[Bug 739419] rtspsrc: not-linked after a while

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=739419

--- Comment #29 from Nicola <lists at svrinformatica.it> ---
I don't fully understand you suggestion sorry :(

suppose you have an rtsp media with audio + video (example here
rtsp://admin:12345@93.63.189.11:1554/Streaming/channels/401)

1) new_manager_pad
(http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtsp/gstrtspsrc.c#n2914)
is called two time, video has pt 96 and audio has pt 0
2) about every two seconds on the rtp video session arrive an rtp packet with a
wrong sequence number and payload set to 112 (so no new pad is added, is the
same pad already created)
3) request_pt_map is called
(http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtsp/gstrtspsrc.c#n3006)
and return null caps since there is no stream with pt 112
4) the rtp buffer with pt 112 is passed to the next elements anyway. How we can
instruct the manager to drop buffer when request-pt-map signal callback return
null? (sorry for the probably trivial question)

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