rtspsrc just stopping playback

[marcin at saepia.net]

Hello,

I encounter odd behaviour with rtspsrc.

I have Gst-backed RTSP server that encodes opus audio stream and rtspsrc on
the other side (on Android) 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 pipeline it just starts to work again so I assume that
problem is on the receiver side.

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 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

Any suggestions what can be the reason for such mysteroius hangs?

How can I enable equivalent of GST_DEBUG env var on Android?

m.
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