When a participant is closing, RTCP readers should be cleaned up from
factory even if the participant is expected to resume. The resumed
participant will be a new participant session and peer connection(s) and
everything will be set up again.
After adding more fields in
https://github.com/livekit/livekit/pull/4105/files, it was not even
logging. Access to one of the added fields must have ended up waiting on
a lock and blocked.
Unfotunately, the deadlock fix in https://github.com/pion/ice/pull/840
did not address the peer connection close hang.
Splitting the logs so that the base log still happens. Ordering after
looking at the code and guessing what could still log to see if we get
more of the logs and learn more about the state and which lock ends up
the first blocking one.
* Add checks for participant and sub-components close.
Looks like there might be some memory leak with participant sessions not
getting closed properly. Adding checks (to be cleaned up later) to see
if there is a consistent place where things might hang.
* init with right type
* Remove unnecessary goroutine, thank you @milos-lk
* clean up
Correct triple-d spelling of "address" field in transport logs.
I’m not sure whether this was intentional, but I noticed it
while creating Grafana queries and filters. This matters because
anyone filtering logs using the correct spelling may
unintentionally miss relevant data, leading to incomplete or
misleading analysis.
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.
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.
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
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.
* Set publisher codec preferences after setting remote description
Munging SDP prior to setting remote description was becoming problematic
in single peer connection mode. In that mode, it is possible that a
subscribe track m-section is added which sets the fmtp of H.265 to a
value that is different from when that client publishes. That gets
locked in as negotiated codecs when pion processes remote description.
Later when the client publishes H.265, the H.265 does only partial
match. So, if we munge offer and send it to SetRemoteDescription, the
H.265 does only a partial match due to different fmtp line and that gets
put at the end of the list. So, the answer does not enforce the
preferred codec. Changing pion to put partial match up front is more
risky given other projects. So, switch codec preferences to after remote
description is set and directly operate on transceiver which is a better
place to make these changes without munging SDP.
This fixes the case of
- firefox joins first
- Chrome preferring H.265 joining next. This causes a subscribe track
m-section (for firefox's tracks) to be created first. So, the
preferred codec munging was not working. Works after this change.
* clean up
* mage generate
* test
* clean up
- Move downTrack instantiation to SubscribedTrack as it should own that
DownTrack. Still more to do here as `DownTrack` is fetched from
`SubscribedTrack` in a few places and used. Would like to avoid that,
but doing this initially.
- Use an interface from sfu.Downtrack and replace a bunch of callbacks.
SubscribedTrack is the implementation for DownTrackListener.
