Gstreamer: No RTP protocol present

[Robert Hensel]

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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