rtsp server, save record stream to a file

Matthew Waters ystreet00 at gmail.com
Mon Sep 3 05:55:43 UTC 2018


On 3/9/18 3:37 pm, Dinh Nguyen wrote:
> Hello Nicolas,
>
> Thanks for your response.
>
> I am trying with your suggestion
> `gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay ! rtspclientsink
> location=rtsp://127.0.0.1:8554/test` <http://127.0.0.1:8554/test`>
>
> however, seems that the pipe itself get error
> `WARNING: erroneous pipeline: could not link rtph264pay0 to
> rtspclientsink0`.
>
> I am not sure I get your suggestion right?


rtspclientsink already includes payloaders so you don't need a
rtph264pay there.

What you're missing is that the name of the elements need to match a
specific format. i.e. the depayloader in your test-record command line
needs to be named 'depay0'.  You also may need to modify the test-record
example to send an EOS on shutdown so that mp4 headers are finalized
correctly as the resulting fiel will not really be playable as-is. 
Another option is to attempt to attempt enable robust-muxing mode.

Cheers
-Matt

> Thanks.
>
>
>
> On Mon, Sep 3, 2018 at 2:09 AM Nicolas Dufresne <nicolas at ndufresne.ca
> <mailto:nicolas at ndufresne.ca>> wrote:
>
>
>
>     Le sam. 1 sept. 2018 23:57, Dinh Nguyen <dinh.nguyen at zinnoinc.com
>     <mailto:dinh.nguyen at zinnoinc.com>> a écrit :
>
>         Hello,
>
>         I am new to GStreamer ecosystem, and nowhere to be good at C;
>         hence, please forgive if I put this silly question here.
>
>         I am using gst-rtsp server, with test-record to save a
>         "record" stream from client to a file (mp4), and this is the
>         error.
>
>         |GST_DEBUG=2 ./build/test-record "( rtph264depay ! h264parse !
>         mp4mux ! filesink location=result3.mp4 )" stream ready at
>         rtsp://127.0.0.1:8554/test <http://127.0.0.1:8554/test> On the
>         sender, send a stream with rtspclientsink: gst-launch-1.0
>         videotestsrc ! x264enc ! rtspclientsink
>         location=rtsp://127.0.0.1:8554/test
>         <http://127.0.0.1:8554/test> 0:00:09.120079875 951 0xd0a190
>         ERROR rtspmedia rtsp-media.c:3637:default_handle_sdp:
>         0x7f93380452b0: Media has more or less streams than SDP (0 /=
>         1) 0:00:09.120101464 951 0xd0a190 ERROR rtspclient
>         rtsp-client.c:2946:handle_sdp: client 0xe551a0: could not
>         handle SDP 0:00:09.120110777 951 0xd0a190 ERROR rtspclient
>         rtsp-client.c:3107:handle_announce_request: client 0xe551a0:
>         can't handle SDP |
>
>>
>         This is the client side:
>
>         |gst-launch-1.0 videotestsrc ! x264enc ! rtspclientsink |
>
>     To rtspclientsink you should provide one or more RTP stream. To
>     change h264 into RTP, you need to add rtph264pay.
>
>         |location=rtsp://127.0.0.1:8554/test
>         <http://127.0.0.1:8554/test> Setting pipeline to PAUSED ...
>         Pipeline is PREROLLED ... Progress: (open) Opening Stream
>         Progress: (connect) Connecting to rtsp://127.0.0.1:8554/test
>         <http://127.0.0.1:8554/test> Redistribute latency... 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 ERROR: from
>         element
>         /GstPipeline:pipeline0/GstRTSPClientSink:rtspclientsink0:
>         Could not read from resource. Additional debug info:
>         gstrtspclientsink.c(3059): gst_rtsp_client_sink_send ():
>         /GstPipeline:pipeline0/GstRTSPClientSink:rtspclientsink0: Got
>         error response: 415 (Unsupported Media Type). Execution ended
>         after 0:00:00.046342782 Setting pipeline to PAUSED ... Setting
>         pipeline to READY ... Setting pipeline to NULL ... Freeing
>         pipeline ... |
>
>>
>         Could you please give me a hint for this problems.
>
>         Thanks in advanced.
>
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