[Bug 764905] New: rtspsrc stops playback after a while

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=764905

            Bug ID: 764905
           Summary: rtspsrc stops playback after a while
    Classification: Platform
           Product: GStreamer
           Version: 1.8.0
                OS: Mac OS
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gstreamer (core)
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: marcin at saepia.net
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

I encounter odd behaviour with rtspsrc.

I have Gst-backed RTSP server that encodes opus audio stream and rtspsrc on the
other side that tries to play it back.

Generally speaking it works fine but it just stops playback after some time
(5-10 min) with no error, warning, message etc. (I am logging everything that
appears on the pipeline's bus).

When I restart the receiver pipeline it just starts to work again so I assume
that problem is on the receiver side.

If sometimes happen that two identical receivers are connected to the same
mount of the same RTSP server and while one is silent, another plays back, so I
assume that RTSP server is sending data correctky.


The receiver's pipeline is

rtspsrc location=rtsp://.... user-agent=... drop-on-latency=true latency=500
tls-database=... protocols=... username=... password=... ! decodebin !
audioconvert ! audioresample ! queue2 ! openslessink

(protocols are  0x00000001 | 0x00000002 | 0x00000004 | 0x00000010 | 0x00000020)

I am using 1.8.0 on both sides.

I am not sure if this is related but during playback on Android I ocassionally
get

04-11 16:38:41.989 21543 21629 W GStreamer+audiobasesink: 0:04:35.920673280
0xb8e045b0
gstaudiobasesink.c:1484:gst_audio_base_sink_skew_slaving:<audio_interface_playback>
correct clock skew +0:00:00.020063566 > +0:00:00.020000000
04-11 16:38:42.006 21543 21629 W GStreamer+audiobasesink: 0:04:35.938068176
0xb8e045b0
gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<audio_interface_playback>
correct clock skew -0:00:00.020290466 < -+0:00:00.020000000


I have managed to reproduce the issue on Mac OS X. 

I've used gst-launch -m -vv to run identical pipeline and the only output was

/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstRtpBin:manager/GstRtpSession:rtpsession0:
stats = "application/x-rtp-session-stats\,\ rtx-drop-count\=\(uint\)0\,\
sent-nack-count\=\(uint\)0\,\ recv-nack-count\=\(uint\)0\,\
source-stats\=\(GValueArray\)NULL\,\ rtx-count\=\(uint\)0\;"

(repeated many times)

and it kept appearing even when audio was not playing.

I have run it with GST_DEBUG=*:4,rtspsrc:5,opus:5 and there were no messages
prior to hanging.

I've added level element to the pipeline to see whether it emits anything when
audio stops playing. As I anticipated, level stops emitting messages when audio
stops playing.

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