Jabber: initial version of Jingle support

11 files changed, 1996 insertions, 0 deletions

diff --git a/include/gst/rtsp/gstrtsp-enumtypes.h b/include/gst/rtsp/gstrtsp-enumtypes.h
new file mode 100644
index 0000000000..47c84d5226
--- /dev/null
+++ b/include/gst/rtsp/gstrtsp-enumtypes.h
@@ -0,0 +1,82 @@
+
+/* This file is generated by glib-mkenums, do not modify it. This code is licensed under the same license as the containing project. Note that it links to GLib, so must comply with the LGPL linking clauses. */
+
+#pragma once
+
+            #include <glib-object.h>
+            #include <gst/rtsp/rtsp-prelude.h>
+
+            G_BEGIN_DECLS
+
+/* enumerations from "gstrtspdefs.h" */
+
+GST_RTSP_API
+GType gst_rtsp_result_get_type (void);
+#define GST_TYPE_RTSP_RESULT (gst_rtsp_result_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_event_get_type (void);
+#define GST_TYPE_RTSP_EVENT (gst_rtsp_event_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_family_get_type (void);
+#define GST_TYPE_RTSP_FAMILY (gst_rtsp_family_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_state_get_type (void);
+#define GST_TYPE_RTSP_STATE (gst_rtsp_state_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_version_get_type (void);
+#define GST_TYPE_RTSP_VERSION (gst_rtsp_version_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_method_get_type (void);
+#define GST_TYPE_RTSP_METHOD (gst_rtsp_method_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_auth_method_get_type (void);
+#define GST_TYPE_RTSP_AUTH_METHOD (gst_rtsp_auth_method_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_header_field_get_type (void);
+#define GST_TYPE_RTSP_HEADER_FIELD (gst_rtsp_header_field_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_status_code_get_type (void);
+#define GST_TYPE_RTSP_STATUS_CODE (gst_rtsp_status_code_get_type())
+
+/* enumerations from "gstrtspmessage.h" */
+
+GST_RTSP_API
+GType gst_rtsp_msg_type_get_type (void);
+#define GST_TYPE_RTSP_MSG_TYPE (gst_rtsp_msg_type_get_type())
+
+/* enumerations from "gstrtsprange.h" */
+
+GST_RTSP_API
+GType gst_rtsp_range_unit_get_type (void);
+#define GST_TYPE_RTSP_RANGE_UNIT (gst_rtsp_range_unit_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_time_type_get_type (void);
+#define GST_TYPE_RTSP_TIME_TYPE (gst_rtsp_time_type_get_type())
+
+/* enumerations from "gstrtsptransport.h" */
+
+GST_RTSP_API
+GType gst_rtsp_trans_mode_get_type (void);
+#define GST_TYPE_RTSP_TRANS_MODE (gst_rtsp_trans_mode_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_profile_get_type (void);
+#define GST_TYPE_RTSP_PROFILE (gst_rtsp_profile_get_type())
+
+GST_RTSP_API
+GType gst_rtsp_lower_trans_get_type (void);
+#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
+
+G_END_DECLS
+
+/* Generated data ends here */
+
diff --git a/include/gst/rtsp/gstrtsp.h b/include/gst/rtsp/gstrtsp.h
new file mode 100644
index 0000000000..eb1370e759
--- /dev/null
+++ b/include/gst/rtsp/gstrtsp.h
@@ -0,0 +1,33 @@
+/* GStreamer RTSP extension
+ * Copyright (C) 2007 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * rtsp.h: Generic include to pull in defs
+ *
+ * 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.
+ */
+#ifndef __GST_RTSP_H__
+#define __GST_RTSP_H__
+
+#include <gst/rtsp/gstrtspconnection.h>
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtsp-enumtypes.h>
+#include <gst/rtsp/gstrtspextension.h>
+#include <gst/rtsp/gstrtspmessage.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtsptransport.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#endif /* __GST_RTSP_H__ */
diff --git a/include/gst/rtsp/gstrtspconnection.h b/include/gst/rtsp/gstrtspconnection.h
new file mode 100644
index 0000000000..2eeb462bda
--- /dev/null
+++ b/include/gst/rtsp/gstrtspconnection.h
@@ -0,0 +1,395 @@
+/* GStreamer
+ * Copyright (C) <2005,2009> Wim Taymans <wim.taymans@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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_CONNECTION_H__
+#define __GST_RTSP_CONNECTION_H__
+
+#include <glib.h>
+
+#include <gst/gstconfig.h>
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtspurl.h>
+#include <gst/rtsp/gstrtspmessage.h>
+#include <gio/gio.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstRTSPConnection:
+ *
+ * Opaque RTSP connection object.
+ */
+typedef struct _GstRTSPConnection GstRTSPConnection;
+
+/* opening/closing a connection */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_create         (const GstRTSPUrl *url, GstRTSPConnection **conn);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_create_from_socket (GSocket * socket,
+                                                       const gchar * ip,
+                                                       guint16 port,
+                                                       const gchar * initial_buffer,
+                                                       GstRTSPConnection ** conn);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_accept                 (GSocket * socket, GstRTSPConnection ** conn, GCancellable * cancellable);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_connect_usec           (GstRTSPConnection * conn, gint64 timeout);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_connect_with_response_usec (GstRTSPConnection * conn, gint64 timeout, GstRTSPMessage * response);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_close                  (GstRTSPConnection *conn);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_free                   (GstRTSPConnection *conn);
+
+/* TLS connections */
+
+GST_RTSP_API
+GTlsConnection *     gst_rtsp_connection_get_tls                  (GstRTSPConnection * conn, GError ** error);
+
+GST_RTSP_API
+gboolean             gst_rtsp_connection_set_tls_validation_flags (GstRTSPConnection * conn, GTlsCertificateFlags flags);
+
+GST_RTSP_API
+GTlsCertificateFlags gst_rtsp_connection_get_tls_validation_flags (GstRTSPConnection * conn);
+
+GST_RTSP_API
+void                 gst_rtsp_connection_set_tls_database (GstRTSPConnection * conn, GTlsDatabase * database);
+
+GST_RTSP_API
+GTlsDatabase *       gst_rtsp_connection_get_tls_database (GstRTSPConnection * conn);
+
+GST_RTSP_API
+void                 gst_rtsp_connection_set_tls_interaction (GstRTSPConnection * conn, GTlsInteraction * interaction);
+
+GST_RTSP_API
+GTlsInteraction *    gst_rtsp_connection_get_tls_interaction (GstRTSPConnection * conn);
+
+typedef gboolean (*GstRTSPConnectionAcceptCertificateFunc) (GTlsConnection *conn,
+                                                            GTlsCertificate *peer_cert,
+                                                            GTlsCertificateFlags errors,
+                                                            gpointer user_data);
+GST_RTSP_API
+void                 gst_rtsp_connection_set_accept_certificate_func (GstRTSPConnection * conn,
+                                                                      GstRTSPConnectionAcceptCertificateFunc func,
+                                                                      gpointer user_data,
+                                                                      GDestroyNotify destroy_notify);
+
+/* sending/receiving raw bytes */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_read_usec      (GstRTSPConnection * conn, guint8 * data,
+                                                       guint size, gint64 timeout);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_write_usec     (GstRTSPConnection * conn, const guint8 * data,
+                                                       guint size, gint64 timeout);
+
+/* sending/receiving messages */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_send_usec      (GstRTSPConnection *conn, GstRTSPMessage *message,
+                                                       gint64 timeout);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_send_messages_usec (GstRTSPConnection *conn, GstRTSPMessage *messages, guint n_messages,
+                                                       gint64 timeout);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_receive_usec    (GstRTSPConnection *conn, GstRTSPMessage *message,
+                                                       gint64 timeout);
+
+/* status management */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_poll_usec      (GstRTSPConnection *conn, GstRTSPEvent events,
+                                                       GstRTSPEvent *revents, gint64 timeout);
+
+/* reset the timeout */
+
+GST_RTSP_API
+gint64             gst_rtsp_connection_next_timeout_usec (GstRTSPConnection *conn);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_reset_timeout  (GstRTSPConnection *conn);
+
+/* flushing state */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_flush          (GstRTSPConnection *conn, gboolean flush);
+
+/* HTTP proxy support */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_set_proxy      (GstRTSPConnection *conn,
+                                                       const gchar *host, guint port);
+
+/* configure authentication data */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_set_auth       (GstRTSPConnection *conn, GstRTSPAuthMethod method,
+                                                       const gchar *user, const gchar *pass);
+
+GST_RTSP_API
+void               gst_rtsp_connection_set_auth_param    (GstRTSPConnection *conn,
+                                                          const gchar * param,
+                                                          const gchar *value);
+
+GST_RTSP_API
+void               gst_rtsp_connection_clear_auth_params (GstRTSPConnection *conn);
+
+/* configure DSCP */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_set_qos_dscp   (GstRTSPConnection *conn,
+                                                       guint qos_dscp);
+
+/* Content-Length limit */
+GST_RTSP_API
+void               gst_rtsp_connection_set_content_length_limit (GstRTSPConnection *conn,
+                                                                 guint limit);
+
+/* accessors */
+
+GST_RTSP_API
+GstRTSPUrl *       gst_rtsp_connection_get_url        (const GstRTSPConnection *conn);
+
+GST_RTSP_API
+const gchar *      gst_rtsp_connection_get_ip         (const GstRTSPConnection *conn);
+
+GST_RTSP_API
+void               gst_rtsp_connection_set_ip         (GstRTSPConnection *conn, const gchar *ip);
+
+GST_RTSP_API
+GSocket *          gst_rtsp_connection_get_read_socket  (const GstRTSPConnection *conn);
+
+GST_RTSP_API
+GSocket *          gst_rtsp_connection_get_write_socket (const GstRTSPConnection *conn);
+
+GST_RTSP_API
+void               gst_rtsp_connection_set_http_mode  (GstRTSPConnection *conn,
+                                                       gboolean enable);
+
+/* tunneling */
+
+GST_RTSP_API
+void               gst_rtsp_connection_set_tunneled   (GstRTSPConnection *conn, gboolean tunneled);
+
+GST_RTSP_API
+gboolean           gst_rtsp_connection_is_tunneled    (const GstRTSPConnection *conn);
+
+GST_RTSP_API
+const gchar *      gst_rtsp_connection_get_tunnelid   (const GstRTSPConnection *conn);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_connection_do_tunnel      (GstRTSPConnection *conn, GstRTSPConnection *conn2);
+
+GST_RTSP_API
+void               gst_rtsp_connection_set_remember_session_id (GstRTSPConnection *conn, gboolean remember);
+
+GST_RTSP_API
+gboolean           gst_rtsp_connection_get_remember_session_id (GstRTSPConnection *conn);
+
+GST_RTSP_API
+void               gst_rtsp_connection_set_ignore_x_server_reply (GstRTSPConnection *conn, gboolean ignore);
+
+GST_RTSP_API
+gboolean           gst_rtsp_connection_get_ignore_x_server_reply (const GstRTSPConnection *conn);
+
+/* async IO */
+
+/**
+ * GstRTSPWatch:
+ *
+ * Opaque RTSP watch object that can be used for asynchronous RTSP
+ * operations.
