[gst-devel] rtp pipeline problem

[bik55 at op.pl]

Hi!
I have problem with streaming sound through network.

I use this pipeline for streaming:

gst-launch audiotestsrc ! lame   ! rtpmpapay   ! udpsink

And client pilepine is (it worked untill I used rtp, evary plugins are in the system):

gst-launch udpsrc caps=application/x-rtp    !  rtpmpadepay  !  mad ! audioconvert !  autoaudiosink -v
Setting pipeline to PAUSED ...
/pipeline0/udpsrc0.src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/pipeline0/rtpmpadepay0.sink: caps = application/x-rtp
ERROR: from element /pipeline0/udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(1416): gst_base_src_loop (): /pipeline0/udpsrc0:
streaming task paused, reason error
Execution ended after 6100000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
/pipeline0/rtpmpadepay0.sink: caps = NULL
/pipeline0/udpsrc0.src: caps = NULL
Setting pipeline to NULL ...
FREEING pipeline ...


Any idea????


Bart