LastSenderReportTime() returned time.Unix(0, srNewest.At). But At is a
mono timestamp, so the result is not a real wall time. Connection quality
compares it to time.Now() to decide if RTCP is stale. If the wall clock
drifts from mono, for example after an NTP step, streams can look stale
when they are not, or the reverse.
- Convert At to wall time: now minus the mono time since the report.
- Return zero time when At is 0, so the scorer skips the stale check
(before, it got the 1970 epoch and saw the stream as stale).
Follow-up to #5005, which fixed the same mix-up in DLSR.
Tests: rtpstats and connectionquality pass. No new test, because a
test process cannot make the mono and wall clocks drift.
Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
* agent: release JobTerminate handler when job ends before registration
JobRequest registered the job's JobTerminate handler after AssignJob
returned, so an ended UpdateJobStatus or worker disconnect handled in
between deregistered nothing and the handler leaked. AssignJob also sent
the assignment before recording the job as running, so an immediate
ended update was dropped.
Record the job before sending the assignment, and after registering the
handler recheck under h.mu that the job is still running and its worker
still registered, deregistering otherwise.
Fixes#4901
* replace utils.CloneProto with proto.CloneOf
* agent: wait on worker readiness and poll for released handlers in test
- prevent some escape to heap
- avoid copying by using a ring buffer for receiver reports (probably
should remove this as this is for debugging only and data so far has
shown clients sending bad data and nothing more.)
Removing some logs which have not been useful in terms of insights other
than saying that there are a bunch of packets missing. Going to start
looking at gaps in terms of time if the inter-packet gap is too high.
Also, using logging these events as first 20 and then every 200.
* Guard against timestamp inversion in RED -> Opus conversion.
Seeing timestamp inversion (sequence number is +1, but timestamp is
-960, i.e. 20ms) in the RED -> Opus conversion path. Not able to spot
any bugs in code. So, logging details upon detection and also dropping
the packet. If not dropped, downstream components like Egress treat it
as a big timestamp jump (because sequence number is moving forward) and
try to adjust pts which ends up causing drops.
* do not log time reversal at the start
* typo
* Optimise some bits in rtpstats_receiver
RTPStatsReceiver.Update is one of the high CPU bits. Taking some
suggestions from Cursor. Makes the `Update` function verbose though :-(
* zap.Inline logging fields
* rename
With audio simulcast codecs, it is possible that the clock rate of the
primary codec is different from the secondary codec. If a subscriber
binds to the secondary codec, the clock rate should be set correctly. Do
it at bind time.
* Refactor receiver and buffer into Base and higher layer.
To be able to share code/functionality with relay.
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* clean up
* deps
* fix test
* fix test
* Add support for RTP stream restart.
When an unhandled packet is encountered, try a restart sequence.
Restart happens when 5 packets with contiguous sequence numbers and same
or increasing time stamps are received. Note that this does not work for
B-frame type of scenarios, but that is true for receive path handling
even before this. As WebRTC does not use B-frames, it is fine. But,
needs to be looked at again if B-frames are necessary.
It is controlled by a config that is disabled by default.
* clean up
* debug log
It is possible that the stream stops just after start and
restarts much later introducing a large gap in sequence number.
That could look like an unhandled case because the wrap back handler
does not have enough packets yet.
Let other checks based on time stamp gap take effect and only if that
also leaves the sequence number unhandled, drop the packet.
- New bucket API to pass in max packet size and sequence number offset
and seequence number size generic type
- Move OWD estimator to mediatransportutil.
* 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
* Make lite version of RTPStatsReceiver called RTPStatsReceiverLite.
Refactor around that.
Will probably make some more flavors to have lighter versions still.
* update deps
* use MarshalLogArray
* use util
* Splitting out rtp stats stuff into its own package.
Going to be making some lighter versions of these.
Will be cleaner to have all of these grouped together.
So, as a first step, just making a package for it.
* tests
