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<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial",sans-serif">Hi, I do something similar by routing my already existing rtp stream via appsink to an rtsp_server.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial",sans-serif">The server is created using gst_rtsp_media_factory_launch and looks like this<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial",sans-serif">Appsrc name=MySrc min-latency=2000000000 is-live=true ! queue ! rtphpay name=pay0 config-interval= 10 pt=96<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial",sans-serif">I create the rtsp server and get a handle to appsrc using element=gst_rtsp_media_get_element(media) followed by gst_bin_get_by_name_recurse_up(element, MySrc);<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial",sans-serif">There are probably better ways but this one works for me.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Von:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> gstreamer-devel [mailto:gstreamer-devel-bounces@lists.freedesktop.org]
<b>Im Auftrag von </b>Martin Jackson<br>
<b>Gesendet:</b> Freitag, 30. Juni 2017 09:51<br>
<b>An:</b> Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org><br>
<b>Betreff:</b> gst-rtsp-server: rebroadcasting existing RTP streams<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Dear gstreamer developers,<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">I'm trying to use the gst-rtsp-server to slurp in existing RTP streams in various formats and make these available to external clients with the RTSP protocol.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">However, the general design of the gst-rtsp-server seems to be that you have some media streams e.g. h264 + aac, for which you create payloader elements, and the server does all the rest for you. This is great, unless you just want to route
some existing RTP streams, in which case it gets in the way a bit! I'm really impressed with the rtsp protocol handling and RTP infrastructure of gstreamer, so I think there's a lot of gain to be had here from gst-rtsp-server.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Does anyone have any advice on how to do such a thing? The best I have come up with so far is to use identity elements as the payloaders, which is obviously less than ideal!<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Best regards, many thanks,<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Martin Jackson<o:p></o:p></p>
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