+ */
+typedef struct _GstRTSPWatch GstRTSPWatch;
+
+/**
+ * GstRTSPWatchFuncs:
+ * @message_received: callback when a message was received
+ * @message_sent: callback when a message was sent
+ * @closed: callback when the connection is closed
+ * @error: callback when an error occurred
+ * @tunnel_start: a client started a tunneled connection. The tunnelid of the
+ *   connection must be saved.
+ * @tunnel_complete: a client finished a tunneled connection. In this callback
+ *   you usually pair the tunnelid of this connection with the saved one using
+ *   gst_rtsp_connection_do_tunnel().
+ * @error_full: callback when an error occurred with more information than
+ *   the @error callback.
+ * @tunnel_lost: callback when the post connection of a tunnel is closed.
+ * @tunnel_http_response: callback when an HTTP response to the GET request
+ *   is about to be sent for a tunneled connection. The response can be
+ *   modified in the callback. Since: 1.4.
+ *
+ * Callback functions from a #GstRTSPWatch.
+ */
+typedef struct {
+  GstRTSPResult     (*message_received) (GstRTSPWatch *watch, GstRTSPMessage *message,
+                                         gpointer user_data);
+  GstRTSPResult     (*message_sent)     (GstRTSPWatch *watch, guint id,
+                                         gpointer user_data);
+  GstRTSPResult     (*closed)           (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*error)            (GstRTSPWatch *watch, GstRTSPResult result,
+                                         gpointer user_data);
+  GstRTSPStatusCode (*tunnel_start)     (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*tunnel_complete)  (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*error_full)       (GstRTSPWatch *watch, GstRTSPResult result,
+                                         GstRTSPMessage *message, guint id,
+                                         gpointer user_data);
+  GstRTSPResult     (*tunnel_lost)      (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*tunnel_http_response) (GstRTSPWatch *watch,
+                                             GstRTSPMessage *request,
+                                             GstRTSPMessage *response,
+                                             gpointer user_data);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING-1];
+} GstRTSPWatchFuncs;
+
+GST_RTSP_API
+GstRTSPWatch *     gst_rtsp_watch_new                (GstRTSPConnection *conn,
+                                                      GstRTSPWatchFuncs *funcs,
+                                                      gpointer user_data,
+                                                      GDestroyNotify notify);
+
+GST_RTSP_API
+void               gst_rtsp_watch_reset              (GstRTSPWatch *watch);
+
+GST_RTSP_API
+void               gst_rtsp_watch_unref              (GstRTSPWatch *watch);
+
+GST_RTSP_API
+guint              gst_rtsp_watch_attach             (GstRTSPWatch *watch,
+                                                      GMainContext *context);
+
+GST_RTSP_API
+void               gst_rtsp_watch_set_send_backlog  (GstRTSPWatch *watch,
+                                                     gsize bytes, guint messages);
+
+GST_RTSP_API
+void               gst_rtsp_watch_get_send_backlog  (GstRTSPWatch *watch,
+                                                     gsize *bytes, guint *messages);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_watch_write_data         (GstRTSPWatch *watch,
+                                                      const guint8 *data,
+                                                      guint size, guint *id);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_watch_send_message       (GstRTSPWatch *watch,
+                                                      GstRTSPMessage *message,
+                                                      guint *id);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_watch_send_messages      (GstRTSPWatch *watch,
+                                                      GstRTSPMessage *messages,
+                                                      guint n_messages,
+                                                      guint *id);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_watch_wait_backlog_usec  (GstRTSPWatch * watch,
+                                                      gint64 timeout);
+
+GST_RTSP_API
+void               gst_rtsp_watch_set_flushing       (GstRTSPWatch * watch,
+                                                      gboolean flushing);
+
+#ifndef GST_DISABLE_DEPRECATED
+
+/* Deprecated */
+G_GNUC_BEGIN_IGNORE_DEPRECATIONS
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_connect_usec)
+GstRTSPResult      gst_rtsp_connection_connect                (GstRTSPConnection * conn, GTimeVal * timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_connect_with_response_usec)
+GstRTSPResult      gst_rtsp_connection_connect_with_response  (GstRTSPConnection * conn, GTimeVal * timeout, GstRTSPMessage * response);
+
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_read_usec)
+GstRTSPResult      gst_rtsp_connection_read           (GstRTSPConnection * conn, guint8 * data,
+                                                       guint size, GTimeVal * timeout);
+
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_write_usec)
+GstRTSPResult      gst_rtsp_connection_write          (GstRTSPConnection * conn, const guint8 * data,
+                                                       guint size, GTimeVal * timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_send_usec)
+GstRTSPResult      gst_rtsp_connection_send           (GstRTSPConnection *conn, GstRTSPMessage *message,
+                                                       GTimeVal *timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_send_messages_usec)
+GstRTSPResult      gst_rtsp_connection_send_messages  (GstRTSPConnection *conn, GstRTSPMessage *messages, guint n_messages,
+                                                       GTimeVal *timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_receive_usec)
+GstRTSPResult      gst_rtsp_connection_receive        (GstRTSPConnection *conn, GstRTSPMessage *message,
+                                                       GTimeVal *timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_poll_usec)
+GstRTSPResult      gst_rtsp_connection_poll           (GstRTSPConnection *conn, GstRTSPEvent events,
+                                                       GstRTSPEvent *revents, GTimeVal *timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_next_timeout_usec)
+GstRTSPResult      gst_rtsp_connection_next_timeout   (GstRTSPConnection *conn, GTimeVal *timeout);
+
+GST_RTSP_DEPRECATED_FOR (gst_rtsp_watch_wait_backlog_usec)
+GstRTSPResult      gst_rtsp_watch_wait_backlog       (GstRTSPWatch * watch,
+                                                      GTimeVal *timeout);
+
+G_GNUC_END_IGNORE_DEPRECATIONS
+
+#endif /* GST_DISABLE_DEPRECATED */
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CONNECTION_H__ */
diff --git a/include/gst/rtsp/gstrtspdefs.h b/include/gst/rtsp/gstrtspdefs.h
new file mode 100644
index 0000000000..8a028c381b
--- /dev/null
+++ b/include/gst/rtsp/gstrtspdefs.h
@@ -0,0 +1,453 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_DEFS_H__
+#define __GST_RTSP_DEFS_H__
+
+#include <gst/gst.h>
+#include <gst/rtsp/rtsp-prelude.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GST_RTSP_CHECK:
+ * @stmt: a statement
+ * @label: a label
+ *
+ * Macro that checks the return value of @stmt and jumps to @label when it does
+ * not equal #GST_RTSP_OK.
