calculate upload and download rates.
We are also able to get transportStats information for browsers that
don't support iceTransport attribute of RTCDtlsTransport.
Added support for getStats in screenshare's service. This works similar
to the getStats for video provider, and the information retrieved from
screenshare is added to the video information for cameras.
We now retrieve update information about active video peers, and calculates
download and upload rates. These rates are the sum of data transfered in
all video peers.
Screenshare stats is not being added to the sum, yet.
- Removed the connection-status history from the user list's gear
icon and now is opened by the connection-status button. Moderators will
render the same modal as before and viewers will only have access to their
own data.
- Added data-savings shortcut at the connection-status modal.
- Added websocket round-trip time.
