gst/rtsp/gstrtspsrc.c

/* GStreamer
 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
 *               <2006> Lutz Mueller <lutz at topfrose dot de>
 *
 * 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.
 */
/**
 * SECTION:element-rtspsrc
 *
 * Makes a connection to an RTSP server and read the data.
 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
 * RealMedia/Quicktime/Microsoft extensions.
 *
 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
 * default rtspsrc will negotiate a connection in the following order:
 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
 * protocols can be controlled with the #GstRTSPSrc:protocols property.
 *
 * rtspsrc currently understands SDP as the format of the session description.
 * For each stream listed in the SDP a new rtp_stream\%d pad will be created
 * with caps derived from the SDP media description. This is a caps of mime type
 * "application/x-rtp" that can be connected to any available RTP depayloader
 * element.
 *
 * rtspsrc will internally instantiate an RTP session manager element
 * that will handle the RTCP messages to and from the server, jitter removal,
 * packet reordering along with providing a clock for the pipeline.
 * This feature is implemented using the gstrtpbin element.
 *
 * rtspsrc acts like a live source and will therefore only generate data in the
 * PLAYING state.
 *
 * <refsect2>
 * <title>Example launch line</title>
 * |[
 * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
 * fakesink.
 * </refsect2>
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#ifdef HAVE_UNISTD_H
#include <unistd.h>
#endif /* HAVE_UNISTD_H */
#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <stdarg.h>

#include <gst/net/gstnet.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/sdp/gstmikey.h>
#include <gst/rtp/rtp.h>

#include "gst/gst-i18n-plugin.h"

#include "gstrtspsrc.h"

GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
#define GST_CAT_DEFAULT (rtspsrc_debug)

static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
    GST_PAD_SRC,
    GST_PAD_SOMETIMES,
    GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));

/* templates used internally */
static GstStaticPadTemplate anysrctemplate =
GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
    GST_PAD_SRC,
    GST_PAD_SOMETIMES,
    GST_STATIC_CAPS_ANY);

static GstStaticPadTemplate anysinktemplate =
GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
    GST_PAD_SINK,
    GST_PAD_SOMETIMES,
    GST_STATIC_CAPS_ANY);

enum
{
  SIGNAL_HANDLE_REQUEST,
  SIGNAL_ON_SDP,
  SIGNAL_SELECT_STREAM,
  SIGNAL_NEW_MANAGER,
  SIGNAL_REQUEST_RTCP_KEY,
  SIGNAL_ACCEPT_CERTIFICATE,
  SIGNAL_BEFORE_SEND,
  SIGNAL_PUSH_BACKCHANNEL_BUFFER,
  LAST_SIGNAL
};

enum _GstRtspSrcRtcpSyncMode
{
  RTCP_SYNC_ALWAYS,
  RTCP_SYNC_INITIAL,
  RTCP_SYNC_RTP
};

enum _GstRtspSrcBufferMode
{
  BUFFER_MODE_NONE,
  BUFFER_MODE_SLAVE,
  BUFFER_MODE_BUFFER,
  BUFFER_MODE_AUTO,
  BUFFER_MODE_SYNCED
};

#define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
static GType
gst_rtsp_src_buffer_mode_get_type (void)
{
  static GType buffer_mode_type = 0;
  static const GEnumValue buffer_modes[] = {
    {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
    {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
    {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
    {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
    {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
    {0, NULL, NULL},
  };

  if (!buffer_mode_type) {
    buffer_mode_type =
        g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
  }
  return buffer_mode_type;
}

enum _GstRtspSrcNtpTimeSource
{
  NTP_TIME_SOURCE_NTP,
  NTP_TIME_SOURCE_UNIX,
  NTP_TIME_SOURCE_RUNNING_TIME,
  NTP_TIME_SOURCE_CLOCK_TIME
};

#define DEBUG_RTSP(__self,msg) gst_rtspsrc_print_rtsp_message (__self, msg)
#define DEBUG_SDP(__self,msg) gst_rtspsrc_print_sdp_message (__self, msg)

#define GST_TYPE_RTSP_SRC_NTP_TIME_SOURCE (gst_rtsp_src_ntp_time_source_get_type())
static GType
gst_rtsp_src_ntp_time_source_get_type (void)
{
  static GType ntp_time_source_type = 0;
  static const GEnumValue ntp_time_source_values[] = {
    {NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
    {NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
    {NTP_TIME_SOURCE_RUNNING_TIME,
          "Running time based on pipeline clock",
        "running-time"},
    {NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
    {0, NULL, NULL},
  };

  if (!ntp_time_source_type) {
    ntp_time_source_type =
        g_enum_register_static ("GstRTSPSrcNtpTimeSource",
        ntp_time_source_values);
  }
  return ntp_time_source_type;
}

enum _GstRtspBackchannel
{
  BACKCHANNEL_NONE,
  BACKCHANNEL_ONVIF
};

#define GST_TYPE_RTSP_BACKCHANNEL (gst_rtsp_backchannel_get_type())
static GType
gst_rtsp_backchannel_get_type (void)
{
  static GType backchannel_type = 0;
  static const GEnumValue backchannel_values[] = {
    {BACKCHANNEL_NONE, "No backchannel", "none"},
    {BACKCHANNEL_ONVIF, "ONVIF audio backchannel", "onvif"},
    {0, NULL, NULL},
  };

  if (G_UNLIKELY (backchannel_type == 0)) {
    backchannel_type =
        g_enum_register_static ("GstRTSPBackchannel", backchannel_values);
  }
  return backchannel_type;
}

#define BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL "www.onvif.org/ver20/backchannel"

#define DEFAULT_LOCATION         NULL
#define DEFAULT_PROTOCOLS        GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG            FALSE
#define DEFAULT_RETRY            20
#define DEFAULT_TIMEOUT          5000000
#define DEFAULT_UDP_BUFFER_SIZE  0x80000
#define DEFAULT_TCP_TIMEOUT      20000000
#define DEFAULT_LATENCY_MS       2000
#define DEFAULT_DROP_ON_LATENCY  FALSE
#define DEFAULT_CONNECTION_SPEED 0
#define DEFAULT_NAT_METHOD       GST_RTSP_NAT_DUMMY
#define DEFAULT_DO_RTCP          TRUE
#define DEFAULT_DO_RTSP_KEEP_ALIVE       TRUE
#define DEFAULT_PROXY            NULL
#define DEFAULT_RTP_BLOCKSIZE    0
#define DEFAULT_USER_ID          NULL
#define DEFAULT_USER_PW          NULL
#define DEFAULT_BUFFER_MODE      BUFFER_MODE_AUTO
#define DEFAULT_PORT_RANGE       NULL
#define DEFAULT_SHORT_HEADER     FALSE
#define DEFAULT_PROBATION        2
#define DEFAULT_UDP_RECONNECT    TRUE
#define DEFAULT_MULTICAST_IFACE  NULL
#define DEFAULT_NTP_SYNC         FALSE
#define DEFAULT_USE_PIPELINE_CLOCK       FALSE
#define DEFAULT_TLS_VALIDATION_FLAGS     G_TLS_CERTIFICATE_VALIDATE_ALL
#define DEFAULT_TLS_DATABASE     NULL
#define DEFAULT_TLS_INTERACTION     NULL
#define DEFAULT_DO_RETRANSMISSION        TRUE
#define DEFAULT_NTP_TIME_SOURCE  NTP_TIME_SOURCE_NTP
#define DEFAULT_USER_AGENT       "GStreamer/" PACKAGE_VERSION
#define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
#define DEFAULT_RFC7273_SYNC         FALSE
#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT   G_GUINT64_CONSTANT(0)
#define DEFAULT_MAX_TS_OFFSET   G_GINT64_CONSTANT(3000000000)
#define DEFAULT_VERSION         GST_RTSP_VERSION_1_0
#define DEFAULT_BACKCHANNEL  GST_RTSP_BACKCHANNEL_NONE

enum
{
  PROP_0,
  PROP_LOCATION,
  PROP_PROTOCOLS,
  PROP_DEBUG,
  PROP_RETRY,
  PROP_TIMEOUT,
  PROP_TCP_TIMEOUT,
  PROP_LATENCY,
  PROP_DROP_ON_LATENCY,
  PROP_CONNECTION_SPEED,
  PROP_NAT_METHOD,
  PROP_DO_RTCP,
  PROP_DO_RTSP_KEEP_ALIVE,
  PROP_PROXY,
  PROP_PROXY_ID,
  PROP_PROXY_PW,
  PROP_RTP_BLOCKSIZE,
  PROP_USER_ID,
  PROP_USER_PW,
  PROP_BUFFER_MODE,
  PROP_PORT_RANGE,
  PROP_UDP_BUFFER_SIZE,
  PROP_SHORT_HEADER,
  PROP_PROBATION,
  PROP_UDP_RECONNECT,
  PROP_MULTICAST_IFACE,
  PROP_NTP_SYNC,
  PROP_USE_PIPELINE_CLOCK,
  PROP_SDES,
  PROP_TLS_VALIDATION_FLAGS,
  PROP_TLS_DATABASE,
  PROP_TLS_INTERACTION,
  PROP_DO_RETRANSMISSION,
  PROP_NTP_TIME_SOURCE,
  PROP_USER_AGENT,
  PROP_MAX_RTCP_RTP_TIME_DIFF,
  PROP_RFC7273_SYNC,
  PROP_MAX_TS_OFFSET_ADJUSTMENT,
  PROP_MAX_TS_OFFSET,
  PROP_DEFAULT_VERSION,
  PROP_BACKCHANNEL,
};

#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
static GType
gst_rtsp_nat_method_get_type (void)
{
  static GType rtsp_nat_method_type = 0;
  static const GEnumValue rtsp_nat_method[] = {
    {GST_RTSP_NAT_NONE, "None", "none"},
    {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
    {0, NULL, NULL},
  };

  if (!rtsp_nat_method_type) {
    rtsp_nat_method_type =
        g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
  }
  return rtsp_nat_method_type;
}

#define RTSP_SRC_RESPONSE_ERROR(src, response_msg, err_cat, err_code, error_message) \
  do { \
    GST_ELEMENT_ERROR_WITH_DETAILS((src), err_cat, err_code, ("%s", error_message), \
        ("%s (%d)", (response_msg)->type_data.response.reason, (response_msg)->type_data.response.code), \
        ("rtsp-status-code", G_TYPE_UINT, (response_msg)->type_data.response.code, \
         "rtsp-status-reason", G_TYPE_STRING, GST_STR_NULL((response_msg)->type_data.response.reason), NULL)); \
  } while (0)

static void gst_rtspsrc_finalize (GObject * object);

static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);

static GstClock *gst_rtspsrc_provide_clock (GstElement * element);

static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
    gpointer iface_data);

static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);

static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
    GstStateChange transition);
static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);

static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
    GstRTSPMessage * response);

static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
    gint mask);
static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
    GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);

static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
    gboolean async, const gchar * seek_style);
static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
    gboolean only_close);

static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
    const gchar * uri, GError ** error);
static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);

static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
    GstRTSPStream * stream, GstEvent * event);
static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
static GstRTSPResult gst_rtsp_conninfo_close (GstRTSPSrc * src,
    GstRTSPConnInfo * info, gboolean free);
static void
gst_rtspsrc_print_rtsp_message (GstRTSPSrc * src, const GstRTSPMessage * msg);
static void
gst_rtspsrc_print_sdp_message (GstRTSPSrc * src, const GstSDPMessage * msg);

static GstFlowReturn gst_rtspsrc_push_backchannel_buffer (GstRTSPSrc * src,
    guint id, GstSample * sample);

typedef struct
{
  guint8 pt;
  GstCaps *caps;
} PtMapItem;

