<html> <head> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <style type="text/css" style="display:none;"> P {margin-top:0;margin-bottom:0;} </style> </head> <body dir="ltr"> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> 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</div> <div> <div id="appendonsend"></div> <div style="font-family:Calibri,Helvetica,sans-serif; font-size:12pt; color:rgb(0,0,0)"> </div> <hr tabindex="-1" style="display:inline-block; width:98%"> <div id="divRplyFwdMsg" dir="ltr">From: gstreamer-devel <gstreamer-devel-bounces@lists.freedesktop.org> on behalf of gstreamer-devel-request@lists.freedesktop.org <gstreamer-devel-request@lists.freedesktop.org> Sent: Friday, December 17, 2021 11:22 AM To: gstreamer-devel@lists.freedesktop.org <gstreamer-devel@lists.freedesktop.org> Subject: gstreamer-devel Digest, Vol 131, Issue 21 <div> </div> </div> <div class="BodyFragment"> <div class="PlainText">Send gstreamer-devel mailing list submissions to gstreamer-devel@lists.freedesktop.org To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel"> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to gstreamer-devel-request@lists.freedesktop.org You can reach the person managing the list at gstreamer-devel-owner@lists.freedesktop.org When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." 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@hotmail.com> To: "gstreamer-devel@lists.freedesktop.org" <gstreamer-devel@lists.freedesktop.org> Subject: Gstreamer: No RTP protocol present Message-ID: <SI2PR04MB5182E8B2DF3B9669E94C5BC1CB789@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 <a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README"> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README</a> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/4681e121/attachment-0001.htm">https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/4681e121/attachment-0001.htm</a>> ------------------------------ Message: 2 Date: Fri, 17 Dec 2021 14:21:49 +1100 From: Robert Hensel <vk3eht@gmail.com> To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Subject: Re: Gstreamer: No RTP protocol present Message-ID: <CAGRXgDxxb4BrjW+yT9fdudoYdiQjs0JCxpLAJQx6_sMjXHoOMQ@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@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 > <a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README"> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README</a> > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... 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