[Bug 760556] New: rtspsrc: interleaved data and srtp don't play well together

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=760556

            Bug ID: 760556
           Summary: rtspsrc: interleaved data and srtp don't play well
                    together
    Classification: Platform
           Product: GStreamer
           Version: unspecified
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: aconchillo at gmail.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

When using RTSP with interleaved data (protocols=tcp) in conjunction with SRTP,
rtspsrc doesn't work (see at the end).

This could be solved by just disabling SRTP. If you have SRTP it probably means
you also encrypt the RTSP channel, so there's no point on having additional
encryption.

But, in any case, this should still work.

----

$ gst-launch-1.0 rtspsrc location=rtsps://localhost:8554/webcam latency=300
protocols=tcp tls-validation-flags=0 ! decodebin ! fakesink silent=false -v

Gives these error message:

ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Internal data
flow error.
Additional debug info:
gstrtspsrc.c(5483): gst_rtspsrc_loop ():
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
streaming task paused, reason not-negotiated (-4)

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