<div> hi all         I use rtp protocol to send a movie. On the "sender" when  I set the value of "async" property of udpsink true, the "receiver" can't receive any packets. Instead,when i set "async" false ,the receiver can receiver can receive the rtp packets and play the movie ,but the audio and the vedio are asynchronous, and the video is faster than normal.         Any suggestion? What could be the problem?      Regards  xuxin    the following is my code  receiver:\      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. ! rtph264depay ! decodebin ! xvimagesink \ udpsrc 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 sender : #include <gst/gst.h> #include <glib.h> #include <unistd.h> #include <stdlib.h> #define SOURCE  "/home/xuxin/desktop/g_p/test/a.avi" #define DEST_HOST  "172.21.29.169" #define VIDEO_PORT 5000 #define AUDIO_PORT 5002 // global declaration GstElement *pipeline,*udpsink0,*udpsink1; GstElement *audio_bin,*video_bin; // gboolean res;//element link GstPadLinkReturn lres;//pad link //bus 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_rtpbin(GstElement *element, GstPad *pad, gpointer data) {     GstCaps *caps=NULL;     GstStructure *str=NULL;     GstPad *t_pad=NULL;         const gchar* media_type=NULL;     gint clock_rate;     gboolean judgment;     const gchar* encoding_name=NULL;     const gchar* name=NULL;     g_print("**************enter into rtpbin callback ****************\n");     g_assert(pad!=NULL);     caps=gst_pad_get_caps(pad);     g_assert(caps!=NULL);     str=gst_caps_get_structure(caps,0);     g_assert(str!=NULL);         media_type=gst_structure_get_string(str,"media");     g_assert(media_type!=NULL);     g_print("the media type is %s\n",media_type);     //**************test*****************************     name=gst_structure_get_name(str);     g_print("the caps is %s\n",name);     judgment=gst_structure_get_int(str,"clock-rate",&clock_rate);     g_print("clock_rate is :%d\n",clock_rate);     encoding_name=gst_structure_get_string(str,"encoding-name");     g_assert(encoding_name!=NULL);     g_print("the encoding_name is %s\n",encoding_name);     //***********************************************                 //media type audio or video          if (g_strrstr (media_type, "video"))          {                          g_print("rtp catch audio\n");             t_pad = gst_element_get_static_pad (udpsink0, "sink");             g_assert(t_pad!=NULL);                   g_print("get the pad of udpsink0\n");                  lres=gst_pad_link (pad, t_pad);           g_assert (lres == GST_PAD_LINK_OK);           g_print("Dynamic pad created, linking rtpbin/udp1\n");          }          else if(g_strrstr (media_type, "audio"))       {                    g_print("rtp catch video\n");             t_pad = gst_element_get_static_pad (udpsink1, "sink");             g_assert(t_pad!=NULL);               g_print("get the pad of udpsink1\n");             lres=gst_pad_link (pad, t_pad);           g_assert (lres == GST_PAD_LINK_OK);           g_print("Dynamic pad created, linking rtpbin/udp0\n");                            }         else         {             gst_caps_unref (caps);             return;         }     } static void on_pad_added_decodebin(GstElement *decodebin,GstPad *pad,gboolean last,gpointer data) {     GstCaps *caps=NULL;       GstStructure *str=NULL;       GstPad *t_pad=NULL;         const gchar* media_type=NULL;     g_print("**************** enter into the decodebin signal callback **********\n");         g_assert(pad!=NULL);         caps = gst_pad_get_caps (pad);     g_assert(caps!=NULL);       str = gst_caps_get_structure (caps, 0);       g_assert(str!=NULL);             media_type=gst_structure_get_name (str);       g_assert(media_type!=NULL);     g_print("the media type is %s\n",media_type);       //check media type audio or video          if (g_strrstr (media_type, "audio"))          {                          g_print("catch audio in decodebin\n");             t_pad = gst_element_get_static_pad (audio_bin, "sink");             g_assert(t_pad!=NULL);             lres=gst_pad_link (pad, t_pad);           g_assert (lres == GST_PAD_LINK_OK);           g_print("Dynamic pad created, linking decoderbin/queue2\n");                                    }          else if(g_strrstr (media_type, "video"))       {                    g_print("catch video in decodebin\n");             t_pad = gst_element_get_static_pad (video_bin, "sink");             g_assert(t_pad!