hehe now that I think of it it should work with the default ....but still it's a good thing to set it just to be sure... try to run it with --gst-debug-level=5 or any level and check what's happening juste before the error... <div>On 6/6/06, Antoine Tremblay <<a href="mailto:hexa00@gmail.com">hexa00@gmail.com</a>> wrote:<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> <div>Well you're missing some args in udpsink /udpsrc Should be like udpsink host=localhost port=7777 udpsrc port=7777   the rest seems ok at 1st glance... Regards </div><div> Antoine</div><div> <div>On 6/6/06, Bebjak, Michal <<a href="mailto:michal.bebjak@siemens.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">michal.bebjak@siemens.com </a>> wrote:<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Hi, I'm developing an client-server aplication which should use GStreamer. I want to endcode the audio into MP3 and send it oved RTP to the client. I first tried to run this following commands: server terminal:  gst-launch-0.10 -v audiotestsrc ! lame ! rtpmpapay ! udpsink client terminal:  gst-launch-0.10 -v udpsrc ! 'application/x-rtp, media=audio, payload=96, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA' !                   rtpmpadepay ! 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' ! mad ! audioconvert ! volume volume=0.2 ! autoaudiosink They both work independently but when I try to send the audio from server to client the client crashes. The terminal output is: $ gst-launch-0.10 -v udpsrc ! 'application/x-rtp, media=audio, payload=96, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA' ! rtpmpadepay ! 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' ! mad ! audioconvert ! volume volume=0.2 ! autoaudiosink Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock /pipeline0/capsfilter0.sink: caps = application/x-rtp, media=(string)audio, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)MPA /pipeline0/capsfilter0.src: caps = application/x-rtp, media=(string)audio, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)MPA /pipeline0/rtpmpadepay0.sink: caps = application/x-rtp, media=(string)audio, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)MPA ERROR: from element /pipeline0/udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(1318): gst_base_src_loop (): /pipeline0/udpsrc0: streaming task paused, reason error Execution ended after 1513470000 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /pipeline0/rtpmpadepay0.sink: caps = NULL /pipeline0/capsfilter0.sink: caps = NULL /pipeline0/capsfilter0.src: caps = NULL Setting pipeline to NULL ... FREEING pipeline ... Can someone please help me to make it work?? Thanks a lot!! Michael _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel </a> </blockquote></div> </div></blockquote></div>