/* commands we send to out loop to notify it of events */
#define CMD_OPEN       (1 << 0)
#define CMD_PLAY       (1 << 1)
#define CMD_PAUSE      (1 << 2)
#define CMD_CLOSE      (1 << 3)
#define CMD_WAIT       (1 << 4)
#define CMD_RECONNECT  (1 << 5)
#define CMD_LOOP       (1 << 6)

/* mask for all commands */
#define CMD_ALL         ((CMD_LOOP << 1) - 1)

#define GST_ELEMENT_PROGRESS(el, type, code, text)      \
G_STMT_START {                                          \
  gchar *__txt = _gst_element_error_printf text;        \
  gst_element_post_message (GST_ELEMENT_CAST (el),      \
      gst_message_new_progress (GST_OBJECT_CAST (el),   \
          GST_PROGRESS_TYPE_ ##type, code, __txt));     \
  g_free (__txt);                                       \
} G_STMT_END

static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };

#define gst_rtspsrc_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
    G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));

#ifndef GST_DISABLE_GST_DEBUG
static inline const char *
cmd_to_string (guint cmd)
{
  switch (cmd) {
    case CMD_OPEN:
      return "OPEN";
    case CMD_PLAY:
      return "PLAY";
    case CMD_PAUSE:
      return "PAUSE";
    case CMD_CLOSE:
      return "CLOSE";
    case CMD_WAIT:
      return "WAIT";
    case CMD_RECONNECT:
      return "RECONNECT";
    case CMD_LOOP:
      return "LOOP";
  }

  return "unknown";
}
#endif

static gboolean
default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
{
  GST_DEBUG_OBJECT (src, "default handler");
  return TRUE;
}

static gboolean
select_stream_accum (GSignalInvocationHint * ihint,
    GValue * return_accu, const GValue * handler_return, gpointer data)
{
  gboolean myboolean;

  myboolean = g_value_get_boolean (handler_return);
  GST_DEBUG ("accum %d", myboolean);
  g_value_set_boolean (return_accu, myboolean);

  /* stop emission if FALSE */
  return myboolean;
}

static gboolean
default_before_send (GstRTSPSrc * src, GstRTSPMessage * msg)
{
  GST_DEBUG_OBJECT (src, "default handler");
  return TRUE;
}

static gboolean
before_send_accum (GSignalInvocationHint * ihint,
    GValue * return_accu, const GValue * handler_return, gpointer data)
{
  gboolean myboolean;

  myboolean = g_value_get_boolean (handler_return);
  g_value_set_boolean (return_accu, myboolean);

  /* prevent send if FALSE */
  return myboolean;
}

static void
gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstBinClass *gstbin_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;
  gstbin_class = (GstBinClass *) klass;

  GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");

  gobject_class->set_property = gst_rtspsrc_set_property;
  gobject_class->get_property = gst_rtspsrc_get_property;

  gobject_class->finalize = gst_rtspsrc_finalize;

  g_object_class_install_property (gobject_class, PROP_LOCATION,
      g_param_spec_string ("location", "RTSP Location",
          "Location of the RTSP url to read",
          DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
      g_param_spec_flags ("protocols", "Protocols",
          "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
          DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_DEBUG,
      g_param_spec_boolean ("debug", "Debug",
          "Dump request and response messages to stdout"
          "(DEPRECATED: Printed all RTSP message to gstreamer log as 'log' level)",
          DEFAULT_DEBUG,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));

  g_object_class_install_property (gobject_class, PROP_RETRY,
      g_param_spec_uint ("retry", "Retry",
          "Max number of retries when allocating RTP ports.",
          0, G_MAXUINT16, DEFAULT_RETRY,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_TIMEOUT,
      g_param_spec_uint64 ("timeout", "Timeout",
          "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
          0, G_MAXUINT64, DEFAULT_TIMEOUT,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
      g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
          "Fail after timeout microseconds on TCP connections (0 = disabled)",
          0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_LATENCY,
      g_param_spec_uint ("latency", "Buffer latency in ms",
          "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
      g_param_spec_boolean ("drop-on-latency",
          "Drop buffers when maximum latency is reached",
          "Tells the jitterbuffer to never exceed the given latency in size",
          DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
      g_param_spec_uint64 ("connection-speed", "Connection Speed",
          "Network connection speed in kbps (0 = unknown)",
          0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
      g_param_spec_enum ("nat-method", "NAT Method",
          "Method to use for traversing firewalls and NAT",
          GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:do-rtcp:
   *
   * Enable RTCP support. Some old server don't like RTCP and then this property
   * needs to be set to FALSE.
   */
  g_object_class_install_property (gobject_class, PROP_DO_RTCP,
      g_param_spec_boolean ("do-rtcp", "Do RTCP",
          "Send RTCP packets, disable for old incompatible server.",
          DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:do-rtsp-keep-alive:
   *
   * Enable RTSP keep alive support. Some old server don't like RTSP
   * keep alive and then this property needs to be set to FALSE.
   */
  g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
      g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
          "Send RTSP keep alive packets, disable for old incompatible server.",
          DEFAULT_DO_RTSP_KEEP_ALIVE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:proxy:
   *
   * Set the proxy parameters. This has to be a string of the format
   * [http://][user:passwd@]host[:port].
   */
  g_object_class_install_property (gobject_class, PROP_PROXY,
      g_param_spec_string ("proxy", "Proxy",
          "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
          DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
  /**
   * GstRTSPSrc:proxy-id:
   *
   * Sets the proxy URI user id for authentication. If the URI set via the
   * "proxy" property contains a user-id already, that will take precedence.
   *
   * Since: 1.2
   */
  g_object_class_install_property (gobject_class, PROP_PROXY_ID,
      g_param_spec_string ("proxy-id", "proxy-id",
          "HTTP proxy URI user id for authentication", "",
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
  /**
   * GstRTSPSrc:proxy-pw:
   *
   * Sets the proxy URI password for authentication. If the URI set via the
   * "proxy" property contains a password already, that will take precedence.
   *
   * Since: 1.2
   */
  g_object_class_install_property (gobject_class, PROP_PROXY_PW,
      g_param_spec_string ("proxy-pw", "proxy-pw",
          "HTTP proxy URI user password for authentication", "",
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:rtp-blocksize:
   *
   * RTP package size to suggest to server.
   */
  g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
      g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
          "RTP package size to suggest to server (0 = disabled)",
          0, 65536, DEFAULT_RTP_BLOCKSIZE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class,
      PROP_USER_ID,
      g_param_spec_string ("user-id", "user-id",
          "RTSP location URI user id for authentication", DEFAULT_USER_ID,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
  g_object_class_install_property (gobject_class, PROP_USER_PW,
      g_param_spec_string ("user-pw", "user-pw",
          "RTSP location URI user password for authentication", DEFAULT_USER_PW,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:buffer-mode:
   *
   * Control the buffering and timestamping mode used by the jitterbuffer.
   */
  g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
      g_param_spec_enum ("buffer-mode", "Buffer Mode",
          "Control the buffering algorithm in use",
          GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:port-range:
   *
   * Configure the client port numbers that can be used to recieve RTP and
   * RTCP.
   */
  g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
      g_param_spec_string ("port-range", "Port range",
          "Client port range that can be used to receive RTP and RTCP data, "
          "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:udp-buffer-size:
   *
   * Size of the kernel UDP receive buffer in bytes.
   */
  g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
      g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
          "Size of the kernel UDP receive buffer in bytes, 0=default",
          0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:short-header:
   *
   * Only send the basic RTSP headers for broken encoders.
   */
  g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
      g_param_spec_boolean ("short-header", "Short Header",
          "Only send the basic RTSP headers for broken encoders",
          DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_PROBATION,
      g_param_spec_uint ("probation", "Number of probations",
          "Consecutive packet sequence numbers to accept the source",
          0, G_MAXUINT, DEFAULT_PROBATION,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
      g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
          "Reconnect to the server if RTSP connection is closed when doing UDP",
          DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
      g_param_spec_string ("multicast-iface", "Multicast Interface",
          "The network interface on which to join the multicast group",
          DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
      g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
          "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
      g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
          "Use the pipeline running-time to set the NTP time in the RTCP SR messages"
          "(DEPRECATED: Use ntp-time-source property)",
          DEFAULT_USE_PIPELINE_CLOCK,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));

  g_object_class_install_property (gobject_class, PROP_SDES,
      g_param_spec_boxed ("sdes", "SDES",
          "The SDES items of this session",
          GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::tls-validation-flags:
   *
   * TLS certificate validation flags used to validate server
   * certificate.
   *
   * Since: 1.2.1
   */
  g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
      g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
          "TLS certificate validation flags used to validate the server certificate",
          G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::tls-database:
   *
   * TLS database with anchor certificate authorities used to validate
   * the server certificate.
   *
   * Since: 1.4
   */
  g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
      g_param_spec_object ("tls-database", "TLS database",
          "TLS database with anchor certificate authorities used to validate the server certificate",
          G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::tls-interaction:
   *
   * A #GTlsInteraction object to be used when the connection or certificate
   * database need to interact with the user. This will be used to prompt the
   * user for passwords where necessary.
   *
   * Since: 1.6
   */
  g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
      g_param_spec_object ("tls-interaction", "TLS interaction",
          "A GTlsInteraction object to promt the user for password or certificate",
          G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::do-retransmission:
   *
   * Attempt to ask the server to retransmit lost packets according to RFC4588.
   *
   * Note: currently only works with SSRC-multiplexed retransmission streams
   *
   * Since: 1.6
   */
  g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
      g_param_spec_boolean ("do-retransmission", "Retransmission",
          "Ask the server to retransmit lost packets",
          DEFAULT_DO_RETRANSMISSION,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::ntp-time-source:
   *
   * allows to select the time source that should be used
   * for the NTP time in RTCP packets
   *
   * Since: 1.6
   */
  g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
      g_param_spec_enum ("ntp-time-source", "NTP Time Source",
          "NTP time source for RTCP packets",
          GST_TYPE_RTSP_SRC_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::user-agent:
   *
   * The string to set in the User-Agent header.
   *
   * Since: 1.6
   */
  g_object_class_install_property (gobject_class, PROP_USER_AGENT,
      g_param_spec_string ("user-agent", "User Agent",
          "The User-Agent string to send to the server",
          DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
      g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
          "Maximum amount of time in ms that the RTP time in RTCP SRs "
          "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
          DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
      g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
          "Synchronize received streams to the RFC7273 clock "
          "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:default-rtsp-version:
   *
   * The preferred RTSP version to use while negotiating the version with the server.
   *
   * Since: 1.14
   */
  g_object_class_install_property (gobject_class, PROP_DEFAULT_VERSION,
      g_param_spec_enum ("default-rtsp-version",
          "The RTSP version to try first",
          "The RTSP version that should be tried first when negotiating version.",
          GST_TYPE_RTSP_VERSION, DEFAULT_VERSION,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc:max-ts-offset-adjustment:
   *
   * Syncing time stamps to NTP time adds a time offset. This parameter
   * specifies the maximum number of nanoseconds per frame that this time offset
   * may be adjusted with. This is used to avoid sudden large changes to time
   * stamps.
   */
  g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
      g_param_spec_uint64 ("max-ts-offset-adjustment",
          "Max Timestamp Offset Adjustment",
          "The maximum number of nanoseconds per frame that time stamp offsets "
          "may be adjusted (0 = no limit).", 0, G_MAXUINT64,
          DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
          G_PARAM_STATIC_STRINGS));

