<div dir="ltr">Hi<div><br></div><div>I found out that when connecting to the camera, ``new_sample`` callback in ``<span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">gst-plugins-good-1.14.1/</span><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">tests/examples/rtsp/test-onvif`` if never called (it gets called when connected to the local rtsp server (``<span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">gst-rtsp-server-1.14.1/</span><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">examples/test-onvif-</span><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">backchannel``). </span></span></div><div><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">I do not understand why this the chain ("audiotestsrc volume=0.99 ! mulawenc ! rtppcmupay ! appsink name=out") already depends on the connection.</span></span></div><div><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></span></div><div><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">Any thoughts?</span></span></div><div><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></span></div><div><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">Best</span></span></div><div><span style="text-align:left;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span style="text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">Stefan</span></span></div><br><div class="gmail_quote"><div dir="ltr">On Fri, Jun 22, 2018 at 5:58 PM Stefan Ulbrich <<a href="mailto:stefan.ulbrich@acceptto.com">stefan.ulbrich@acceptto.com</a>> wrote:<br></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">Hi List<div><br></div><div>I want to send audio to an onvif-compliant IP camera (Amcrest) and therefore, I have compiled gstreamer-1.14.1-* on Ubuntu 16.04 (installed to /usr/local). I now tried:</div><div><ol><li>./gst-plugins-good-1.14.1/tests/examples/rtsp/test-onvif rtsp://<a href="http://admin:XXXX@192.168.2.116:554/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvi" target="_blank">admin:XXXX@192.168.2.116:554/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvi</a><br>But nothing is audible on the camera. While the program is running, I cannot send audio via the manufacturer's android app so I think the connection is established.<br></li><li>I changed the audio signal in test-onvif.c to a sine signal (by removing `wave=red-noise`). Without suggess</li><li>Wireshark shows that there is data exchanged between ubuntu and the camera<br><span style="color:rgb(0,0,0);font-family:Consolas,"Lucida Console",monospace;font-size:12.8px;text-align:start;background-color:rgb(240,247,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">`58422     149.274903247   192.168.2.125   192.168.2.116   RTCP    126     Receiver Report</span><span style="color:rgb(0,0,0);font-family:Consolas,"Lucida Console",monospace;font-size:12.8px;text-align:start;background-color:rgb(240,247,255);text-decoration-style:initial;text-decoration-color:initial">  </span><span style="color:rgb(0,0,0);font-family:Consolas,"Lucida Console",monospace;font-size:12.8px;text-align:start;background-color:rgb(240,247,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><span> </span>Source description`</span><span style="color:rgb(0,0,0);font-family:Consolas,"Lucida Console",monospace;font-size:12.8px;text-align:start;background-color:rgb(240,247,255);text-decoration-style:initial;text-decoration-color:initial"> </span><br></li><li>I build the gst-rtsp-server-1.14.1/examples/test-onvif-backchannel and replaced `fakesink` with `pulseaudiosink` and confirmed `/usr/local/bin/gst-launch-1.0 audiotestsrc ! pulsesink` works. `test-onvif` connects but again, I cannot hear any sound.</li></ol><div>How can I trace/narrow down the problem? </div><div><br></div><div>Best</div><div>Stefan</div></div></div>
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