rtpsession: Don't relay more than one PLI request per RTT

[platform/upstream/gstreamer.git] / gst / rtpmanager / rtpsession.c

/* GStreamer
 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

#include <string.h>

#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/netbuffer/gstnetbuffer.h>

#include "gstrtpbin-marshal.h"
#include "rtpsession.h"

GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
#define GST_CAT_DEFAULT rtp_session_debug

/* signals and args */
enum
{
  SIGNAL_GET_SOURCE_BY_SSRC,
  SIGNAL_ON_NEW_SSRC,
  SIGNAL_ON_SSRC_COLLISION,
  SIGNAL_ON_SSRC_VALIDATED,
  SIGNAL_ON_SSRC_ACTIVE,
  SIGNAL_ON_SSRC_SDES,
  SIGNAL_ON_BYE_SSRC,
  SIGNAL_ON_BYE_TIMEOUT,
  SIGNAL_ON_TIMEOUT,
  SIGNAL_ON_SENDER_TIMEOUT,
  SIGNAL_ON_SENDING_RTCP,
  SIGNAL_ON_FEEDBACK_RTCP,
  LAST_SIGNAL
};

#define DEFAULT_INTERNAL_SOURCE      NULL
#define DEFAULT_BANDWIDTH            RTP_STATS_BANDWIDTH
#define DEFAULT_RTCP_FRACTION        (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
#define DEFAULT_RTCP_RR_BANDWIDTH    -1
#define DEFAULT_RTCP_RS_BANDWIDTH    -1
#define DEFAULT_RTCP_MTU             1400
#define DEFAULT_SDES                 NULL
#define DEFAULT_NUM_SOURCES          0
#define DEFAULT_NUM_ACTIVE_SOURCES   0
#define DEFAULT_SOURCES              NULL
#define DEFAULT_RTCP_MIN_INTERVAL    (RTP_STATS_MIN_INTERVAL * GST_SECOND)
#define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)

enum
{
  PROP_0,
  PROP_INTERNAL_SSRC,
  PROP_INTERNAL_SOURCE,
  PROP_BANDWIDTH,
  PROP_RTCP_FRACTION,
  PROP_RTCP_RR_BANDWIDTH,
  PROP_RTCP_RS_BANDWIDTH,
  PROP_RTCP_MTU,
  PROP_SDES,
  PROP_NUM_SOURCES,
  PROP_NUM_ACTIVE_SOURCES,
  PROP_SOURCES,
  PROP_FAVOR_NEW,
  PROP_RTCP_MIN_INTERVAL,
  PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
  PROP_LAST
};

/* update average packet size */
#define INIT_AVG(avg, val) \
   (avg) = (val);
#define UPDATE_AVG(avg, val)            \
  if ((avg) == 0)                       \
   (avg) = (val);                       \
  else                                  \
   (avg) = ((val) + (15 * (avg))) >> 4;


/* The number RTCP intervals after which to timeout entries in the
 * collision table
 */
#define RTCP_INTERVAL_COLLISION_TIMEOUT 10

/* GObject vmethods */
static void rtp_session_finalize (GObject * object);
static void rtp_session_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void rtp_session_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);

static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
    GstBuffer * buffer, gboolean early);


static guint rtp_session_signals[LAST_SIGNAL] = { 0 };

G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);

static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
    gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
    const gchar * reason, GstClockTime current_time);
static GstClockTime calculate_rtcp_interval (RTPSession * sess,
    gboolean deterministic, gboolean first);

static gboolean
accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
    const GValue * handler_return, gpointer data)
{
  if (g_value_get_boolean (handler_return))
    g_value_set_boolean (return_accu, TRUE);

  return TRUE;
}

static void
rtp_session_class_init (RTPSessionClass * klass)
{
  GObjectClass *gobject_class;

  gobject_class = (GObjectClass *) klass;

  gobject_class->finalize = rtp_session_finalize;
  gobject_class->set_property = rtp_session_set_property;
  gobject_class->get_property = rtp_session_get_property;

  /**
   * RTPSession::get-source-by-ssrc:
   * @session: the object which received the signal
   * @ssrc: the SSRC of the RTPSource
   *
   * Request the #RTPSource object with SSRC @ssrc in @session.
   */
  rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
      g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
          get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
      RTP_TYPE_SOURCE, 1, G_TYPE_UINT);

  /**
   * RTPSession::on-new-ssrc:
   * @session: the object which received the signal
   * @src: the new RTPSource
   *
   * Notify of a new SSRC that entered @session.
   */
  rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
      g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-ssrc-collision:
   * @session: the object which received the signal
   * @src: the #RTPSource that caused a collision
   *
   * Notify when we have an SSRC collision
   */
  rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
      g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-ssrc-validated:
   * @session: the object which received the signal
   * @src: the new validated RTPSource
   *
   * Notify of a new SSRC that became validated.
   */
  rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
      g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-ssrc-active:
   * @session: the object which received the signal
   * @src: the active RTPSource
   *
   * Notify of a SSRC that is active, i.e., sending RTCP.
   */
  rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
      g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-ssrc-sdes:
   * @session: the object which received the signal
   * @src: the RTPSource
   *
   * Notify that a new SDES was received for SSRC.
   */
  rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
      g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-bye-ssrc:
   * @session: the object which received the signal
   * @src: the RTPSource that went away
   *
   * Notify of an SSRC that became inactive because of a BYE packet.
   */
  rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
      g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-bye-timeout:
   * @session: the object which received the signal
   * @src: the RTPSource that timed out
   *
   * Notify of an SSRC that has timed out because of BYE
   */
  rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
      g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-timeout:
   * @session: the object which received the signal
   * @src: the RTPSource that timed out
   *
   * Notify of an SSRC that has timed out
   */
  rtp_session_signals[SIGNAL_ON_TIMEOUT] =
      g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);
  /**
   * RTPSession::on-sender-timeout:
   * @session: the object which received the signal
   * @src: the RTPSource that timed out
   *
   * Notify of an SSRC that was a sender but timed out and became a receiver.
   */
  rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
      g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
      RTP_TYPE_SOURCE);

  /**
   * RTPSession::on-sending-rtcp
   * @session: the object which received the signal
   * @buffer: the #GstBuffer containing the RTCP packet about to be sent
   * @early: %TRUE if the packet is early, %FALSE if it is regular
   *
   * This signal is emitted before sending an RTCP packet, it can be used
   * to add extra RTCP Packets.
   *
   * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
   * if suppressing it is acceptable
   */
  rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
      g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
      accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__POINTER_BOOLEAN,
      G_TYPE_BOOLEAN, 2, G_TYPE_POINTER, G_TYPE_BOOLEAN);