+ */
+#define GST_RTSP_CHECK(stmt, label)  \
+G_STMT_START { \
+  if (G_UNLIKELY ((res = (stmt)) != GST_RTSP_OK)) \
+    goto label; \
+} G_STMT_END
+
+/**
+ * GstRTSPResult:
+ * @GST_RTSP_OK: no error
+ * @GST_RTSP_ERROR: some unspecified error occurred
+ * @GST_RTSP_EINVAL: invalid arguments were provided to a function
+ * @GST_RTSP_EINTR: an operation was canceled
+ * @GST_RTSP_ENOMEM: no memory was available for the operation
+ * @GST_RTSP_ERESOLV: a host resolve error occurred
+ * @GST_RTSP_ENOTIMPL: function not implemented
+ * @GST_RTSP_ESYS: a system error occurred, errno contains more details
+ * @GST_RTSP_EPARSE: a parsing error occurred
+ * @GST_RTSP_EWSASTART: windows networking could not start
+ * @GST_RTSP_EWSAVERSION: windows networking stack has wrong version
+ * @GST_RTSP_EEOF: end-of-file was reached
+ * @GST_RTSP_ENET: a network problem occurred, h_errno contains more details
+ * @GST_RTSP_ENOTIP: the host is not an IP host
+ * @GST_RTSP_ETIMEOUT: a timeout occurred
+ * @GST_RTSP_ETGET: the tunnel GET request has been performed
+ * @GST_RTSP_ETPOST: the tunnel POST request has been performed
+ * @GST_RTSP_ELAST: last error
+ *
+ * Result codes from the RTSP functions.
+ */
+typedef enum {
+  GST_RTSP_OK          =  0,
+  /* errors */
+  GST_RTSP_ERROR       = -1,
+  GST_RTSP_EINVAL      = -2,
+  GST_RTSP_EINTR       = -3,
+  GST_RTSP_ENOMEM      = -4,
+  GST_RTSP_ERESOLV     = -5,
+  GST_RTSP_ENOTIMPL    = -6,
+  GST_RTSP_ESYS        = -7,
+  GST_RTSP_EPARSE      = -8,
+  GST_RTSP_EWSASTART   = -9,
+  GST_RTSP_EWSAVERSION = -10,
+  GST_RTSP_EEOF        = -11,
+  GST_RTSP_ENET        = -12,
+  GST_RTSP_ENOTIP      = -13,
+  GST_RTSP_ETIMEOUT    = -14,
+  GST_RTSP_ETGET       = -15,
+  GST_RTSP_ETPOST      = -16,
+
+  GST_RTSP_ELAST       = -17
+} GstRTSPResult;
+
+/**
+ * GstRTSPEvent:
+ * @GST_RTSP_EV_READ: connection is readable
+ * @GST_RTSP_EV_WRITE: connection is writable
+ *
+ * The possible events for the connection.
+ */
+typedef enum {
+  GST_RTSP_EV_READ  = (1 << 0),
+  GST_RTSP_EV_WRITE = (1 << 1)
+} GstRTSPEvent;
+
+/**
+ * GstRTSPFamily:
+ * @GST_RTSP_FAM_NONE: unknown network family
+ * @GST_RTSP_FAM_INET: internet
+ * @GST_RTSP_FAM_INET6: internet V6
+ *
+ * The possible network families.
+ */
+typedef enum {
+  GST_RTSP_FAM_NONE,
+  GST_RTSP_FAM_INET,
+  GST_RTSP_FAM_INET6
+} GstRTSPFamily;
+
+/**
+ * GstRTSPState:
+ * @GST_RTSP_STATE_INVALID: invalid state
+ * @GST_RTSP_STATE_INIT: initializing
+ * @GST_RTSP_STATE_READY: ready for operation
+ * @GST_RTSP_STATE_SEEKING: seeking in progress
+ * @GST_RTSP_STATE_PLAYING: playing
+ * @GST_RTSP_STATE_RECORDING: recording
+ *
+ * The different RTSP states.
+ */
+typedef enum {
+  GST_RTSP_STATE_INVALID,
+  GST_RTSP_STATE_INIT,
+  GST_RTSP_STATE_READY,
+  GST_RTSP_STATE_SEEKING,
+  GST_RTSP_STATE_PLAYING,
+  GST_RTSP_STATE_RECORDING
+} GstRTSPState;
+
+/**
+ * GstRTSPVersion:
+ * @GST_RTSP_VERSION_INVALID: unknown/invalid version
+ * @GST_RTSP_VERSION_1_0: version 1.0
+ * @GST_RTSP_VERSION_1_1: version 1.1.
+ * @GST_RTSP_VERSION_2_0: version 2.0.
+ *
+ * The supported RTSP versions.
+ */
+typedef enum {
+  GST_RTSP_VERSION_INVALID = 0x00,
+  GST_RTSP_VERSION_1_0     = 0x10,
+  GST_RTSP_VERSION_1_1     = 0x11,
+  GST_RTSP_VERSION_2_0     = 0x20
+} GstRTSPVersion;
+
+/**
+ * GstRTSPMethod:
+ * @GST_RTSP_INVALID: invalid method
+ * @GST_RTSP_DESCRIBE: the DESCRIBE method
+ * @GST_RTSP_ANNOUNCE: the ANNOUNCE method
+ * @GST_RTSP_GET_PARAMETER: the GET_PARAMETER method
+ * @GST_RTSP_OPTIONS: the OPTIONS method
+ * @GST_RTSP_PAUSE: the PAUSE method
+ * @GST_RTSP_PLAY: the PLAY method
+ * @GST_RTSP_RECORD: the RECORD method
+ * @GST_RTSP_REDIRECT: the REDIRECT method
+ * @GST_RTSP_SETUP: the SETUP method
+ * @GST_RTSP_SET_PARAMETER: the SET_PARAMETER method
+ * @GST_RTSP_TEARDOWN: the TEARDOWN method
+ * @GST_RTSP_GET: the GET method (HTTP).
+ * @GST_RTSP_POST: the POST method (HTTP).
+ *
+ * The different supported RTSP methods.
