<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Hi Fabio,<div class=""><br class=""></div><div class="">First of all it could be a bandwidth problem.</div><div class="">Because the GStreamer implementation does not do adaptive streaming. In general your RTSP camera will try to use much more bandwidth than the videotestsrc (even for the same resolution).</div><div class=""><br class=""></div><div class="">Also I think RSTP and WebRTC are both using RTP but in slightly different ways. So you might need to add rtph264depay ! rtph264pay to make it work. You can try to dump the SDP offer in both case to compare.</div><div class=""><br class=""></div><div class="">No idea about your problem on ARM.<br class=""><div><br class=""><blockquote type="cite" class=""><div class="">Le 18 avr. 2020 à 17:14, Fabio Zingaro <<a href="mailto:fabio.zingaro@gmail.com" class="">fabio.zingaro@gmail.com</a>> a écrit :</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><div dir="ltr" class="">Hi all,</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">no one has ideas and suggestions?<br class=""></div></div><br class=""><div class="gmail_quote"><div dir="ltr" class="gmail_attr">Il giorno mar 14 apr 2020 alle ore 16:18 Fabio Zingaro <<a href="mailto:fabio.zingaro@gmail.com" class="">fabio.zingaro@gmail.com</a>> ha scritto:<br class=""></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr" class=""><div dir="ltr" class="">Hi Stephenwei,<div class=""><br class=""></div><div class="">yes, i've no error on webrtc part.</div><div class="">Please note that with this chain i receive the video test stream without problems.</div><div class=""><br class=""></div><div class=""><div class=""><i class=""><font color="#666666" class="">receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "</font></i></div><div class=""><i class=""><font color="#666666" class=""> " videotestsrc ! "</font></i></div><div class=""><i class=""><font color="#666666" class=""> " video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max$</font></i></div><div class=""><i class=""><font color="#666666" class=""> "rtph264pay config-interval=-1 name=payloader ! "</font></i></div><div class=""><i class=""><font color="#666666" class=""> "application/x-rtp,media=video,encoding-name=H264,payload="</font></i></div><div class=""><i class=""><font color="#666666" class=""> RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);</font></i></div></div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><br class=""></div></div></div><br class=""><div class="gmail_quote"><div dir="ltr" class="gmail_attr">Il giorno mar 14 apr 2020 alle ore 03:21 Stephenwei <<a href="mailto:lofy.stephen@gmail.com" target="_blank" class="">lofy.stephen@gmail.com</a>> ha scritto:<br class=""></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">receiver_entry->pipeline = gst_parse_launch (" rtspsrc location=//" RTSPSRC "<br class="">
"<br class="">
"! webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "<br class="">
"! queue ! rtph264depay ! rtph264pay !<br class="">
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin. ",<br class="">
&error);<br class="">
<br class="">
webrtcbin <--- Make sure gst-inspect-1.0 can find this element.<br class="">
<br class="">
<br class="">
<br class="">
-----<br class="">
GStreamer is a convenient multimedia platform, I like it. <br class="">
Develop the NVR system on ARM/x86(c/python)<br class="">
Use python to generate NVR is crazy, of course works fine.<br class="">
<br class="">
--<br class="">
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