* Handle large jumps in RTCP sender report timestamp.
Seeing cases of RTCP Sender Report spaced apart by more than half the
RTP Timestamp range. Maybe a case of laptop going to sleep and waking
up. Handle it using time diff from last report and calculating expected
timestamp.
* try go 1.22
* Connection quality LOST only if RTCP is also not available.
It is possible that sender stops all layers of video due to some
constraint (CPU or bandwidth). Packet reception going dry due to
that should not trigger `LOST` quality.
Add last received RTCP time also to distinguish the case
of real `LOST` and sender stopping traffic.
Some bits to watch for
- With audio, RTCP reports could be more than 5 seconds apart (5 seconds
is the default interval for connection quality scorer), but audio
senders usually send silence packets even when there is no input.
So audio completely stopping can be considered `LOST`.
- With video, have to observe if all clients continue to send RTCP even
if all layers are stopped.
- RTCP bandwidth is not supposed to exceed the primary stream bandwidth.
libwebrtc calculates that and spaces out RTCP reports accordingly.
That is the reason why audio reports are that far apart. If a video
stream is encoded at a very low bit rate, it could also be sending
RTCP rarely. So, there is the case of LOST being indistinguishable
from sender stopping all layers. But, this should be a rare case.
* typo
* Do codec munging when munging RTP header.
It was possible for probe packets to get in between RTP munging and
codec munging and throw off sequence number while dropping packets.
Affected only VP8 as it does codec munging.
* do not pass in buffer as it is created anyway
* flip fields
* flip order
* fix test
* call translate for all tracks
* simplify
If the first packet of keyframe has template structure is lost then
subsequent packets rely on it will report invalid tempalte error which
is expected.
* Add support for "abs-capture-time" extension.
Currently, it is just passed through from publisher -> subscriber side.
TODO: Need to store in sequencer and restore for retransmission.
* abs-capture-time in retransmissions
* clean up
* fix test
* more test fixes
* more test fixes
* more test fixes
* log only when size is non-zero
* log on both sides for debugging
* add marshal/unmarshal
* normalize abs capture time to SFU clock
* comment out adding abs-capture-time from registered extensions
* Disable audio loss proxying.
Added a config which is off by default.
With audio NACKs, that is the preferred repair mechanism.
With RED, repair is built in via packet redundancy to recover from
isolated losses.
So, proxying is not required. But, leaving it in there with a config
that is disabled by default.
* fix test
- Had arguments reversed.
- Also, cannot take away reference layer from state as a new layer
as reference could have a time stamp that is widely different from
expected. So, put that back.
* Move caching of publisher sender report to subscriber side.
Please see inline for descriptive comments on why. Basically,
pause/unpause using replaceTrack(null)/replaceTrack(actualTrack) can
cause time stamp in sender report sent to subscribers jump ahead.
This prevents that.
With the caching on subscriber side, cleaning up the caching on
publisher side.
* fix compile, test still failing, need to debug
* skip reference TS for testing
* Prevent large spikes in propagation delay
A few tweaks
- Large spike in propagation delay due to congested channel results in
long term estimate getting high value. Ignore outliers in long term
estimate.
- Introduce a new field for adjusted arrival time as adjusting the
arrival time in place meant it got applied again across the relay and
that caused different propagation delay on remote nodes.
- Reset path change counters as long as there is any sample that is not
higher than the multiple of long term. There was a case of
o Sample with high value that triggered path change start.
o Then some samples with high enough delta, but did not meet the
criteria for increasing counter further.
o Some time later, another sample met the threshold and that triggered
a path change re-init.
* do not adapt to large delta
With the change in https://github.com/livekit/livekit/pull/2611,
the dummy receiver was replaced with real receiver. But, the close check
was using the dummy receiver.
Doing two things
- Use the dummy receiver post upgrade also
(NOTE: this is not needed, but just keeping old behaviour)
- Fix the close check to count number of open receivers.
Raja Subramanian