+ */
+typedef enum {
+  GST_RTSP_INVALID          = 0,
+  GST_RTSP_DESCRIBE         = (1 <<  0),
+  GST_RTSP_ANNOUNCE         = (1 <<  1),
+  GST_RTSP_GET_PARAMETER    = (1 <<  2),
+  GST_RTSP_OPTIONS          = (1 <<  3),
+  GST_RTSP_PAUSE            = (1 <<  4),
+  GST_RTSP_PLAY             = (1 <<  5),
+  GST_RTSP_RECORD           = (1 <<  6),
+  GST_RTSP_REDIRECT         = (1 <<  7),
+  GST_RTSP_SETUP            = (1 <<  8),
+  GST_RTSP_SET_PARAMETER    = (1 <<  9),
+  GST_RTSP_TEARDOWN         = (1 << 10),
+  GST_RTSP_GET              = (1 << 11),
+  GST_RTSP_POST             = (1 << 12)
+} GstRTSPMethod;
+
+/**
+ * GstRTSPAuthMethod:
+ * @GST_RTSP_AUTH_NONE: no authentication
+ * @GST_RTSP_AUTH_BASIC: basic authentication
+ * @GST_RTSP_AUTH_DIGEST: digest authentication
+ *
+ * Authentication methods, ordered by strength
+ */
+typedef enum {
+  GST_RTSP_AUTH_NONE    = 0x00,
+  GST_RTSP_AUTH_BASIC   = 0x01,
+  GST_RTSP_AUTH_DIGEST  = 0x02
+} GstRTSPAuthMethod;
+
+/**
+ * GST_RTSP_AUTH_MAX:
+ *
+ * Strongest available authentication method
+ */
+#define GST_RTSP_AUTH_MAX GST_RTSP_AUTH_DIGEST
+
+/**
+ * GstRTSPHeaderField:
+ *
+ * Enumeration of rtsp header fields
+ */
+typedef enum {
+  GST_RTSP_HDR_INVALID,
+
+  /*
+   * R = Request
+   * r = response
+   * g = general
+   * e = entity
+   */
+  GST_RTSP_HDR_ACCEPT,              /* Accept               R      opt.      entity */
+  GST_RTSP_HDR_ACCEPT_ENCODING,     /* Accept-Encoding      R      opt.      entity */
+  GST_RTSP_HDR_ACCEPT_LANGUAGE,     /* Accept-Language      R      opt.      all */
+  GST_RTSP_HDR_ALLOW,               /* Allow                r      opt.      all */
+  GST_RTSP_HDR_AUTHORIZATION,       /* Authorization        R      opt.      all */
+  GST_RTSP_HDR_BANDWIDTH,           /* Bandwidth            R      opt.      all */
+  GST_RTSP_HDR_BLOCKSIZE,           /* Blocksize            R      opt.      all but OPTIONS, TEARDOWN */
+  GST_RTSP_HDR_CACHE_CONTROL,       /* Cache-Control        g      opt.      SETUP */
+  GST_RTSP_HDR_CONFERENCE,          /* Conference           R      opt.      SETUP */
+  GST_RTSP_HDR_CONNECTION,          /* Connection           g      req.      all */
+  GST_RTSP_HDR_CONTENT_BASE,        /* Content-Base         e      opt.      entity */
+  GST_RTSP_HDR_CONTENT_ENCODING,    /* Content-Encoding     e      req.      SET_PARAMETER, DESCRIBE, ANNOUNCE */
+  GST_RTSP_HDR_CONTENT_LANGUAGE,    /* Content-Language     e      req.      DESCRIBE, ANNOUNCE */
+  GST_RTSP_HDR_CONTENT_LENGTH,      /* Content-Length       e      req.      SET_PARAMETER, ANNOUNCE, entity */
+  GST_RTSP_HDR_CONTENT_LOCATION,    /* Content-Location     e      opt.      entity */
+  GST_RTSP_HDR_CONTENT_TYPE,        /* Content-Type         e      req.      SET_PARAMETER, ANNOUNCE, entity */
+  GST_RTSP_HDR_CSEQ,                /* CSeq                 g      req.      all */
+  GST_RTSP_HDR_DATE,                /* Date                 g      opt.      all */
+  GST_RTSP_HDR_EXPIRES,             /* Expires              e      opt.      DESCRIBE, ANNOUNCE */
+  GST_RTSP_HDR_FROM,                /* From                 R      opt.      all */
+  GST_RTSP_HDR_IF_MODIFIED_SINCE,   /* If-Modified-Since    R      opt.      DESCRIBE, SETUP */
+  GST_RTSP_HDR_LAST_MODIFIED,       /* Last-Modified        e      opt.      entity */
+  GST_RTSP_HDR_PROXY_AUTHENTICATE,  /* Proxy-Authenticate */
+  GST_RTSP_HDR_PROXY_REQUIRE,       /* Proxy-Require        R      req.      all */
+  GST_RTSP_HDR_PUBLIC,              /* Public               r      opt.      all */
+  GST_RTSP_HDR_RANGE,               /* Range                Rr     opt.      PLAY, PAUSE, RECORD */
+  GST_RTSP_HDR_REFERER,             /* Referrer              R      opt.      all */
+  GST_RTSP_HDR_REQUIRE,             /* Require              R      req.      all */
+  GST_RTSP_HDR_RETRY_AFTER,         /* Retry-After          r      opt.      all */
+  GST_RTSP_HDR_RTP_INFO,            /* RTP-Info             r      req.      PLAY */
+  GST_RTSP_HDR_SCALE,               /* Scale                Rr     opt.      PLAY, RECORD */
+  GST_RTSP_HDR_SESSION,             /* Session              Rr     req.      all but SETUP, OPTIONS */
+  GST_RTSP_HDR_SERVER,              /* Server               r      opt.      all */
+  GST_RTSP_HDR_SPEED,               /* Speed                Rr     opt.      PLAY */
+  GST_RTSP_HDR_TRANSPORT,           /* Transport            Rr     req.      SETUP */
+  GST_RTSP_HDR_UNSUPPORTED,         /* Unsupported          r      req.      all */
+  GST_RTSP_HDR_USER_AGENT,          /* User-Agent           R      opt.      all */
+  GST_RTSP_HDR_VIA,                 /* Via                  g      opt.      all */
+  GST_RTSP_HDR_WWW_AUTHENTICATE,    /* WWW-Authenticate     r      opt.      all */
+
+  /* Real extensions */
+  GST_RTSP_HDR_CLIENT_CHALLENGE,    /* ClientChallenge */
+  GST_RTSP_HDR_REAL_CHALLENGE1,     /* RealChallenge1 */
+  GST_RTSP_HDR_REAL_CHALLENGE2,     /* RealChallenge2 */
+  GST_RTSP_HDR_REAL_CHALLENGE3,     /* RealChallenge3 */
+  GST_RTSP_HDR_SUBSCRIBE,           /* Subscribe */
+  GST_RTSP_HDR_ALERT,               /* Alert */
+  GST_RTSP_HDR_CLIENT_ID,           /* ClientID */
+  GST_RTSP_HDR_COMPANY_ID,          /* CompanyID */
+  GST_RTSP_HDR_GUID,                /* GUID */
+  GST_RTSP_HDR_REGION_DATA,         /* RegionData */
+  GST_RTSP_HDR_MAX_ASM_WIDTH,       /* SupportsMaximumASMBandwidth */
+  GST_RTSP_HDR_LANGUAGE,            /* Language */
+  GST_RTSP_HDR_PLAYER_START_TIME,   /* PlayerStarttime */
+
+  GST_RTSP_HDR_LOCATION,            /* Location */
+
+  GST_RTSP_HDR_ETAG,                /* ETag */
+  GST_RTSP_HDR_IF_MATCH,            /* If-Match */
+
+  /* WM extensions [MS-RTSP] */
+  GST_RTSP_HDR_ACCEPT_CHARSET,      /* Accept-Charset */
+  GST_RTSP_HDR_SUPPORTED,           /* Supported */
+  GST_RTSP_HDR_VARY,                /* Vary */
+  GST_RTSP_HDR_X_ACCELERATE_STREAMING,    /* X-Accelerate-Streaming */
+  GST_RTSP_HDR_X_ACCEPT_AUTHENT,    /* X-Accept-Authentication */
+  GST_RTSP_HDR_X_ACCEPT_PROXY_AUTHENT,    /* X-Accept-Proxy-Authentication */
+  GST_RTSP_HDR_X_BROADCAST_ID,      /* X-Broadcast-Id */
+  GST_RTSP_HDR_X_BURST_STREAMING,   /* X-Burst-Streaming */
+  GST_RTSP_HDR_X_NOTICE,            /* X-Notice */
+  GST_RTSP_HDR_X_PLAYER_LAG_TIME,   /* X-Player-Lag-Time */
+  GST_RTSP_HDR_X_PLAYLIST,          /* X-Playlist */
+  GST_RTSP_HDR_X_PLAYLIST_CHANGE_NOTICE,  /* X-Playlist-Change-Notice */
+  GST_RTSP_HDR_X_PLAYLIST_GEN_ID,   /* X-Playlist-Gen-Id */
+  GST_RTSP_HDR_X_PLAYLIST_SEEK_ID,  /* X-Playlist-Seek-Id */
+  GST_RTSP_HDR_X_PROXY_CLIENT_AGENT,      /* X-Proxy-Client-Agent */
+  GST_RTSP_HDR_X_PROXY_CLIENT_VERB, /* X-Proxy-Client-Verb */
+  GST_RTSP_HDR_X_RECEDING_PLAYLISTCHANGE, /* X-Receding-PlaylistChange */
+  GST_RTSP_HDR_X_RTP_INFO,          /* X-RTP-Info */
+  GST_RTSP_HDR_X_STARTUPPROFILE,    /* X-StartupProfile */
+
+  GST_RTSP_HDR_TIMESTAMP,           /* Timestamp */
+
+  GST_RTSP_HDR_AUTHENTICATION_INFO, /* Authentication-Info */
