hi all
    Hi, I was trying to transfer video and audio using network

sender------->server---------------->receiver_1
             
when i use the gst-launch tool to test my commends,it succeede. But when i wrote the "server" in c language and run the project again , i got this
   
    Error: internal data flow error.

on "server".the flowing is my commends and source code of server. Please help me,thank you very much

sender:
gst-launch -v gstrtpbin name=rtpbin \
filesrc location=filesrc location=/home/xuxin/desktop/g_p/a.avi ! decodebin name=dec \
dec. ! queue ! x264enc byte-stream=false ! rtph264pay ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! udpsink port=5000 host=172.21.29.177 name=vrtpsink \
dec. ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_1 \
rtpbin.send_rtp_src_1 ! udpsink port=5002 host=172.21.29.177 ts-offset=0 name=artpsink

Server( ip:172.21.29.177)
gst-launch -v gstrtpbin name=rtpbin latency=200 \
udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin ! udpsink port=5000 host=224.0.0.1 sync=false ts-offset=0 \
udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! udpsink port=5002 host=224.0.0.1 sync=false ts-offset=0

receiver (in multigroup)

gst-launch -v gstrtpbin name=rtpbin latency=200 \
udpsrc multigroup="224.0.0.1" caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! rtph264depay ! decodebin ! xvimagesink \
udpsrc multigroup="224.0.0.1" caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! rtppcmadepay ! decodebin ! audioconvert ! audioresample ! alsasink

 then I write "Server" in C language ,the code is showed below
 
#include <gst/gst.h>
#include <glib.h>
#include <unistd.h>
#include <stdlib.h>

static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
{
  GMainLoop *loop = (GMainLoop *) data;
  switch (GST_MESSAGE_TYPE (msg)) {
    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;
    case GST_MESSAGE_ERROR: {
      gchar *debug;
      GError *error;
      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);
      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);
      g_main_loop_quit (loop);
      break;
    }
    default:
      break;
  }
  return TRUE;
}

static void on_pad_added(GstElement *element, GstPad *pad, gpointer data)
{
    GstPad *sinkpad;
    GstElement *udpsink = (GstElement *)data;
  
    g_print("Dynamic pad created, linking demuxer/decoder\n");
    sinkpad = gst_element_get_static_pad(udpsink, "sink");
    gst_pad_link(pad, sinkpad);
    gst_object_unref(sinkpad);
}

int main(int argc, char **argv)
{
    GMainLoop *loop;
    GstBus *bus;
    GstPad *pad;
    GstCaps *videocap, *audiocap;
    GstElement *pipeline, *gstrtpbin, *udpsrc1, *udpsrc2,
        *udpsink1, *udpsink2;
  
    gst_init(&argc, &argv);
    loop = g_main_loop_new(NULL, FALSE);
  
    pipeline = gst_pipeline_new("server");
    gstrtpbin = gst_element_factory_make("gstrtpbin", "gst_rtpbin");
    udpsrc1 = gst_element_factory_make("udpsrc", "udpsrc1");
    udpsrc2 = gst_element_factory_make("udpsrc", "udpsrc2");
    udpsink1 = gst_element_factory_make("udpsink", "udpsink1");
    udpsink2 = gst_element_factory_make("udpsink", "udpsink2"); 
  
    bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
    gst_bus_add_watch(bus, bus_call, loop);
    gst_object_unref(bus);
  
    videocap = gst_caps_new_simple("application/x-rtp",
        "media", G_TYPE_STRING, "video",
        "clock-rate", G_TYPE_LONG, 90000,
        "encoding-name", G_TYPE_STRING, "H264", NULL);
      
    audiocap = gst_caps_new_simple("application/x-rtp",
        "media", G_TYPE_STRING, "audio",
        "clock-rate", G_TYPE_LONG, 8000,
        "encoding-name", G_TYPE_STRING, "PCMA", NULL);
  
    g_object_set(G_OBJECT(udpsrc1), "caps", videocap, NULL);
    g_object_set(G_OBJECT(udpsrc2), "caps", audiocap, NULL);
    g_object_set(G_OBJECT(udpsrc1), "port", 5000, NULL);
    g_object_set(G_OBJECT(udpsrc2), "port", 5002, NULL);
    g_object_set(G_OBJECT(udpsink1), "port", 5000, NULL);
    g_object_set(G_OBJECT(udpsink2), "port", 5002, NULL);
    g_object_set(G_OBJECT(udpsink1), "host", "172.21.29.177", NULL);
    g_object_set(G_OBJECT(udpsink2), "host", "172.21.29.177", NULL);
  
    gst_caps_unref(videocap);
    gst_caps_unref(audiocap);
      
    gst_bin_add_many(GST_BIN(pipeline), udpsrc1, udpsrc2, gstrtpbin, udpsink1, udpsink2, NULL);
  
    pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_0");
    gst_pad_link(gst_element_get_pad(udpsrc1, "src"), pad);
  
    pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_1");
    gst_pad_link(gst_element_get_pad(udpsrc2, "src"), pad);
  
    g_signal_connect(gstrtpbin, "pad-added", G_CALLBACK(on_pad_added), udpsink1);
    g_signal_connect(gstrtpbin, "pad_added", G_CALLBACK(on_pad_added), udpsink2);
  
    gst_element_set_state(pipeline, GST_STATE_PLAYING);
  
    g_print("Running...\n");
    g_main_loop_run(loop);
  
    /* Out of the main loop, clean up nicely */
    g_print("Returned, stopping playback\n");
    gst_element_set_state(pipeline, GST_STATE_NULL);
  
    g_print("Deleting pipeline\n");
    gst_object_unref(GST_OBJECT(pipeline));
  
    return 0;
}




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