[gst-devel] Faulty pipeline..

[Miguel Luis]

Hi!

I'm trying to stream some audio with rtp protocol, from the server part 
it's fine (well.. I think!), the problem is the client side..

SERVER:
$gst-launch-0.10 -v filesrc location=foo.mp3  ! mad ! audioconvert ! 
lame ! rtpmpapay ! udpsink host=127.0.0.1 port=9090


CLIENT:
$gst-launch-0.10 udpsrc port=9090 caps="SENDER_CAPS" ! rtpmpadepay ! mad 
! alsasink

I don't know what do I have to put between rtpmpadepay and udpsrc..
(I almost can't control myself... kidding yes I can)

Can anyone give me a hand please? :)

Best regards,
Miguel Luis.