[Bug 790412] unit test failure: rtspclientsink test_record fails with GST_MESSAGE_TYPE (msg) != GST_MESSAGE_EOS

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=790412

--- Comment #22 from Patricia Muscalu <patricia at axis.com> ---
(In reply to Sebastian Dröge (slomo) from comment #21)
> (In reply to Patricia Muscalu from comment #20)
> 
> > > 2) In PAUSED->PLAYING the sink is doing SETUP(s) and RECORD. Between the
> > > last SETUP and RECORD, all transports are known and can be configured on the
> > > rtsp-streams. Which can then be "completed" (-> add udpsink, appsink, etc.
> > > as needed for the RTP streams but also for the RTCP streams. How would we
> > > know before this where to send the RTCP?), and once RECORD has happened we
> > > can unblock the rtsp-streams so that data gets send to the RTP sinks, and
> > > thus to the RTSP server.
> > 
> > Yes, before SETUP we need to allocate UDP ports for RTP and RTCP and complete
> > the streams with the receiver and the sender parts.
> 
> Hm but *before* SETUP we might not know yet where to send the UDP packets as
> the server will tell us in its reply to SETUP. How would that work?
> 
Good point. I think this problem already exists in 1.12 as _join_bin() ->
alloc_ports() is called before SETUP. Let me check how it works ;)

I have a patch that actually works <=> the test passes. However, I need to
check all the details before providing a proper solution.

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