<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:12pt"><div>Hello everyone.</div><div><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;">I'm now trying to establish a rtsp server based on GStreamer 0.10.7 ( Ossbuild ) and gst-rtsp-server-RELEASE-0.10.8. My platform is Windows7 and develop tool is VS2010 Express. I had adopt a sort part of code to Winsock and compile without problem.</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;"><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica,
Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;">I wrote a simple console problem, basically exactly the same with http://0rz.tw/GZuvZ but without audio pipeline.</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;"><span style="background-color: transparent;"><br></span></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;"><span style="background-color: transparent;">Then as I start to run server. I always got client says timeout. I tried GStreamer 0.10.7, & 1.2 & VLC 2.1.0</span></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande',
sans-serif; background-color: transparent; font-style: normal;"><span style="background-color: transparent;"><br></span></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;"><span style="background-color: transparent;">From different RTSP Clients, I got fellow messages:</span></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;"><span style="background-color: transparent;"><br></span></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; background-color: transparent; font-style: normal;"><span style="background-color: transparent;">GStreamer 0.10.7:</span></div><div
style="background-color: transparent;">gst-launch -v rtspsrc location=rtsp://127.0.0.1:554 ! rtph264depay ! queue2 ! h264parse ! ffdec_h264 ! d3dvideosink</div><div style="background-color: transparent;">ImportError: No module named pygtk</div><div style="background-color: transparent;">ImportError: No module named pygtk</div><div style="background-color: transparent;">Setting pipeline to PAUSED ...</div><div style="background-color: transparent;">ERROR: Pipeline doesn't want to pause.</div><div style="background-color: transparent;">ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Resource not found.</div><div style="background-color: transparent;">Additional debug info:</div><div style="background-color: transparent;">..\..\..\..\..\Source\gst-plugins-good\gst\rtsp\gstrtspsrc.c(4637): gst_rtspsrc_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:</div><div style="background-color: transparent;">Not Found</div><div
style="background-color: transparent;"><span style="background-color: transparent;"></span></div><div style="background-color: transparent;">Freeing pipeline ...</div><div style="background-color: transparent;"><br></div><div style="background-color: transparent; color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal;">GStreamer 1.2:<br></div><div style="background-color: transparent;">gst-launch rtspsrc location=rtsp://10.100.0.84:554 ! rtph264depay ! queue2 ! h264parse ! avdec_h264 ! d3dvideosink</div><div style="background-color: transparent;">Setting pipeline to PAUSED ...</div><div style="background-color: transparent;">Pipeline is live and does not need PREROLL ...</div><div style="background-color: transparent;">Progress: (open) Opening Stream</div><div style="background-color: transparent;">Progress: (connect) Connecting to
rtsp://10.100.0.84:554</div><div style="background-color: transparent;">Progress: (open) Retrieving server options</div><div style="background-color: transparent;">Progress: (open) Retrieving media info</div><div style="background-color: transparent;">ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not read from resource.</div><div style="background-color: transparent;">Additional debug info:</div><div style="background-color: transparent;">gstrtspsrc.c(4955): gst_rtspsrc_try_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:</div><div style="background-color: transparent;">Could not receive message. (Timeout while waiting for server response)</div><div style="background-color: transparent;">ERROR: pipeline doesn't want to preroll.</div><div style="background-color: transparent;">Setting pipeline to PAUSED ...</div><div style="background-color: transparent;">Setting pipeline to READY ...</div><div style="background-color:
transparent;">Setting pipeline to NULL ...</div><div style="background-color: transparent;">Freeing pipeline ...</div><div style="background-color: transparent;"><br></div><div style="background-color: transparent; color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal;">VLC 2.1.0:</div><div style="background-color: transparent;">main debug: processing request item: rtsp://10.100.0.84:554/test, node: 播放清單, skip: 0</div><div style="background-color: transparent;">main debug: resyncing on rtsp://10.100.0.84:554/test</div><div style="background-color: transparent;">main debug: rtsp://10.100.0.84:554/test is at 4</div><div style="background-color: transparent;">main debug: starting playback of the new playlist item</div><div style="background-color: transparent;">main debug: resyncing on rtsp://10.100.0.84:554/test</div><div style="background-color:
transparent;">main debug: rtsp://10.100.0.84:554/test is at 4</div><div style="background-color: transparent;">main debug: creating new input thread</div><div style="background-color: transparent;">main debug: Creating an input for 'rtsp://10.100.0.84:554/test'</div><div style="background-color: transparent;">main debug: using timeshift granularity of 50 MiB, in path 'C:\Users\John Smith\AppData\Local\Temp'</div><div style="background-color: transparent;">main debug: `rtsp://10.100.0.84:554/test' gives access `rtsp' demux `' path `10.100.0.84:554/test'</div><div style="background-color: transparent;">main debug: creating demux: access='rtsp' demux='' location='10.100.0.84:554/test' file='\\10.100.0.84:554\test'</div><div style="background-color: transparent;">main debug: looking for access_demux module matching "rtsp": 12 candidates</div><div style="background-color: transparent;">live555 debug: version 2012.12.18</div><div style="background-color:
transparent;">qt4 debug: IM: Setting an input</div><div style="background-color: transparent;">live555 debug: connection timeout</div><div style="background-color: transparent;">live555 error: Failed to connect with rtsp://10.100.0.84:554/test</div><div style="background-color: transparent;">main debug: no access_demux modules matched</div><div style="background-color: transparent;">main debug: creating access 'rtsp' location='10.100.0.84:554/test', path='\\10.100.0.84:554\test'</div><div style="background-color: transparent;">main debug: looking for access module matching "rtsp": 20 candidates</div><div style="background-color: transparent;">main debug: net: connecting to 10.100.0.84 port 554</div><div style="background-color: transparent;">main debug: connection succeeded (socket = 1572)</div><div style="background-color: transparent;">access_realrtsp debug: rtsp connected</div><div style="background-color: transparent;">access_realrtsp warning: only
real/helix rtsp servers supported for now</div><div style="background-color: transparent;">main debug: no access modules matched</div><div style="background-color: transparent;">main error: open of `rtsp://10.100.0.84:554/test' failed</div><div style="background-color: transparent;">main debug: finished input</div><div style="background-color: transparent;">main debug: dead input</div><div style="background-color: transparent;">main debug: changing item without a request (current 4/5)</div><div style="background-color: transparent;">main debug: nothing to play</div><div style="background-color: transparent;">qt4 debug: IM: Deleting the input</div><div style="background-color: transparent;"><br></div><div style="background-color: transparent; color: rgb(0, 0, 0); font-size: 16px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal;">It seems like that client is waiting for sever sending media
related information but timeout. Did anyone have succeeded running GStreamer on Windows before? Did I missed something? Thanks. </div></div></body></html>