[Bug 752716] New: audio rtp L16 streaming to alsa

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=752716

            Bug ID: 752716
           Summary: audio rtp L16 streaming to alsa
    Classification: Platform
           Product: GStreamer
           Version: 0.10.x
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: info at itpox.de
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

I want to stream audio to a alsa device, which is capable of playing 16 bit,
one channel, signed audio at a rate of 6000.

This is working for a test pipeline:

gst-launch -vvv filesrc location=tst.wav ! wavparse ! alsasink
device=plughw:1,0
pipeline return:
alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234,
channels=(int)1, width=(int)16, depth=(int)16, signed=(boolean)true,
rate=(int)8000

With the following pipeling sound is played scrambled:
gst-launch -vvv audiotestsrc ! audioresample !
"audio/x-raw-int,width=16,channels=1,rate=8000" ! alsasink device=hw:1,0
sync=false 

At the end I need a pipeline like this, but this also does not work (audio
output is understandable very bad, has a lot of diturbances):
 gst-launch -vvv udpsrc port=5006
caps="application/x-rtp,clock-rate=44100,payload=11" ! rtpL16depay !
audioconvert ! "audio/x-raw-int,endianness=1234" ! audioresample quality=10 !
"audio/x-raw-int,rate=8000,clock-rate=8000" ! alsasink device=plughw:1,0
sync=false
An example stream for this pipeline is created using rtp tools:
rtpplay.exe -f hu_to_rfsb2.rtpdump -p rtp.type 192.168.2.255/5006

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