C code for rtp SPEEX decoding. I can't find how to solve the error

[Puyol]

I have two source codes one for client and the other for the server, when I
want to compile the source code of the client (gstreamer) I found this error
: 

ERROR:client.c:196:main: assertion failed: (lres == GST_PAD_LINK_OK)
Abandon (core dumped)

and this is the code of the client : 

#include <string.h>
#include <math.h>

#include <gst/gst.h>


/* the caps of the sender RTP stream. This is usually negotiated out of band
with
 * SDP or RTSP. */
#define AUDIO_CAPS
"audio/x-raw-int,media=(string)audio,clock-rate=(int)16000,width=(int)16,channels=(int)1"

#define AUDIO_DEPAY "rtpspeexdepay"
#define AUDIO_DEC   "speexdec"
#define AUDIO_SINK  "autoaudiosink"

/* the destination machine to send RTCP to. This is the address of the
sender and
 * is used to send back the RTCP reports of this receiver. If the data is
sent
 * from another machine, change this address. */
#define DEST_HOST "127.0.0.1"

/* print the stats of a source */
static void
print_source_stats (GObject * source)
{
  GstStructure *stats;
  gchar *str;

  g_return_if_fail (source != NULL);

  /* get the source stats */
  g_object_get (source, "stats", &stats, NULL);

  /* simply dump the stats structure */
  str = gst_structure_to_string (stats);
  g_print ("source stats: %s\n", str);

  gst_structure_free (stats);
  g_free (str);
}

/* will be called when gstrtpbin signals on-ssrc-active. It means that an
RTCP
 * packet was received from another source. */
static void
on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc,
    GstElement * depay)
{
  GObject *session, *isrc, *osrc;

  g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);

  /* get the right session */
  g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);

  /* get the internal source (the SSRC allocated to us, the receiver */
  g_object_get (session, "internal-source", &isrc, NULL);
  print_source_stats (isrc);

  /* get the remote source that sent us RTCP */
  g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
  print_source_stats (osrc);
}

/* will be called when rtpbin has validated a payload that we can depayload
*/
static void
pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
{
  GstPad *sinkpad;
  GstPadLinkReturn lres;

  g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));

  sinkpad = gst_element_get_static_pad (depay, "sink");
  g_assert (sinkpad);

  lres = gst_pad_link (new_pad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);
}


int
main (int argc, char *argv[])
{
  GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
  GstElement *audiodepay, *audiodec, *audiores, *audioconv, *audiosink;
  GstElement *pipeline;
  GMainLoop *loop;
  GstCaps *caps;
  gboolean res;
  GstPadLinkReturn lres;
  GstPad *srcpad, *sinkpad;

  /* always init first */
  gst_init (&argc, &argv);

  /* the pipeline to hold everything */
  pipeline = gst_parse_launch ("playbin2
uri=http://www.opus-codec.org/examples/samples/speech_orig.wav", NULL);
/*pipeline = gst_pipeline_new (NULL);*/
  g_assert (pipeline);

  /* the udp src and source we will use for RTP and RTCP */
  rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc");
  g_assert (rtpsrc);
  g_object_set (rtpsrc, "port", 5002, NULL);
  /* we need to set caps on the udpsrc for the RTP data */
  caps = gst_caps_from_string (AUDIO_CAPS);
  g_object_set (rtpsrc, "caps", caps, NULL);
  gst_caps_unref (caps);

  rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
  g_assert (rtcpsrc);
  g_object_set (rtcpsrc, "port", 5003, NULL);

  rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
  g_assert (rtcpsink);
  g_object_set (rtcpsink, "port", 5007, "host", DEST_HOST, NULL);
  /* no need for synchronisation or preroll on the RTCP sink */
  g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);

  gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);

  /* the depayloading and decoding */
  audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay");
  g_assert (audiodepay);
  audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec");
  g_assert (audiodec);
  /* the audio playback and format conversion */
  audioconv = gst_element_factory_make ("audioconvert", "audioconv");
  g_assert (audioconv);
  audiores = gst_element_factory_make ("audioresample", "audiores");
  g_assert (audiores);
  audiosink = gst_element_factory_make (AUDIO_SINK, "audiosink");
  g_assert (audiosink);

  /* add depayloading and playback to the pipeline and link */
  gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv,
      audiores, audiosink, NULL);

  res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores,
      audiosink, NULL);
  g_assert (res == TRUE);

  /* the rtpbin element */
  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
  g_assert (rtpbin);

  gst_bin_add (GST_BIN (pipeline), rtpbin);

  /* now link all to the rtpbin, start by getting an RTP sinkpad for session
0 */
  srcpad = gst_element_get_static_pad (rtpsrc, "src");
  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (srcpad);

  /* get an RTCP sinkpad in session 0 */
  srcpad = gst_element_get_static_pad (rtcpsrc, "src");
  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (srcpad);
  gst_object_unref (sinkpad);

  /* get an RTCP srcpad for sending RTCP back to the sender */
  srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
  sinkpad = gst_element_get_static_pad (rtcpsink, "sink");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);

  /* the RTP pad that we have to connect to the depayloader will be created
   * dynamically so we connect to the pad-added signal, pass the depayloader
as
   * user_data so that we can link to it. */
  g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb),
audiodepay);

  /* give some stats when we receive RTCP */
  g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK
(on_ssrc_active_cb),
      audiodepay);

  /* set the pipeline to playing */
  g_print ("starting receiver pipeline\n");
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* we need to run a GLib main loop to get the messages */
  loop = g_main_loop_new (NULL, FALSE);
  g_main_loop_run (loop);

  g_print ("stopping receiver pipeline\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  gst_object_unref (pipeline);

  return 0;
}

How I can solve the problem? Do you have any ideas? 
thanks



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