rtspsrc just stopping playback

[marcin at saepia.net]

The same happens with playbin. I have run it with
GST_DEBUG=*:4,rtspsrc:5,opus:5 and there was no output prior to hang.

m.

2016-04-11 17:27 GMT+02:00 marcin at saepia.net <marcin at saepia.net>:

> I was able to reproduce the same behaviour 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
>
>
> am I doing something wrong or is it a bug?
>
> m.
>
> 2016-04-11 16:48 GMT+02:00 marcin at saepia.net <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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