* Make connection quality not too optimistic.
With score normalization, the quality indicator showed good
under conditions which should have normally showed some badness.
So, a few things in this PR
- Do not normalize scores
- Pick the weakest link as the representative score (moving away from
averaging)
- For down track direction, when reporting delta stats, take the number
of packets sent actually. If there are holes in the feed (upstream
packet loss), down tracks should not be penalised for that loss.
State of things in connection quality feature
- Audio uses rtcscore-go (with a change to accommodate RED codec). This
follows the E-model.
- Camera uses rtcscore-go. No change here. NOTE: THe rtscore here is
purely based on bits per pixel per frame (bpf). This has the following
existing issues (no change, these were already there)
o Does not take packet loss, jitter, rtt into account
o Expected frame rate is not available. So, measured frame rate is
used as expected frame rate also. If expected frame rate were available,
the score could be reduced for lower frame rates.
- Screen share tracks: No change. This uses the very old simple loss
based thresholding for scoring. As the bit rate varies a lot based on
content and rtcscore video algorithm used for camera relies on
bits per pixel per frame, this could produce a very low value
(large width/height encoded in a small number of bits because of static content)
and hence a low score. So, the old loss based thresholding is used.
* clean up
* update rtcscore pointer
* fix tests
* log lines reformat
* WIP commit
* WIP commit
* update mute of receiver
* WIP commit
* WIP commit
* start adding tests
* take min score if quality matches
* start adding bytes based scoring
* clean up
* more clean up
* Use Fuse
* log quality drop
* clean up debug log
* - Use number of windows for wait to make things simpler
- track no layer expected case
- always update transition
- always call updateScore
Due to the order of events in MediaTrackReceiver and friends, SubscribedTrack
will be closed before the track is removed from RoomTrackManager.
Because of this, when a track is unpublished, it's possible to be subscribed
to the track as it's closing.
By introducing a closing state, we'd prevent accidental subscription to
closing tracks.
* add prometheus stats for rtt/jitter/packet loss
* add track source to metrics
* better packet loss bins
* add track type to metrics
* remove source from AnalyticsStat
* regenerate telemetry service fake
* compute loss from per stream packet count
Related to livekit/protocol#273
This PR adds:
- ParticipantResumed - for when ICE restart or migration had occurred
- TrackPublishRequested - when we initiate a publication
- TrackSubscribeRequested - when we initiate a subscription
- TrackMuted - publisher muted track
- TrackUnmuted - publisher unmuted track
- TrackPublish/TrackSubcribe events will indicate when those actions have been successful, to differentiate.
* Split stream tracker impl from base
* slight re-arrangement of code
* fps based stream tracker
* MinFPS config
* switch back to packet based tracker
* use video config by default to handle sources without type
* Fix rtcp lost for downtrack used incorrect buffer factory
In buffer factory change(#1173), every pariticipant has its own
buffer factory, can't use publisher's bufferfactory to create
DownTrack
* clean code
* Cache RTPStats and seed on re-use
When a cached down track is re-used, RTPStats was not cached.
This caused sender reports getting out-of-sync with the remote side.
Cache RTPStats and seed it on re-use.
* staticcheck
* Prevent track subscriptions/adding receivers after close
With subscribe/unsubscribe queuing, a subscribe may be
attempted after a call to `RemoveAllSubscribers`.
So, renaming `RemoveAllSubscribers` to `InitiateClose`
and maintaining state that track is in the process of closing.
* Mime specific remove
* Remove unused error
* do not add receiver when closing
* Limit dynacast to video and media loss proxy to audio
Was looking at keeping the track type out of those modules
and do a check at a higher level, but it is a bit unwieldy.
So, adding checks to the modules.
Also, ensuring that media loss proxy does not reset unconditionally
every second. Audio RTCP happens once in 5 seconds or so.
So, if server proxied let say 2% at t = 5, t = 6 would have
proxied 0 loss which may or may not be true. So, ensure that
a report was received and proxy value is updated by an actual
report.
* Remove track type from modules
* WIP commit
* Refactor media loss proxy
* Use DynacastQuality and MediaLossProxy from MediaTrack
* fix test
* Remove unused param
* Remove unused interfaces
* Move interface methods to local
* Split out DynacastManager
* have to add codec to dynacast manager
* RUnlock
* fix restart
* Adding API to force quality and also maintain closed state
* Address PR comments
* Move subscribe/unsubscribe queue to participant.
As subscribe/unsubscribe operation can come from both
local media track or remote media track, participant
needs to have it.
* Remove comment
* Stop reneg timer on close
* address comments
* Fixed unclean DownTrack close when removed before bound.
When a DownTrack is closed before it had a chance to be bound to a
transceiver, we'd skip close and leave it hanging. This is unlikely in
normal operations. However, it can be seen with permissions and
subscription APIs.
* remove remaining peerID references
* 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>
* Telemetry and webhook improvements.
* avoid blocking on telemetry channel - increase channel size and drop when full
* send ParticipantJoined webhook when fully joined (i.e. on ParticipantActive)
* send TrackPublished & TrackUnpublished webhooks
* increase number of parallel webhook workers to 50
* update protocol
* Consolidating PLI throttle
Use the throttler in `sfu.WebRTCReceiver`.
Does change shape of config object.
* Move PLIThrottleConfig to sfu.WebRTCReceiver
* fix test compile
* Cleaning up unused stuff
* improve readability
* RTT
- Calculate down track RTT using RTCP Receiver report
- Surface it back to the participant
- Participant updates all its published trackes
(throttled to limit update to once in 5 seconds)
- That propagates to all the upstream sfu.Buffer and the nacker.
So, we will have RTT throttled NACKs.
* rtt callback
* Consolidating PLI throttle
Use the throttler in `sfu.WebRTCReceiver`.
Does change shape of config object.
* Move PLIThrottleConfig to sfu.WebRTCReceiver
* fix test compile
* Cleaning up unused stuff
* readability improvement
* 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
David Zhao