  /**
   * GstRtpBin:max-ts-offset:
   *
   * Used to set an upper limit of how large a time offset may be. This
   * is used to protect against unrealistic values as a result of either
   * client,server or clock issues.
   */
  g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
      g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
          "The maximum absolute value of the time offset in (nanoseconds). "
          "Note, if the ntp-sync parameter is set the default value is "
          "changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRtpSrc:backchannel
   *
   * Select a type of backchannel to setup with the RTSP server.
   * Default value is "none". Allowed values are "none" and "onvif".
   *
   * Since: 1.14
   */
  g_object_class_install_property (gobject_class, PROP_BACKCHANNEL,
      g_param_spec_enum ("backchannel", "Backchannel type",
          "The type of backchannel to setup. Default is 'none'.",
          GST_TYPE_RTSP_BACKCHANNEL, BACKCHANNEL_NONE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTSPSrc::handle-request:
   * @rtspsrc: a #GstRTSPSrc
   * @request: a #GstRTSPMessage
   * @response: a #GstRTSPMessage
   *
   * Handle a server request in @request and prepare @response.
   *
   * This signal is called from the streaming thread, you should therefore not
   * do any state changes on @rtspsrc because this might deadlock. If you want
   * to modify the state as a result of this signal, post a
   * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
   * in some other way.
   *
   * Since: 1.2
   */
  gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
      g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
      0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
      G_TYPE_POINTER, G_TYPE_POINTER);

  /**
   * GstRTSPSrc::on-sdp:
   * @rtspsrc: a #GstRTSPSrc
   * @sdp: a #GstSDPMessage
   *
   * Emited when the client has retrieved the SDP and before it configures the
   * streams in the SDP. @sdp can be inspected and modified.
   *
   * This signal is called from the streaming thread, you should therefore not
   * do any state changes on @rtspsrc because this might deadlock. If you want
   * to modify the state as a result of this signal, post a
   * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
   * in some other way.
   *
   * Since: 1.2
   */
  gst_rtspsrc_signals[SIGNAL_ON_SDP] =
      g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
      0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
      GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);

  /**
   * GstRTSPSrc::select-stream:
   * @rtspsrc: a #GstRTSPSrc
   * @num: the stream number
   * @caps: the stream caps
   *
   * Emited before the client decides to configure the stream @num with
   * @caps.
   *
   * Returns: %TRUE when the stream should be selected, %FALSE when the stream
   * is to be ignored.
   *
   * Since: 1.2
   */
  gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
      g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
      (GCallback) default_select_stream, select_stream_accum, NULL,
      g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
      GST_TYPE_CAPS);
  /**
   * GstRTSPSrc::new-manager:
   * @rtspsrc: a #GstRTSPSrc
   * @manager: a #GstElement
   *
   * Emited after a new manager (like rtpbin) was created and the default
   * properties were configured.
   *
   * Since: 1.4
   */
  gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
      g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
      g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);

  /**
   * GstRTSPSrc::request-rtcp-key:
   * @rtspsrc: a #GstRTSPSrc
   * @num: the stream number
   *
   * Signal emited to get the crypto parameters relevant to the RTCP
   * stream. User should provide the key and the RTCP encryption ciphers
   * and authentication, and return them wrapped in a GstCaps.
   *
   * Since: 1.4
   */
  gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY] =
      g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);

  /**
   * GstRTSPSrc::accept-certificate:
   * @rtspsrc: a #GstRTSPSrc
   * @peer_cert: the peer's #GTlsCertificate
   * @errors: the problems with @peer_cert
   * @user_data: user data set when the signal handler was connected.
   *
   * This will directly map to #GTlsConnection 's "accept-certificate"
   * signal and be performed after the default checks of #GstRTSPConnection
   * (checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
   * have failed. If no #GTlsDatabase is set on this connection, only this
   * signal will be emitted.
   *
   * Since: 1.14
   */
  gst_rtspsrc_signals[SIGNAL_ACCEPT_CERTIFICATE] =
      g_signal_new ("accept-certificate", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, 0, g_signal_accumulator_true_handled, NULL, NULL,
      G_TYPE_BOOLEAN, 3, G_TYPE_TLS_CONNECTION, G_TYPE_TLS_CERTIFICATE,
      G_TYPE_TLS_CERTIFICATE_FLAGS);

  /*
   * GstRTSPSrc::before-send
   * @rtspsrc: a #GstRTSPSrc
   * @num: the stream number
   *
   * Emitted before each RTSP request is sent, in order to allow
   * the application to modify send parameters or to skip the message entirely.
   * This can be used, for example, to work with ONVIF Profile G servers,
   * which need a different/additional range, rate-control, and intra/x
   * parameters.
   *
   * Returns: %TRUE when the command should be sent, %FALSE when the
   * command should be dropped.
   *
   * Since: 1.14
   */
  gst_rtspsrc_signals[SIGNAL_BEFORE_SEND] =
      g_signal_new_class_handler ("before-send", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
      (GCallback) default_before_send, before_send_accum, NULL,
      g_cclosure_marshal_generic, G_TYPE_BOOLEAN,
      1, GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);

  /**
   * GstRTSPSrc::push-backchannel-buffer:
   * @rtspsrc: a #GstRTSPSrc
   * @buffer: RTP buffer to send back
   *
   *
   */
  gst_rtspsrc_signals[SIGNAL_PUSH_BACKCHANNEL_BUFFER] =
      g_signal_new ("push-backchannel-buffer", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTSPSrcClass,
          push_backchannel_buffer), NULL, NULL, NULL, GST_TYPE_FLOW_RETURN, 2,
      G_TYPE_UINT, GST_TYPE_BUFFER);

  gstelement_class->send_event = gst_rtspsrc_send_event;
  gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
  gstelement_class->change_state = gst_rtspsrc_change_state;

  gst_element_class_add_static_pad_template (gstelement_class, &rtptemplate);

  gst_element_class_set_static_metadata (gstelement_class,
      "RTSP packet receiver", "Source/Network",
      "Receive data over the network via RTSP (RFC 2326)",
      "Wim Taymans <wim@fluendo.com>, "
      "Thijs Vermeir <thijs.vermeir@barco.com>, "
      "Lutz Mueller <lutz@topfrose.de>");

  gstbin_class->handle_message = gst_rtspsrc_handle_message;

  klass->push_backchannel_buffer = gst_rtspsrc_push_backchannel_buffer;

  gst_rtsp_ext_list_init ();
}

static void
gst_rtspsrc_init (GstRTSPSrc * src)
{
  src->conninfo.location = g_strdup (DEFAULT_LOCATION);
  src->protocols = DEFAULT_PROTOCOLS;
  src->debug = DEFAULT_DEBUG;
  src->retry = DEFAULT_RETRY;
  src->udp_timeout = DEFAULT_TIMEOUT;
  gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
  src->latency = DEFAULT_LATENCY_MS;
  src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
  src->connection_speed = DEFAULT_CONNECTION_SPEED;
  src->nat_method = DEFAULT_NAT_METHOD;
  src->do_rtcp = DEFAULT_DO_RTCP;
  src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
  gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
  src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
  src->user_id = g_strdup (DEFAULT_USER_ID);
  src->user_pw = g_strdup (DEFAULT_USER_PW);
  src->buffer_mode = DEFAULT_BUFFER_MODE;
  src->client_port_range.min = 0;
  src->client_port_range.max = 0;
  src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
  src->short_header = DEFAULT_SHORT_HEADER;
  src->probation = DEFAULT_PROBATION;
  src->udp_reconnect = DEFAULT_UDP_RECONNECT;
  src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
  src->ntp_sync = DEFAULT_NTP_SYNC;
  src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
  src->sdes = NULL;
  src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
  src->tls_database = DEFAULT_TLS_DATABASE;
  src->tls_interaction = DEFAULT_TLS_INTERACTION;
  src->do_retransmission = DEFAULT_DO_RETRANSMISSION;
  src->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
  src->user_agent = g_strdup (DEFAULT_USER_AGENT);
  src->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
  src->rfc7273_sync = DEFAULT_RFC7273_SYNC;
  src->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
  src->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
  src->max_ts_offset_is_set = FALSE;
  src->default_version = DEFAULT_VERSION;
  src->version = GST_RTSP_VERSION_INVALID;

  /* get a list of all extensions */
  src->extensions = gst_rtsp_ext_list_get ();

  /* connect to send signal */
  gst_rtsp_ext_list_connect (src->extensions, "send",
      (GCallback) gst_rtspsrc_send_cb, src);

  /* protects the streaming thread in interleaved mode or the polling
   * thread in UDP mode. */
  g_rec_mutex_init (&src->stream_rec_lock);

  /* protects our state changes from multiple invocations */
  g_rec_mutex_init (&src->state_rec_lock);

  src->state = GST_RTSP_STATE_INVALID;

  g_mutex_init (&src->conninfo.send_lock);
  g_mutex_init (&src->conninfo.recv_lock);

  GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
  gst_bin_set_suppressed_flags (GST_BIN (src),
      GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
}

static void
gst_rtspsrc_finalize (GObject * object)
{
  GstRTSPSrc *rtspsrc;

  rtspsrc = GST_RTSPSRC (object);

  gst_rtsp_ext_list_free (rtspsrc->extensions);
  g_free (rtspsrc->conninfo.location);
  gst_rtsp_url_free (rtspsrc->conninfo.url);
  g_free (rtspsrc->conninfo.url_str);
  g_free (rtspsrc->user_id);
  g_free (rtspsrc->user_pw);
  g_free (rtspsrc->multi_iface);
  g_free (rtspsrc->user_agent);

  if (rtspsrc->sdp) {
    gst_sdp_message_free (rtspsrc->sdp);
    rtspsrc->sdp = NULL;
  }
  if (rtspsrc->provided_clock)
    gst_object_unref (rtspsrc->provided_clock);

  if (rtspsrc->sdes)
    gst_structure_free (rtspsrc->sdes);

  if (rtspsrc->tls_database)
    g_object_unref (rtspsrc->tls_database);

  if (rtspsrc->tls_interaction)
    g_object_unref (rtspsrc->tls_interaction);

