pion no longer starts the repair stream reader when a custom BufferFactory
is set, so the mid/rid/rsid extensions were never observed and simulcast RTX
streams were never paired with their primary streams.
Extract the extensions on the buffer write path instead. Migrated publishers
send no extensions at all, so pair those from SimTracks.
Adds an integration test covering both paths, and moves the vnet setup it
shares with the downtrack test into pkg/testutils/vnettest.
Co-authored-by: Claude Opus 5 <noreply@anthropic.com>
Flush the downtrack sequencer on stream restart (Resync, ReceiverRestart,
codec change) so NACK retransmissions can't use metadata that no longer
matches the resynced bucket. Add a defensive bounds guard on the RTX and
forward payload slicing.
Cap the PacketHistory and FrameIntegrityChecker catch-up loops to the ring
size so a large sequence/frame-number jump can't drive a big per-packet
iteration count.
Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* Fix skipped packets accounting.
No need to copy unskipped packet RTP header to skipped packet.
That was causing padding bytes to be counted.
Also use Header.PaddingSize as base PaddingSize is deprecated.
* PaddingSize in header in utils
Reverting
- https://github.com/livekit/livekit/pull/4521
- https://github.com/livekit/livekit/pull/4525
There are TWCC feedback packets that are larger than MTU. Seems to
happen under a couple of conditions
1. Bad client data, i. e. severely out-of-order packets, bad sequence
numbers, etc.
2. On an ICE restart - this is rare, but it seemed to be flaky network
with some packets arriving and some not and causing a lot of gaps.
Either case, not much to do. If fargmentation/re-assembly back to
publisher works, the feedback will make it through. If not, feedbacks
will be missed and clients have to work with some missing data which is
not unexpected and the protocol is designed to handle.
However, filed pion/interceptor issue just in case - https://github.com/pion/interceptor/issues/416
Not a major issue, but just avoiding duplicate creation of NACK module.
RTCP feedback of `nack` and `nack pli` end up getting treated as `nack`
and was double creating.
* Log large packets receive/send.
Seeing cases of servers reporting need for segmentation/re-assembly of
packets. So, logging packet receive/send for RTP/RTCP to check if
anything is seeing more than 1400 byte packets.
* log downtrack RTCP too
* Add debug for receiver restart.
Have a suspicion that something is deadlocking between restart receiver
and buffer bind during replay. Adding debug to get a better picture of
state of receiver restart.
* consistent logging
* Set up audio config in audio level module when config is updated.
It is possible to get audio config after bind (bind is where the audio
level module is created) for remote tracks. So, split out setting audio
level config in audio level module and invoke it when config is updated.
* coderabbit review
* prevent divide-by-0
* not active before config
* Reducing some info level logs.
Also, relaxing the check for runaway RTCP receiver report to allow for
rollover to catch up if it is not too far away.
* set logger
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.
* Do not restart receiver on codec change mid-session.
This is not supported and was an erroneous change during the
receiver_base/buffer_base + RTP stream restart consolidation. Also make
the codec munger creation more resilient.
* fix test
In some paths, it is better to preserve pre-recorded time. So, make the
base implementations preserve the RTCP Sender Report receive time.
Also, add a method to enable forwarding packet arrival time. Could be
used across relay.
* Rework receiver restart.
- Protect against concurrent restarts
- Clean up and consolidate code around restarts
- Use `RestartStream` of buffer rather than creating new buffers.
* fix test
* Minor refactor in buffer base and audio level
- Make a function for `restartStream`. Will be useful
when external signal needs to restart a stream. Also restart all the
bits (audio level, dd parser and frame rate calculator)
- make an audio level mode with RTP timestamp so that some state can be
moved out of buffer base
* clean up
* log restart
This is not all of it as it is not possible (or at least I do not know
of a way) to get all suggestions for a repo/project. Did this via loop
searching mainly and taking the modernize suggestions.
David Chen