RTSP Server with digest auth

kiroru-cast.c

/*
 * Kiroru-cast is derived work from gst-rtsp-server examples[1]. 
 * Because of this nature, Kiroru-cast is licensed under LGPL v2
 *
 * [1]: https://cgit.freedesktop.org/gstreamer/gst-rtsp-server/tree/examples
 */

/*
 * Original license 
 *
 * GStreamer
 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

#include <gst/gst.h>

#include <gst/rtsp-server/rtsp-server.h>

#define DEFAULT_RTSP_PORT "8554"
#define DEFAULT_RTSP_USER "user"
#define DEFAULT_RTSP_PASSWORD "password"

static char *port = (char *) DEFAULT_RTSP_PORT;
static char *user = (char *) DEFAULT_RTSP_USER;
static char *password = (char *) DEFAULT_RTSP_PASSWORD;

static GOptionEntry entries[] = {
    {"port", 'p', 0, G_OPTION_ARG_STRING, &port, "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
    {"user", 'u', 0, G_OPTION_ARG_STRING, &user, "Username for the digest authorication (default: " DEFAULT_RTSP_USER ")", "USER"},
    {"password", 'a', 0, G_OPTION_ARG_STRING, &password, "Password for the digest authorication (default: " DEFAULT_RTSP_PASSWORD ")", "PASSWORD"},
    {NULL}
};

/* called when a stream has received an RTCP packet from the client */
static void
on_ssrc_active (GObject * session, GObject * source, GstRTSPMedia * media)
{
  GstStructure *stats;

  (void)media;

  GST_INFO ("source %p in session %p is active", source, session);

  g_object_get (source, "stats", &stats, NULL);
  if (stats) {
    gchar *sstr;

    sstr = gst_structure_to_string (stats);
    g_print ("structure: %s\n", sstr);
    g_free (sstr);

    gst_structure_free (stats);
  }
}

static void
on_sender_ssrc_active (GObject * session, GObject * source,
                       GstRTSPMedia * media)
{
  GstStructure *stats;

  (void)media;

  GST_INFO ("source %p in session %p is active", source, session);

  g_object_get (source, "stats", &stats, NULL);
  if (stats) {
    gchar *sstr;

    sstr = gst_structure_to_string (stats);
    g_print ("Sender stats:\nstructure: %s\n", sstr);
    g_free (sstr);

    gst_structure_free (stats);
  }
}

/* signal callback when the media is prepared for streaming. We can get the
 * session manager for each of the streams and connect to some signals. */
static void
media_prepared_cb (GstRTSPMedia * media)
{
  guint i, n_streams;

  n_streams = gst_rtsp_media_n_streams (media);

  GST_INFO ("media %p is prepared and has %u streams", media, n_streams);

  for (i = 0; i < n_streams; i++) {
    GstRTSPStream *stream;
    GObject *session;

    stream = gst_rtsp_media_get_stream (media, i);
    if (stream == NULL)
      continue;

    session = gst_rtsp_stream_get_rtpsession (stream);
    GST_INFO ("watching session %p on stream %u", session, i);

    g_signal_connect (session, "on-ssrc-active",
                      (GCallback) on_ssrc_active, media);
    g_signal_connect (session, "on-sender-ssrc-active",
                      (GCallback) on_sender_ssrc_active, media);
  }
}

static void
media_configure_cb (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
{
  /* connect our prepared signal so that we can see when this media is
   * prepared for streaming */
  g_signal_connect (media, "prepared", (GCallback) media_prepared_cb, factory);
}

static gboolean
remove_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
    GstRTSPServer * server)
{
  (void)pool;
  (void)session;
  (void)server;

  return GST_RTSP_FILTER_REMOVE;
}

static gboolean
remove_sessions (GstRTSPServer * server)
{
  GstRTSPSessionPool *pool;

  g_print ("removing all sessions\n");
  pool = gst_rtsp_server_get_session_pool (server);
  gst_rtsp_session_pool_filter (pool,
      (GstRTSPSessionPoolFilterFunc) remove_func, server);
  g_object_unref (pool);

  return FALSE;
}

static gboolean
timeout (GstRTSPServer * server)
{
  GstRTSPSessionPool *pool;

  pool = gst_rtsp_server_get_session_pool (server);
  gst_rtsp_session_pool_cleanup (pool);
  g_object_unref (pool);

  return TRUE;
}

int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
  GstRTSPAuth *auth;
  GstRTSPToken *token;
  GOptionContext *optctx;
  GError *error = NULL;
  gchar *str;

  optctx = g_option_context_new ("<filename.mp4> - Test RTSP Server, MP4");
  g_option_context_add_main_entries (optctx, entries, NULL);
  g_option_context_add_group (optctx, gst_init_get_option_group ());
  if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
    g_printerr ("Error parsing options: %s\n", error->message);
    g_option_context_free (optctx);
    g_clear_error (&error);
    return -1;
  }

  if (argc < 2) {
    g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
    return 1;
  }
  g_option_context_free (optctx);

  //gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();
  g_object_set (server, "service", port, NULL);

  /* get the mounts for this server, every server has a default mapper object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  str = g_strdup_printf ("( "
                         "filesrc location=\"%s\" ! qtdemux name=d "
                         "d. ! queue ! rtph264pay pt=96 name=pay0 "
                         "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);
  g_print("-> %s\n", argv[1]);

  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines.
   * any launch line works as long as it contains elements named pay%d. Each
   * element with pay%d names will be a stream */
  factory = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory, str);
  g_signal_connect (factory, "media-configure", (GCallback) media_configure_cb,
                    factory);
  g_free (str);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/sample", factory);

  /* allow user and admin to access this resource */
  gst_rtsp_media_factory_add_role (factory, "user",
      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
  gst_rtsp_media_factory_add_role (factory, "anonymous",
      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* make a new authentication manager */
  auth = gst_rtsp_auth_new ();
  gst_rtsp_auth_set_supported_methods (auth, GST_RTSP_AUTH_DIGEST);

  /* make default token, it has no permissions */
  token =
      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
      "anonymous", NULL);
  gst_rtsp_auth_set_default_token (auth, token);
  gst_rtsp_token_unref (token);

  /* make user token */
  token =
      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
      "user", NULL);
  gst_rtsp_auth_add_digest (auth, user, password, token);
  gst_rtsp_token_unref (token);

  /* set as the server authentication manager */
  gst_rtsp_server_set_auth (server, auth);
  g_object_unref (auth);

  /* attach the server to the default maincontext */
  if (gst_rtsp_server_attach (server, NULL) == 0)
    goto failed;

  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
  g_timeout_add_seconds (10, (GSourceFunc) remove_sessions, server);

  /* start serving */
  g_print ("stream with %s:%s ready at rtsp://127.0.0.1:8554/sample\n", user, password);
  g_main_loop_run (loop);

  return 0;

  /* ERRORS */
failed:
  {
    g_print ("failed to attach the server\n");
    return -1;
  }
}