[gst-devel] sound problems with RTP in GStreamer-0.10.13

[ANTONIO MARQUES MARQUES]

Hi,

I am working with a RTP communication trying to encode and send sound in
the same machine. I launch two pipelines with gst-launch in order to
test it(one encodes the audiotestsrc and send it to a local port by
udpsink and the other decodes it and outputs it to alsasink). The
commands are the next ones:

gst-launch audiotestsrc !
audio/x-raw-int,rate=16000,channels=1,endianness=1234,width=16,depth=16
! audioconvert ! speexenc ! rtpspeexpay ! udpsink port=10500 host=127.0.0.1

gst-launch udpsrc port=10500 typefind=true caps="application/x-rtp" !
rtpspeexdepay ! speexdec ! audioconvert ! alsasink sync=false

When I tried it with gstreamer-0.10.11 it worked fine, but with
gstreamer-0.10.13 the pipeline decoding the audio dies.
The error shown is:
GST_EVENT gstpad.c:4240:gst_pad_send_event:<rtpspeexdepay0:sink>
Received event on flushing pad. Discarding
I don't know if the problem may be due to a problem with rtpspeexdepay.
If I merge both pipelines in just one (without the udpsink and udpsrc)
it works ok.

Thanks in advance.

Antonio