Rtp stream, audio stops after <1 sec

[Ferdinand Wörister]

Hi guys,
I've got the following pipeline:
gst-launch -v udpsrc port=6000 name=udpSrcAudio  ! application/x-rtp, 
media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMU, 
payload=(int)0 ! gstrtpjitterbuffer drop-on-latency=true latency=200 !  
rtppcmudepay ! mulawdec ! autoaudiosink provide clock=false

Which is supposed to play the following stream:
gst-launch -v audiotestsrc is-live=true !  audio/x-raw-int, rate=8000, 
channels=1 ! mulawenc ! rtppcmupay ! queue max-size-buffers=0 
max-size-time=0 max-size-bytes=0 leaky=1 ! udpsink host=127.0.0.1 
port=6000 sync=false async=false

Often, but not always I can hear sound for about half a second, it then 
cuts out and I get the same warning periodically.
0:00:00.789046000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
correct clock skew -23617542 < -20000000
0:00:01.429082000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
correct clock skew -33369589 < -20000000
0:00:01.941111000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
correct clock skew -23957663 < -20000000
0:00:02.837163000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
correct clock skew -23524410 < -20000000

I'm using the following version of Gstreamer on Windows 7/64
gst-launch-0.10 version 0.10.31
GStreamer 0.10.31
http://code.google.com/p/ossbuild/

Is something wrong in my pipelines? Any advice?
In the past, I have never received any responses from this board. If it 
is about my Windows build or me doing something wrong please let me know.
In any case, thanks!


P.S. This is the full output of the receiving pipe:
gst-launch -v  udpsrc port=6000 name=udpSrcAudio  !   applicatio
n/x-rtp, media=(string)audio, clock-rate=(int)8000, 
encoding-name=(string)PCMU,
payload=(int)0, ssrc=(uint)1364371436, clock-base=(uint)1619660721, 
seqnum-base=
(uint)17178 !  gstrtpjitterbuffer  drop-on-latency=true latency=200 !  
rtppcmude
pay ! mulawdec ! autoaudiosink provide-clock=false
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
0:00:00.204012000  5208   007F8DE0 WARN                     bin 
gstbin.c:2378:gs
t_bin_do_latency_func:<pipeline0> failed to query latency
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = 
application/
x-rtp, media=(string)audio, clock-rate=(int)8000, 
encoding-name=(string)PCMU, pa
yload=(int)0, ssrc=(uint)1364371436, clock-base=(uint)1619660721, 
seqnum-base=(u
int)17178
/GstPipeline:pipeline0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:src: 
caps = ap
plication/x-rtp, media=(string)audio, clock-rate=(int)8000, 
encoding-name=(strin
g)PCMU, payload=(int)0, ssrc=(uint)1364371436, 
clock-base=(uint)1619660721, seqn
um-base=(uint)17178
/GstPipeline:pipeline0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink: 
caps = a
pplication/x-rtp, media=(string)audio, clock-rate=(int)8000, 
encoding-name=(stri
ng)PCMU, payload=(int)0, ssrc=(uint)1364371436, 
clock-base=(uint)1619660721, seq
num-base=(uint)17178
/GstPipeline:pipeline0/GstRtpPcmuDepay:rtppcmudepay0.GstPad:src: caps = 
audio/x-
mulaw, channels=(int)1, rate=(int)8000
/GstPipeline:pipeline0/GstRtpPcmuDepay:rtppcmudepay0.GstPad:sink: caps = 
applica
tion/x-rtp, media=(string)audio, clock-rate=(int)8000, 
encoding-name=(string)PCM
U, payload=(int)0, ssrc=(uint)1364371436, clock-base=(uint)1619660721, 
seqnum-ba
se=(uint)17178
/GstPipeline:pipeline0/GstMuLawDec:mulawdec0.GstPad:src: caps = 
audio/x-raw-int,
  width=(int)16, depth=(int)16, endianness=(int)1234, 
signed=(boolean)true, rate=
(int)8000, channels=(int)1
/GstPipeline:pipeline0/GstMuLawDec:mulawdec0.GstPad:sink: caps = 
audio/x-mulaw,
channels=(int)1, rate=(int)8000
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstWaveFormSink:autoaudio
sink0-actual-sink-waveform.GstPad:sink: caps = audio/x-raw-int, 
width=(int)16, d
epth=(int)16, endianness=(int)1234, signed=(boolean)true, 
rate=(int)8000, channe
ls=(int)1
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink: 
caps =
audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, 
signed=(boo
lean)true, rate=(int)8000, channels=(int)1
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink.GstProxy
Pad:proxypad0: caps = audio/x-raw-int, width=(int)16, depth=(int)16, 
endianness=
(int)1234, signed=(boolean)true, rate=(int)8000, channels=(int)1
0:00:00.789046000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink
.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
c
orrect clock skew -23617542 < -20000000
0:00:01.429082000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink
.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
c
orrect clock skew -33369589 < -20000000
0:00:01.941111000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink
.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
c
orrect clock skew -23957663 < -20000000
0:00:02.837163000  5208   025B4168 WARN           baseaudiosink 
gstbaseaudiosink
.c:1113:gst_base_audio_sink_skew_slaving:<autoaudiosink0-actual-sink-waveform> 
c
orrect clock skew -23524410 < -20000000