rtsp-server: Add support for tunneling

Add support for tunneling over HTTP.
Use new connection methods to retrieve the url.
Dispatch messages based on the message type instead of blindly
assuming it's always a request.
Keep track of the watch id so that we can remove it later.
Set the media pipeline to NULL before unreffing the pipeline.

gst/rtsp-server/rtsp-client.c

@@ -26,6 +26,9 @@
 
 #define DEFAULT_TIMEOUT  60
 
+static GMutex *tunnels_lock;
+static GHashTable *tunnels;
+
 enum
 {
   PROP_0,
@@ -67,6 +70,9 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass)
       g_param_spec_object ("media-mapping", "Media Mapping",
           "The media mapping to use for client session",
           GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  tunnels = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
+  tunnels_lock = g_mutex_new ();
 }
 
 static void
@@ -486,6 +492,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
   guint streamid;
   GstRTSPSessionMedia *media;
   gboolean need_session;
+  GstRTSPUrl *url;
 
   /* the uri contains the stream number we added in the SDP config, which is
    * always /stream=%d so we need to strip that off 
@@ -535,7 +542,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
     goto unsupported_transports;
 
   supported = GST_RTSP_LOWER_TRANS_UDP |
-	GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
+	GST_RTSP_LOWER_TRANS_UDP_MCAST;
   if (!(ct->lower_transport & supported))
     goto unsupported_transports;
 
@@ -544,7 +551,8 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
 
   /* we have a valid transport now, set the destination of the client. */
   g_free (ct->destination);
-  ct->destination = g_strdup (client->connection->url->host);
+  url = gst_rtsp_connection_get_url (client->connection);
+  ct->destination = g_strdup (url->host);
 
   if (session) {
     g_object_ref (session);
@@ -996,8 +1004,17 @@ message_received (GstRTSPWatch *watch, GstRTSPMessage *message, gpointer user_da
 
   g_message ("client %p: received a message", client);
 
-  handle_request (client, message);
+  switch (message->type) {
-
+    case GST_RTSP_MESSAGE_REQUEST:
+      handle_request (client, message);
+      break;
+    case GST_RTSP_MESSAGE_RESPONSE:
+      break;
+    case GST_RTSP_MESSAGE_DATA:
+      break;
+    default:
+      break;
+  }
   return GST_RTSP_OK;
 }
 
@@ -1015,9 +1032,15 @@ static GstRTSPResult
 closed (GstRTSPWatch *watch, gpointer user_data)
 {
   GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  const gchar *tunnelid;
 
   g_message ("client %p: connection closed", client);
 
+  tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+  g_mutex_lock (tunnels_lock);
+  g_hash_table_remove (tunnels, tunnelid);
+  g_mutex_unlock (tunnels_lock);
+
   return GST_RTSP_OK;
 }
 
@@ -1034,11 +1057,90 @@ error (GstRTSPWatch *watch, GstRTSPResult result, gpointer user_data)
   return GST_RTSP_OK;
 }
 
+static GstRTSPStatusCode
+tunnel_start (GstRTSPWatch *watch, gpointer user_data)
+{
+  GstRTSPClient *client;
+  const gchar *tunnelid;
+
+  client = GST_RTSP_CLIENT (user_data);
+
+  g_message ("client %p: tunnel start", client);
+
+  /* store client in the pending tunnels */
+  tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+
+  g_message ("client %p: inserting %s", client, tunnelid);
+
+  /* we can't have two clients connecting with the same tunnelid */
+  g_mutex_lock (tunnels_lock);
+  if (g_hash_table_lookup (tunnels, tunnelid))
+    goto tunnel_existed;
+
+  g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+  g_mutex_unlock (tunnels_lock);
+
+  return GST_RTSP_STS_OK;
+
+  /* ERRORS */
+tunnel_existed:
+  {
+    g_mutex_unlock (tunnels_lock);
+    g_message ("client %p: tunnel session %s existed", client, tunnelid);
+    return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+  }
+}
+
+static GstRTSPResult
+tunnel_complete (GstRTSPWatch *watch, gpointer user_data)
+{
+  const gchar *tunnelid;
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  GstRTSPClient *oclient;
+
+  g_message ("client %p: tunnel complete", client);
+
+  /* find previous tunnel */
+  tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+
+  g_mutex_lock (tunnels_lock);
+  if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
+    goto no_tunnel;
+
+  /* remove the old client from the table. ref before because removing it will
+   * remove the ref to it. */
+  g_object_ref (oclient);
+  g_hash_table_remove (tunnels, tunnelid);
+  g_mutex_unlock (tunnels_lock);
+
+  g_message ("client %p: found tunnel %p", client, oclient);
+
+  /* merge the tunnels into the first client */
+  gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
+  gst_rtsp_watch_reset (oclient->watch);
+  g_object_unref (oclient);
+
+  /* we don't need this watch anymore */
+  g_source_remove (client->watchid);
+
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+no_tunnel:
+  {
+    g_mutex_unlock (tunnels_lock);
+    g_message ("client %p: tunnel session %s not found", client, tunnelid);
+    return GST_RTSP_OK;
+  }
+}
+
 static GstRTSPWatchFuncs watch_funcs = {
   message_received,
   message_sent,
   closed,
-  error
+  error,
+  tunnel_start,
+  tunnel_complete
 };
 
 /**
@@ -1061,14 +1163,16 @@ gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
   GstRTSPResult res;
   GSource *source;
   GMainContext *context;
+  GstRTSPUrl *url;
 
   /* a new client connected. */
   sock = g_io_channel_unix_get_fd (channel);
 
   GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
 
-  g_message ("added new client %p ip %s:%d with fd %d", client,
+  url = gst_rtsp_connection_get_url (conn);
-	        conn->url->host, conn->url->port, conn->fd.fd);
+  g_message ("added new client %p ip %s:%d", client,
+	        url->host, url->port);
 
   client->connection = conn;
 
@@ -1084,7 +1188,7 @@ gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
 
   g_message ("attaching to context %p", context);
 
-  gst_rtsp_watch_attach (client->watch, context);
+  client->watchid = gst_rtsp_watch_attach (client->watch, context);
   gst_rtsp_watch_unref (client->watch);
 
   return TRUE;

gst/rtsp-server/rtsp-client.h

@@ -71,6 +71,7 @@ struct _GstRTSPClient {
 
   GstRTSPConnection *connection;
   GstRTSPWatch      *watch;
+  guint              watchid;
 
   guint                 timeout;
   GstRTSPSessionPool   *session_pool;

gst/rtsp-server/rtsp-media.c

@@ -110,8 +110,10 @@ gst_rtsp_media_finalize (GObject * obj)
     g_source_unref (media->source);
   }
 
-  if (media->pipeline)
+  if (media->pipeline) {
+    gst_element_set_state (media->pipeline, GST_STATE_NULL);
     gst_object_unref (media->pipeline);
+  }
 
   G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
 }