[Bug 784094] gst-rtsp-server: huge delay for the client to connect sometimes

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=784094

--- Comment #6 from Julien Isorce <julien.isorce at gmail.com> ---
While trying to make a unit test I found that the issue happens only if the
aux-rtp buffer is the one buffer used to generate the rtsp url. Basically the
rtsp server get the seqnum of the current buffer to write it in the rtsp url,
ex: url=rtsp://0.0.0.0:34605/test/stream=0;seq=24345;rtptime=2832971436 . I
suppose the client will parse this seqnum and wait for rtp buffers starting
from this seqnum. So if the seqnum error/diff is really big then the client
will wait a lot before accepting the rtp buffers.

So it cannot be reproduced with the rtx elements because it will necessarly not
be the first packets sent.

In the production app, the issue happened because the first packet sent was a
FEC rtp packet, with a different ssrc and a different seqnum base. And the rtsp
server was using this seqnum to generate the rtsp url.

For a unit test, one can try to inject, into the payloader, a fake rtp packet
with a magic seqnum. And then the test will fail if the this magic seqnum
appears in the rtsp url. The attached patch allow the rtsp server to ignore
this packet and use the payloader's infos instead.

The test can be added in gst-rtsp-server/test/checks/gst/rtspserver.c and run
with:
cd gst-rtsp-server/test/checks && make gst/rtspserver.check

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