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Bug 550801 - rtsp client disconnects from server after few seconds normal of working
rtsp client disconnects from server after few seconds normal of working
Status: RESOLVED FIXED
Product: GStreamer
Classification: Platform
Component: gst-plugins-good
git master
Other All
: Normal major
: 0.10.10
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2008-09-04 09:06 UTC by Ilia S. Smelykh
Modified: 2008-09-25 14:27 UTC
See Also:
GNOME target: ---
GNOME version: 2.21/2.22


Attachments
result of GST_DEBUG=*rtp*:5,*rtsp*:5 gst-launch rtspsrc location="url" debug=1 ! fakesink -v >debug.log 2>&1 (19.44 KB, application/x-gzip)
2008-09-04 09:45 UTC, Ilia S. Smelykh
Details

Description Ilia S. Smelykh 2008-09-04 09:06:24 UTC
Please describe the problem:
I tried to use gstreamer as rtsp client, but after simple test (`gst-launch rtspsrc location="url" ! fakesink`) I could see that server close the connection after few seconds of playing, i think that gstreamer rtsp module doesn't send rtcp requests for supporting session. During the playing video stream pass though gstreamer elements well. This bug cloud be checked on the live555MediaServer.

Before start playing(before rtsp gstreamer module send PLAY request and server send response OK) all rtsp command is valid, you can assure in it by specifying debug=1 parameter in rtspsrc.

Steps to reproduce:
1. run streaming server, live555MediaServer for example
2. execute next command: "gst-launch rtspsrc location="url" debug=1 ! fakesink

or you can just try to connect using totem video player


Actual results:
video streaming server disconnects

Expected results:
video streaming server doesn't disconnects

Does this happen every time?
This happen every time

Other information:
gstreamer-plugins-good-0.10.10 has this bug too
Comment 1 Wim Taymans 2008-09-04 09:31:21 UTC
I can't reproduce. Can you make and attach a debug log like:

GST_DEBUG=*rtp*:5,*rtsp*:5 gst-launch rtspsrc location="url" debug=1 ! fakesink -v >debug.log 2>&1

then gzip debug.log and attach. Thanks.
Comment 2 Ilia S. Smelykh 2008-09-04 09:45:31 UTC
Created attachment 117992 [details]
result of GST_DEBUG=*rtp*:5,*rtsp*:5 gst-launch rtspsrc location="url" debug=1 ! fakesink -v >debug.log 2>&1
Comment 3 Ilia S. Smelykh 2008-09-10 11:33:19 UTC
Wireshark with rtcp filter enabled shows that no RTCP Receiver Response comes from gstreamer.
Comment 4 Wim Taymans 2008-09-25 14:27:27 UTC
I believe this bug was fixed with the following patch that is in >= 0.10.9 of gst-plugins-good:

  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
  Call getsockname() after the call to bind() to get updated values
  for the port, etc. This fixes the usage of udpsrc on anonymous
  binding and it's usage by rtspsrc. Fixes bugs #539372, #539548.
  Thanks to Aurelien Grimaud for pointing out the obvious fix.