[EXTERNAL] Re: Working example of RTMP stream to Gstreamer RTSP server?

[evaluat0r]

John Deutscher wrote
> Thanks Tim,
> 
> I tried the gst-play-1.0, but unfortunately I am running this on a
> container in a cloud instance and would need to setup a linux box to test
> that out with. 
> 
> I did check the test-uri example and got the following output. Looks like
> it "failed reading a tag" as a warning, and then "removed the /test mount
> point" immediately after that.  The stream is not live any longer for
> testing, but I can turn it on again.  Stream path is just OBS Studio->
> ngnix RTMP server module -> test-uri right now...
> 
> 
> ./test-uri rtmp://johndeu-gstreamer.westus.azurecontainer.io/live/test 
> stream ready at rtsp://127.0.0.1:8554/test
> 0:00:09.452060177   610 0x7fbfd401ed90 WARN                flvdemux
> gstflvdemux.c:659:gst_flv_demux_parse_tag_script:
> <flvdemux0>
>  failed reading a tag, skipping
> removing /test mount point
> 0:00:10.230779826   610 0x5563212ea190 ERROR             rtspclient
> rtsp-client.c:565:find_media: client 0x556321127580: no factory for path
> /test/stream=0
> 0:00:10.230940728   610 0x5563212ea190 ERROR             rtspclient
> rtsp-client.c:1592:handle_setup_request: client 0x556321127580: media
> '/test/stream=0' not found
> 0:00:10.255076503   610 0x5563212ea190 ERROR             rtspclient
> rtsp-client.c:565:find_media: client 0x556321127580: no factory for path
> /test/stream=1
> 0:00:10.255181503   610 0x5563212ea190 ERROR             rtspclient
> rtsp-client.c:1592:handle_setup_request: client 0x556321127580: media
> '/test/stream=1' not found
> 0:00:10.311535712   610 0x5563212ea190 ERROR             rtspclient
> rtsp-client.c:565:find_media: client 0x7fbfd8061f70: no factory for path
> /test
> 0:00:10.311642713   610 0x5563212ea190 ERROR             rtspclient
> rtsp-client.c:1592:handle_setup_request: client 0x7fbfd8061f70: media
> '/test' not found
> 
> -----Original Message-----
> From: Tim Müller <

> tim@

> > 
> Sent: Wednesday, July 29, 2020 5:02 PM
> To: 

> gstreamer-devel at .freedesktop

> Subject: [EXTERNAL] Re: Working example of RTMP stream to Gstreamer RTSP
> server?
> 
> On Mon, 2020-07-27 at 23:30 +0000, John Deutscher wrote:
> 
> Hi John,
> 
> Bit difficult to know what's going on without having the input stream to
> test with at hand, so just a couple of questions / comments /
> suggestions:
> 
> - Did you try playing back the stream with GStreamer (e.g. gst-play-
> 1.0) instead of VLC?
> 
> - Have you tried the test-uri example for comparison? You should be able
> to pass it the rtmp:// URI directly, and it should relay the data without
> transcoding.
> 
> - Does the rtmp input stream play with gst-play-1.0 rtmp://.. (quick
> sanity check)
> 
> - Instead of using flvdemux and two decodebins you should be able to use a
> single decodebin which will give you a decoded audio + video stream
> (hopefully); or uridecodebin which includes the rtmpsrc as well then. Or
> even the (uri)decodebin3 variants.
> 
> - How does VLC "fail"? Any errors? Does it show a window in the right size
> but without the video?
> 
> - If you don't want the audio, you *should* be able to just not link the
> audio pad - that *should* work fine as long as the video pad is linked.
> (Though I'm only 90% sure that flvdemux will handle this right, I have a
> vague memory of a bug about that at some point).
> 
> - That fakesink in your rtsp server pipeline could mess up some things a
> little, maybe setting it to async=false helps (if just dropping the audio
> branch is not possible)
> 
> Cheers
>  Tim
> 
> --
> Tim Müller, Centricular Ltd -
> https://nam06.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.centricular.com%2F&data=02%7C01%7CJohn.Deutscher%40microsoft.com%7Ce53cb0b136d149e6df5c08d8341bd86e%7C72f988bf86f141af91ab2d7cd011db47%7C1%7C0%7C637316641483449525&sdata=7VQZWEgaZ3620Qt5%2BrtCQHvYQBbcHy0Hv5eAFEnIFoc%3D&reserved=0
> 
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You should bump up your logging so you can see if a failure happens
somewhere else. Run your command with GST_DEBUG="*rtsp*:4,*rtmp*:4"
./the_command

Increasing logging on relevant elements to verify everything is working as
expected (ingest rtmp, demux to get video and audio, repackage video and
audio in rtp to be sent out on behalf of rtsp)



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