  /* free locks */
  g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
  g_rec_mutex_clear (&rtspsrc->state_rec_lock);

  g_mutex_clear (&rtspsrc->conninfo.send_lock);
  g_mutex_clear (&rtspsrc->conninfo.recv_lock);

  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static GstClock *
gst_rtspsrc_provide_clock (GstElement * element)
{
  GstRTSPSrc *src = GST_RTSPSRC (element);
  GstClock *clock;

  if ((clock = src->provided_clock) != NULL)
    return gst_object_ref (clock);

  return GST_ELEMENT_CLASS (parent_class)->provide_clock (element);
}

/* a proxy string of the format [user:passwd@]host[:port] */
static gboolean
gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
{
  gchar *p, *at, *col;

  g_free (rtsp->proxy_user);
  rtsp->proxy_user = NULL;
  g_free (rtsp->proxy_passwd);
  rtsp->proxy_passwd = NULL;
  g_free (rtsp->proxy_host);
  rtsp->proxy_host = NULL;
  rtsp->proxy_port = 0;

  p = (gchar *) proxy;

  if (p == NULL)
    return TRUE;

  /* we allow http:// in front but ignore it */
  if (g_str_has_prefix (p, "http://"))
    p += 7;

  at = strchr (p, '@');
  if (at) {
    /* look for user:passwd */
    col = strchr (proxy, ':');
    if (col == NULL || col > at)
      return FALSE;

    rtsp->proxy_user = g_strndup (p, col - p);
    col++;
    rtsp->proxy_passwd = g_strndup (col, at - col);

    /* move to host */
    p = at + 1;
  } else {
    if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
      rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
    if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
      rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
    if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
      GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
          GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
    }
  }
  col = strchr (p, ':');

  if (col) {
    /* everything before the colon is the hostname */
    rtsp->proxy_host = g_strndup (p, col - p);
    p = col + 1;
    rtsp->proxy_port = strtoul (p, (char **) &p, 10);
  } else {
    rtsp->proxy_host = g_strdup (p);
    rtsp->proxy_port = 8080;
  }
  return TRUE;
}

static void
gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
{
  rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
  rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;

  if (timeout != 0)
    rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
  else
    rtspsrc->ptcp_timeout = NULL;
}

static void
gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
    GParamSpec * pspec)
{
  GstRTSPSrc *rtspsrc;

  rtspsrc = GST_RTSPSRC (object);

  switch (prop_id) {
    case PROP_LOCATION:
      gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
          g_value_get_string (value), NULL);
      break;
    case PROP_PROTOCOLS:
      rtspsrc->protocols = g_value_get_flags (value);
      break;
    case PROP_DEBUG:
      rtspsrc->debug = g_value_get_boolean (value);
      break;
    case PROP_RETRY:
      rtspsrc->retry = g_value_get_uint (value);
      break;
    case PROP_TIMEOUT:
      rtspsrc->udp_timeout = g_value_get_uint64 (value);
      break;
    case PROP_TCP_TIMEOUT:
      gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
      break;
    case PROP_LATENCY:
      rtspsrc->latency = g_value_get_uint (value);
      break;
    case PROP_DROP_ON_LATENCY:
      rtspsrc->drop_on_latency = g_value_get_boolean (value);
      break;
    case PROP_CONNECTION_SPEED:
      rtspsrc->connection_speed = g_value_get_uint64 (value);
      break;
    case PROP_NAT_METHOD:
      rtspsrc->nat_method = g_value_get_enum (value);
      break;
    case PROP_DO_RTCP:
      rtspsrc->do_rtcp = g_value_get_boolean (value);
      break;
    case PROP_DO_RTSP_KEEP_ALIVE:
      rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
      break;
    case PROP_PROXY:
      gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
      break;
    case PROP_PROXY_ID:
      g_free (rtspsrc->prop_proxy_id);
      rtspsrc->prop_proxy_id = g_value_dup_string (value);
      break;
    case PROP_PROXY_PW:
      g_free (rtspsrc->prop_proxy_pw);
      rtspsrc->prop_proxy_pw = g_value_dup_string (value);
      break;
    case PROP_RTP_BLOCKSIZE:
      rtspsrc->rtp_blocksize = g_value_get_uint (value);
      break;
    case PROP_USER_ID:
      g_free (rtspsrc->user_id);
      rtspsrc->user_id = g_value_dup_string (value);
      break;
    case PROP_USER_PW:
      g_free (rtspsrc->user_pw);
      rtspsrc->user_pw = g_value_dup_string (value);
      break;
    case PROP_BUFFER_MODE:
      rtspsrc->buffer_mode = g_value_get_enum (value);
      break;
    case PROP_PORT_RANGE:
    {
      const gchar *str;

      str = g_value_get_string (value);
      if (sscanf (str, "%u-%u", &rtspsrc->client_port_range.min,
              &rtspsrc->client_port_range.max) != 2) {
        rtspsrc->client_port_range.min = 0;
        rtspsrc->client_port_range.max = 0;
      }
      break;
    }
    case PROP_UDP_BUFFER_SIZE:
      rtspsrc->udp_buffer_size = g_value_get_int (value);
      break;
    case PROP_SHORT_HEADER:
      rtspsrc->short_header = g_value_get_boolean (value);
      break;
    case PROP_PROBATION:
      rtspsrc->probation = g_value_get_uint (value);
      break;
    case PROP_UDP_RECONNECT:
      rtspsrc->udp_reconnect = g_value_get_boolean (value);
      break;
    case PROP_MULTICAST_IFACE:
      g_free (rtspsrc->multi_iface);

      if (g_value_get_string (value) == NULL)
        rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
      else
        rtspsrc->multi_iface = g_value_dup_string (value);
      break;
    case PROP_NTP_SYNC:
      rtspsrc->ntp_sync = g_value_get_boolean (value);
      /* The default value of max_ts_offset depends on ntp_sync. If user
       * hasn't set it then change default value */
      if (!rtspsrc->max_ts_offset_is_set) {
        if (rtspsrc->ntp_sync) {
          rtspsrc->max_ts_offset = 0;
        } else {
          rtspsrc->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
        }
      }
      break;
    case PROP_USE_PIPELINE_CLOCK:
      rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
      break;
    case PROP_SDES:
      rtspsrc->sdes = g_value_dup_boxed (value);
      break;
    case PROP_TLS_VALIDATION_FLAGS:
      rtspsrc->tls_validation_flags = g_value_get_flags (value);
      break;
    case PROP_TLS_DATABASE:
      g_clear_object (&rtspsrc->tls_database);
      rtspsrc->tls_database = g_value_dup_object (value);
      break;
    case PROP_TLS_INTERACTION:
      g_clear_object (&rtspsrc->tls_interaction);
      rtspsrc->tls_interaction = g_value_dup_object (value);
      break;
    case PROP_DO_RETRANSMISSION:
      rtspsrc->do_retransmission = g_value_get_boolean (value);
      break;
    case PROP_NTP_TIME_SOURCE:
      rtspsrc->ntp_time_source = g_value_get_enum (value);
      break;
    case PROP_USER_AGENT:
      g_free (rtspsrc->user_agent);
      rtspsrc->user_agent = g_value_dup_string (value);
      break;
    case PROP_MAX_RTCP_RTP_TIME_DIFF:
      rtspsrc->max_rtcp_rtp_time_diff = g_value_get_int (value);
      break;
    case PROP_RFC7273_SYNC:
      rtspsrc->rfc7273_sync = g_value_get_boolean (value);
      break;
    case PROP_MAX_TS_OFFSET_ADJUSTMENT:
      rtspsrc->max_ts_offset_adjustment = g_value_get_uint64 (value);
      break;
    case PROP_MAX_TS_OFFSET:
      rtspsrc->max_ts_offset = g_value_get_int64 (value);
      rtspsrc->max_ts_offset_is_set = TRUE;
      break;
    case PROP_DEFAULT_VERSION:
      rtspsrc->default_version = g_value_get_enum (value);
      break;
    case PROP_BACKCHANNEL:
      rtspsrc->backchannel = g_value_get_enum (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstRTSPSrc *rtspsrc;

  rtspsrc = GST_RTSPSRC (object);

  switch (prop_id) {
    case PROP_LOCATION:
      g_value_set_string (value, rtspsrc->conninfo.location);
      break;
    case PROP_PROTOCOLS:
      g_value_set_flags (value, rtspsrc->protocols);
      break;
    case PROP_DEBUG:
      g_value_set_boolean (value, rtspsrc->debug);
      break;
    case PROP_RETRY:
      g_value_set_uint (value, rtspsrc->retry);
      break;
    case PROP_TIMEOUT:
      g_value_set_uint64 (value, rtspsrc->udp_timeout);
      break;
    case PROP_TCP_TIMEOUT:
    {
      guint64 timeout;

      timeout = ((guint64) rtspsrc->tcp_timeout.tv_sec) * G_USEC_PER_SEC +
          rtspsrc->tcp_timeout.tv_usec;
      g_value_set_uint64 (value, timeout);
      break;
    }
    case PROP_LATENCY:
      g_value_set_uint (value, rtspsrc->latency);
      break;
    case PROP_DROP_ON_LATENCY:
      g_value_set_boolean (value, rtspsrc->drop_on_latency);
      break;
    case PROP_CONNECTION_SPEED:
      g_value_set_uint64 (value, rtspsrc->connection_speed);
      break;
    case PROP_NAT_METHOD:
      g_value_set_enum (value, rtspsrc->nat_method);
      break;
    case PROP_DO_RTCP:
      g_value_set_boolean (value, rtspsrc->do_rtcp);
      break;
    case PROP_DO_RTSP_KEEP_ALIVE:
      g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
      break;
    case PROP_PROXY:
    {
      gchar *str;

      if (rtspsrc->proxy_host) {
        str =
            g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
      } else {
        str = NULL;
      }
      g_value_take_string (value, str);
      break;
    }
    case PROP_PROXY_ID:
      g_value_set_string (value, rtspsrc->prop_proxy_id);
      break;
    case PROP_PROXY_PW:
      g_value_set_string (value, rtspsrc->prop_proxy_pw);
      break;
    case PROP_RTP_BLOCKSIZE:
      g_value_set_uint (value, rtspsrc->rtp_blocksize);
      break;
    case PROP_USER_ID:
      g_value_set_string (value, rtspsrc->user_id);
      break;
    case PROP_USER_PW:
      g_value_set_string (value, rtspsrc->user_pw);
      break;
    case PROP_BUFFER_MODE:
      g_value_set_enum (value, rtspsrc->buffer_mode);
      break;
    case PROP_PORT_RANGE:
    {
      gchar *str;

