I've been using gstreamer for a while - I am impressed how well it works. But recently a requirement has surfaced to examine a content of RTSP requests and responses. I was thinking that I could use handle-request callback feature of rtspclientsink - correct me if I am wrong. But I have been unable to get the callback working at all. Here is the code example I've been playing with. Any ideas?
static void handle_request_callback (GstElement  rtsp_client_sink,
                         gpointer  request,
                         gpointer  response,
                         gpointer  udata)
{
    printf("here\n\r");
}
int main (int   argc,   char *argv[])
{
    GMainLoop *loop;
    GstElement *pipeline, *videosrc, *sink, *videofilter, *videoencoder, *encoderfilter, *h264parse;
    GstCaps *filtercaps;
    GstPad *pad;
    /* init GStreamer */
    gst_init (&argc, &argv);
    loop = g_main_loop_new (NULL, FALSE);
    /* build */
    pipeline = gst_pipeline_new ("my-pipeline");
    videosrc = gst_element_factory_make ("v4l2src", "videosrc");
    if (videosrc == NULL)
        g_error ("Could not create 'v4l2src' element");
    videofilter = gst_element_factory_make ("capsfilter", "videofilter");
    g_assert (videofilter != NULL); /* should always exist */
    encoderfilter = gst_element_factory_make ("capsfilter", "encoderfilter");
    g_assert (encoderfilter != NULL); /* should always exist */
    h264parse = gst_element_factory_make ("h264parse", "parser");
    if (h264parse == NULL)
        g_error ("Could not create 'h264parse' element");
    videoencoder = gst_element_factory_make ("omxh264enc", "videoencoder");
    if (videoencoder == NULL)
        g_error ("Could not create 'omxh264enc' element");
    sink = gst_element_factory_make ("rtspclientsink", "sink");
    if (sink == NULL)
        g_error ("Could not create 'rtspclientsink' element");
    g_object_set(G_OBJECT(sink), "protocols", 0x00000004, NULL);
    g_object_set(G_OBJECT(sink), "latency", "100", NULL);
    g_object_set(G_OBJECT(sink), "location", "rtspt://192.169.1.2:1935/video", NULL);
    g_object_set(G_OBJECT(sink), "name", "rtspsink", NULL);
    //g_object_set(G_OBJECT(sink), "debug", true, NULL);
    gst_bin_add_many (GST_BIN (pipeline), videosrc, videofilter, videoencoder, h264parse, sink, NULL);
    gst_element_link_many (videosrc, videofilter, videoencoder, h264parse, sink, NULL);
    filtercaps = gst_caps_new_simple ("video/x-raw",
           "width", G_TYPE_INT, 1280,
           "height", G_TYPE_INT, 720,
           "framerate", GST_TYPE_FRACTION, 30, 1,
           NULL);
    g_object_set (G_OBJECT (videofilter), "caps", filtercaps, NULL);
    gst_caps_unref (filtercaps);
    filtercaps = gst_caps_new_simple ("video/x-h264",
           "width", G_TYPE_INT, 1280,
           "height", G_TYPE_INT, 720,
           "framerate", GST_TYPE_FRACTION, 30, 1,
           "profile", G_TYPE_STRING, "high",
           NULL);
    g_object_set (G_OBJECT (encoderfilter), "caps", filtercaps, NULL);
    gst_caps_unref (filtercaps);    
    gulong ret = g_signal_connect(sink, "handle-request", G_CALLBACK(handle_request_callback), sink);
    printf("%d\r\n", ret);
    /* run */
    gst_element_set_state (pipeline, GST_STATE_PLAYING);
    /* wait until it's up and running or failed */
    if (gst_element_get_state (pipeline, NULL, NULL, -1) == GST_STATE_CHANGE_FAILURE) {
        g_error ("Failed to go into PLAYING state");
    }
    g_print ("Running ...\n");
    g_main_loop_run (loop);
    /* exit */
    gst_element_set_state (pipeline, GST_STATE_NULL);
    gst_object_unref (pipeline);
    return 0;
}
here is rtspclientsink information:
$ gst-inspect-1.0 rtspclientsink 
Factory Details:
  Rank                     none (0)
  Long-name                RTSP RECORD client
  Klass                    Sink/Network
  Description              Send data over the network via RTSP RECORD(RFC 2326)
  Author                   Jan Schmidt <jan@centricular.com>
Plugin Details:
  Name                     rtspclientsink
  Description              RTSP client sink element
  Filename                 /usr/lib/arm-linux-gnueabihf/gstreamer-1.0/libgstrtspclientsink.so
  Version                  1.14.4
  License                  LGPL
  Source module            gst-rtsp-server
  Source release date      2018-10-02
  Binary package           GStreamer RTSP Server Library source release
  Origin URL               Unknown package origin
I've figured it out by looking the gstrtspclientsink.c file (https://github.com/GStreamer/gst-rtsp-server/blob/master/tests/check/gst/rtspclientsink.c). The signal emit function for handle-request is called from a part of the code meant to work as rtsp server while rtspclientsink is a RTSP client that starts by sending ANNOUNCE + sdp (instead of DESCRIBE).
Basically rtspclientsink "handle-request" signal only works for requests originated from server. In the context of RTSP ANNOUNCE where initial request is generated on a device producing media content that signal will never be triggered.
Anyway - g_signal_emit() for "handle-request" like signal could be added to pre and post processing logic of RTSP commands to allow to manipulate RTSP requests and monitor RTSP responses.
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