* WIP commit
* Connection quality changes
- Fix Firefox showing poor quality
o The issue was that we were using max available layer and
calculating quality. The rationale being that even if
server sends dynacast messages, client may not implement
dynacast and still stream all layers. But, with Firefox
(maybe a Firefox bug), it sends some small amount of
data on layer 2 even when that layer is disabled.
Guessing it is probing (or actually we might be using
some small value for high layers as Firefox cannot turn off
layers). That higher layer gets used in quality calculation.
As the bit rate on that layer is extremely low, it yields low
score.
Fixed by considering the max expected layer. That is of most
interest. Yes, clients may ignore dynacast and stream all layers,
but, max expected is the one of interest. So, look for
quality in the max expected layer and not max available layer.
- Lots of clean up around connection quality stuff
o Use a dynamic scaling thing to ensure that we do not get bitten
by absolute values. Calculate best possible scenario score and
map that to maximum MOS score. This will ensure that different
codecs, different settings do not mess up the scoring. For example,
a client might use 1 Mbps for 720p, but a different client could
use 2 Mbps for 720p. As an SFU/infrastructure middlebox, we do
not have control over quality at those rates. We can only ensure
that streaming happens smoothly at those rates. So, in that
example, for client 1, 1 Mbps will map to MOS 5.0 and for client 2,
2 Mbps will map to MOS 5.0. Any impairments after that will
reflect in the score.
o Penalise for missing target layer by one level for one layer missed.
o Move tests to connection quality directory. The participant test
was not super useful.
* Add missed file
* Remove debug code
* use more constants and initialise normalisation factor
* rtcscore pointer
* Use media payload size in scoring.
Subtract out header bytes when calculating score.
This does not seem to affect the score (under perfect conditions),
but, using header bytes will inflate the bit rate and
will affect scoring.
* Add header bytes to ToProto
* protocol pointer
* fix test
With a small window, the quality is volatile even on small disturbances.
For example losing 2 audio packets in a 2 second window could
drop the quality metric.
* Set DtxDisabled from TrackInfo in score calculation.
Also, fix sending connection quality upate on a new subscription.
* comments tweaks
* Move TrackInfo into StreamTrackerManager as this is used by cloud as well
* WIP commit
* WIP commit
* Remove debug
* Revert to reduce diff
* Fix tests
* Determine spatial layer from track info quality if non-simulcast
* Adjust for invalid layer on no rid, previously that function was returning 0 for no rid case
* Fall back to top level width/height if there are no layers
* Use duration from RTPDeltaInfo
* Use rtcscore-go to calculate audio/video score
Signed-off-by: shishir gowda <shishir@livekit.io>
* Get max expected layer and find max actual layer from stream
Signed-off-by: shishir gowda <shishir@livekit.io>
* Cleanup unused methods
Signed-off-by: shishir gowda <shishir@livekit.io>
* Cleanup code - address review comments
Signed-off-by: shishir gowda <shishir@livekit.io>
* get expected layer info instead of just quality
Signed-off-by: shishir gowda <shishir@livekit.io>
* Move SpatialLayerForQuality to utils/helpers
method is required in rtc,sfu and connectionstats pkg
Moved to utils/helpers.go to remove cyclic deps
Signed-off-by: shishir gowda <shishir@livekit.io>
* update tests
Signed-off-by: shishir gowda <shishir@livekit.io>
* Pick stream stats with max layer
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update rtcscore-go pkg to make rtt/jitter optional
when passing 0, rtcscore-go was setting default values
Signed-off-by: shishir gowda <shishir@livekit.io>
* update score to rating
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update rtcscore-go pkg to use simulcast layer info for score
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update score ratings to reflect rtcscore range
Signed-off-by: shishir gowda <shishir@livekit.io>
* update test params for new rtcscore
Signed-off-by: shishir gowda <shishir@livekit.io>
* Delay sending scores to connections only till full data is available
first interval can have partial data leading to lower scores
Signed-off-by: shishir gowda <shishir@livekit.io>
* Check for inf values in quality params
Signed-off-by: shishir gowda <shishir@livekit.io>
* Clean up initial score calculation. Default to 5
Signed-off-by: shishir gowda <shishir@livekit.io>
Co-authored-by: David Zhao <dz@livekit.io>
* Improve frequency of stats update
Prometheus stats are updated as the data becomes available, instead of
aggregated along with telemetry batches. Node availability decisions can
now react much faster to these stats.
* use the same intervals for connection quality updates
* Use delta stats throughout and avoid calculating deltas in telemetry
* Fix a few things after testing
* Remove debug
* Fix tests
* delete instead of setting to nil
* Point to the latest protocol
* Add a resync API to sfu.DownTrack
Also passing in logger with context into sfu package. More to do here
with proper logging context in all modules, but this is a start
* Remove debug code
* fix tests
* audio connection quality mos for publisher stats
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update tests
Signed-off-by: shishir gowda <shishir@livekit.io>
* Change ratings range, increase default rtt to 80
Signed-off-by: shishir gowda <shishir@livekit.io>
* Use stats worker to get total packets to find %lost in window
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update go dep
Signed-off-by: shishir gowda <shishir@livekit.io>
* Increase interval of score cal to 5 seconds
Signed-off-by: shishir gowda <shishir@livekit.io>
* use lastSequenceNumber in reports to find total packets
Signed-off-by: shishir gowda <shishir@livekit.io>
* Account for delay while calculating scores
Signed-off-by: shishir gowda <shishir@livekit.io>
* Fix minor typo
Signed-off-by: shishir gowda <shishir@livekit.io>
* Add connection stats/score to subscribed audio tracks
Signed-off-by: shishir gowda <shishir@livekit.io>
* Cleanup
Signed-off-by: shishir gowda <shishir@livekit.io>
* Ignore duplicate LastSequenceNumbers in rtcp reports
Ignore if sequence number is less than what was recieved
Signed-off-by: shishir gowda <shishir@livekit.io>
* Move video track score calc to media/downtracks
Signed-off-by: shishir gowda <shishir@livekit.io>
* Deprecate SubscribeLossPercentage() as score calc is now handled downstream
Signed-off-by: shishir gowda <shishir@livekit.io>
* Initialize connection score to excellent
score is calc at 5sec interval. Client fetches score before first
score is computed
* Update test cases for connection quality
Signed-off-by: shishir gowda <shishir@livekit.io>
Raja Subramanian