rtsp server, save record stream to a file
Matthew Waters
ystreet00 at gmail.com
Mon Sep 3 15:00:41 UTC 2018
That is known from experience and looking at various outputs such as
'gst-inspect-1.0 $ELEMENT' or GST_DEBUG=3 or even GST_DEBUG=*PAD*:5,3
For the element naming, the test-record --help output is useful for this.
Cheers
-Matt
On 4/9/18 12:57 am, Dinh Nguyen wrote:
> Hello Matthew,
>
> Thanks a lot, it is working for me know.
>
> Well, the pipeline elements are so confusing for me. How could one
> know that after this element should be that element; is there
> something like interface?
>
> Thanks again.
>
>
> On Mon, Sep 3, 2018 at 12:55 PM Matthew Waters <ystreet00 at gmail.com
> <mailto:ystreet00 at gmail.com>> wrote:
>
> 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.
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org
>> <mailto:gstreamer-devel at lists.freedesktop.org>
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org
>> <mailto:gstreamer-devel at lists.freedesktop.org>
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org <mailto:gstreamer-devel at lists.freedesktop.org>
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20180904/3083155f/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 488 bytes
Desc: OpenPGP digital signature
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20180904/3083155f/attachment-0001.sig>
More information about the gstreamer-devel
mailing list