+  GST_RTSP_HDR_HOST,                /* Host */
+  GST_RTSP_HDR_PRAGMA,              /* Pragma */
+  GST_RTSP_HDR_X_SERVER_IP_ADDRESS, /* X-Server-IP-Address */
+  GST_RTSP_HDR_X_SESSIONCOOKIE,     /* x-sessioncookie */
+
+  GST_RTSP_HDR_RTCP_INTERVAL,       /* RTCP-Interval */
+
+  /* Since: 1.4 */
+  GST_RTSP_HDR_KEYMGMT,             /* KeyMgmt */
+
+  /* Since: 1.14 */
+  GST_RTSP_HDR_PIPELINED_REQUESTS,  /* Pipelined-Requests Rr     opt.      SETUP */
+  GST_RTSP_HDR_MEDIA_PROPERTIES,    /* Media-Properties   Rr     opt.      SETUP */
+  GST_RTSP_HDR_SEEK_STYLE,          /* Seek-Type          Rr     opt.      PLAY */
+  GST_RTSP_HDR_ACCEPT_RANGES,       /* Accept-Ranges      Rr     opt.      SETUP, GET_PARAMETER */
+
+  /* Onvif extensions */
+  GST_RTSP_HDR_FRAMES,              /* Frames             Rr     opt.      PLAY */
+  GST_RTSP_HDR_RATE_CONTROL,        /* Rate-control       Rr     opt.      PLAY */
+
+  GST_RTSP_HDR_LAST
+} GstRTSPHeaderField;
+
+/**
+ * GstRTSPStatusCode:
+ *
+ * Enumeration of rtsp status codes
+ */
+typedef enum {
+  GST_RTSP_STS_INVALID                              = 0,
+  GST_RTSP_STS_CONTINUE                             = 100,
+  GST_RTSP_STS_OK                                   = 200,
+  GST_RTSP_STS_CREATED                              = 201,
+  GST_RTSP_STS_LOW_ON_STORAGE                       = 250,
+  GST_RTSP_STS_MULTIPLE_CHOICES                     = 300,
+  GST_RTSP_STS_MOVED_PERMANENTLY                    = 301,
+  GST_RTSP_STS_MOVE_TEMPORARILY                     = 302,
+  GST_RTSP_STS_SEE_OTHER                            = 303,
+  GST_RTSP_STS_NOT_MODIFIED                         = 304,
+  GST_RTSP_STS_USE_PROXY                            = 305,
+  GST_RTSP_STS_BAD_REQUEST                          = 400,
+  GST_RTSP_STS_UNAUTHORIZED                         = 401,
+  GST_RTSP_STS_PAYMENT_REQUIRED                     = 402,
+  GST_RTSP_STS_FORBIDDEN                            = 403,
+  GST_RTSP_STS_NOT_FOUND                            = 404,
+  GST_RTSP_STS_METHOD_NOT_ALLOWED                   = 405,
+  GST_RTSP_STS_NOT_ACCEPTABLE                       = 406,
+  GST_RTSP_STS_PROXY_AUTH_REQUIRED                  = 407,
+  GST_RTSP_STS_REQUEST_TIMEOUT                      = 408,
+  GST_RTSP_STS_GONE                                 = 410,
+  GST_RTSP_STS_LENGTH_REQUIRED                      = 411,
+  GST_RTSP_STS_PRECONDITION_FAILED                  = 412,
+  GST_RTSP_STS_REQUEST_ENTITY_TOO_LARGE             = 413,
+  GST_RTSP_STS_REQUEST_URI_TOO_LARGE                = 414,
+  GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE               = 415,
+  GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD             = 451,
+  GST_RTSP_STS_CONFERENCE_NOT_FOUND                 = 452,
+  GST_RTSP_STS_NOT_ENOUGH_BANDWIDTH                 = 453,
+  GST_RTSP_STS_SESSION_NOT_FOUND                    = 454,
+  GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE       = 455,
+  GST_RTSP_STS_HEADER_FIELD_NOT_VALID_FOR_RESOURCE  = 456,
+  GST_RTSP_STS_INVALID_RANGE                        = 457,
+  GST_RTSP_STS_PARAMETER_IS_READONLY                = 458,
+  GST_RTSP_STS_AGGREGATE_OPERATION_NOT_ALLOWED      = 459,
+  GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED     = 460,
+  GST_RTSP_STS_UNSUPPORTED_TRANSPORT                = 461,
+  GST_RTSP_STS_DESTINATION_UNREACHABLE              = 462,
+  GST_RTSP_STS_KEY_MANAGEMENT_FAILURE               = 463, /* since 1.4 */
+  GST_RTSP_STS_INTERNAL_SERVER_ERROR                = 500,
+  GST_RTSP_STS_NOT_IMPLEMENTED                      = 501,
+  GST_RTSP_STS_BAD_GATEWAY                          = 502,
+  GST_RTSP_STS_SERVICE_UNAVAILABLE                  = 503,
+  GST_RTSP_STS_GATEWAY_TIMEOUT                      = 504,
+  GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED           = 505,
+  GST_RTSP_STS_OPTION_NOT_SUPPORTED                 = 551
+} GstRTSPStatusCode;
+
+GST_RTSP_API
+gchar*             gst_rtsp_strresult          (GstRTSPResult result);
+
+GST_RTSP_API
+const gchar*       gst_rtsp_method_as_text     (GstRTSPMethod method);
+
+GST_RTSP_API
+const gchar*       gst_rtsp_version_as_text    (GstRTSPVersion version);
+
+GST_RTSP_API
+const gchar*       gst_rtsp_header_as_text     (GstRTSPHeaderField field);
+
+GST_RTSP_API
+const gchar*       gst_rtsp_status_as_text     (GstRTSPStatusCode code);
+
+GST_RTSP_API
+gchar*             gst_rtsp_options_as_text    (GstRTSPMethod options);
+
+GST_RTSP_API
+GstRTSPMethod      gst_rtsp_options_from_text  (const gchar *options);
+
+GST_RTSP_API
+GstRTSPHeaderField gst_rtsp_find_header_field  (const gchar *header);
+
+GST_RTSP_API
+GstRTSPMethod      gst_rtsp_find_method        (const gchar *method);
+
+GST_RTSP_API
+gboolean           gst_rtsp_header_allow_multiple (GstRTSPHeaderField field);
+
+GST_RTSP_API
+gchar *            gst_rtsp_generate_digest_auth_response (const gchar *algorithm,
+                                                           const gchar *method,
+                                                           const gchar *realm,
+                                                           const gchar *username,
+                                                           const gchar *password,
+                                                           const gchar *uri,
+                                                           const gchar *nonce);
+
+GST_RTSP_API
+gchar *            gst_rtsp_generate_digest_auth_response_from_md5 (const gchar *algorithm,
+                                                                    const gchar * method,
+                                                                    const gchar * md5,
+                                                                    const gchar * uri,
+                                                                    const gchar * nonce);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_DEFS_H__ */
diff --git a/include/gst/rtsp/gstrtspextension.h b/include/gst/rtsp/gstrtspextension.h
new file mode 100644
index 0000000000..60b129b46a
--- /dev/null
+++ b/include/gst/rtsp/gstrtspextension.h
@@ -0,0 +1,122 @@
+/* GStreamer RTSP Extension
+ * Copyright (C) 2007 Wim Taymans <wim@fluendo.com>
+ *
+ * gstrtspextension.h: RTSP Extension interface.
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTSP_EXTENSION_H__
+#define __GST_RTSP_EXTENSION_H__
+
+#include <gst/gst.h>
+
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/rtsp/gstrtsptransport.h>
+#include <gst/rtsp/gstrtspmessage.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_EXTENSION \
+  (gst_rtsp_extension_get_type ())
+#define GST_RTSP_EXTENSION(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_EXTENSION, GstRTSPExtension))
+#define GST_IS_RTSP_EXTENSION(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_EXTENSION))
+#define GST_RTSP_EXTENSION_GET_INTERFACE(inst) \
+  (G_TYPE_INSTANCE_GET_INTERFACE ((inst), GST_TYPE_RTSP_EXTENSION, GstRTSPExtensionInterface))
+
+/**
+ * GstRTSPExtensionInterface:
+ *
+ * An interface representing RTSP extensions.
