The browser could send rtp packets of svc encoding without
DD extension while the sdp negotiates it, sfu detects extension
in rtp packet for this case.
* Call Broadcast in lock scope.
Seems like there is a possible window where things can hang forever
if a goroutine enters the Wait) after the lock is released but before
Broadcast gets called, it will never see that broadcast and will hang forever.
* RLock
* Normalize mime type and add utilities.
An attempt to normalize mime type and avoid string compares remembering
to do case insensitive search.
Not the best solution. Open to ideas. But, define our own mime types
(just in case Pion changes things and Pion also does not have red mime
type defined which should be easy to add though) and tried to use it everywhere.
But, as we get a bunch of callbacks and info from Pion, needed conversion in
more places than I anticipated. And also makes it necessary to carry
that cognitive load of what comes from Pion and needing to process it
properly.
* more locations
* test
* Paul feedback
* MimeType type
* more consolidation
* Remove unused
* test
* test
* mime type as int
* use string method
* Pass error details and timeouts. (#3402)
* go mod tidy (#3408)
* Rename CHANGELOG to CHANGELOG.md (#3391)
Enables markdown features in this otherwise already markdown'ish formatted document
* Update config.go to properly process bool env vars (#3382)
Fixes issue https://github.com/livekit/livekit/issues/3381
* fix(deps): update go deps (#3341)
Generated by renovateBot
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
* Use a Twirp server hook to send API call details to telemetry. (#3401)
* Use a Twirp server hook to send API call details to telemetry.
* mage generate and clean up
* Add project_id
* deps
* - Redact requests
- Do not store responses
- Extract top level fields room_name, room_id, participant_identity,
participant_id, track_id as appropriate
- Store status as int
* deps
* Update pkg/sfu/mime/mimetype.go
* Fix prefer codec test
* handle down track mime changes
---------
Co-authored-by: Denys Smirnov <dennwc@pm.me>
Co-authored-by: Philzen <Philzen@users.noreply.github.com>
Co-authored-by: Pablo Fuente Pérez <pablofuenteperez@gmail.com>
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
Co-authored-by: Paul Wells <paulwe@gmail.com>
Co-authored-by: cnderrauber <zengjie9004@gmail.com>
* move unrolled mime type check for broader use
* Use in IsSvcCodec and make MimeType exported
* test
* tidy branches
* tidy
---------
Co-authored-by: Paul Wells <paulwe@gmail.com>
* 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
* Cache RTCP sender report in forwarder state.
To be used in migration.
TODO: need to check more places to operate pure in unix nano rather than
converting.
* match name
It is possible that old packets arrive on receiver. If subscriber starts
on that, the first packet time would be incorrect. Do not start
forwarding on out-of-order packets.
Seeing cases of huge jumps in sender erport rtp time stamp
(of the order of minutes) a few hundred ms after start of track.
Only less than 20 packets have been published at that time as seen by
server. Adding these to sender report to check if client thinks it has
sent much more.
* Handle cases of long mute/rollover of time stamp.
There are cases where the track is muted for long enough for timestamp
roll over to happen. There are no packets in that window (typically
there should be black frames (for video) or silence (for audio)). But,
maybe the pause based implementation of mute is causing this.
Anyhow, use time since last packet to gauge how much roll over should
have happened and use that to update time stamp. There will be really
edge cases where this could also fail (for e. g. packet time is affected
by propagation delay, so it could theoretically happen that mute/unmute
+ packet reception could happen exactly around that rollover point and
miscalculate, but should be rare).
As this happen per packet on receive side, changing time to `UnixNano()`
to make it more efficient to check this.
* spelling
* tests
* test util
* tests
With RTX, some clients use very old packets for probing. Check for
duplicate before logging warning about old packet/negative sequence
number jump.
Also, double the history so that duplicate tracking is better. Adds
about 1/2 KB per RTP stream.
Remove the return when encountering invalid packet.
Also, log more sparesely.
Proper error returns from util so that we can selectively drop packets
based on error type, for example SSRC mismatches are okay type of thing.
* Validate RTP packets.
Check version, payload type (if available) and SSRC (if available)
and drop bad packets. And let repair mechanisms take effect for those
packets.
* address data race reported by test
* fix an unlock and test packets
With Read and ReadExtended waiting (they are two different goroutines),
use Broadcast always. In theory, they both should not be waiting at the
same time, but just being safe.
* 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
* 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