  /**
   * RTPSession::on-feedback-rtcp:
   * @session: the object which received the signal
   * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
   *  %GST_RTCP_TYPE_RTPFB
   * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
   * @sender_ssrc: The SSRC of the sender
   * @media_ssrc: The SSRC of the media this refers to
   * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
   * there was no FCI
   *
   * Notify that a RTCP feedback packet has been received
   */

  rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
      g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_POINTER,
      G_TYPE_NONE, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
      G_TYPE_POINTER);

  g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
      g_param_spec_uint ("internal-ssrc", "Internal SSRC",
          "The internal SSRC used for the session",
          0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
      g_param_spec_object ("internal-source", "Internal Source",
          "The internal source element of the session",
          RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
      g_param_spec_double ("bandwidth", "Bandwidth",
          "The bandwidth of the session (0 for auto-discover)",
          0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
      g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
          "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
          0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
      g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
          "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
          -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
      g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
          "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
          -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
      g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
          "The maximum size of the RTCP packets",
          16, G_MAXINT16, DEFAULT_RTCP_MTU,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  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));

  g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
      g_param_spec_uint ("num-sources", "Num Sources",
          "The number of sources in the session", 0, G_MAXUINT,
          DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
      g_param_spec_uint ("num-active-sources", "Num Active Sources",
          "The number of active sources in the session", 0, G_MAXUINT,
          DEFAULT_NUM_ACTIVE_SOURCES,
          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
  /**
   * RTPSource::sources
   *
   * Get a GValue Array of all sources in the session.
   *
   * <example>
   * <title>Getting the #RTPSources of a session
   * <programlisting>
   * {
   *   GValueArray *arr;
   *   GValue *val;
   *   guint i;
   *
   *   g_object_get (sess, "sources", &arr, NULL);
   *
   *   for (i = 0; i < arr->n_values; i++) {
   *     RTPSource *source;
   *
   *     val = g_value_array_get_nth (arr, i);
   *     source = g_value_get_object (val);
   *   }
   *   g_value_array_free (arr);
   * }
   * </programlisting>
   * </example>
   */
  g_object_class_install_property (gobject_class, PROP_SOURCES,
      g_param_spec_boxed ("sources", "Sources",
          "An array of all known sources in the session",
          G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
      g_param_spec_boolean ("favor-new", "Favor new sources",
          "Resolve SSRC conflict in favor of new sources", FALSE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
      g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
          "Minimum interval between Regular RTCP packet (in ns)",
          0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class,
      PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
      g_param_spec_uint64 ("rtcp-feedback-retention-window",
          "RTCP Feedback retention window",
          "Duration during which RTCP Feedback packets are retained (in ns)",
          0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));


  klass->get_source_by_ssrc =
      GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
  klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);

  GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
}

static void
rtp_session_init (RTPSession * sess)
{
  gint i;
  gchar *str;

  sess->lock = g_mutex_new ();
  sess->key = g_random_int ();
  sess->mask_idx = 0;
  sess->mask = 0;

  for (i = 0; i < 32; i++) {
    sess->ssrcs[i] =
        g_hash_table_new_full (NULL, NULL, NULL,
        (GDestroyNotify) g_object_unref);
  }
  sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);

  rtp_stats_init_defaults (&sess->stats);

  sess->recalc_bandwidth = TRUE;
  sess->bandwidth = DEFAULT_BANDWIDTH;
  sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
  sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
  sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;

  /* create an active SSRC for this session manager */
  sess->source = rtp_session_create_source (sess);
  sess->source->validated = TRUE;
  sess->source->internal = TRUE;
  sess->stats.active_sources++;
  INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);

  /* default UDP header length */
  sess->header_len = 28;
  sess->mtu = DEFAULT_RTCP_MTU;

  /* some default SDES entries */
  str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
  rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
  g_free (str);

  rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
      g_get_real_name ());
  rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");

  sess->first_rtcp = TRUE;
  sess->allow_early = TRUE;
  sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;

  sess->rtcp_pli_requests = g_array_new (FALSE, FALSE, sizeof (guint32));

  GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
}

static void
rtp_session_finalize (GObject * object)
{
  RTPSession *sess;
  gint i;

  sess = RTP_SESSION_CAST (object);

  g_mutex_free (sess->lock);
  for (i = 0; i < 32; i++)
    g_hash_table_destroy (sess->ssrcs[i]);

  g_free (sess->bye_reason);

  g_hash_table_destroy (sess->cnames);
  g_object_unref (sess->source);

  g_array_free (sess->rtcp_pli_requests, TRUE);

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

static void
copy_source (gpointer key, RTPSource * source, GValueArray * arr)
{
  GValue value = { 0 };

  g_value_init (&value, RTP_TYPE_SOURCE);
  g_value_take_object (&value, source);
  /* copies the value */
  g_value_array_append (arr, &value);
}

static GValueArray *
rtp_session_create_sources (RTPSession * sess)
{
  GValueArray *res;
  guint size;

  RTP_SESSION_LOCK (sess);
  /* get number of elements in the table */
  size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
  /* create the result value array */
  res = g_value_array_new (size);

  /* and copy all values into the array */
  g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
  RTP_SESSION_UNLOCK (sess);

  return res;
}

static void
rtp_session_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  RTPSession *sess;

  sess = RTP_SESSION (object);

  switch (prop_id) {
    case PROP_INTERNAL_SSRC:
      rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
      break;
    case PROP_BANDWIDTH:
      sess->bandwidth = g_value_get_double (value);
      sess->recalc_bandwidth = TRUE;
      break;
    case PROP_RTCP_FRACTION:
      sess->rtcp_bandwidth = g_value_get_double (value);
      sess->recalc_bandwidth = TRUE;
      break;
    case PROP_RTCP_RR_BANDWIDTH:
      sess->rtcp_rr_bandwidth = g_value_get_int (value);
      sess->recalc_bandwidth = TRUE;
      break;
    case PROP_RTCP_RS_BANDWIDTH:
      sess->rtcp_rs_bandwidth = g_value_get_int (value);
      sess->recalc_bandwidth = TRUE;
      break;
    case PROP_RTCP_MTU:
      sess->mtu = g_value_get_uint (value);
      break;
    case PROP_SDES:
      rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
      break;
    case PROP_FAVOR_NEW:
      sess->favor_new = g_value_get_boolean (value);
      break;
    case PROP_RTCP_MIN_INTERVAL:
      rtp_stats_set_min_interval (&sess->stats,
          (gdouble) g_value_get_uint64 (value) / GST_SECOND);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
rtp_session_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  RTPSession *sess;

  sess = RTP_SESSION (object);

  switch (prop_id) {
    case PROP_INTERNAL_SSRC:
      g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
      break;
    case PROP_INTERNAL_SOURCE:
      g_value_take_object (value, rtp_session_get_internal_source (sess));
      break;
    case PROP_BANDWIDTH:
      g_value_set_double (value, sess->bandwidth);
      break;
    case PROP_RTCP_FRACTION:
      g_value_set_double (value, sess->rtcp_bandwidth);
      break;
    case PROP_RTCP_RR_BANDWIDTH:
      g_value_set_int (value, sess->rtcp_rr_bandwidth);
      break;
    case PROP_RTCP_RS_BANDWIDTH:
      g_value_set_int (value, sess->rtcp_rs_bandwidth);
      break;
    case PROP_RTCP_MTU:
      g_value_set_uint (value, sess->mtu);
      break;
    case PROP_SDES:
      g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
      break;
    case PROP_NUM_SOURCES:
      g_value_set_uint (value, rtp_session_get_num_sources (sess));
      break;
    case PROP_NUM_ACTIVE_SOURCES:
      g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
      break;
    case PROP_SOURCES:
      g_value_take_boxed (value, rtp_session_create_sources (sess));
      break;
    case PROP_FAVOR_NEW:
      g_value_set_boolean (value, sess->favor_new);
      break;
    case PROP_RTCP_MIN_INTERVAL:
      g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
on_new_ssrc (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_ssrc_collision (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
      source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_ssrc_validated (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
      source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_ssrc_active (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_ssrc_sdes (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_bye_ssrc (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_bye_timeout (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_timeout (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

static void
on_sender_timeout (RTPSession * sess, RTPSource * source)
{
  g_object_ref (source);
  RTP_SESSION_UNLOCK (sess);
  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
      source);
  RTP_SESSION_LOCK (sess);
  g_object_unref (source);
}