+ */
+typedef struct _GstRTSPExtension GstRTSPExtension;
+typedef struct _GstRTSPExtensionInterface GstRTSPExtensionInterface;
+
+struct _GstRTSPExtensionInterface {
+  GTypeInterface parent;
+
+  /* vfunctions */
+  gboolean      (*detect_server)    (GstRTSPExtension *ext, GstRTSPMessage *resp);
+
+  GstRTSPResult (*before_send)      (GstRTSPExtension *ext, GstRTSPMessage *req);
+  GstRTSPResult (*after_send)       (GstRTSPExtension *ext, GstRTSPMessage *req, GstRTSPMessage *resp);
+
+  GstRTSPResult (*parse_sdp)        (GstRTSPExtension *ext, GstSDPMessage *sdp, GstStructure *s);
+  GstRTSPResult (*setup_media)      (GstRTSPExtension *ext, GstSDPMedia *media);
+
+  gboolean      (*configure_stream) (GstRTSPExtension *ext, GstCaps *caps);
+
+  GstRTSPResult (*get_transports)   (GstRTSPExtension *ext, GstRTSPLowerTrans protocols, gchar **transport);
+
+  GstRTSPResult (*stream_select)    (GstRTSPExtension *ext, GstRTSPUrl *url);
+
+  /* signals */
+  GstRTSPResult (*send)             (GstRTSPExtension *ext, GstRTSPMessage *req, GstRTSPMessage *resp);
+
+  /* ABI: more vfunctions added later */
+  GstRTSPResult (*receive_request)  (GstRTSPExtension *ext, GstRTSPMessage *req);
+
+  /*< private >*/
+  gpointer                 _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_API
+GType           gst_rtsp_extension_get_type          (void);
+
+/* invoke vfunction on interface */
+
+GST_RTSP_API
+gboolean        gst_rtsp_extension_detect_server     (GstRTSPExtension *ext, GstRTSPMessage *resp);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_before_send       (GstRTSPExtension *ext, GstRTSPMessage *req);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_after_send        (GstRTSPExtension *ext, GstRTSPMessage *req,
+                                                      GstRTSPMessage *resp);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_parse_sdp         (GstRTSPExtension *ext, GstSDPMessage *sdp,
+                                                      GstStructure *s);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_setup_media       (GstRTSPExtension *ext, GstSDPMedia *media);
+
+GST_RTSP_API
+gboolean        gst_rtsp_extension_configure_stream  (GstRTSPExtension *ext, GstCaps *caps);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_get_transports    (GstRTSPExtension *ext, GstRTSPLowerTrans protocols,
+                                                      gchar **transport);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_stream_select     (GstRTSPExtension *ext, GstRTSPUrl *url);
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_receive_request   (GstRTSPExtension *ext, GstRTSPMessage *req);
+
+/* signal emission */
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_extension_send              (GstRTSPExtension *ext, GstRTSPMessage *req,
+                                                      GstRTSPMessage *resp);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_EXTENSION_H__ */
diff --git a/include/gst/rtsp/gstrtspmessage.h b/include/gst/rtsp/gstrtspmessage.h
new file mode 100644
index 0000000000..ed6a284c91
--- /dev/null
+++ b/include/gst/rtsp/gstrtspmessage.h
@@ -0,0 +1,347 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_MESSAGE_H__
+#define __GST_RTSP_MESSAGE_H__
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspdefs.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstRTSPMsgType:
+ * @GST_RTSP_MESSAGE_INVALID: invalid message type
+ * @GST_RTSP_MESSAGE_REQUEST: RTSP request message
+ * @GST_RTSP_MESSAGE_RESPONSE: RTSP response message
+ * @GST_RTSP_MESSAGE_HTTP_REQUEST: HTTP request message.
+ * @GST_RTSP_MESSAGE_HTTP_RESPONSE: HTTP response message.
+ * @GST_RTSP_MESSAGE_DATA: data message
+ *
+ * The type of a message.
+ */
+typedef enum
+{
+  GST_RTSP_MESSAGE_INVALID,
+  GST_RTSP_MESSAGE_REQUEST,
+  GST_RTSP_MESSAGE_RESPONSE,
+  GST_RTSP_MESSAGE_HTTP_REQUEST,
+  GST_RTSP_MESSAGE_HTTP_RESPONSE,
+  GST_RTSP_MESSAGE_DATA
+} GstRTSPMsgType;
+
+typedef struct _GstRTSPMessage GstRTSPMessage;
+
+/**
+ * GstRTSPMessage:
+ * @type: the message type
+ *
+ * An RTSP message containing request, response or data messages. Depending on
+ * the @type, the appropriate structure may be accessed.
+ */
+struct _GstRTSPMessage
+{
+  GstRTSPMsgType    type;
+
+  union {
+    struct {
+      GstRTSPMethod      method;
+      gchar             *uri;
+      GstRTSPVersion     version;
+    } request;
+    struct {
+      GstRTSPStatusCode  code;
+      gchar             *reason;
+      GstRTSPVersion     version;
+    } response;
+    struct {
+      guint8             channel;
+    } data;
+  } type_data;
+
+  /*< private >*/
+  GArray        *hdr_fields;
+
+  guint8        *body;
+  guint          body_size;
+
+  GstBuffer     *body_buffer;
+  gpointer _gst_reserved[GST_PADDING-1];
+};
+
+GST_RTSP_API
+GType                   gst_rtsp_msg_get_type            (void);
+
+#define GST_TYPE_RTSP_MESSAGE           (gst_rtsp_msg_get_type())
+#define GST_RTSP_MESSAGE_CAST(object)   ((GstRTSPMessage *)(object))
+#define GST_RTSP_MESSAGE(object)        (GST_RTSP_MESSAGE_CAST(object))
+
+/* memory management */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_new             (GstRTSPMessage **msg);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_init            (GstRTSPMessage *msg);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_unset           (GstRTSPMessage *msg);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_free            (GstRTSPMessage *msg);
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_copy            (const GstRTSPMessage *msg,
+                                                     GstRTSPMessage **copy);
+
+GST_RTSP_API
+GstRTSPMsgType     gst_rtsp_message_get_type        (GstRTSPMessage *msg);
+
+/* request */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_new_request     (GstRTSPMessage **msg,
+                                                     GstRTSPMethod method,
+                                                     const gchar *uri);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_init_request    (GstRTSPMessage *msg,
+                                                     GstRTSPMethod method,
+                                                     const gchar *uri);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_parse_request   (GstRTSPMessage *msg,
+                                                     GstRTSPMethod *method,
+                                                     const gchar **uri,
+                                                     GstRTSPVersion *version);
+
+/* response */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_new_response    (GstRTSPMessage **msg,
+                                                     GstRTSPStatusCode code,
+                                                     const gchar *reason,
+                                                     const GstRTSPMessage *request);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_init_response   (GstRTSPMessage *msg,
+                                                     GstRTSPStatusCode code,
+                                                     const gchar *reason,
+                                                     const GstRTSPMessage *request);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_parse_response  (GstRTSPMessage *msg,
+                                                     GstRTSPStatusCode *code,
+                                                     const gchar **reason,
+                                                     GstRTSPVersion *version);
+
+/* data */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_new_data        (GstRTSPMessage **msg,
+                                                     guint8 channel);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_init_data       (GstRTSPMessage *msg,
+                                                     guint8 channel);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_parse_data      (GstRTSPMessage *msg,
+                                                     guint8 *channel);
+
+/* headers */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_add_header      (GstRTSPMessage *msg,
+                                                     GstRTSPHeaderField field,
+                                                     const gchar *value);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_take_header     (GstRTSPMessage *msg,
+                                                     GstRTSPHeaderField field,
+                                                     gchar *value);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_remove_header   (GstRTSPMessage *msg,
+                                                     GstRTSPHeaderField field,
+                                                     gint indx);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_get_header      (const GstRTSPMessage *msg,
+                                                     GstRTSPHeaderField field,
+                                                     gchar **value,
+                                                     gint indx);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_add_header_by_name    (GstRTSPMessage * msg,
