<div dir="ltr">The problem is in the server pipeline. It looks like you are generating straight MP2T but your client is decoding MP2T encapsulated in RTP, which gives the error. Since you seem to want to multiplex (mux) the audio and video, use straight MP2T. Your server pipeline might be ok. Your client will be receiving something like application/x-MP2T (not sure of the exact syntax), no payload, no ssrc, timestamp or seqno. Use the MP2T demuxer and 'T' for the audio and video. I use 0.10 so the element names have changed. You will have to use gst-inspect | grep -i mp2t to find the proper element name for the MP2T demuxer. There are examples of this but most are for 0.10, not the 1.x code. </div><div class="gmail_extra"> <div class="gmail_quote">On Tue, Apr 22, 2014 at 7:20 PM, kayoss12 <<a href="mailto:noureddinetariba@gmail.com" target="_blank">noureddinetariba@gmail.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">thank you for you answer , to so i try run pepepline client the first , and second i run the commande on RPI , but still not working , the Camera run on PI , but i can't receive the stream on my PC client there are any fault in the pepeline ?? thanks in advance for your answer client : gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp, media=(string)video, \ clock-rate=(int)90000, encoding-name=(string)MP2T, payload=(int)33, ssrc=(uint)4251539348, \ timestamp-offset=(uint)<a href="tel:2033210929" value="+12033210929">2033210929</a>, seqnum-offset=(uint)34671" ! rtpmp2tdepay \ ! tsparse ! tsdemux ! "video/x-h264" ! avdec_h264 ! autovideosink RPI : gst-launch-1.0 rpicamsrc bitrate=1000000 ! "video/x-h264,width=640,height=480" !\ queue ! h264parse ! queue ! mux. alsasrc device=hw:1,0 ! audioconvert ! \ lamemp3enc ! "audio/mpeg"! queue ! mux. mpegtsmux name=mux ! queue ! \ udpsink host=192.168.1.9 port=5000 -v i got this warning Warnig client : Additional debug info: gstrtpbasedepayload.c(381): gst_rtp_base_depayload_chain (): /GstPipeline:pipeli ne0/GstRtpMP2TDepay:rtpmp2tdepay0: Received invalid RTP payload, dropping WARNING: from element /GstPipeline:pipeline0/GstRtpMP2TDepay:rtpmp2tdepay0: Coul d not decode stream. also i have warning in RPI WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough Additional debug info: gstaudiobasesrc.c(848): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Dropped 8820 samples. This is most likely because downstream can't keep up and is consuming samples too slowly. -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/tsdemux-ffec-h264-ffmpeg-0-non-existing-PPS-referenced-tp4665890p4666552.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/tsdemux-ffec-h264-ffmpeg-0-non-existing-PPS-referenced-tp4665890p4666552.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div>