/**
 * rtp_session_new:
 *
 * Create a new session object.
 *
 * Returns: a new #RTPSession. g_object_unref() after usage.
 */
RTPSession *
rtp_session_new (void)
{
  RTPSession *sess;

  sess = g_object_new (RTP_TYPE_SESSION, NULL);

  return sess;
}

/**
 * rtp_session_set_callbacks:
 * @sess: an #RTPSession
 * @callbacks: callbacks to configure
 * @user_data: user data passed in the callbacks
 *
 * Configure a set of callbacks to be notified of actions.
 */
void
rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
    gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  if (callbacks->process_rtp) {
    sess->callbacks.process_rtp = callbacks->process_rtp;
    sess->process_rtp_user_data = user_data;
  }
  if (callbacks->send_rtp) {
    sess->callbacks.send_rtp = callbacks->send_rtp;
    sess->send_rtp_user_data = user_data;
  }
  if (callbacks->send_rtcp) {
    sess->callbacks.send_rtcp = callbacks->send_rtcp;
    sess->send_rtcp_user_data = user_data;
  }
  if (callbacks->sync_rtcp) {
    sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
    sess->sync_rtcp_user_data = user_data;
  }
  if (callbacks->clock_rate) {
    sess->callbacks.clock_rate = callbacks->clock_rate;
    sess->clock_rate_user_data = user_data;
  }
  if (callbacks->reconsider) {
    sess->callbacks.reconsider = callbacks->reconsider;
    sess->reconsider_user_data = user_data;
  }
  if (callbacks->request_key_unit) {
    sess->callbacks.request_key_unit = callbacks->request_key_unit;
    sess->request_key_unit_user_data = user_data;
  }
}

/**
 * rtp_session_set_process_rtp_callback:
 * @sess: an #RTPSession
 * @callback: callback to set
 * @user_data: user data passed in the callback
 *
 * Configure only the process_rtp callback to be notified of the process_rtp action.
 */
void
rtp_session_set_process_rtp_callback (RTPSession * sess,
    RTPSessionProcessRTP callback, gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  sess->callbacks.process_rtp = callback;
  sess->process_rtp_user_data = user_data;
}

/**
 * rtp_session_set_send_rtp_callback:
 * @sess: an #RTPSession
 * @callback: callback to set
 * @user_data: user data passed in the callback
 *
 * Configure only the send_rtp callback to be notified of the send_rtp action.
 */
void
rtp_session_set_send_rtp_callback (RTPSession * sess,
    RTPSessionSendRTP callback, gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  sess->callbacks.send_rtp = callback;
  sess->send_rtp_user_data = user_data;
}

/**
 * rtp_session_set_send_rtcp_callback:
 * @sess: an #RTPSession
 * @callback: callback to set
 * @user_data: user data passed in the callback
 *
 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
 */
void
rtp_session_set_send_rtcp_callback (RTPSession * sess,
    RTPSessionSendRTCP callback, gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  sess->callbacks.send_rtcp = callback;
  sess->send_rtcp_user_data = user_data;
}

/**
 * rtp_session_set_sync_rtcp_callback:
 * @sess: an #RTPSession
 * @callback: callback to set
 * @user_data: user data passed in the callback
 *
 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
 */
void
rtp_session_set_sync_rtcp_callback (RTPSession * sess,
    RTPSessionSyncRTCP callback, gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  sess->callbacks.sync_rtcp = callback;
  sess->sync_rtcp_user_data = user_data;
}

/**
 * rtp_session_set_clock_rate_callback:
 * @sess: an #RTPSession
 * @callback: callback to set
 * @user_data: user data passed in the callback
 *
 * Configure only the clock_rate callback to be notified of the clock_rate action.
 */
void
rtp_session_set_clock_rate_callback (RTPSession * sess,
    RTPSessionClockRate callback, gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  sess->callbacks.clock_rate = callback;
  sess->clock_rate_user_data = user_data;
}

/**
 * rtp_session_set_reconsider_callback:
 * @sess: an #RTPSession
 * @callback: callback to set
 * @user_data: user data passed in the callback
 *
 * Configure only the reconsider callback to be notified of the reconsider action.
 */
void
rtp_session_set_reconsider_callback (RTPSession * sess,
    RTPSessionReconsider callback, gpointer user_data)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  sess->callbacks.reconsider = callback;
  sess->reconsider_user_data = user_data;
}

/**
 * rtp_session_set_bandwidth:
 * @sess: an #RTPSession
 * @bandwidth: the bandwidth allocated
 *
 * Set the session bandwidth in bytes per second.
 */
void
rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  RTP_SESSION_LOCK (sess);
  sess->stats.bandwidth = bandwidth;
  RTP_SESSION_UNLOCK (sess);
}

/**
 * rtp_session_get_bandwidth:
 * @sess: an #RTPSession
 *
 * Get the session bandwidth.
 *
 * Returns: the session bandwidth.
 */
gdouble
rtp_session_get_bandwidth (RTPSession * sess)
{
  gdouble result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), 0);

  RTP_SESSION_LOCK (sess);
  result = sess->stats.bandwidth;
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_set_rtcp_fraction:
 * @sess: an #RTPSession
 * @bandwidth: the RTCP bandwidth
 *
 * Set the bandwidth in bytes per second that should be used for RTCP
 * messages.
 */
void
rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
{
  g_return_if_fail (RTP_IS_SESSION (sess));

  RTP_SESSION_LOCK (sess);
  sess->stats.rtcp_bandwidth = bandwidth;
  RTP_SESSION_UNLOCK (sess);
}

/**
 * rtp_session_get_rtcp_fraction:
 * @sess: an #RTPSession
 *
 * Get the session bandwidth used for RTCP.
 *
 * Returns: The bandwidth used for RTCP messages.
 */
gdouble
rtp_session_get_rtcp_fraction (RTPSession * sess)
{
  gdouble result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);

  RTP_SESSION_LOCK (sess);
  result = sess->stats.rtcp_bandwidth;
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_set_sdes_string:
 * @sess: an #RTPSession
 * @type: the type of the SDES item
 * @item: a null-terminated string to set.
 *
 * Store an SDES item of @type in @sess.
 *
 * Returns: %FALSE if the data was unchanged @type is invalid.
 */
gboolean
rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
    const gchar * item)
{
  gboolean result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);

  RTP_SESSION_LOCK (sess);
  result = rtp_source_set_sdes_string (sess->source, type, item);
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_get_sdes_string:
 * @sess: an #RTPSession
 * @type: the type of the SDES item
 *
 * Get the SDES item of @type from @sess.
 *
 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
 * valid. g_free() after usage.
 */
gchar *
rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
{
  gchar *result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);