      if (rtspsrc->client_port_range.min != 0) {
        str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
            rtspsrc->client_port_range.max);
      } else {
        str = NULL;
      }
      g_value_take_string (value, str);
      break;
    }
    case PROP_UDP_BUFFER_SIZE:
      g_value_set_int (value, rtspsrc->udp_buffer_size);
      break;
    case PROP_SHORT_HEADER:
      g_value_set_boolean (value, rtspsrc->short_header);
      break;
    case PROP_PROBATION:
      g_value_set_uint (value, rtspsrc->probation);
      break;
    case PROP_UDP_RECONNECT:
      g_value_set_boolean (value, rtspsrc->udp_reconnect);
      break;
    case PROP_MULTICAST_IFACE:
      g_value_set_string (value, rtspsrc->multi_iface);
      break;
    case PROP_NTP_SYNC:
      g_value_set_boolean (value, rtspsrc->ntp_sync);
      break;
    case PROP_USE_PIPELINE_CLOCK:
      g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
      break;
    case PROP_SDES:
      g_value_set_boxed (value, rtspsrc->sdes);
      break;
    case PROP_TLS_VALIDATION_FLAGS:
      g_value_set_flags (value, rtspsrc->tls_validation_flags);
      break;
    case PROP_TLS_DATABASE:
      g_value_set_object (value, rtspsrc->tls_database);
      break;
    case PROP_TLS_INTERACTION:
      g_value_set_object (value, rtspsrc->tls_interaction);
      break;
    case PROP_DO_RETRANSMISSION:
      g_value_set_boolean (value, rtspsrc->do_retransmission);
      break;
    case PROP_NTP_TIME_SOURCE:
      g_value_set_enum (value, rtspsrc->ntp_time_source);
      break;
    case PROP_USER_AGENT:
      g_value_set_string (value, rtspsrc->user_agent);
      break;
    case PROP_MAX_RTCP_RTP_TIME_DIFF:
      g_value_set_int (value, rtspsrc->max_rtcp_rtp_time_diff);
      break;
    case PROP_RFC7273_SYNC:
      g_value_set_boolean (value, rtspsrc->rfc7273_sync);
      break;
    case PROP_MAX_TS_OFFSET_ADJUSTMENT:
      g_value_set_uint64 (value, rtspsrc->max_ts_offset_adjustment);
      break;
    case PROP_MAX_TS_OFFSET:
      g_value_set_int64 (value, rtspsrc->max_ts_offset);
      break;
    case PROP_DEFAULT_VERSION:
      g_value_set_enum (value, rtspsrc->default_version);
      break;
    case PROP_BACKCHANNEL:
      g_value_set_enum (value, rtspsrc->backchannel);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static gint
find_stream_by_id (GstRTSPStream * stream, gint * id)
{
  if (stream->id == *id)
    return 0;

  return -1;
}

static gint
find_stream_by_channel (GstRTSPStream * stream, gint * channel)
{
  /* ignore unconfigured channels here (e.g., those that
   * were explicitly skipped during SETUP) */
  if ((stream->channelpad[0] != NULL) &&
      (stream->channel[0] == *channel || stream->channel[1] == *channel))
    return 0;

  return -1;
}

static gint
find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
{
  GstElement *src = (GstElement *) a;

  if (stream->udpsrc[0] == src)
    return 0;
  if (stream->udpsrc[1] == src)
    return 0;

  return -1;
}

static gint
find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
{
  if (stream->conninfo.location) {
    /* check qualified setup_url */
    if (!strcmp (stream->conninfo.location, (gchar *) a))
      return 0;
  }
  if (stream->control_url) {
    /* check original control_url */
    if (!strcmp (stream->control_url, (gchar *) a))
      return 0;

    /* check if qualified setup_url ends with string */
    if (g_str_has_suffix (stream->control_url, (gchar *) a))
      return 0;
  }

  return -1;
}

static GstRTSPStream *
find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
{
  GList *lstream;

  /* find and get stream */
  if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
    return (GstRTSPStream *) lstream->data;

  return NULL;
}

static const GstSDPBandwidth *
gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
    const GstSDPMedia * media, const gchar * type)
{
  guint i, len;

  /* first look in the media specific section */
  len = gst_sdp_media_bandwidths_len (media);
  for (i = 0; i < len; i++) {
    const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);

    if (strcmp (bw->bwtype, type) == 0)
      return bw;
  }
  /* then look in the message specific section */
  len = gst_sdp_message_bandwidths_len (sdp);
  for (i = 0; i < len; i++) {
    const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);

    if (strcmp (bw->bwtype, type) == 0)
      return bw;
  }
  return NULL;
}

static void
gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
    const GstSDPMedia * media, GstRTSPStream * stream)
{
  const GstSDPBandwidth *bw;

  if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
    stream->as_bandwidth = bw->bandwidth;
  else
    stream->as_bandwidth = -1;

  if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
    stream->rr_bandwidth = bw->bandwidth;
  else
    stream->rr_bandwidth = -1;

  if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
    stream->rs_bandwidth = bw->bandwidth;
  else
    stream->rs_bandwidth = -1;
}

static void
gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
    const GstSDPConnection * conn)
{
  if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
    return;

  if (conn->addrtype == NULL)
    return;

  /* check for IPV6 */
  if (strcmp (conn->addrtype, "IP4") == 0)
    stream->is_ipv6 = FALSE;
  else if (strcmp (conn->addrtype, "IP6") == 0)
    stream->is_ipv6 = TRUE;
  else
    return;

  /* save address */
  g_free (stream->destination);
  stream->destination = g_strdup (conn->address);

  /* check for multicast */
  stream->is_multicast =
      gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
      conn->address);
  stream->ttl = conn->ttl;
}

/* Go over the connections for a stream.
 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
 *   receiving.
 * - If we are dealing with a localhost address, we disable multicast
 */
static void
gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
    const GstSDPMedia * media, GstRTSPStream * stream)
{
  const GstSDPConnection *conn;
  guint i, len;

  /* first look in the media specific section */
  len = gst_sdp_media_connections_len (media);
  for (i = 0; i < len; i++) {
    conn = gst_sdp_media_get_connection (media, i);

    gst_rtspsrc_do_stream_connection (src, stream, conn);
  }
  /* then look in the message specific section */
  if ((conn = gst_sdp_message_get_connection (sdp))) {
    gst_rtspsrc_do_stream_connection (src, stream, conn);
  }
}

static gchar *
make_stream_id (GstRTSPStream * stream, const GstSDPMedia * media)
{
  gchar *stream_id =
      g_strdup_printf ("%s:%d:%d:%s:%d", media->media, media->port,
      media->num_ports, media->proto, stream->default_pt);

  g_strcanon (stream_id, G_CSET_a_2_z G_CSET_A_2_Z G_CSET_DIGITS, ':');

  return stream_id;
}

/*   m=<media> <UDP port> RTP/AVP <payload>
 */
static void
gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
    const GstSDPMedia * media, GstRTSPStream * stream)
{
  guint i, len;
  const gchar *proto;
  GstCaps *global_caps;

  /* get proto */
  proto = gst_sdp_media_get_proto (media);
  if (proto == NULL)
    goto no_proto;

  if (g_str_equal (proto, "RTP/AVP"))
    stream->profile = GST_RTSP_PROFILE_AVP;
  else if (g_str_equal (proto, "RTP/SAVP"))
    stream->profile = GST_RTSP_PROFILE_SAVP;
  else if (g_str_equal (proto, "RTP/AVPF"))
    stream->profile = GST_RTSP_PROFILE_AVPF;
  else if (g_str_equal (proto, "RTP/SAVPF"))
    stream->profile = GST_RTSP_PROFILE_SAVPF;
  else
    goto unknown_proto;

  if (gst_sdp_media_get_attribute_val (media, "sendonly") != NULL &&
      /* We want to setup caps for streams configured as backchannel */
      !stream->is_backchannel && src->backchannel != BACKCHANNEL_NONE)
    goto sendonly_media;

  /* Parse global SDP attributes once */
  global_caps = gst_caps_new_empty_simple ("application/x-unknown");
  GST_DEBUG ("mapping sdp session level attributes to caps");
  gst_sdp_message_attributes_to_caps (sdp, global_caps);
  GST_DEBUG ("mapping sdp media level attributes to caps");
  gst_sdp_media_attributes_to_caps (media, global_caps);

  /* Keep a copy of the SDP key management */
  gst_sdp_media_parse_keymgmt (media, &stream->mikey);
  if (stream->mikey == NULL)
    gst_sdp_message_parse_keymgmt (sdp, &stream->mikey);

  len = gst_sdp_media_formats_len (media);
  for (i = 0; i < len; i++) {
    gint pt;
    GstCaps *caps, *outcaps;
    GstStructure *s;
    const gchar *enc;
    PtMapItem item;

    pt = atoi (gst_sdp_media_get_format (media, i));

    GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);

    /* convert caps */
    caps = gst_sdp_media_get_caps_from_media (media, pt);
    if (caps == NULL) {
      GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
      continue;
    }

    /* do some tweaks */
    s = gst_caps_get_structure (caps, 0);
    if ((enc = gst_structure_get_string (s, "encoding-name"))) {
      stream->is_real = (strstr (enc, "-REAL") != NULL);
      if (strcmp (enc, "X-ASF-PF") == 0)
        stream->container = TRUE;
    }

    /* Merge in global caps */
    /* Intersect will merge in missing fields to the current caps */
    outcaps = gst_caps_intersect (caps, global_caps);
    gst_caps_unref (caps);

    /* the first pt will be the default */
    if (stream->ptmap->len == 0)
      stream->default_pt = pt;

    item.pt = pt;
    item.caps = outcaps;

    g_array_append_val (stream->ptmap, item);
  }

  stream->stream_id = make_stream_id (stream, media);

  gst_caps_unref (global_caps);
  return;

no_proto:
  {
    GST_ERROR_OBJECT (src, "can't find proto in media");
    return;
  }
unknown_proto:
  {
    GST_ERROR_OBJECT (src, "unknown proto in media: '%s'", proto);
    return;
  }
sendonly_media:
  {
    GST_DEBUG_OBJECT (src, "sendonly media ignored, no backchannel");
    return;
  }
}

static const gchar *
get_aggregate_control (GstRTSPSrc * src)
{
  const gchar *base;

  if (src->control)
    base = src->control;
  else if (src->content_base)
    base = src->content_base;
  else if (src->conninfo.url_str)
    base = src->conninfo.url_str;
  else
    base = "/";

  return base;
}

static void
clear_ptmap_item (PtMapItem * item)
{
  if (item->caps)
    gst_caps_unref (item->caps);
}

static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx,
    gint n_streams)
{
  GstRTSPStream *stream;
  const gchar *control_url;
  const GstSDPMedia *media;

