Using Gstreamer authentication issue

[mram0509]

I am trying to follow the Readme to publish using Gstreamer. I tried two ways of using docker:

docker run --rm -it-network=host54 aler9/rtsp-simple-server

docker run --rm -it -e RTSP_PROTOCOLS=tcp -p 8554:8554 aler9/rtsp-simple-server

I am getting the following when I publish:
gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream filesrc
location=test.mp4 ! qtdemux name=d d.video_0 ! queue ! s.sink_0 d.audio_0 ! queue ! s.sink_1
Setting pipeline to PAUSED ...
Pipeline is PREROLLED ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://localhost:8554/mystream
WARNING: from element /GstPipeline:pipeline0/GstQTDemux:d: Delayed linking failed.
Additional debug info:
./grammar.y(510): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstQTDemux:d:
failed delayed linking pad audio_0 of GstQTDemux named d to some pad of GstQueue named queue1
Progress: (open) Retrieving server options
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending RECORD request

And if I try reading I get authentication error:
gst-launch-1.0 rtspsrc location=rtsp://localhost:8554/mystream name=s
s. ! application/x-rtp,media=video ! decodebin ! autovideosink s. ! application/x-rtp,media=a
udio ! decodebin ! audioconvert ! audioresample ! autoaudiosink
Setting pipeline to PAUSED ...
XDG_RUNTIME_DIR (/run/user/1000) is not owned by us (uid 0), but by uid 1000! (This could e g happen if you try to connect to a non-root PulseAudio as a root user, over the native protocol.
Don't do that.)
XDG_RUNTIME_DIR (/run/user/1000) is not owned by us (uid 0), but by uid 1000! (This could e g happen if you try to connect to a non-root PulseAudio as a root user, over the native protocol.
Don't do that.)
pci id for fd 9: 1a03:2000, driver (null)
Pipeline is live and does not need PREROLL ...
WARNING: from element /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0: Failed to connect: Connection refused
Additional debug info:
pulsesink.c(615): gst_pulseringbuffer_open_device (): /GstPulseSink:autoaudiosink0-actual-sink-pulse
Got context from element 'autovideosink0': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"(GstGLDisplayGBM)\ gldisplaygbm0";
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://localhost:8554/mystream
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:s: Could not open resource for reading.
Additional debug info:
gstrtspsrc.c(5829): gst_rtspsrc_setup_auth (): /GstPipeline:pipeline0/GstRTSPSrc:s:
No supported authentication protocol was found
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

Am I missing anything? I did follow the instructions in the README. Do I need to change any config options?

Thanks!

[dorinclisu]

WARNING: from element /GstPipeline:pipeline0/GstQTDemux:d: Delayed linking failed.

Your first pipeline for publishing seems to be failing, and this has nothing to do with rtsp server. Try with gst-launch-1.0 -v to possibly get more clues or set GST_DEBUG environment variable to "DEBUG".

Since you are not publishing anything, of course the second pipeline also fails to read, and again this has little to do with the rtsp server itself.

[aler9]

In my opinion, the problem is that your sample file doesn't contain an audio track, but only a video track, therefore the pipeline must be modified to transmit the video track only, by removing the last part

gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream filesrc
location=test.mp4 ! qtdemux name=d d.video_0 ! queue ! s.sink_0

[mram0509]

I have tried with above pipeline and am seeing the same issue. So I changed test file to a video with audio. and turend on debug level to 2 for Gstreamer. The ffmpeg pipeline work beautifully. Maybe there is an issue with my Gstreamer setup.
but here's the log I get with a video with audio:
gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream filesrc location=bigbuckbunny.mp4 ! qtdemux name=d d.video_0 ! queue ! s.sink_0 d.audio_0 ! queue ! s.sink_1
Setting pipeline to PAUSED ...
0:00:00.019775650 7195 0x556243d4cd50 WARN basesrc gstbasesrc.c:3583:gst_base_src_start_complete: pad not activated yet
Pipeline is PREROLLED ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://localhost:8554/mystream
0:00:00.020325451 7195 0x556243d4f8a0 WARN qtdemux qtdemux.c:3031:qtdemux_parse_trex: failed to find fragment defaults for stream 1
0:00:00.020474694 7195 0x556243d4f8a0 WARN qtdemux qtdemux.c:3031:qtdemux_parse_trex: failed to find fragment defaults for stream 2
Progress: (open) Retrieving server options
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending RECORD request
Progress: (record) Sending server stream info
Progress: (request) SETUP stream 1
0:00:00.044645805 7195 0x556243d4fa80 ERROR rtspclientsink gstrtspclientsink.c:3679:gst_rtsp_client_sink_create_transports_string: No supported profiles configured
0:00:00.044683065 7195 0x556243d4fa80 WARN rtspclientsink gstrtspclientsink.c:4119:gst_rtsp_client_sink_setup_streams: error: Could
not setup transport.
0:00:00.044735894 7195 0x556243d4fa80 ERROR rtspclientsink gstrtspclientsink.c:4430:gst_rtsp_client_sink_record: setup failed
ERROR: from element /GstPipeline:pipeline0/GstRTSPClientSink:s: Could not get/set settings from/on resource.
Additional debug info:
gstrtspclientsink.c(4119): gst_rtsp_client_sink_setup_streams (): /GstPipeline:pipeline0/GstRTSPClientSink:s:
Could not setup transport.
0:00:00.044807672 7195 0x556243d4fa80 WARN rtspclientsink gstrtspclientsink.c:2569:gst_rtsp_client_sink_loop: we are not connected
Execution ended after 0:00:00.011429772
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
0:00:00.046091500 7195 0x556243d4fa80 WARN rtspclientsink gstrtspclientsink.c:2893:gst_rtsp_client_sink_try_send: receive interrupted
0:00:00.046157441 7195 0x556243d4fa80 WARN rtspclientsink gstrtspclientsink.c:3272:gst_rtsp_client_sink_open: Failed to connect to server
Setting pipeline to NULL ...
Freeing pipeline ...

[cmiller-nflux]

@CACUser
If you are looking for an answer to this, just run the following.

gst-launch-1.0 videotestsrc ! x264enc ! rtspclientsink protocols=tcp location=rtsp://localhost:8554/app/test

You just need to add tcp protocol to the sink.

[sachitha2]

How can I add webcam feed or mp4 to the feed ?

[aler9]

@sachitha2 please follow Gstreamer tutorials for that.