  RTP_SESSION_LOCK (sess);
  result = rtp_source_get_sdes_string (sess->source, type);
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_get_sdes_struct:
 * @sess: an #RTSPSession
 *
 * Get the SDES data as a #GstStructure
 *
 * Returns: a GstStructure with SDES items for @sess. This function returns a
 * copy of the SDES structure, use gst_structure_free() after usage.
 */
GstStructure *
rtp_session_get_sdes_struct (RTPSession * sess)
{
  const GstStructure *sdes;
  GstStructure *result = NULL;

  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);

  RTP_SESSION_LOCK (sess);
  sdes = rtp_source_get_sdes_struct (sess->source);
  if (sdes)
    result = gst_structure_copy (sdes);
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_set_sdes_struct:
 * @sess: an #RTSPSession
 * @sdes: a #GstStructure
 *
 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
 */
void
rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
{
  g_return_if_fail (sdes);
  g_return_if_fail (RTP_IS_SESSION (sess));

  RTP_SESSION_LOCK (sess);
  rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
  RTP_SESSION_UNLOCK (sess);
}

static GstFlowReturn
source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
{
  GstFlowReturn result = GST_FLOW_OK;

  if (source == session->source) {
    GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);

    RTP_SESSION_UNLOCK (session);

    if (session->callbacks.send_rtp)
      result =
          session->callbacks.send_rtp (session, source, data,
          session->send_rtp_user_data);
    else {
      gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
    }
  } else {
    GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
    RTP_SESSION_UNLOCK (session);

    if (session->callbacks.process_rtp)
      result =
          session->callbacks.process_rtp (session, source,
          GST_BUFFER_CAST (data), session->process_rtp_user_data);
    else
      gst_buffer_unref (GST_BUFFER_CAST (data));
  }
  RTP_SESSION_LOCK (session);

  return result;
}

static gint
source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
{
  gint result;

  RTP_SESSION_UNLOCK (session);

  if (session->callbacks.clock_rate)
    result =
        session->callbacks.clock_rate (session, pt,
        session->clock_rate_user_data);
  else
    result = -1;

  RTP_SESSION_LOCK (session);

  GST_DEBUG ("got clock-rate %d for pt %d", result, pt);

  return result;
}

static RTPSourceCallbacks callbacks = {
  (RTPSourcePushRTP) source_push_rtp,
  (RTPSourceClockRate) source_clock_rate,
};

static gboolean
check_collision (RTPSession * sess, RTPSource * source,
    RTPArrivalStats * arrival, gboolean rtp)
{
  /* If we have no arrival address, we can't do collision checking */
  if (!arrival->have_address)
    return FALSE;

  if (sess->source != source) {
    GstNetAddress *from;
    gboolean have_from;

    /* This is not our local source, but lets check if two remote
     * source collide
     */

    if (rtp) {
      from = &source->rtp_from;
      have_from = source->have_rtp_from;
    } else {
      from = &source->rtcp_from;
      have_from = source->have_rtcp_from;
    }

    if (have_from) {
      if (gst_netaddress_equal (from, &arrival->address)) {
        /* Address is the same */
        return FALSE;
      } else {
        GST_LOG ("we have a third-party collision or loop ssrc:%x",
            rtp_source_get_ssrc (source));
        if (sess->favor_new) {
          if (rtp_source_find_conflicting_address (source,
                  &arrival->address, arrival->current_time)) {
            gchar buf1[40];
            gst_netaddress_to_string (&arrival->address, buf1, 40);
            GST_LOG ("Known conflict on %x for %s, dropping packet",
                rtp_source_get_ssrc (source), buf1);
            return TRUE;
          } else {
            gchar buf1[40], buf2[40];

            /* Current address is not a known conflict, lets assume this is
             * a new source. Save old address in possible conflict list
             */
            rtp_source_add_conflicting_address (source, from,
                arrival->current_time);

            gst_netaddress_to_string (from, buf1, 40);
            gst_netaddress_to_string (&arrival->address, buf2, 40);
            GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
                " saving old as known conflict",
                rtp_source_get_ssrc (source), buf1, buf2);

            if (rtp)
              rtp_source_set_rtp_from (source, &arrival->address);
            else
              rtp_source_set_rtcp_from (source, &arrival->address);
            return FALSE;
          }
        } else {
          /* Don't need to save old addresses, we ignore new sources */
          return TRUE;
        }
      }
    } else {
      /* We don't already have a from address for RTP, just set it */
      if (rtp)
        rtp_source_set_rtp_from (source, &arrival->address);
      else
        rtp_source_set_rtcp_from (source, &arrival->address);
      return FALSE;
    }

    /* FIXME: Log 3rd party collision somehow
     * Maybe should be done in upper layer, only the SDES can tell us
     * if its a collision or a loop
     */

    /* If the source has been inactive for some time, we assume that it has
     * simply changed its transport source address. Hence, there is no true
     * third-party collision - only a simulated one. */
    if (arrival->current_time > source->last_activity) {
      GstClockTime inactivity_period =
          arrival->current_time - source->last_activity;
      if (inactivity_period > 1 * GST_SECOND) {
        /* Use new network address */
        if (rtp) {
          g_assert (source->have_rtp_from);
          rtp_source_set_rtp_from (source, &arrival->address);
        } else {
          g_assert (source->have_rtcp_from);
          rtp_source_set_rtcp_from (source, &arrival->address);
        }
        return FALSE;
      }
    }
  } else {
    /* This is sending with our ssrc, is it an address we already know */

    if (rtp_source_find_conflicting_address (source, &arrival->address,
            arrival->current_time)) {
      /* Its a known conflict, its probably a loop, not a collision
       * lets just drop the incoming packet
       */
      GST_DEBUG ("Our packets are being looped back to us, dropping");
    } else {
      /* Its a new collision, lets change our SSRC */

      rtp_source_add_conflicting_address (source, &arrival->address,
          arrival->current_time);

      GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
      on_ssrc_collision (sess, source);

      rtp_session_schedule_bye_locked (sess, "SSRC Collision",
          arrival->current_time);

      sess->change_ssrc = TRUE;
    }
  }

  return TRUE;
}


/* must be called with the session lock, the returned source needs to be
 * unreffed after usage. */
static RTPSource *
obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
    RTPArrivalStats * arrival, gboolean rtp)
{
  RTPSource *source;

  source =
      g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
  if (source == NULL) {
    /* make new Source in probation and insert */
    source = rtp_source_new (ssrc);

    /* for RTP packets we need to set the source in probation. Receiving RTCP
     * packets of an SSRC, on the other hand, is a strong indication that we
     * are dealing with a valid source. */
    if (rtp)
      source->probation = RTP_DEFAULT_PROBATION;
    else
      source->probation = 0;

    /* store from address, if any */
    if (arrival->have_address) {
      if (rtp)
        rtp_source_set_rtp_from (source, &arrival->address);
      else
        rtp_source_set_rtcp_from (source, &arrival->address);
    }

    /* configure a callback on the source */
    rtp_source_set_callbacks (source, &callbacks, sess);

    g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
        source);

