[Bug 770568] New: rtspsrc: fails to connect when the 'control' attribute is omitted in a single media description

GStreamer (GNOME Bugzilla) bugzilla at gnome.org
Mon Aug 29 17:19:32 UTC 2016


https://bugzilla.gnome.org/show_bug.cgi?id=770568

            Bug ID: 770568
           Summary: rtspsrc: fails to connect when the 'control' attribute
                    is omitted in a single media description
    Classification: Platform
           Product: GStreamer
           Version: 1.8.2
                OS: Linux
            Status: NEW
          Severity: major
          Priority: Normal
         Component: gst-plugins-good
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: brain at jikos.cz
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

If the "a=control:..." attribute in the SDP description is missing rtspsrc
fails to connect to the stream with the message "skipping stream
0x7f686c025940, no setup".

RFC2326 allows to omit the 'control' attribute in case only a single media
description is sent, so this is a perfectly valid scenario. See the chapter
C.3:

'If the presentation comprises only a single stream, the media-level
   "a=control:" attribute may be omitted altogether.'


Debug messages:
---------------
0:00:00.021400885 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1507:gst_rtspsrc_collect_payloads: mapping sdp session level
attributes to caps
0:00:00.021419822 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1509:gst_rtspsrc_collect_payloads: mapping sdp media level
attributes to caps
0:00:00.021427636 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1527:gst_rtspsrc_collect_payloads:<src>  looking at 0 pt: 98
0:00:00.021444514 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1649:gst_rtspsrc_create_stream:<src> stream 0, (0x7f686c025940)
0:00:00.021449581 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1650:gst_rtspsrc_create_stream:<src>  port: 5004
0:00:00.021453207 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1651:gst_rtspsrc_create_stream:<src>  container: 0
0:00:00.021456657 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1652:gst_rtspsrc_create_stream:<src>  control: (NULL)
0:00:00.021460299 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:1682:gst_rtspsrc_create_stream:<src>  setup: (NULL)
0:00:00.021467684 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:398:default_select_stream:<src> default handler
0:00:00.021471936 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:409:select_stream_accum: accum 1
0:00:00.021476157 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:398:default_select_stream:<src> default handler
0:00:00.021480444 29500      0x1fafe30 DEBUG                rtspsrc
gstrtspsrc.c:6101:gst_rtspsrc_setup_streams:<src> skipping stream
0x7f686c025940, no setup
0:00:00.021501131 29500      0x1fafe30 WARN                 rtspsrc
gstrtspsrc.c:6413:gst_rtspsrc_setup_streams:<src> error: No supported stream
was found. You might need to allow more transport protocols or may otherwise be
missing the right GStreamer RTSP extension plugin.



Tested with the following SDP:
------------------------------
v=0
o=- 1472462152047337 0 IN IP4 192.168.56.101
s=JADE01 (on IE11WIN7) streamed by "Lawo Jade"
t=0 0
a=clock-domain:PTPv2 0
a=recvonly
m=audio 5004 RTP/AVP 98
c=IN IP4 224.100.1.1/255
a=rtpmap:98 L24/44100/2
a=sync-time:0
a=framecount:48
a=source-filter: incl IN IP4 224.100.1.1 192.168.56.101
a=ptime:1.08

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