hlssink with live camera

[Arjen Veenhuizen]

As you can see  here
<https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtspsrc.html> 
, rtspsrc has a sometimes pad for every elementary stream of type
application/x-rtp. This means that at the very least you should remove
qtdemux from the pipeline, add an rtph264depay element after the rtspsrc and
remove all audio related stuff (assuming there is only video).

In case there is audio you will have to do the same for that elementary
stream. Figure out how it is payloaded (you can find the various
depayloaders  here
<https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-plugin-rtp.html> 
) insert it and add the appropriate parser before hooking it up to the
mpegtsmux multiplexer.

You can run the pipeline after setting GST_DEBUG=2 (or 3) in order to yield
more verbose logging on stdout and see what goes wrong (you will probably
see one or multiple "not negotiated" errors as it is right now).



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