* Broadcast cond var on RTX write.
High forwarding latency logs all show high queuing delay so far. From
code inspection, RTX writes were not signaling the cond var. Not sure if
that is the reason, but adding a signal there for further tests.
* Remove return values from writeRTX as they are not used
* if RingingTimeout is provided, deadline should be set to that timeout.
This is because the SIP bridge will not return until RingingTimeout
which may be longer than the 30 second default deadline.
* handle Deadline being "before" timeout.
* Prevent leakage of previous codec after codec regression.
In the window between forwarder restart and determining codec, the old
codec packet could leak through. Prevent tha by doing the restart and
codec determination atomically on a codec regression.
* tidy
* use locked function
* Use the optimal allocation function for opportunistic allocation.
Allocation functions set the `lastAllocation` state also.
This might have been causing an e2e failure with v1 client on migration.
* annotate args
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