  /* get media, should not return NULL */
  media = gst_sdp_message_get_media (sdp, idx);
  if (media == NULL)
    return NULL;

  stream = g_new0 (GstRTSPStream, 1);
  stream->parent = src;
  /* we mark the pad as not linked, we will mark it as OK when we add the pad to
   * the element. */
  stream->last_ret = GST_FLOW_NOT_LINKED;
  stream->added = FALSE;
  stream->setup = FALSE;
  stream->skipped = FALSE;
  stream->id = idx;
  stream->eos = FALSE;
  stream->discont = TRUE;
  stream->seqbase = -1;
  stream->timebase = -1;
  stream->send_ssrc = g_random_int ();
  stream->profile = GST_RTSP_PROFILE_AVP;
  stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
  stream->mikey = NULL;
  stream->stream_id = NULL;
  stream->is_backchannel = FALSE;
  g_mutex_init (&stream->conninfo.send_lock);
  g_mutex_init (&stream->conninfo.recv_lock);
  g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);

  /* stream is sendonly and onvif backchannel is requested */
  if (gst_sdp_media_get_attribute_val (media, "sendonly") != NULL &&
      src->backchannel != BACKCHANNEL_NONE)
    stream->is_backchannel = TRUE;

  /* collect bandwidth information for this steam. FIXME, configure in the RTP
   * session manager to scale RTCP. */
  gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);

  /* collect connection info */
  gst_rtspsrc_collect_connections (src, sdp, media, stream);

  /* make the payload type map */
  gst_rtspsrc_collect_payloads (src, sdp, media, stream);

  /* collect port number */
  stream->port = gst_sdp_media_get_port (media);

  /* get control url to construct the setup url. The setup url is used to
   * configure the transport of the stream and is used to identity the stream in
   * the RTP-Info header field returned from PLAY. */
  control_url = gst_sdp_media_get_attribute_val (media, "control");
  if (control_url == NULL)
    control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);

  GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
  GST_DEBUG_OBJECT (src, " port: %d", stream->port);
  GST_DEBUG_OBJECT (src, " container: %d", stream->container);
  GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));

  /* RFC 2326, C.3: missing control_url permitted in case of a single stream */
  if (control_url == NULL && n_streams == 1) {
    control_url = "";
  }

  if (control_url != NULL) {
    stream->control_url = g_strdup (control_url);
    /* Build a fully qualified url using the content_base if any or by prefixing
     * the original request.
     * If the control_url starts with a '/' or a non rtsp: protocol we will most
     * likely build a URL that the server will fail to understand, this is ok,
     * we will fail then. */
    if (g_str_has_prefix (control_url, "rtsp://"))
      stream->conninfo.location = g_strdup (control_url);
    else {
      const gchar *base;
      gboolean has_slash;

      if (g_strcmp0 (control_url, "*") == 0)
        control_url = "";

      base = get_aggregate_control (src);

      /* check if the base ends or control starts with / */
      has_slash = g_str_has_prefix (control_url, "/");
      has_slash = has_slash || g_str_has_suffix (base, "/");

      /* concatenate the two strings, insert / when not present */
      stream->conninfo.location =
          g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
    }
  }
  GST_DEBUG_OBJECT (src, " setup: %s",
      GST_STR_NULL (stream->conninfo.location));

  /* we keep track of all streams */
  src->streams = g_list_append (src->streams, stream);

  return stream;

  /* ERRORS */
}

static void
gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
{
  gint i;

  GST_DEBUG_OBJECT (src, "free stream %p", stream);

  g_array_free (stream->ptmap, TRUE);

  g_free (stream->destination);
  g_free (stream->control_url);
  g_free (stream->conninfo.location);
  g_free (stream->stream_id);

  for (i = 0; i < 2; i++) {
    if (stream->udpsrc[i]) {
      gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
      gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
      gst_object_unref (stream->udpsrc[i]);
    }
    if (stream->channelpad[i])
      gst_object_unref (stream->channelpad[i]);

    if (stream->udpsink[i]) {
      gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
      gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
      gst_object_unref (stream->udpsink[i]);
    }
  }
  if (stream->rtpsrc) {
    gst_element_set_state (stream->rtpsrc, GST_STATE_NULL);
    gst_bin_remove (GST_BIN_CAST (src), stream->rtpsrc);
    gst_object_unref (stream->rtpsrc);
  }
  if (stream->srcpad) {
    gst_pad_set_active (stream->srcpad, FALSE);
    if (stream->added)
      gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
  }
  if (stream->srtpenc)
    gst_object_unref (stream->srtpenc);
  if (stream->srtpdec)
    gst_object_unref (stream->srtpdec);
  if (stream->srtcpparams)
    gst_caps_unref (stream->srtcpparams);
  if (stream->mikey)
    gst_mikey_message_unref (stream->mikey);
  if (stream->rtcppad)
    gst_object_unref (stream->rtcppad);
  if (stream->session)
    g_object_unref (stream->session);
  if (stream->rtx_pt_map)
    gst_structure_free (stream->rtx_pt_map);

  g_mutex_clear (&stream->conninfo.send_lock);
  g_mutex_clear (&stream->conninfo.recv_lock);

  g_free (stream);
}

static void
gst_rtspsrc_cleanup (GstRTSPSrc * src)
{
  GList *walk;

  GST_DEBUG_OBJECT (src, "cleanup");

  for (walk = src->streams; walk; walk = g_list_next (walk)) {
    GstRTSPStream *stream = (GstRTSPStream *) walk->data;

    gst_rtspsrc_stream_free (src, stream);
  }
  g_list_free (src->streams);
  src->streams = NULL;
  if (src->manager) {
    if (src->manager_sig_id) {
      g_signal_handler_disconnect (src->manager, src->manager_sig_id);
      src->manager_sig_id = 0;
    }
    gst_element_set_state (src->manager, GST_STATE_NULL);
    gst_bin_remove (GST_BIN_CAST (src), src->manager);
    src->manager = NULL;
  }
  if (src->props)
    gst_structure_free (src->props);
  src->props = NULL;

  g_free (src->content_base);
  src->content_base = NULL;

  g_free (src->control);
  src->control = NULL;

  if (src->range)
    gst_rtsp_range_free (src->range);
  src->range = NULL;

  /* don't clear the SDP when it was used in the url */
  if (src->sdp && !src->from_sdp) {
    gst_sdp_message_free (src->sdp);
    src->sdp = NULL;
  }

  src->need_segment = FALSE;

  if (src->provided_clock) {
    gst_object_unref (src->provided_clock);
    src->provided_clock = NULL;
  }
}

static gboolean
gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
    gint * rtpport, gint * rtcpport)
{
  GstRTSPSrc *src;
  GstStateChangeReturn ret;
  GstElement *udpsrc0, *udpsrc1;
  gint tmp_rtp, tmp_rtcp;
  guint count;
  const gchar *host;

  src = stream->parent;

  udpsrc0 = NULL;
  udpsrc1 = NULL;
  count = 0;

  /* Start at next port */
  tmp_rtp = src->next_port_num;

  if (stream->is_ipv6)
    host = "udp://[::0]";
  else
    host = "udp://0.0.0.0";

  /* try to allocate 2 UDP ports, the RTP port should be an even
   * number and the RTCP port should be the next (uneven) port */
again:

  if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
      tmp_rtp >= src->client_port_range.max)
    goto no_ports;

  udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
  if (udpsrc0 == NULL)
    goto no_udp_protocol;
  g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);

  if (src->udp_buffer_size != 0)
    g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
        NULL);

  ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
  if (ret == GST_STATE_CHANGE_FAILURE) {
    if (tmp_rtp != 0) {
      GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);

      tmp_rtp += 2;
      if (++count > src->retry)
        goto no_ports;

      GST_DEBUG_OBJECT (src, "free RTP udpsrc");
      gst_element_set_state (udpsrc0, GST_STATE_NULL);
      gst_object_unref (udpsrc0);
      udpsrc0 = NULL;

      GST_DEBUG_OBJECT (src, "retry %d", count);
      goto again;
    }
    goto no_udp_protocol;
  }

  g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
  GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);

  /* check if port is even */
  if ((tmp_rtp & 0x01) != 0) {
    /* port not even, close and allocate another */
    if (++count > src->retry)
      goto no_ports;

    GST_DEBUG_OBJECT (src, "RTP port not even");

    GST_DEBUG_OBJECT (src, "free RTP udpsrc");
    gst_element_set_state (udpsrc0, GST_STATE_NULL);
    gst_object_unref (udpsrc0);
    udpsrc0 = NULL;

    GST_DEBUG_OBJECT (src, "retry %d", count);
    tmp_rtp++;
    goto again;
  }

  /* allocate port+1 for RTCP now */
  udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
  if (udpsrc1 == NULL)
    goto no_udp_rtcp_protocol;

  /* set port */
  tmp_rtcp = tmp_rtp + 1;
  if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
    goto no_ports;

  g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);

  GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
  ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
  /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
  if (ret == GST_STATE_CHANGE_FAILURE) {
    GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);

    if (++count > src->retry)
      goto no_ports;

    GST_DEBUG_OBJECT (src, "free RTP udpsrc");
    gst_element_set_state (udpsrc0, GST_STATE_NULL);
    gst_object_unref (udpsrc0);
    udpsrc0 = NULL;

    GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
    gst_element_set_state (udpsrc1, GST_STATE_NULL);
    gst_object_unref (udpsrc1);
    udpsrc1 = NULL;

    tmp_rtp += 2;
    GST_DEBUG_OBJECT (src, "retry %d", count);
    goto again;
  }

  /* all fine, do port check */
  g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
  g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);

  /* this should not happen... */
  if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
    goto port_error;

  /* we keep these elements, we configure all in configure_transport when the
   * server told us to really use the UDP ports. */
  stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
  stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
  gst_element_set_locked_state (stream->udpsrc[0], TRUE);
  gst_element_set_locked_state (stream->udpsrc[1], TRUE);

  /* keep track of next available port number when we have a range
   * configured */
  if (src->next_port_num != 0)
    src->next_port_num = tmp_rtcp + 1;

  return TRUE;

