gstreamer-devel Digest, Vol 131, Issue 21

[Howling wong]

Hi it is my mistake ... it should be width=640 instead of width=6400 in writing the post... the pipeline do not work with width=640

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From: gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> on behalf of gstreamer-devel-request at lists.freedesktop.org <gstreamer-devel-request at lists.freedesktop.org>
Sent: Friday, December 17, 2021 11:22 AM
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Subject: gstreamer-devel Digest, Vol 131, Issue 21

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Today's Topics:

   1. Gstreamer: No RTP protocol present (Howling wong)
   2. Re: Gstreamer: No RTP protocol present (Robert Hensel)


----------------------------------------------------------------------

Message: 1
Date: Fri, 17 Dec 2021 02:55:33 +0000
From: Howling wong <watertreader at hotmail.com>
To: "gstreamer-devel at lists.freedesktop.org"
        <gstreamer-devel at lists.freedesktop.org>
Subject: Gstreamer: No RTP protocol present
Message-ID:
        <SI2PR04MB5182E8B2DF3B9669E94C5BC1CB789 at SI2PR04MB5182.apcprd04.prod.outlook.com>

Content-Type: text/plain; charset="iso-8859-1"

I am having some issues with the following gstreamer command

Sender (on embeeded system)

gst-launch-1.0 videotestsrc ! video/x-raw, width=6400, height=480 ! queue ! vpuenc_h264 ! rtph264pay ! udpsink host=192.168.60.5 port=5555


Receiver(on windows)

gst-launch-1.0 udpsrc port=5555 ! queue ! rtph264depay  ! decodebin  ! autovideosink


But I have got the following response


   Setting pipeline to PAUSED ...

    Pipeline is live and does not need PREROLL ...

    Got context from element 'autovideosink0': gst.d3d11.device.handle=context, device=(GstD3D11Device)"\(GstD3D11Device\)\ d3d11device4", adapter=(uint)0, device-id=(uint)6429, vendor-id=(uint)32902, hardware=(boolean)true, description=(string)"Intel\(R\)\ HD\ Graphics\ P530";

    Pipeline is PREROLLED ...

    Setting pipeline to PLAYING ...

    New clock: GstSystemClock

     ERROR: from element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: No RTP format was negotiated.

    Additional debug info:

      ../gst-libs/gst/rtp/gstrtpbasedepayload.c(538): gst_rtp_base_depayload_handle_buffer (): /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:

     Input buffers need to have RTP caps set on them. This is usually achieved by setting the 'caps' property of the upstream source element (often udpsrc or appsrc), or by putting a capsfilter element before the depayloader and setting the 'caps' property on that. Also see http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README

     Execution ended after 0:00:00.019641000
    Setting pipeline to NULL ...
    ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data stream error.
    Additional debug info:
    ../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0
    streaming stopped, reason not-negotiated (-4)
    Freeing pipeline ..


The complaint seem to be about the incoming stream is not in rtp format and the rtpdepayh264 should not be placed in the pipeline. This assumption is proven to be correct when i replaced the whole pipeline with a fakesink


Receiver

gst-launch-1.0 udpsrc port=5555 ! queue ! fakesink


The pipeline work. However when i observed the packets exchange in wireshark, it show the communication exchange but the protocol is in udp. Though I know that RTP could be based upon UDP protocol but have thought that Wireshark is entirely capable of showing protocol in RTP format


I have thought that the sender has already wrapped the video in rtp format before sending the package out. Like to have some ideas on what is wrong here and how to proceed

Regards

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Message: 2
Date: Fri, 17 Dec 2021 14:21:49 +1100
From: Robert Hensel <vk3eht at gmail.com>
To: Discussion of the development of and with GStreamer
        <gstreamer-devel at lists.freedesktop.org>
Subject: Re: Gstreamer: No RTP protocol present
Message-ID:
        <CAGRXgDxxb4BrjW+yT9fdudoYdiQjs0JCxpLAJQx6_sMjXHoOMQ at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

width=6400?
Suggest try with width=640

Rob

On Fri, 17 Dec 2021 at 13:55, Howling wong via gstreamer-devel <
gstreamer-devel at lists.freedesktop.org> wrote:

> I am having some issues with the following gstreamer command
>
> Sender (on embeeded system)
>
> gst-launch-1.0 videotestsrc ! video/x-raw, width=6400, height=480 ! queue ! vpuenc_h264 ! rtph264pay ! udpsink host=192.168.60.5 port=5555
>
> Receiver(on windows)
>
> gst-launch-1.0 udpsrc port=5555 ! queue ! rtph264depay  ! decodebin  ! autovideosink
>
> But I have got the following response
>
>
>    Setting pipeline to PAUSED ...
>
>     Pipeline is live and does not need PREROLL ...
>
>     Got context from element 'autovideosink0':
> gst.d3d11.device.handle=context,
> device=(GstD3D11Device)"\(GstD3D11Device\)\ d3d11device4", adapter=(uint)0,
> device-id=(uint)6429, vendor-id=(uint)32902, hardware=(boolean)true,
> description=(string)"Intel\(R\)\ HD\ Graphics\ P530";
>
>     Pipeline is PREROLLED ...
>
>     Setting pipeline to PLAYING ...
>
>     New clock: GstSystemClock
>
>      ERROR: from element
> /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: No RTP format was
> negotiated.
>
>     Additional debug info:
>
>       ../gst-libs/gst/rtp/gstrtpbasedepayload.c(538):
> gst_rtp_base_depayload_handle_buffer ():
> /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:
>
>      Input buffers need to have RTP caps set on them. This is usually
> achieved by setting the 'caps' property of the upstream source element
> (often udpsrc or appsrc), or by putting a capsfilter element before the
> depayloader and setting the 'caps' property on that. Also see
> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README
>      Execution ended after 0:00:00.019641000
>     Setting pipeline to NULL ...
>     ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal
> data stream error.
>     Additional debug info:
>     ../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop ():
> /GstPipeline:pipeline0/GstUDPSrc:udpsrc0
>     streaming stopped, reason not-negotiated (-4)
>     Freeing pipeline ..
>
>
> The complaint seem to be about the incoming stream is not in rtp format
> and the rtpdepayh264 should not be placed in the pipeline. This assumption
> is proven to be correct when i replaced the whole pipeline with a fakesink
>
>
> Receiver
>
> gst-launch-1.0 udpsrc port=5555 ! queue ! fakesink
>
> The pipeline work. However when i observed the packets exchange in
> wireshark, it show the communication exchange but the protocol is in udp.
> Though I know that RTP could be based upon UDP protocol but have thought
> that Wireshark is entirely capable of showing protocol in RTP format
>
>
> I have thought that the sender has already wrapped the video in rtp format
> before sending the package out. Like to have some ideas on what is wrong
> here and how to proceed
>
> Regards
>
>
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