This will end up as a "received" packet, due to the code in
source_push_rtp, which will think this is a packet being received.
Instead drop the packet and hope that either:
1. Something upstream responds to the GstRTPCollision event and changes
SSRC used for sending.
2. That the application responds to the "on-ssrc-collision" signal, and
forces the sender (payloader) to change its SSRC.
3. That the BYE sent to the existing user of this SSRC will respond to
the BYE, and that we timeout this source, so we can continue sending
using the chosen SSRC.
The test reproduces a scenario where we previously would have sent
on a non-internal source.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
If the conflict is detected when sending a packet, then also send an
upstream event to tell the source to reconfigure itself.
Also ignore the collision if we see more than one collision from the same
remote source to avoid problems on loops.
We recently added code to remove outdate NACK to avoid using bandwidth
for packet that have no chance of arriving on time. Though, this had a
side effect, which is that it was to get an early RTCP packet with no
feedback into it. This was pretty useless but also had a side effect,
which is that the RTX RTT value would never be updated. So we we stared
having late RTX request due to high RTT, we'd never manage to recover.
This fixes the regression by making sure we keep at least one NACK in
this situation. This is really light on the bandwidth and allow for
quick recover after the RTT have spiked higher then the jitterbuffer
capacity.
Right now, we may call on-new-ssrc after we have processed the first
RTP packet. This prevents properly configuring the source as some
property like "probation" are copied internally for use as a
decreasing counter. For this specific property, it prevents the
application from disabling probation on auxiliary sparse stream.
Probation is harmful on sparse streams since the probation algorithm
assume frequent and contiguous RTP packets.
We used to split the NACK if a smaller seqnum of a range of seqnum was
submited. This test also make sure that the three operations (append,
prepend, update) works properly.
Calling rtp_session_send_rtcp before marking the source as requiring a
pli/fir/nack meant the rtcp_thread could be scheduled and start running
before the source was updated. This meant the request would not be sent
early but instead was transmitted with the next regular RTCP packet.
Add test for nack generation.
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.
We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.
The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
This patch simplifies the tests (44% less code) and
makes them much more readable.
The provided SessionHarness also makes it much easier
to write new tests for rtpsession.
https://bugzilla.gnome.org/show_bug.cgi?id=791070
Some endpoints (like Tandberg E20) can send BYE packet containing our
internal SSRC. I this case we would detect SSRC collision and get rid
of the source at some point. But because we are still sending packets
with that SSRC the source will be recreated immediately.
This brand new internal source will not have some variables incorrectly
set in its state. For example 'seqnum-base` and `clock-rate` values will be
-1.
The fix is not to act on BYE RTCP if it contains internal or unknown
SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=762219
Keeping the lock while emitting the stats signal introduces potential
deadlock in those situations when the signal callback wants the access
to rtpsession's properties which also requre the lock.
https://bugzilla.gnome.org/show_bug.cgi?id=762216
The time of the first RTCP packet is semi-random, so
sometimes it was produced before enough packets from
the second SSRC were received. First drop queued RTCP
packets, then advance the clock enough to ensure
that at least one new RTCP packet is produced.
https://bugzilla.gnome.org/show_bug.cgi?id=750731