  /* ERRORS */
no_udp_protocol:
  {
    GST_DEBUG_OBJECT (src, "could not get UDP source");
    goto cleanup;
  }
no_ports:
  {
    GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
        count);
    goto cleanup;
  }
no_udp_rtcp_protocol:
  {
    GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
    goto cleanup;
  }
port_error:
  {
    GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
        tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
    goto cleanup;
  }
cleanup:
  {
    if (udpsrc0) {
      gst_element_set_state (udpsrc0, GST_STATE_NULL);
      gst_object_unref (udpsrc0);
    }
    if (udpsrc1) {
      gst_element_set_state (udpsrc1, GST_STATE_NULL);
      gst_object_unref (udpsrc1);
    }
    return FALSE;
  }
}

static void
gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
{
  GList *walk;

  if (src->manager)
    gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);

  for (walk = src->streams; walk; walk = g_list_next (walk)) {
    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
    gint i;

    for (i = 0; i < 2; i++) {
      if (stream->udpsrc[i])
        gst_element_set_state (stream->udpsrc[i], state);
    }
  }
}

static void
gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
{
  GstEvent *event;
  gint cmd;
  GstState state;

  if (flush) {
    event = gst_event_new_flush_start ();
    GST_DEBUG_OBJECT (src, "start flush");
    cmd = CMD_WAIT;
    state = GST_STATE_PAUSED;
  } else {
    event = gst_event_new_flush_stop (FALSE);
    GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
    cmd = CMD_LOOP;
    if (playing)
      state = GST_STATE_PLAYING;
    else
      state = GST_STATE_PAUSED;
  }
  gst_rtspsrc_push_event (src, event);
  gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
  gst_rtspsrc_set_state (src, state);
}

static GstRTSPResult
gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
    GstRTSPMessage * message, GTimeVal * timeout)
{
  GstRTSPResult ret;

  if (conninfo->connection) {
    g_mutex_lock (&conninfo->send_lock);
    ret = gst_rtsp_connection_send (conninfo->connection, message, timeout);
    g_mutex_unlock (&conninfo->send_lock);
  } else {
    ret = GST_RTSP_ERROR;
  }

  return ret;
}

static GstRTSPResult
gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
    GstRTSPMessage * message, GTimeVal * timeout)
{
  GstRTSPResult ret;

  if (conninfo->connection) {
    g_mutex_lock (&conninfo->recv_lock);
    ret = gst_rtsp_connection_receive (conninfo->connection, message, timeout);
    g_mutex_unlock (&conninfo->recv_lock);
  } else {
    ret = GST_RTSP_ERROR;
  }

  return ret;
}

static void
gst_rtspsrc_get_position (GstRTSPSrc * src)
{
  GstQuery *query;
  GList *walk;

  query = gst_query_new_position (GST_FORMAT_TIME);
  /*  should be known somewhere down the stream (e.g. jitterbuffer) */
  for (walk = src->streams; walk; walk = g_list_next (walk)) {
    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
    GstFormat fmt;
    gint64 pos;

    if (stream->srcpad) {
      if (gst_pad_query (stream->srcpad, query)) {
        gst_query_parse_position (query, &fmt, &pos);
        GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
            GST_TIME_ARGS (pos));
        src->last_pos = pos;
        goto out;
      }
    }
  }

  src->last_pos = 0;

out:

  gst_query_unref (query);
}

static gboolean
gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
{
  gdouble rate;
  GstFormat format;
  GstSeekFlags flags;
  GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
  gint64 cur, stop;
  gboolean flush, skip;
  gboolean update;
  gboolean playing;
  GstSegment seeksegment = { 0, };
  GList *walk;
  const gchar *seek_style = NULL;

  if (event) {
    GST_DEBUG_OBJECT (src, "doing seek with event %" GST_PTR_FORMAT, event);

    gst_event_parse_seek (event, &rate, &format, &flags,
        &cur_type, &cur, &stop_type, &stop);

    /* no negative rates yet */
    if (rate < 0.0)
      goto negative_rate;

    /* we need TIME format */
    if (format != src->segment.format)
      goto no_format;

    /* Check if we are not at all seekable */
    if (src->seekable == -1.0)
      goto not_seekable;

    /* Additional seeking-to-beginning-only check */
    if (src->seekable == 0.0 && cur != 0)
      goto not_seekable;
  } else {
    GST_DEBUG_OBJECT (src, "doing seek without event");
    flags = 0;
    cur_type = GST_SEEK_TYPE_SET;
    stop_type = GST_SEEK_TYPE_SET;
  }

  /* get flush flag */
  flush = flags & GST_SEEK_FLAG_FLUSH;
  skip = flags & GST_SEEK_FLAG_SKIP;

  /* now we need to make sure the streaming thread is stopped. We do this by
   * either sending a FLUSH_START event downstream which will cause the
   * streaming thread to stop with a WRONG_STATE.
   * For a non-flushing seek we simply pause the task, which will happen as soon
   * as it completes one iteration (and thus might block when the sink is
   * blocking in preroll). */
  if (flush) {
    GST_DEBUG_OBJECT (src, "starting flush");
    gst_rtspsrc_flush (src, TRUE, FALSE);
  } else {
    if (src->task) {
      gst_task_pause (src->task);
    }
  }

  /* we should now be able to grab the streaming thread because we stopped it
   * with the above flush/pause code */
  GST_RTSP_STREAM_LOCK (src);

  GST_DEBUG_OBJECT (src, "stopped streaming");

  /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
  gst_rtspsrc_connection_flush (src, FALSE);

  /* copy segment, we need this because we still need the old
   * segment when we close the current segment. */
  memcpy (&seeksegment, &src->segment, sizeof (GstSegment));

  /* configure the seek parameters in the seeksegment. We will then have the
   * right values in the segment to perform the seek */
  if (event) {
    GST_DEBUG_OBJECT (src, "configuring seek");
    gst_segment_do_seek (&seeksegment, rate, format, flags,
        cur_type, cur, stop_type, stop, &update);
  }

  /* figure out the last position we need to play. If it's configured (stop !=
   * -1), use that, else we play until the total duration of the file */
  if ((stop = seeksegment.stop) == -1)
    stop = seeksegment.duration;

  /* if we were playing, pause first */
  playing = (src->state == GST_RTSP_STATE_PLAYING);
  if (playing) {
    /* obtain current position in case seek fails */
    gst_rtspsrc_get_position (src);
    gst_rtspsrc_pause (src, FALSE);
  }
  src->skip = skip;

  src->state = GST_RTSP_STATE_SEEKING;

  /* PLAY will add the range header now. */
  src->need_range = TRUE;

  /* prepare for streaming again */
  if (flush) {
    /* if we started flush, we stop now */
    GST_DEBUG_OBJECT (src, "stopping flush");
    gst_rtspsrc_flush (src, FALSE, playing);
  }

  /* now we did the seek and can activate the new segment values */
  memcpy (&src->segment, &seeksegment, sizeof (GstSegment));

  /* if we're doing a segment seek, post a SEGMENT_START message */
  if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
    gst_element_post_message (GST_ELEMENT_CAST (src),
        gst_message_new_segment_start (GST_OBJECT_CAST (src),
            src->segment.format, src->segment.position));
  }

  /* now create the newsegment */
  GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
      " to %" G_GINT64_FORMAT, src->segment.position, stop);

  /* mark discont */
  GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
  for (walk = src->streams; walk; walk = g_list_next (walk)) {
    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
    stream->discont = TRUE;
  }

  /* and continue playing if needed */
  GST_OBJECT_LOCK (src);
  playing = (GST_STATE_PENDING (src) == GST_STATE_VOID_PENDING
      && GST_STATE (src) == GST_STATE_PLAYING)
      || (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
  GST_OBJECT_UNLOCK (src);

  if (src->version >= GST_RTSP_VERSION_2_0) {
    if (flags & GST_SEEK_FLAG_ACCURATE)
      seek_style = "RAP";
    else if (flags & GST_SEEK_FLAG_KEY_UNIT)
      seek_style = "CoRAP";
    else if (flags & GST_SEEK_FLAG_KEY_UNIT
        && flags & GST_SEEK_FLAG_SNAP_BEFORE)
      seek_style = "First-Prior";
    else if (flags & GST_SEEK_FLAG_KEY_UNIT && flags & GST_SEEK_FLAG_SNAP_AFTER)
      seek_style = "Next";
  }

  if (playing)
    gst_rtspsrc_play (src, &seeksegment, FALSE, seek_style);

  GST_RTSP_STREAM_UNLOCK (src);

  return TRUE;

  /* ERRORS */
negative_rate:
  {
    GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
    return FALSE;
  }
no_format:
  {
    GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
    return FALSE;
  }
not_seekable:
  {
    GST_DEBUG_OBJECT (src, "stream is not seekable");
    return FALSE;
  }
}

static gboolean
gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  GstRTSPSrc *src;
  gboolean res = TRUE;
  gboolean forward;

  src = GST_RTSPSRC_CAST (parent);

  GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
      GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEEK:
      res = gst_rtspsrc_perform_seek (src, event);
      forward = FALSE;
      break;
    case GST_EVENT_QOS:
    case GST_EVENT_NAVIGATION:
    case GST_EVENT_LATENCY:
    default:
      forward = TRUE;
      break;
  }
  if (forward) {
    GstPad *target;

    if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
      res = gst_pad_send_event (target, event);
      gst_object_unref (target);
    } else {
      gst_event_unref (event);
    }
  } else {
    gst_event_unref (event);
  }

  return res;
}

static gboolean
gst_rtspsrc_handle_src_sink_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  GstRTSPStream *stream;

  stream = gst_pad_get_element_private (pad);

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_STREAM_START:{
      const gchar *upstream_id;
      gchar *stream_id;

      gst_event_parse_stream_start (event, &upstream_id);
      stream_id = g_strdup_printf ("%s/%s", upstream_id, stream->stream_id);

      gst_event_unref (event);
      event = gst_event_new_stream_start (stream_id);
      g_free (stream_id);
      break;
    }
    default:
      break;
  }

  return gst_pad_push_event (stream->srcpad, event);
}

/* this is the final event function we receive on the internal source pad when
 * we deal with TCP connections */
static gboolean
gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  gboolean res;

  GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEEK:
    case GST_EVENT_QOS:
    case GST_EVENT_NAVIGATION:
    case GST_EVENT_LATENCY:
    default:
      gst_event_unref (event);
      res = TRUE;
      break;
  }
  return res;
}

/* this is the final query function we receive on the internal source pad when
 * we deal with TCP connections */
static gboolean
gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
    GstQuery * query)
{
  GstRTSPSrc *src;
  gboolean res = TRUE;

  src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));

  GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
      GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));

  switch (GST_QUERY_TYPE (query)) {
    case GST_QUERY_POSITION:
    {
      /* no idea */
      break;
    }
    case GST_QUERY_DURATION:
    {
      GstFormat format;

      gst_query_parse_duration (query, &format, NULL);

      switch (format) {
        case GST_FORMAT_TIME:
          gst_query_set_duration (query, format, src->segment.duration);
          break;
        default:
          res = FALSE;
          break;
      }
      break;
    }
    case GST_QUERY_LATENCY:
    {
      /* we are live with a min latency of 0 and unlimited max latency, this
       * result will be updated by the session manager if there is any. */
      gst_query_set_latency (query, TRUE, 0, -1);
      break;
    }
    default:
      break;
  }

  return res;
}

/* this query is executed on the ghost source pad exposed on rtspsrc. */
static gboolean
gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
    GstQuery * query)
{
  GstRTSPSrc *src;
  gboolean res = FALSE;

  src = GST_RTSPSRC_CAST (parent);

  GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
      GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));

  switch (GST_QUERY_TYPE (query)) {
    case GST_QUERY_DURATION:
    {
      GstFormat format;

      gst_query_parse_duration (query, &format, NULL);

      switch (format) {
        case GST_FORMAT_TIME:
          gst_query_set_duration (query, format, src->segment.duration);
          res = TRUE;
          break;
        default:
          break;
      }
      break;
    }
    case GST_QUERY_SEEKING:
    {
      GstFormat format;

      gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
      if (format == GST_FORMAT_TIME) {
        gboolean seekable =
            src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
        GstClockTime start = 0, duration = src->segment.duration;

        /* seeking without duration is unlikely */
        seekable = seekable && src->seekable >= 0.0 && src->segment.duration &&
            GST_CLOCK_TIME_IS_VALID (src->segment.duration);

        if (seekable) {
          if (src->seekable > 0.0) {
            start = src->last_pos - src->seekable * GST_SECOND;
          } else {
            /* src->seekable == 0 means that we can only seek to 0 */
            start = 0;
            duration = 0;
          }
        }

        GST_LOG_OBJECT (src, "seekable : %d", seekable);

        gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, start,
            duration);
        res = TRUE;
      }
      break;
    }
    case GST_QUERY_URI:
    {
      gchar *uri;

      uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
      if (uri != NULL) {
        gst_query_set_uri (query, uri);
        g_free (uri);
        res = TRUE;
      }
      break;
    }
    default:
    {
      GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));

      /* forward the query to the proxy target pad */
      if (target) {
        res = gst_pad_query (target, query);
        gst_object_unref (target);
      }
      break;
    }
  }

  return res;
}

/* callback for RTCP messages to be sent to the server when operating in TCP
 * mode. */
static GstFlowReturn
gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
  GstRTSPSrc *src;
  GstRTSPStream *stream;
  GstFlowReturn res = GST_FLOW_OK;
  GstMapInfo map;
  guint8 *data;
  guint size;
  GstRTSPResult ret;
  GstRTSPMessage message = { 0 };
  GstRTSPConnInfo *conninfo;

  stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
  src = stream->parent;

  gst_buffer_map (buffer, &map, GST_MAP_READ);
  size = map.size;
  data = map.data;

  gst_rtsp_message_init_data (&message, stream->channel[1]);

  /* lend the body data to the message */
  gst_rtsp_message_take_body (&message, data, size);

  if (stream->conninfo.connection)
    conninfo = &stream->conninfo;
  else
    conninfo = &src->conninfo;

  GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
  ret = gst_rtspsrc_connection_send (src, conninfo, &message, NULL);
  GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);

  /* and steal it away again because we will free it when unreffing the
   * buffer */
  gst_rtsp_message_steal_body (&message, &data, &size);
  gst_rtsp_message_unset (&message);

  gst_buffer_unmap (buffer, &map);
  gst_buffer_unref (buffer);

  return res;
}

static GstFlowReturn
gst_rtspsrc_push_backchannel_buffer (GstRTSPSrc * src, guint id,
    GstSample * sample)
{
  GstFlowReturn res = GST_FLOW_OK;
  GstRTSPStream *stream;

  if (!src->conninfo.connected || src->state != GST_RTSP_STATE_PLAYING)
    goto out;

  stream = find_stream (src, &id, (gpointer) find_stream_by_id);
  if (stream == NULL) {
    GST_ERROR_OBJECT (src, "no stream with id %u", id);
    goto out;
  }

  if (src->interleaved) {
    GstBuffer *buffer;
    GstMapInfo map;
    guint8 *data;
    guint size;
    GstRTSPResult ret;
    GstRTSPMessage message = { 0 };
    GstRTSPConnInfo *conninfo;

    buffer = gst_sample_get_buffer (sample);

    gst_buffer_map (buffer, &map, GST_MAP_READ);
    size = map.size;
    data = map.data;

    gst_rtsp_message_init_data (&message, stream->channel[0]);

    /* lend the body data to the message */
    gst_rtsp_message_take_body (&message, data, size);

    if (stream->conninfo.connection)
      conninfo = &stream->conninfo;
    else
      conninfo = &src->conninfo;

    GST_DEBUG_OBJECT (src, "sending %u bytes backchannel RTP", size);
    ret = gst_rtspsrc_connection_send (src, conninfo, &message, NULL);
    GST_DEBUG_OBJECT (src, "sent backchannel RTP, %d", ret);

    /* and steal it away again because we will free it when unreffing the
     * buffer */
    gst_rtsp_message_steal_body (&message, &data, &size);
    gst_rtsp_message_unset (&message);

    gst_buffer_unmap (buffer, &map);

    res = GST_FLOW_OK;
  } else {
    g_signal_emit_by_name (stream->rtpsrc, "push-sample", sample, &res);
    GST_DEBUG_OBJECT (src, "sent backchannel RTP sample %p: %s", sample,
        gst_flow_get_name (res));
  }

out:
  gst_sample_unref (sample);

  return res;
}

static GstPadProbeReturn
pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
  GstRTSPSrc *src = user_data;

  GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
      GST_DEBUG_PAD_NAME (pad));

  /* activate the streams */
  GST_OBJECT_LOCK (src);
  if (!src->need_activate)
    goto was_ok;

  src->need_activate = FALSE;
  GST_OBJECT_UNLOCK (src);

  gst_rtspsrc_activate_streams (src);

  return GST_PAD_PROBE_OK;

was_ok:
  {
    GST_OBJECT_UNLOCK (src);
    return GST_PAD_PROBE_OK;
  }
}

static gboolean
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
{
  GstPad *gpad = GST_PAD_CAST (user_data);

  GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
  gst_pad_store_sticky_event (gpad, *event);

  return TRUE;
}

static gboolean
add_backchannel_fakesink (GstRTSPSrc * src, GstRTSPStream * stream,
    GstPad * srcpad)
{
  GstPad *sinkpad;
  GstElement *fakesink;

  fakesink = gst_element_factory_make ("fakesink", NULL);
  if (fakesink == NULL) {
    GST_ERROR_OBJECT (src, "no fakesink");
    return FALSE;
  }

  sinkpad = gst_element_get_static_pad (fakesink, "sink");

  GST_DEBUG_OBJECT (src, "backchannel stream %p, hooking fakesink", stream);

  gst_bin_add (GST_BIN_CAST (src), fakesink);
  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
    GST_WARNING_OBJECT (src, "could not link to fakesink");
    return FALSE;
  }

  gst_object_unref (sinkpad);

  gst_element_sync_state_with_parent (fakesink);
  return TRUE;
}

/* this callback is called when the session manager generated a new src pad with
 * payloaded RTP packets. We simply ghost the pad here. */
static void
new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
{
  gchar *name;
  GstPadTemplate *template;
  gint id, ssrc, pt;
  GList *ostreams;
  GstRTSPStream *stream;
  gboolean all_added;
  GstPad *internal_src;

  GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);

  GST_RTSP_STATE_LOCK (src);
  /* find stream */
  name = gst_object_get_name (GST_OBJECT_CAST (pad));
  if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
    goto unknown_stream;

  GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);

  stream = find_stream (src, &id, (gpointer) find_stream_by_id);
  if (stream == NULL)
    goto unknown_stream;

  /* save SSRC */
  stream->ssrc = ssrc;

  /* we'll add it later see below */
  stream->added = TRUE;

  /* check if we added all streams */
  all_added = TRUE;
  for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
    GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;

    GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
        ostream, ostream->container, ostream->added, ostream->setup);

    /* if we find a stream for which we did a setup that is not added, we
     * need to wait some more */
    if (ostream->setup && !ostream->added) {
      all_added = FALSE;
      break;
    }
  }
  GST_RTSP_STATE_UNLOCK (src);

  /* create a new pad we will use to stream to */
  template = gst_static_pad_template_get (&rtptemplate);
  stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
  gst_object_unref (template);
  g_free (name);

  /* We intercept and modify the stream start event */
  internal_src =
      GST_PAD (gst_proxy_pad_get_internal (GST_PROXY_PAD (stream->srcpad)));
  gst_pad_set_element_private (internal_src, stream);
  gst_pad_set_event_function (internal_src, gst_rtspsrc_handle_src_sink_event);
  gst_object_unref (internal_src);

  gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
  gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
  gst_pad_set_active (stream->srcpad, TRUE);
  gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);

  /* don't add the srcpad if this is a sendonly stream */
  if (stream->is_backchannel)
    add_backchannel_fakesink (src, stream, stream->srcpad);
  else
    gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);

  if (all_added) {
    GST_DEBUG_OBJECT (src, "We added all streams");
    /* when we get here, all stream are added and we can fire the no-more-pads
     * signal. */
    gst_element_no_more_pads (GST_ELEMENT_CAST (src));
  }

  return;

  /* ERRORS */
unknown_stream:
  {
    GST_DEBUG_OBJECT (src, "ignoring unknown stream");
    GST_RTSP_STATE_UNLOCK (src);
    g_free (name);
    return;
  }
}

static GstCaps *
stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
{
  guint i, len;

  len = stream->ptmap->len;
  for (i = 0; i < len; i++) {
    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
    if (item->pt == pt)
      return item->caps;
  }
  return NULL;
}

static GstCaps *
request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
{
  GstRTSPStream *stream;
  GstCaps *caps;

  GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);

  GST_RTSP_STATE_LOCK (src);
  stream = find_stream (src, &session, (gpointer) find_stream_by_id);
  if (!stream)
    goto unknown_stream;

  if ((caps = stream_get_caps_for_pt (stream, pt)))
    gst_caps_ref (caps);
  GST_RTSP_STATE_UNLOCK (src);

  return caps;

unknown_stream:
  {
    GST_DEBUG_OBJECT (src, "unknown stream %d", session);
    GST_RTSP_STATE_UNLOCK (src);
    return NULL;
  }
}

static void
gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
{
  GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);

  if (stream->eos)
    goto was_eos;

  stream->eos = TRUE;
  gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
  return;

  /* ERRORS */
was_eos:
  {
    GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
    return;
  }
}

static void
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
  GstRTSPSrc *src = stream->parent;
  guint ssrc;

  g_object_get (source, "ssrc", &ssrc, NULL);

  GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
      ssrc, stream->ssrc, stream->id);

  if (ssrc == stream->ssrc)
    gst_rtspsrc_do_stream_eos (src, stream);
}

static void
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
  GstRTSPSrc *src = stream->parent;
  guint ssrc;

  g_object_get (source, "ssrc", &ssrc, NULL);

  GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
      ssrc, stream->ssrc, stream->id);

  if (ssrc == stream->ssrc)
    gst_rtspsrc_do_stream_eos (src, stream);
}

static void
on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
{
  GstRTSPStream *stream;

  GST_DEBUG_OBJECT (src, "source in session %u reached NPT st