When SetMaxSpatialLayer() is called with target/current layers in
InvalidLayerSpatial state, opportunistically initialize the target
layer to avoid dropped packets during async stream allocator
initialization.
Guards:
- Only sets target if not congestion-throttled (isDeficientLocked)
- Does not set current layer (deferred to keyframe-based forwarder start)
- Logs at Debug level to avoid log noise
This prevents undefined layer state during manual subscription
with immediate quality upgrades (WithAutoSubscribe(false) +
SetVideoQuality(HIGH)).
Pion does not protect the stats getter and using it after close could
cause nil de-reference. Do a couple of things
1. Stop timer that access peer connection stats before closing peer
connection.
2. Do not access stats if peer connection is already closed
Currently, the signal requests are counted on media side and signal
responses are counted on controller side. This does not provide the
granularity to check how many response messages each media node is
sending.
Seeing some cases where track subscriptions are slow under load. This
would be good to see if the media node is doing a lot of signal response
messages.
When doing code changes for dynamic rid, inadventently relied on
ordering of quality in track info layers to pick the highest layer if
the requested quality is higher than available qualities.
@cnderrauber addressed it in
https://github.com/livekit/livekit/pull/3998. Just adding some more
robustness behind that by doing a full search when requested quality is
not available.
Tested using JS SDK demo app and picking different qualities from
subscriber side with adaptive streaming turned off.
Effectively reverts https://github.com/livekit/livekit/pull/3984.
Using padding only packets for audio dummy start introduces dependencies
on other services and is not a necessary change. Would have been good to
use padding only for audio also from t=0. We can re-visit this for
better compatbility down the line.
* Added optional "Power of Two Random Choices" algorithm for the node selector sort_by feature. The current, default behavior of picking the lowest-valued node remains.
Seeing cases of `ConnectionTimeout` and `ResponseTimeout`.
So, logging destination identity in RPC request and also logging ACK and
response. Will pare back logs/log level of these messages after gettnig
some data.
Also a small change I noticed and had sitting in my local tree to set
the previous RTP marker on a padding packet.
Even when encrypted, can set up opus as the second codec to support the
case of RED interspersed with Opus packets when the RED packet is too
big to fit in one packet.
The change here is to not go through all up stream codecs when trying to
find a match in DownTrack.Bind when source is encrypted. When encrypted,
the down track codec should match the primary upstream codec, i. e. the
codec at index 0.
It is posible that a subscriber joins when a publisher has reconnected
and has received a flood of retransmitted packets due to NACKing the
gap caused by the publisher reconnecting. Starting on that spurt means
the subscriber gets a burst of unpaced packets that could lead to issues
with calculating render time (especially obvious in cases like egress).
The following scenario produced silence audio in egress
1. JS SDK publishing audio/red
2. Egress joins
3. Egress picks the RED primary receiver -> this converts RED to Opus in SFU
4. At the same time, codec regression is triggered back to the publisher. Publisher switches to publishing audio/opus and stops audio/red.
5. But, egress is still attached to audio/red and pulls down only silence.
Fix by checking if the negotiated codec needs publish before sending
subscribed codec udpate.
publisher peer connection.
While cleaning up during single peer connection changes, unintentionally
removed handler.
Also, another small change to log first packet time adjustment after
increment.
* Adjust for hold time when fowarding RTCP report.
When passing through RTCP sender report, holding it for some time before
sending means the remote receiver could see varying amount of
propagation delay if the remote uses something like local_clock -
ntp_sender_report_time and adapting to it.
Ideally, SFU should just forward RTCP Sender Report, but the current pull model to
group RTCP sender reports makes it a bigger change. So, adjust it by
hold time.
Also add a initial condition for one-way-delay estimator which can init
with a smaller value of latency if the first sample to measure
one-way-delay itself experienced higher delay than the prevailing
conditions.
* variable name
* log as duration
* Adjust stream allocator ping interval based on state.
In steady state, does a 15 second ping.
While deficient, to be able to react to probes faster, it pings at 100ms
interval.
* clean up
* log ops queue not able to wake up
* Tweaks tresholds for logging high forwarding latency/jitter.
Previous attempt showed skewed jitter (i. e. more than 10x latency),
But, no large latency.
So, reducing the latency treshold to declare high latency.
And also keeping track of lowest/highest per reporting window and
logging those along with short term and long term measurements.
NOTE: previously short term and long term were separate calls with locks
acquired. Now, it is all in one lock. So, it does increase the lock
duration a bit, but hopefully not by too much as the welford merge for
short term would go over 20 samples (at 50 ms sampling interval and 1 s
reporting window).
* revert skew factor
* Log some information around high forwarding latency.
Latency is not 0 after switching to microseconds resolution.
But, still seeing high jitter. Logging a bit more to understand under
what conditions it happens.
More notes inline.
* compact
Latency is always 0, but jitter is high.
Not sure how that happens as latency is the welford mean and jitter is
welford standard deviation. Feels like some mis-labeling.
Anyhow, switching to microseconds units to get better resolution.
Due to SDP ordering, Pion did not provide track ID on a receiver.
Pion needs a=msid line to be before a=ssrc line -> need to check if this
is a spec requirement
Because of the above, it had empty id for the receiver in onTrack.
That matched a published track because we do not duplicate SdpCid in
TrackInfo if the SDP cid matches the signal cid. But, the search checks
both and it matched on empty SDP cid.
Do not accept empty ids in searches to prevent this from happening.
* Add debugging from DD frame number wrap around.
On a DD parser restart, the extended highest sequence number oes not
seem to be updated. Adding some debug to understand it better.
* more logs
* log incoming sequence number and frame number