    /* we have one more source now */
    sess->total_sources++;
    *created = TRUE;
  } else {
    *created = FALSE;
    /* check for collision, this updates the address when not previously set */
    if (check_collision (sess, source, arrival, rtp)) {
      return NULL;
    }
  }
  /* update last activity */
  source->last_activity = arrival->current_time;
  if (rtp)
    source->last_rtp_activity = arrival->current_time;
  g_object_ref (source);

  return source;
}

/**
 * rtp_session_get_internal_source:
 * @sess: a #RTPSession
 *
 * Get the internal #RTPSource of @sess.
 *
 * Returns: The internal #RTPSource. g_object_unref() after usage.
 */
RTPSource *
rtp_session_get_internal_source (RTPSession * sess)
{
  RTPSource *result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);

  result = g_object_ref (sess->source);

  return result;
}

/**
 * rtp_session_set_internal_ssrc:
 * @sess: a #RTPSession
 * @ssrc: an SSRC
 *
 * Set the SSRC of @sess to @ssrc.
 */
void
rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
{
  RTP_SESSION_LOCK (sess);
  if (ssrc != sess->source->ssrc) {
    g_hash_table_steal (sess->ssrcs[sess->mask_idx],
        GINT_TO_POINTER (sess->source->ssrc));

    GST_DEBUG ("setting internal SSRC to %08x", ssrc);
    /* After this call, any receiver of the old SSRC either in RTP or RTCP
     * packets will timeout on the old SSRC, we could potentially schedule a
     * BYE RTCP for the old SSRC... */
    sess->source->ssrc = ssrc;
    rtp_source_reset (sess->source);

    /* rehash with the new SSRC */
    g_hash_table_insert (sess->ssrcs[sess->mask_idx],
        GINT_TO_POINTER (sess->source->ssrc), sess->source);
  }
  RTP_SESSION_UNLOCK (sess);

  g_object_notify (G_OBJECT (sess), "internal-ssrc");
}

/**
 * rtp_session_get_internal_ssrc:
 * @sess: a #RTPSession
 *
 * Get the internal SSRC of @sess.
 *
 * Returns: The SSRC of the session.
 */
guint32
rtp_session_get_internal_ssrc (RTPSession * sess)
{
  guint32 ssrc;

  RTP_SESSION_LOCK (sess);
  ssrc = sess->source->ssrc;
  RTP_SESSION_UNLOCK (sess);

  return ssrc;
}

/**
 * rtp_session_add_source:
 * @sess: a #RTPSession
 * @src: #RTPSource to add
 *
 * Add @src to @session.
 *
 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
 * existed in the session.
 */
gboolean
rtp_session_add_source (RTPSession * sess, RTPSource * src)
{
  gboolean result = FALSE;
  RTPSource *find;

  g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
  g_return_val_if_fail (src != NULL, FALSE);

  RTP_SESSION_LOCK (sess);
  find =
      g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
      GINT_TO_POINTER (src->ssrc));
  if (find == NULL) {
    g_hash_table_insert (sess->ssrcs[sess->mask_idx],
        GINT_TO_POINTER (src->ssrc), src);
    /* we have one more source now */
    sess->total_sources++;
    result = TRUE;
  }
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_get_num_sources:
 * @sess: an #RTPSession
 *
 * Get the number of sources in @sess.
 *
 * Returns: The number of sources in @sess.
 */
guint
rtp_session_get_num_sources (RTPSession * sess)
{
  guint result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);

  RTP_SESSION_LOCK (sess);
  result = sess->total_sources;
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_get_num_active_sources:
 * @sess: an #RTPSession
 *
 * Get the number of active sources in @sess. A source is considered active when
 * it has been validated and has not yet received a BYE RTCP message.
 *
 * Returns: The number of active sources in @sess.
 */
guint
rtp_session_get_num_active_sources (RTPSession * sess)
{
  guint result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), 0);

  RTP_SESSION_LOCK (sess);
  result = sess->stats.active_sources;
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_get_source_by_ssrc:
 * @sess: an #RTPSession
 * @ssrc: an SSRC
 *
 * Find the source with @ssrc in @sess.
 *
 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
 * g_object_unref() after usage.
 */
RTPSource *
rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
{
  RTPSource *result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);

  RTP_SESSION_LOCK (sess);
  result =
      g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
  if (result)
    g_object_ref (result);
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_get_source_by_cname:
 * @sess: a #RTPSession
 * @cname: an CNAME
 *
 * Find the source with @cname in @sess.
 *
 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
 * g_object_unref() after usage.
 */
RTPSource *
rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
{
  RTPSource *result;

  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
  g_return_val_if_fail (cname != NULL, NULL);

  RTP_SESSION_LOCK (sess);
  result = g_hash_table_lookup (sess->cnames, cname);
  if (result)
    g_object_ref (result);
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/* should be called with the SESSION lock */
static guint32
rtp_session_create_new_ssrc (RTPSession * sess)
{
  guint32 ssrc;

  while (TRUE) {
    ssrc = g_random_int ();

    /* see if it exists in the session, we're done if it doesn't */
    if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
            GINT_TO_POINTER (ssrc)) == NULL)
      break;
  }
  return ssrc;
}


/**
 * rtp_session_create_source:
 * @sess: an #RTPSession
 *
 * Create an #RTPSource for use in @sess. This function will create a source
 * with an ssrc that is currently not used by any participants in the session.
 *
 * Returns: an #RTPSource.
 */
RTPSource *
rtp_session_create_source (RTPSession * sess)
{
  guint32 ssrc;
  RTPSource *source;

  RTP_SESSION_LOCK (sess);
  ssrc = rtp_session_create_new_ssrc (sess);
  source = rtp_source_new (ssrc);
  rtp_source_set_callbacks (source, &callbacks, sess);
  /* we need an additional ref for the source in the hashtable */
  g_object_ref (source);
  g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
      source);
  /* we have one more source now */
  sess->total_sources++;
  RTP_SESSION_UNLOCK (sess);

  return source;
}

/* update the RTPArrivalStats structure with the current time and other bits
 * about the current buffer we are handling.
 * This function is typically called when a validated packet is received.
 * This function should be called with the SESSION_LOCK
 */
static void
update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
    gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
    GstClockTime running_time)
{
  /* get time of arrival */
  arrival->current_time = current_time;
  arrival->running_time = running_time;

  /* get packet size including header overhead */
  arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;

  if (rtp) {
    arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
  } else {
    arrival->payload_len = 0;
  }

  /* for netbuffer we can store the IP address to check for collisions */
  arrival->have_address = GST_IS_NETBUFFER (buffer);
  if (arrival->have_address) {
    GstNetBuffer *netbuf = (GstNetBuffer *) buffer;

    memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
  }
}

/**
 * rtp_session_process_rtp:
 * @sess: and #RTPSession
 * @buffer: an RTP buffer
 * @current_time: the current system time
 * @running_time: the running_time of @buffer
 *
 * Process an RTP buffer in the session manager. This function takes ownership
 * of @buffer.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
    GstClockTime current_time, GstClockTime running_time)
{
  GstFlowReturn result;
  guint32 ssrc;
  RTPSource *source;
  gboolean created;
  gboolean prevsender, prevactive;
  RTPArrivalStats arrival;
  guint32 csrcs[16];
  guint8 i, count;
  guint64 oldrate;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);

  if (!gst_rtp_buffer_validate (buffer))
    goto invalid_packet;