+                                                           const gchar    * header,
+                                                           const gchar    * value);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_take_header_by_name   (GstRTSPMessage * msg,
+                                                           const gchar    * header,
+                                                           gchar          * value);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_remove_header_by_name (GstRTSPMessage * msg,
+                                                           const gchar    * header,
+                                                           gint             index);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_get_header_by_name    (GstRTSPMessage * msg,
+                                                           const gchar    * header,
+                                                           gchar         ** value,
+                                                           gint             index);
+
+/* header serialization */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_append_headers  (const GstRTSPMessage *msg,
+                                                     GString *str);
+
+/* handling the body */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_set_body        (GstRTSPMessage *msg,
+                                                     const guint8 *data,
+                                                     guint size);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_take_body       (GstRTSPMessage *msg,
+                                                     guint8 *data,
+                                                     guint size);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_get_body        (const GstRTSPMessage *msg,
+                                                     guint8 **data,
+                                                     guint *size);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_steal_body      (GstRTSPMessage *msg,
+                                                     guint8 **data,
+                                                     guint *size);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_set_body_buffer (GstRTSPMessage *msg,
+                                                     GstBuffer * buffer);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_take_body_buffer(GstRTSPMessage *msg,
+                                                     GstBuffer * buffer);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_get_body_buffer (const GstRTSPMessage *msg,
+                                                     GstBuffer ** buffer);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_steal_body_buffer(GstRTSPMessage *msg,
+                                                      GstBuffer ** buffer);
+
+GST_RTSP_API
+gboolean           gst_rtsp_message_has_body_buffer(const GstRTSPMessage *msg);
+
+typedef struct _GstRTSPAuthCredential GstRTSPAuthCredential;
+typedef struct _GstRTSPAuthParam GstRTSPAuthParam;
+
+/**
+ * GstRTSPAuthCredential:
+ * @scheme: a #GstRTSPAuthMethod
+ * @params: A NULL-terminated array of #GstRTSPAuthParam
+ * @authorization: The authorization for the basic schem
+ *
+ * RTSP Authentication credentials
+ *
+ * Since: 1.12
+ */
+struct _GstRTSPAuthCredential {
+  GstRTSPAuthMethod scheme;
+
+  /* For Basic/Digest WWW-Authenticate and Digest
+   * Authorization */
+  GstRTSPAuthParam **params; /* NULL terminated */
+
+  /* For Basic Authorization */
+  gchar *authorization;
+};
+
+/**
+ * GstRTSPAuthParam:
+ * @name: The name of the parameter
+ * @value: The value of the parameter
+ *
+ * RTSP Authentication parameter
+ *
+ * Since: 1.12
+ */
+struct _GstRTSPAuthParam {
+  gchar *name;
+  gchar *value;
+};
+
+GST_RTSP_API
+GstRTSPAuthParam *       gst_rtsp_auth_param_copy (GstRTSPAuthParam * param);
+GST_RTSP_API
+void                     gst_rtsp_auth_param_free (GstRTSPAuthParam * param);
+
+GST_RTSP_API
+GstRTSPAuthCredential ** gst_rtsp_message_parse_auth_credentials (GstRTSPMessage * msg, GstRTSPHeaderField field);
+
+GST_RTSP_API
+void                     gst_rtsp_auth_credentials_free (GstRTSPAuthCredential ** credentials);
+
+#define GST_TYPE_RTSP_AUTH_CREDENTIAL gst_rtsp_auth_credential_get_type()
+
+GST_RTSP_API
+GType                    gst_rtsp_auth_credential_get_type (void);
+
+#define GST_TYPE_RTSP_AUTH_PARAM gst_rtsp_auth_param_get_type()
+
+GST_RTSP_API
+GType                    gst_rtsp_auth_param_get_type (void);
+
+/* debug */
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_message_dump            (GstRTSPMessage *msg);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MESSAGE_H__ */
diff --git a/include/gst/rtsp/gstrtsprange.h b/include/gst/rtsp/gstrtsprange.h
new file mode 100644
index 0000000000..c14bf0f103
--- /dev/null
+++ b/include/gst/rtsp/gstrtsprange.h
@@ -0,0 +1,164 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_RANGE_H__
+#define __GST_RTSP_RANGE_H__
+
+#include <glib.h>
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtspdefs.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstRTSPRangeUnit:
+ * @GST_RTSP_RANGE_SMPTE: SMPTE timecode
+ * @GST_RTSP_RANGE_SMPTE_30_DROP: 29.97 frames per second
+ * @GST_RTSP_RANGE_SMPTE_25: 25 frames per second
+ * @GST_RTSP_RANGE_NPT: Normal play time
+ * @GST_RTSP_RANGE_CLOCK: Absolute time expressed as ISO 8601 timestamps
+ *
+ * Different possible time range units.
+ */
+typedef enum
+{
+  GST_RTSP_RANGE_SMPTE,
+  GST_RTSP_RANGE_SMPTE_30_DROP,
+  GST_RTSP_RANGE_SMPTE_25,
+  GST_RTSP_RANGE_NPT,
+  GST_RTSP_RANGE_CLOCK
+} GstRTSPRangeUnit;
+
+typedef struct _GstRTSPTimeRange GstRTSPTimeRange;
+typedef struct _GstRTSPTime GstRTSPTime;
+typedef struct _GstRTSPTime2 GstRTSPTime2;
+
+/**
+ * GstRTSPTimeType:
+ * @GST_RTSP_TIME_SECONDS: seconds
+ * @GST_RTSP_TIME_NOW: now
+ * @GST_RTSP_TIME_END: end
+ * @GST_RTSP_TIME_FRAMES: frames and subframes
+ * @GST_RTSP_TIME_UTC: UTC time
+ *
+ * Possible time types.
+ */
+typedef enum {
+  GST_RTSP_TIME_SECONDS,
+  GST_RTSP_TIME_NOW,
+  GST_RTSP_TIME_END,
+  GST_RTSP_TIME_FRAMES,
+  GST_RTSP_TIME_UTC
+} GstRTSPTimeType;
+
+/**
+ * GstRTSPTime:
+ * @type: the time of the time
+ * @seconds: seconds when @type is GST_RTSP_TIME_SECONDS,
+ *           GST_RTSP_TIME_UTC and GST_RTSP_TIME_FRAMES
+ *
+ * A time indication.
+ */
+struct _GstRTSPTime {
+  GstRTSPTimeType type;
+  gdouble         seconds;
+};
+
+/**
+ * GstRTSPTime2:
+ * @frames: frames and subframes when type in GstRTSPTime is
+ *          GST_RTSP_TIME_FRAMES
+ * @year: year when type is GST_RTSP_TIME_UTC
+ * @month: month when type is GST_RTSP_TIME_UTC
+ * @day: day when type is GST_RTSP_TIME_UTC
+ *
+ * Extra fields for a time indication.
+ *
+ * Since: 1.2
+ */
+struct _GstRTSPTime2 {
+  gdouble         frames;
+  guint           year;
+  guint           month;
+  guint           day;
+};
+
+/**
+ * GstRTSPTimeRange:
+ * @unit: the time units used
+ * @min: the minimum interval
+ * @max: the maximum interval
+ * @min2: extra fields in the minimum interval (Since: 1.2)
+ * @max2: extra fields in the maximum interval (Since: 1.2)
+ *
+ * A time range.
+ */
+struct _GstRTSPTimeRange {
+  GstRTSPRangeUnit unit;
+
+  GstRTSPTime  min;
+  GstRTSPTime  max;
+  GstRTSPTime2 min2;
+  GstRTSPTime2 max2;
+};
+
+GST_RTSP_API
+GstRTSPResult   gst_rtsp_range_parse        (const gchar *rangestr, GstRTSPTimeRange **range);
+
+GST_RTSP_API
+gchar *         gst_rtsp_range_to_string    (const GstRTSPTimeRange *range);
+
+GST_RTSP_API
+void            gst_rtsp_range_free         (GstRTSPTimeRange *range);
+
+GST_RTSP_API
+gboolean        gst_rtsp_range_get_times     (const GstRTSPTimeRange *range,
+                                              GstClockTime *min, GstClockTime *max);
+
+GST_RTSP_API
+gboolean        gst_rtsp_range_convert_units (GstRTSPTimeRange * range,
+                                              GstRTSPRangeUnit unit);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_RANGE_H__ */
diff --git a/include/gst/rtsp/gstrtsptransport.h b/include/gst/rtsp/gstrtsptransport.h
new file mode 100644
index 0000000000..909edecd7a
--- /dev/null
+++ b/include/gst/rtsp/gstrtsptransport.h
@@ -0,0 +1,201 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_TRANSPORT_H__
+#define __GST_RTSP_TRANSPORT_H__
+
+#include <gst/gstconfig.h>
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtsp-enumtypes.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstRTSPTransMode:
+ * @GST_RTSP_TRANS_UNKNOWN: invalid tansport mode
+ * @GST_RTSP_TRANS_RTP: transfer RTP data
+ * @GST_RTSP_TRANS_RDT: transfer RDT (RealMedia) data
+ *
+ * The transfer mode to use.
+ */
+typedef enum {
+  GST_RTSP_TRANS_UNKNOWN =  0,
+  GST_RTSP_TRANS_RTP     = (1 << 0),
+  GST_RTSP_TRANS_RDT     = (1 << 1)
+} GstRTSPTransMode;
+
+/**
+ * GstRTSPProfile:
+ * @GST_RTSP_PROFILE_UNKNOWN: invalid profile
+ * @GST_RTSP_PROFILE_AVP: the Audio/Visual profile (RFC 3551)
+ * @GST_RTSP_PROFILE_SAVP: the secure Audio/Visual profile (RFC 3711)
+ * @GST_RTSP_PROFILE_AVPF: the Audio/Visual profile with feedback (RFC 4585)
+ * @GST_RTSP_PROFILE_SAVPF: the secure Audio/Visual profile with feedback (RFC 5124)
+ *
+ * The transfer profile to use.
