[gstreamer-bugs] [Bug 549409] New: gstrtpbin don't stop at the end of a stream

GStreamer (bugzilla.gnome.org) bugzilla-daemon at bugzilla.gnome.org
Tue Aug 26 02:34:14 PDT 2008


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  GStreamer | gst-plugins-bad | Ver: 0.10.x
           Summary: gstrtpbin don't stop at the end of a stream
           Product: GStreamer
           Version: 0.10.x
          Platform: Other
        OS/Version: All
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-bad
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: remi.buisson at viotech.net
         QAContact: gstreamer-bugs at lists.sourceforge.net
     GNOME version: Unspecified
   GNOME milestone: Unspecified


Please describe the problem:
When a file is send using gstrtpbin and when the end of it is reached,
gstrtpbin don't stop and I have to send gstreamer a SIGTERM signal.

Steps to reproduce:
on th server side :
gst-launch -v gstrtpbin name=rtpbin \
    filesrc location=filesrc
location=../../../partage/Videos/superman_originale.avi ! decodebin name=dec \
    dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay !
rtpbin.send_rtp_sink_0 \
    rtpbin.send_rtp_src_0 ! udpsink port=5000 host=127.0.0.1 ts-offset=0
name=vrtpsink \
    rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=127.0.0.1 sync=false
async=false name=vrtcpsink \
    udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \
    dec. ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay !
rtpbin.send_rtp_sink_1 \
    rtpbin.send_rtp_src_1 ! udpsink port=5002 host=127.0.0.1 ts-offset=0
name=artpsink \
    rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=127.0.0.1 sync=false
async=false name=artcpsink \
    udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1

on the client side :
gst-launch -vvv playbin uri=file:///home/user/test/gstreamer/client.sdp

with client.sdp:
v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
m=audio 5002 RTP/AVP 8
c=IN IP4 127.0.0.1

Actual results:
the server side don't stop

Expected results:
the server stop

Does this happen every time?
yes

Other information:


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