Gstreamer: dump the incoming rtp payload into a local buffer using rtpL16depay

I am using Gstreamer1.0 to extract the rtp packet's payload data into a buffer.

To test this i have created these pipelines (host/client)

host:

gst-launch-1.0 audiotestsrc ! audioconvert ! audio/x-raw,channels=1,depth=16,width=16,rate=44100 ! rtpL16pay ! udpsink host=192.168.151.99 port=5000

client:

gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96" ! rtpL16depay ! audioconvert ! autoaudiosink sync=false

This worked fine in terminal as well as using C++ code. Below is the source code for client in C++.

gint
main (gint   argc,
      gchar *argv[])
{
    GstElement *pipeline, *udp, *sinkcaps, *depay, *conv, *audiosink, *abcd;

    GMainLoop *loop;
    // init GStreamer
    gst_init (&argc, &argv);
    loop = g_main_loop_new (NULL, FALSE);

    // setup pipeline
    pipeline = gst_pipeline_new ("pipeline");

    udp = gst_element_factory_make("udpsrc", "udp");
    g_object_set(G_OBJECT(udp), "port", 5000, NULL);

    sinkcaps = gst_element_factory_make("capsfilter", "caps");
    g_object_set(sinkcaps, "caps", gst_caps_new_simple("application/x-rtp",
                                                        "media", G_TYPE_STRING, "audio",
                                                        "clock-rate", G_TYPE_INT, 44100,
                                                        "width", G_TYPE_INT, 16,
                                                        "height", G_TYPE_INT, 16,
                                                        "encoding-name", G_TYPE_STRING, "L16",
                                                        "encoding-params", G_TYPE_STRING, "1",
                                                        "channels", G_TYPE_INT, 1,
                                                        "channel-positions", G_TYPE_INT, 1,
                                                        "payload", G_TYPE_INT, 96,NULL), NULL);

    depay = gst_element_factory_make("rtpL16depay", "depay");

    conv = gst_element_factory_make ("audioconvert", "conv");

    audiosink = gst_element_factory_make ("multifilesink", "sink");
    g_object_set(G_OBJECT(audiosink), "sync", FALSE, NULL);


    gst_bin_add_many (GST_BIN (pipeline), udp, sinkcaps, depay, audiosink, NULL);

    if (gst_element_link_many (udp, sinkcaps, depay, conv, audiosink, NULL) != TRUE)
    {
        return -1;
    }

    // play
    gst_element_set_state (pipeline, GST_STATE_PLAYING);

    //Runs a main loop until g_main_loop_quit() is called on the loop.
    g_main_loop_run (loop);

    // clean up
    gst_element_set_state (pipeline, GST_STATE_NULL);
    gst_object_unref (GST_OBJECT (pipeline));
    g_main_loop_unref (loop);

    return 0;
}

I am getting the audio from speakers for this.

Now I am trying to accomplish the following objective: Dump the rtp payload (only) in sequence in a buffer.

I have tried finding the appropriate element to get the payload data to buffer. The closest I could find is "appsink". For this I have made this following pipeline.

client:

gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96" ! rtpL16depay ! appsink

The pipeline is created successfully but I am not sure where the data is getting dumped and how can I retrieve it?

Thank you any help :)