+ */
+/* FIXME 2.0: This should probably be an enum, not flags and maybe be replaced
+ * by GstRTPTransport */
+typedef enum {
+  GST_RTSP_PROFILE_UNKNOWN =  0,
+  GST_RTSP_PROFILE_AVP     = (1 << 0),
+  GST_RTSP_PROFILE_SAVP    = (1 << 1),
+  GST_RTSP_PROFILE_AVPF    = (1 << 2),
+  GST_RTSP_PROFILE_SAVPF   = (1 << 3),
+} GstRTSPProfile;
+
+/**
+ * GstRTSPLowerTrans:
+ * @GST_RTSP_LOWER_TRANS_UNKNOWN: invalid transport flag
+ * @GST_RTSP_LOWER_TRANS_UDP: stream data over UDP
+ * @GST_RTSP_LOWER_TRANS_UDP_MCAST: stream data over UDP multicast
+ * @GST_RTSP_LOWER_TRANS_TCP: stream data over TCP
+ * @GST_RTSP_LOWER_TRANS_HTTP: stream data tunneled over HTTP.
+ * @GST_RTSP_LOWER_TRANS_TLS: encrypt TCP and HTTP with TLS
+ *
+ * The different transport methods.
+ */
+typedef enum {
+  GST_RTSP_LOWER_TRANS_UNKNOWN   = 0,
+  GST_RTSP_LOWER_TRANS_UDP       = (1 << 0),
+  GST_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
+  GST_RTSP_LOWER_TRANS_TCP       = (1 << 2),
+  GST_RTSP_LOWER_TRANS_HTTP      = (1 << 4),
+  GST_RTSP_LOWER_TRANS_TLS       = (1 << 5)
+} GstRTSPLowerTrans;
+
+typedef struct _GstRTSPRange GstRTSPRange;
+typedef struct _GstRTSPTransport GstRTSPTransport;
+
+/**
+ * GstRTSPRange:
+ * @min: minimum value of the range
+ * @max: maximum value of the range
+ *
+ * A type to specify a range.
+ */
+
+struct _GstRTSPRange {
+  gint min;
+  gint max;
+};
+
+/**
+ * GstRTSPTransport:
+ * @trans: the transport mode
+ * @profile: the tansport profile
+ * @lower_transport: the lower transport
+ * @destination: the destination ip/hostname
+ * @source: the source ip/hostname
+ * @layers: the number of layers
+ * @mode_play: if play mode was selected
+ * @mode_record: if record mode was selected
+ * @append: is append mode was selected
+ * @interleaved: the interleave range
+ * @ttl: the time to live for multicast UDP
+ * @port: the port pair for multicast sessions
+ * @client_port: the client port pair for receiving data. For TCP
+ *   based transports, applications can use this field to store the
+ *   sender and receiver ports of the client.
+ * @server_port: the server port pair for receiving data. For TCP
+ *   based transports, applications can use this field to store the
+ *   sender and receiver ports of the server.
+ * @ssrc: the ssrc that the sender/receiver will use
+ *
+ * A structure holding the RTSP transport values.
+ */
+
+struct _GstRTSPTransport {
+  GstRTSPTransMode  trans;
+  GstRTSPProfile    profile;
+  GstRTSPLowerTrans lower_transport;
+
+  gchar         *destination;
+  gchar         *source;
+  guint          layers;
+  gboolean       mode_play;
+  gboolean       mode_record;
+  gboolean       append;
+  GstRTSPRange   interleaved;
+
+  /* multicast specific */
+  guint  ttl;
+  GstRTSPRange   port;
+
+  /* UDP/TCP specific */
+  GstRTSPRange   client_port;
+  GstRTSPRange   server_port;
+  /* RTP specific */
+  guint          ssrc;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_transport_new          (GstRTSPTransport **transport);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_transport_init         (GstRTSPTransport *transport);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_transport_parse        (const gchar *str, GstRTSPTransport *transport);
+
+GST_RTSP_API
+gchar*             gst_rtsp_transport_as_text      (GstRTSPTransport *transport);
+
+GST_RTSP_DEPRECATED_FOR(gst_rtsp_transport_get_media_type)
+GstRTSPResult      gst_rtsp_transport_get_mime     (GstRTSPTransMode trans, const gchar **mime);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_transport_get_manager  (GstRTSPTransMode trans, const gchar **manager, guint option);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_transport_get_media_type (GstRTSPTransport *transport,
+                                                      const gchar **media_type);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_transport_free         (GstRTSPTransport *transport);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_TRANSPORT_H__ */
diff --git a/include/gst/rtsp/gstrtspurl.h b/include/gst/rtsp/gstrtspurl.h
new file mode 100644
index 0000000000..8e95a1567f
--- /dev/null
+++ b/include/gst/rtsp/gstrtspurl.h
@@ -0,0 +1,121 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_URL_H__
+#define __GST_RTSP_URL_H__
+
+#include <glib.h>
+#include <glib-object.h>
+
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtsptransport.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GST_RTSP_DEFAULT_PORT:
+ *
+ * The default RTSP port to connect to.
+ */
+#define GST_RTSP_DEFAULT_PORT       554
+
+#define GST_TYPE_RTSP_URL  (gst_rtsp_url_get_type())
+
+typedef struct _GstRTSPUrl GstRTSPUrl;
+
+/**
+ * GstRTSPUrl:
+ * @transports: the transports allowed
+ * @family: the family
+ * @user: the user
+ * @passwd: the password
+ * @host: the host
+ * @port: the port
+ * @abspath: the absolute path
+ * @query: additional query parameters
+ *
+ * This structure contains the result of a parsed RTSP URL
+ */
+struct _GstRTSPUrl {
+  GstRTSPLowerTrans  transports;
+  GstRTSPFamily      family;
+  gchar             *user;
+  gchar             *passwd;
+  gchar             *host;
+  guint16            port;
+  gchar             *abspath;
+  gchar             *query;
+};
+
+GST_RTSP_API
+GType gst_rtsp_url_get_type (void);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_url_parse           (const gchar *urlstr, GstRTSPUrl **url);
+
+GST_RTSP_API
+GstRTSPUrl*        gst_rtsp_url_copy            (const GstRTSPUrl *url);
+
+GST_RTSP_API
+void               gst_rtsp_url_free            (GstRTSPUrl *url);
+
+GST_RTSP_API
+gchar*             gst_rtsp_url_get_request_uri (const GstRTSPUrl *url);
+
+GST_RTSP_API
+gchar *            gst_rtsp_url_get_request_uri_with_control (const GstRTSPUrl * url,
+                                                              const gchar * control_path);
+
+GST_RTSP_API
+gchar**            gst_rtsp_url_decode_path_components
+                                                (const GstRTSPUrl *url);
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_url_set_port        (GstRTSPUrl *url, guint16 port);
+
+GST_RTSP_API
+GstRTSPResult      gst_rtsp_url_get_port        (const GstRTSPUrl *url, guint16 *port);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPUrl, gst_rtsp_url_free)
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_URL_H__ */
diff --git a/include/gst/rtsp/rtsp-prelude.h b/include/gst/rtsp/rtsp-prelude.h
new file mode 100644
index 0000000000..1086091179
--- /dev/null
+++ b/include/gst/rtsp/rtsp-prelude.h
@@ -0,0 +1,41 @@
+/* GStreamer Rtsp Library
+ * Copyright (C) 2018 GStreamer developers
+ *
+ * rtsp-prelude.h: prelude include header for gst-rtsp library
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTSP_PRELUDE_H__
+#define __GST_RTSP_PRELUDE_H__
+
+#include <gst/gst.h>
+
+#ifdef BUILDING_GST_RTSP
+#define GST_RTSP_API GST_API_EXPORT         /* from config.h */
+#else
+#define GST_RTSP_API GST_API_IMPORT
+#endif
+
+#ifndef GST_DISABLE_DEPRECATED
+#define GST_RTSP_DEPRECATED GST_RTSP_API
+#define GST_RTSP_DEPRECATED_FOR(f) GST_RTSP_API
+#else
+#define GST_RTSP_DEPRECATED G_DEPRECATED GST_RTSP_API
+#define GST_RTSP_DEPRECATED_FOR(f) G_DEPRECATED_FOR(f) GST_RTSP_API
+#endif
+
+#endif /* __GST_RTSP_PRELUDE_H__ */
diff --git a/include/gst/rtsp/rtsp.h b/include/gst/rtsp/rtsp.h
new file mode 100644
index 0000000000..a44b650b09
--- /dev/null
+++ b/include/gst/rtsp/rtsp.h
@@ -0,0 +1,37 @@
+/* GStreamer
+ * Copyright (C) 2012 GStreamer developers
+ *
+ * gstrtsp.h: single include header for gst-rtsp library
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTSP__H__
+#define __GST_RTSP__H__
+
+#include <gst/rtsp/rtsp-prelude.h>
+
+#include <gst/rtsp/gstrtsp.h>
+#include <gst/rtsp/gstrtspconnection.h>
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtsp-enumtypes.h>
+#include <gst/rtsp/gstrtspextension.h>
+#include <gst/rtsp/gstrtspmessage.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtsptransport.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#endif /* __GST_RTSP__H__ */