=NULL);             lres=gst_pad_link (pad, t_pad);           g_assert (lres == GST_PAD_LINK_OK);           g_print("Dynamic pad created, linking decoderbin/queue1\n");                                   }         else         {             g_print("have not get suitable type\n");             gst_caps_unref (caps);             return;         }     } int main(int argc, char **argv) {     GMainLoop *loop;     GstBus *bus;         //pad of rtpbin     GstPad *sink_pad;        GstPad *bin_pad;            GstPad *videopad_src,*videopad_sink;     GstPad *audiopad_src,*audiopad_sink;     //declare elements     GstElement  *filesrc,*decodebin,*gstrtpbin;     //element for video     GstElement *queue1,*x264enc,*rtph264pay;     //element for audio     GstElement *queue2,*audioresample,*audioconvert,*alawenc,*rtppcmapay;         gst_init(&argc, &argv);     loop = g_main_loop_new(NULL, FALSE);         //generate elements     pipeline = gst_pipeline_new("server");     filesrc=gst_element_factory_make("filesrc", "filesrc");     decodebin=gst_element_factory_make("decodebin", "decodebin");     gstrtpbin = gst_element_factory_make("gstrtpbin", "gstrtpbin");     //for video     queue1=gst_element_factory_make("queue", "queue1");     x264enc=gst_element_factory_make("x264enc", "x264enc");     rtph264pay=gst_element_factory_make("rtph264pay", "rtph264pay");     udpsink0=gst_element_factory_make("udpsink", "udpsink0");     //for audio     queue2 = gst_element_factory_make("queue", "queue2");     audioresample = gst_element_factory_make("audioresample", "audioresample");     audioconvert = gst_element_factory_make("audioconvert", "audioconvert");     alawenc = gst_element_factory_make("alawenc", "alawenc");        rtppcmapay = gst_element_factory_make("rtppcmapay", "rtppcmapay");     udpsink1=gst_element_factory_make("udpsink", "udpsink1");         bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));     gst_bus_add_watch(bus, bus_call, loop);     gst_object_unref(bus);         //set properties of elements     g_object_set(G_OBJECT(filesrc), "location",SOURCE, NULL);        g_object_set(G_OBJECT(udpsink0), "host",DEST_HOST,"port",VIDEO_PORT, "sync",FALSE,             "async", FALSE,NULL);     g_object_set(G_OBJECT(udpsink1), "host",DEST_HOST,"port",AUDIO_PORT, "sync",FALSE,             "async", FALSE,NULL);     //g_object_set(G_OBJECT(udpsink0),  NULL);     //g_object_set(G_OBJECT(udpsink1), "host",DEST_HOST, NULL);     //g_object_set(G_OBJECT(udpsink0), "sync",FALSE, NULL);     //g_object_set(G_OBJECT(udpsink1), "sync",FALSE, NULL);         //create video_bin     video_bin = gst_bin_new ("videobin");     gst_bin_add_many (GST_BIN (video_bin),queue1,x264enc,rtph264pay,NULL);     res=gst_element_link_many(queue1,x264enc,rtph264pay,NULL);     g_assert (res == TRUE);     //add pad to video_bin     videopad_sink = gst_element_get_static_pad (queue1, "sink");     videopad_src= gst_element_get_static_pad (rtph264pay, "src");        gst_element_add_pad (video_bin,gst_ghost_pad_new ("sink", videopad_sink));     gst_element_add_pad (video_bin,gst_ghost_pad_new ("src",videopad_src));       gst_object_unref (videopad_sink);       gst_object_unref (videopad_src);             //create audio_bin       audio_bin = gst_bin_new ("audiobin");     gst_bin_add_many (GST_BIN (audio_bin),queue2,audioresample,audioconvert,alawenc,rtppcmapay,NULL);     res=gst_element_link_many(queue2,audioresample,audioconvert,alawenc,rtppcmapay,NULL);     g_assert (res == TRUE);     //add pad to audio_bin     audiopad_sink = gst_element_get_static_pad (queue2, "sink");     audiopad_src= gst_element_get_static_pad (rtppcmapay, "src");        res=gst_element_add_pad (audio_bin,gst_ghost_pad_new ("sink", audiopad_sink));     g_assert (res == TRUE);     res=gst_element_add_pad (audio_bin,gst_ghost_pad_new ("src", audiopad_src));     g_assert (res == TRUE);       gst_object_unref (audiopad_sink);       gst_object_unref (audiopad_src);         //add elements into pipeline     gst_bin_add_many(GST_BIN(pipeline),             filesrc,decodebin,audio_bin,video_bin,gstrtpbin,udpsink0,udpsink1,NULL);         //static link filesrc and decoderbin     res=gst_element_link(filesrc,decodebin);        g_assert (res == TRUE);            //get request pad from gstrtpbin and connect with video_bin     bin_pad=gst_element_get_pad(video_bin, "src");     g_assert(bin_pad!=NULL);        sink_pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_0");     g_assert(sink_pad!=NULL);        lres=gst_pad_link(bin_pad, sink_pad);     g_assert (lres == GST_PAD_LINK_OK);             //get request pad from gstrtpbin and connect with audio_bin     bin_pad=gst_element_get_pad(audio_bin, "src");     g_assert(bin_pad!=NULL);     sink_pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_1");     g_assert(sink_pad!=NULL);     lres=gst_pad_link(bin_pad, sink_pad);     g_assert (lres == GST_PAD_LINK_OK);     //signal link     g_signal_connect(decodebin, "new-decoded-pad", G_CALLBACK(on_pad_added_decodebin),NULL);            g_signal_connect(gstrtpbin, "pad-added", G_CALLBACK(on_pad_added_rtpbin),NULL);            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; } </div>  <hr/> <a href="http://qiye.163.com/?ft=3">��ҵ�ʣ��ר��</a>