  RTP_SESSION_LOCK (sess);
  /* update arrival stats */
  update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
      running_time);

  /* ignore more RTP packets when we left the session */
  if (sess->source->received_bye)
    goto ignore;

  /* get SSRC and look up in session database */
  ssrc = gst_rtp_buffer_get_ssrc (buffer);
  source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
  if (!source)
    goto collision;

  prevsender = RTP_SOURCE_IS_SENDER (source);
  prevactive = RTP_SOURCE_IS_ACTIVE (source);
  oldrate = source->bitrate;

  /* copy available csrc for later */
  count = gst_rtp_buffer_get_csrc_count (buffer);
  /* make sure to not overflow our array. An RTP buffer can maximally contain
   * 16 CSRCs */
  count = MIN (count, 16);

  for (i = 0; i < count; i++)
    csrcs[i] = gst_rtp_buffer_get_csrc (buffer, i);

  /* let source process the packet */
  result = rtp_source_process_rtp (source, buffer, &arrival);

  /* source became active */
  if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
    sess->stats.active_sources++;
    GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
        sess->stats.active_sources);
    on_ssrc_validated (sess, source);
  }
  if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
    sess->stats.sender_sources++;
    GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
        sess->stats.sender_sources);
  }
  if (oldrate != source->bitrate)
    sess->recalc_bandwidth = TRUE;

  if (created)
    on_new_ssrc (sess, source);

  if (source->validated) {
    gboolean created;

    /* for validated sources, we add the CSRCs as well */
    for (i = 0; i < count; i++) {
      guint32 csrc;
      RTPSource *csrc_src;

      csrc = csrcs[i];

      /* get source */
      csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
      if (!csrc_src)
        continue;

      if (created) {
        GST_DEBUG ("created new CSRC: %08x", csrc);
        rtp_source_set_as_csrc (csrc_src);
        if (RTP_SOURCE_IS_ACTIVE (csrc_src))
          sess->stats.active_sources++;
        on_new_ssrc (sess, csrc_src);
      }
      g_object_unref (csrc_src);
    }
  }
  g_object_unref (source);

  RTP_SESSION_UNLOCK (sess);

  return result;

  /* ERRORS */
invalid_packet:
  {
    gst_buffer_unref (buffer);
    GST_DEBUG ("invalid RTP packet received");
    return GST_FLOW_OK;
  }
ignore:
  {
    gst_buffer_unref (buffer);
    RTP_SESSION_UNLOCK (sess);
    GST_DEBUG ("ignoring RTP packet because we are leaving");
    return GST_FLOW_OK;
  }
collision:
  {
    gst_buffer_unref (buffer);
    RTP_SESSION_UNLOCK (sess);
    GST_DEBUG ("ignoring packet because its collisioning");
    return GST_FLOW_OK;
  }
}

static void
rtp_session_process_rb (RTPSession * sess, RTPSource * source,
    GstRTCPPacket * packet, RTPArrivalStats * arrival)
{
  guint count, i;

  count = gst_rtcp_packet_get_rb_count (packet);
  for (i = 0; i < count; i++) {
    guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
    guint8 fractionlost;
    gint32 packetslost;

    gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
        &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);

    GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);

    if (ssrc == sess->source->ssrc) {
      /* only deal with report blocks for our session, we update the stats of
       * the sender of the RTCP message. We could also compare our stats against
       * the other sender to see if we are better or worse. */
      rtp_source_process_rb (source, arrival->current_time, fractionlost,
          packetslost, exthighestseq, jitter, lsr, dlsr);
    }
  }
  on_ssrc_active (sess, source);
}

/* A Sender report contains statistics about how the sender is doing. This
 * includes timing informataion such as the relation between RTP and NTP
 * timestamps and the number of packets/bytes it sent to us.
 *
 * In this report is also included a set of report blocks related to how this
 * sender is receiving data (in case we (or somebody else) is also sending stuff
 * to it). This info includes the packet loss, jitter and seqnum. It also
 * contains information to calculate the round trip time (LSR/DLSR).
 */
static void
rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
    RTPArrivalStats * arrival, gboolean * do_sync)
{
  guint32 senderssrc, rtptime, packet_count, octet_count;
  guint64 ntptime;
  RTPSource *source;
  gboolean created, prevsender;

  gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
      &packet_count, &octet_count);

  GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
      senderssrc, GST_TIME_ARGS (arrival->current_time));

  source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
  if (!source)
    return;

  /* don't try to do lip-sync for sources that sent a BYE */
  if (rtp_source_received_bye (source))
    *do_sync = FALSE;
  else
    *do_sync = TRUE;

  prevsender = RTP_SOURCE_IS_SENDER (source);

  /* first update the source */
  rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
      packet_count, octet_count);

  if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
    sess->stats.sender_sources++;
    GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
        sess->stats.sender_sources);
  }

  if (created)
    on_new_ssrc (sess, source);

  rtp_session_process_rb (sess, source, packet, arrival);
  g_object_unref (source);
}

/* A receiver report contains statistics about how a receiver is doing. It
 * includes stuff like packet loss, jitter and the seqnum it received last. It
 * also contains info to calculate the round trip time.
 *
 * We are only interested in how the sender of this report is doing wrt to us.
 */
static void
rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
    RTPArrivalStats * arrival)
{
  guint32 senderssrc;
  RTPSource *source;
  gboolean created;

  senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);

  GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);

  source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
  if (!source)
    return;

  if (created)
    on_new_ssrc (sess, source);

  rtp_session_process_rb (sess, source, packet, arrival);
  g_object_unref (source);
}

/* Get SDES items and store them in the SSRC */
static void
rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
    RTPArrivalStats * arrival)
{
  guint items, i, j;
  gboolean more_items, more_entries;

  items = gst_rtcp_packet_sdes_get_item_count (packet);
  GST_DEBUG ("got SDES packet with %d items", items);

  more_items = gst_rtcp_packet_sdes_first_item (packet);
  i = 0;
  while (more_items) {
    guint32 ssrc;
    gboolean changed, created, validated;
    RTPSource *source;
    GstStructure *sdes;

    ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);

    GST_DEBUG ("item %d, SSRC %08x", i, ssrc);

    changed = FALSE;

    /* find src, no probation when dealing with RTCP */
    source = obtain_source (sess, ssrc, &created, arrival, FALSE);
    if (!source)
      return;

    sdes = gst_structure_new ("application/x-rtp-source-sdes", NULL);

    more_entries = gst_rtcp_packet_sdes_first_entry (packet);
    j = 0;
    while (more_entries) {
      GstRTCPSDESType type;
      guint8 len;
      guint8 *data;
      gchar *name;
      gchar *value;

      gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);

      GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
          data);

      if (type == GST_RTCP_SDES_PRIV) {
        name = g_strndup ((const gchar *) &data[1], data[0]);
        len -= data[0] + 1;
        data += data[0] + 1;
      } else {
        name = g_strdup (gst_rtcp_sdes_type_to_name (type));
      }

      value = g_strndup ((const gchar *) data, len);

      gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);

      g_free (name);
      g_free (value);

      more_entries = gst_rtcp_packet_sdes_next_entry (packet);
      j++;
    }

    /* takes ownership of sdes */
    changed = rtp_source_set_sdes_struct (source, sdes);

    validated = !RTP_SOURCE_IS_ACTIVE (source);
    source->validated = TRUE;

    if (created)
      on_new_ssrc (sess, source);
    if (validated)
      on_ssrc_validated (sess, source);
    if (changed)
      on_ssrc_sdes (sess, source);

    g_object_unref (source);

    more_items = gst_rtcp_packet_sdes_next_item (packet);
    i++;
  }
}

/* BYE is sent when a client leaves the session
 */
static void
rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
    RTPArrivalStats * arrival)
{
  guint count, i;
  gchar *reason;
  gboolean reconsider = FALSE;

  reason = gst_rtcp_packet_bye_get_reason (packet);
  GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));

  count = gst_rtcp_packet_bye_get_ssrc_count (packet);
  for (i = 0; i < count; i++) {
    guint32 ssrc;
    RTPSource *source;
    gboolean created, prevactive, prevsender;
    guint pmembers, members;

    ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
    GST_DEBUG ("SSRC: %08x", ssrc);

    /* find src and mark bye, no probation when dealing with RTCP */
    source = obtain_source (sess, ssrc, &created, arrival, FALSE);
    if (!source)
      return;

    /* store time for when we need to time out this source */
    source->bye_time = arrival->current_time;

    prevactive = RTP_SOURCE_IS_ACTIVE (source);
    prevsender = RTP_SOURCE_IS_SENDER (source);

    /* let the source handle the rest */
    rtp_source_process_bye (source, reason);

    pmembers = sess->stats.active_sources;

    if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
      sess->stats.active_sources--;
      GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
          sess->stats.active_sources);
    }
    if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
      sess->stats.sender_sources--;
      GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
          sess->stats.sender_sources);
    }
    members = sess->stats.active_sources;

    if (!sess->source->received_bye && members < pmembers) {
      /* some members went away since the previous timeout estimate.
       * Perform reverse reconsideration but only when we are not scheduling a
       * BYE ourselves. */
      if (arrival->current_time < sess->next_rtcp_check_time) {
        GstClockTime time_remaining;

        time_remaining = sess->next_rtcp_check_time - arrival->current_time;
        sess->next_rtcp_check_time =
            gst_util_uint64_scale (time_remaining, members, pmembers);

        GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
            GST_TIME_ARGS (sess->next_rtcp_check_time));

        sess->next_rtcp_check_time += arrival->current_time;

        /* mark pending reconsider. We only want to signal the reconsideration
         * once after we handled all the source in the bye packet */
        reconsider = TRUE;
      }
    }

    if (created)
      on_new_ssrc (sess, source);

    on_bye_ssrc (sess, source);

    g_object_unref (source);
  }
  if (reconsider) {
    RTP_SESSION_UNLOCK (sess);
    /* notify app of reconsideration */
    if (sess->callbacks.reconsider)
      sess->callbacks.reconsider (sess, sess->reconsider_user_data);
    RTP_SESSION_LOCK (sess);
  }
  g_free (reason);
}

static void
rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
    RTPArrivalStats * arrival)
{
  GST_DEBUG ("received APP");
}

static void
rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
    guint32 media_ssrc, GstClockTime current_time)
{
  RTPSource *src;
  guint32 round_trip = 0;

  if (!sess->callbacks.request_key_unit)
    return;

  src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
      GINT_TO_POINTER (sender_ssrc));

  if (!src)
    return;

  if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
      rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL,
          &round_trip)) {
    GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
        GST_SECOND, 65536);

    if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
        current_time - sess->last_keyframe_request < round_trip_in_ns) {
      GST_DEBUG ("Ignoring PLI because one was send without one RTT (%"
          GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
          GST_TIME_ARGS (current_time - sess->last_keyframe_request),
          GST_TIME_ARGS (round_trip_in_ns));;
      return;
    }
  }

  sess->last_keyframe_request = current_time;

  GST_LOG ("received PLI from %X %p(%p)", sender_ssrc,
      sess->callbacks.process_rtp, sess->callbacks.request_key_unit);

  sess->callbacks.request_key_unit (sess, FALSE,
      sess->request_key_unit_user_data);
}

static void
rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
    RTPArrivalStats * arrival, GstClockTime current_time)
{
  GstRTCPType type = gst_rtcp_packet_get_type (packet);
  GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
  guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
  guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
  guint length = 4 * (gst_rtcp_packet_get_length (packet) - 2);

  GST_DEBUG ("received feedback %d:%d from %08X about %08X"
      " with FCI of length %d", type, fbtype, sender_ssrc, media_ssrc, length);

  if (g_signal_has_handler_pending (sess,
          rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
    GstBuffer *fci = NULL;

    if (length) {
      fci = gst_buffer_create_sub (packet->buffer, packet->offset + 72, length);
      GST_BUFFER_TIMESTAMP (fci) = arrival->running_time;
    }

    RTP_SESSION_UNLOCK (sess);
    g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
        type, fbtype, sender_ssrc, media_ssrc, fci);
    RTP_SESSION_LOCK (sess);

    if (fci)
      gst_buffer_unref (fci);
  }

  if (sess->rtcp_feedback_retention_window) {
    RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
        GINT_TO_POINTER (media_ssrc));

    if (src)
      rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
  }

  if (rtp_source_get_ssrc (sess->source) == media_ssrc) {
    switch (type) {
      case GST_RTCP_TYPE_PSFB:
        switch (fbtype) {
          case GST_RTCP_PSFB_TYPE_PLI:
            rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
                current_time);
            break;
          default:
            break;
        }
        break;
      case GST_RTCP_TYPE_RTPFB:
      default:
        break;
    }
  }
}

/**
 * rtp_session_process_rtcp:
 * @sess: and #RTPSession
 * @buffer: an RTCP buffer
 * @current_time: the current system time
 *
 * Process an RTCP buffer in the session manager. This function takes ownership
 * of @buffer.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
    GstClockTime current_time)
{
  GstRTCPPacket packet;
  gboolean more, is_bye = FALSE, do_sync = FALSE;
  RTPArrivalStats arrival;
  GstFlowReturn result = GST_FLOW_OK;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);

  if (!gst_rtcp_buffer_validate (buffer))
    goto invalid_packet;

  GST_DEBUG ("received RTCP packet");

  RTP_SESSION_LOCK (sess);
  /* update arrival stats */
  update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1);

  if (sess->sent_bye)
    goto ignore;

  /* start processing the compound packet */
  more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
  while (more) {
    GstRTCPType type;

    type = gst_rtcp_packet_get_type (&packet);

    /* when we are leaving the session, we should ignore all non-BYE messages */
    if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
      GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
      goto next;
    }

    switch (type) {
      case GST_RTCP_TYPE_SR:
        rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
        break;
      case GST_RTCP_TYPE_RR:
        rtp_session_process_rr (sess, &packet, &arrival);
        break;
      case GST_RTCP_TYPE_SDES:
        rtp_session_process_sdes (sess, &packet, &arrival);
        break;
      case GST_RTCP_TYPE_BYE:
        is_bye = TRUE;
        /* don't try to attempt lip-sync anymore for streams with a BYE */
        do_sync = FALSE;
        rtp_session_process_bye (sess, &packet, &arrival);
        break;
      case GST_RTCP_TYPE_APP:
        rtp_session_process_app (sess, &packet, &arrival);
        break;
      case GST_RTCP_TYPE_RTPFB:
      case GST_RTCP_TYPE_PSFB:
        rtp_session_process_feedback (sess, &packet, &arrival, current_time);
        break;
      default:
        GST_WARNING ("got unknown RTCP packet");
        break;
    }
  next:
    more = gst_rtcp_packet_move_to_next (&packet);
  }

  /* if we are scheduling a BYE, we only want to count bye packets, else we
   * count everything */
  if (sess->source->received_bye) {
    if (is_bye) {
      sess->stats.bye_members++;
      UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
    }
  } else {
    /* keep track of average packet size */
    UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
  }
  GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
      sess->stats.avg_rtcp_packet_size, arrival.bytes);
  RTP_SESSION_UNLOCK (sess);

  /* notify caller of sr packets in the callback */
  if (do_sync && sess->callbacks.sync_rtcp) {
    /* make writable, we might want to change the buffer */
    buffer = gst_buffer_make_metadata_writable (buffer);

    result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
        sess->sync_rtcp_user_data);
  } else
    gst_buffer_unref (buffer);

  return result;

  /* ERRORS */
invalid_packet:
  {
    GST_DEBUG ("invalid RTCP packet received");
    gst_buffer_unref (buffer);
    return GST_FLOW_OK;
  }
ignore:
  {
    gst_buffer_unref (buffer);
    RTP_SESSION_UNLOCK (sess);
    GST_DEBUG ("ignoring RTP packet because we left");
    return GST_FLOW_OK;
  }
}

/**
 * rtp_session_send_rtp:
 * @sess: an #RTPSession
 * @data: pointer to either an RTP buffer or a list of RTP buffers
 * @is_list: TRUE when @data is a buffer list
 * @current_time: the current system time
 * @running_time: the running time of @data
 *
 * Send the RTP buffer in the session manager. This function takes ownership of
 * @buffer.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
    GstClockTime current_time, GstClockTime running_time)
{
  GstFlowReturn result;
  RTPSource *source;
  gboolean prevsender;
  gboolean valid_packet;
  guint64 oldrate;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
  g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);

  if (is_list) {
    valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
  } else {
    valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
  }

  if (!valid_packet)
    goto invalid_packet;

  GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");

  RTP_SESSION_LOCK (sess);
  source = sess->source;

  /* update last activity */
  source->last_rtp_activity = current_time;

  prevsender = RTP_SOURCE_IS_SENDER (source);
  oldrate = source->bitrate;

  /* we use our own source to send */
  result = rtp_source_send_rtp (source, data, is_list, running_time);

  if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
    sess->stats.sender_sources++;
  if (oldrate != source->bitrate)
    sess->recalc_bandwidth = TRUE;
  RTP_SESSION_UNLOCK (sess);

  return result;

  /* ERRORS */
invalid_packet:
  {
    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
    GST_DEBUG ("invalid RTP packet received");
    return GST_FLOW_OK;
  }
}

static void
add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
{
  *bandwidth += source->bitrate;
}

static GstClockTime
calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
    gboolean first)
{
  GstClockTime result;

  /* recalculate bandwidth when it changed */
  if (sess->recalc_bandwidth) {
    gdouble bandwidth;

    if (sess->bandwidth > 0)
      bandwidth = sess->bandwidth;
    else {
      /* If it is <= 0, then try to estimate the actual bandwidth */
      bandwidth = sess->source->bitrate;

      g_hash_table_foreach (sess->cnames, (GHFunc) add_bitrates, &bandwidth);
      bandwidth /= 8.0;
    }
    if (bandwidth == 0)
      bandwidth = RTP_STATS_BANDWIDTH;

    rtp_stats_set_bandwidths (&sess->stats, bandwidth,
        sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);

    sess->recalc_bandwidth = FALSE;
  }

  if (sess->source->received_bye) {
    result = rtp_stats_calculate_bye_interval (&sess->stats);
  } else {
    result = rtp_stats_calculate_rtcp_interval (&sess->stats,
        RTP_SOURCE_IS_SENDER (sess->source), first);
  }

  GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
      GST_TIME_ARGS (result), first);

  if (!deterministic && result != GST_CLOCK_TIME_NONE)
    result = rtp_stats_add_rtcp_jitter (&sess->stats, result);

  GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));

  return result;
}

/* Stop the current @sess and schedule a BYE message for the other members.
 * One must have the session lock to call this function
 */
static GstFlowReturn
rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
    GstClockTime current_time)
{
  GstFlowReturn result = GST_FLOW_OK;
  RTPSource *source;
  GstClockTime interval;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);

  source = sess->source;

  /* ignore more BYEs */
  if (source->received_bye)
    goto done;

  /* we have BYE now */
  source->received_bye = TRUE;
  /* at least one member wants to send a BYE */
  g_free (sess->bye_reason);
  sess->bye_reason = g_strdup (reason);
  INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
  sess->stats.bye_members = 1;
  sess->first_rtcp = TRUE;
  sess->sent_bye = FALSE;
  sess->allow_early = TRUE;

  /* reschedule transmission */
  sess->last_rtcp_send_time = current_time;
  interval = calculate_rtcp_interval (sess, FALSE, TRUE);
  sess->next_rtcp_check_time = current_time + interval;

  GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
      GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));

  RTP_SESSION_UNLOCK (sess);
  /* notify app of reconsideration */
  if (sess->callbacks.reconsider)
    sess->callbacks.reconsider (sess, sess->reconsider_user_data);
  RTP_SESSION_LOCK (sess);
done:

  return result;
}

/**
 * rtp_session_schedule_bye:
 * @sess: an #RTPSession
 * @reason: a reason or NULL
 * @current_time: the current system time
 *
 * Stop the current @sess and schedule a BYE message for the other members.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
    GstClockTime current_time)
{
  GstFlowReturn result = GST_FLOW_OK;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);

  RTP_SESSION_LOCK (sess);
  result = rtp_session_schedule_bye_locked (sess, reason, current_time);
  RTP_SESSION_UNLOCK (sess);

  return result;
}

/**
 * rtp_session_next_timeout:
 * @sess: an #RTPSession
 * @current_time: the current system time
 *
 * Get the next time we should perform session maintenance tasks.
 *
 * Returns: a time when rtp_session_on_timeout() should be called with the
 * current system time.
 */
GstClockTime
rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
{
  GstClockTime result, interval = 0;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);

  RTP_SESSION_LOCK (sess);

  if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
    result = sess->next_early_rtcp_time;
    goto early_exit;
  }

  result = sess->next_rtcp_check_time;

  GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
      GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));

  if (result < current_time) {
    GST_DEBUG ("take current time as base");
    /* our previous check time expired, start counting from the current time
     * again. */
    result = current_time;
  }

  if (sess->source->received_bye) {
    if (sess->sent_bye) {
      GST_DEBUG ("we sent BYE already");
      interval = GST_CLOCK_TIME_NONE;
    } else if (sess->stats.active_sources >= 50) {
      GST_DEBUG ("reconsider BYE, more than 50 sources");
      /* reconsider BYE if members >= 50 */
      interval = calculate_rtcp_interval (sess, FALSE, TRUE);
    }
  } else {
    if (sess->first_rtcp) {
      GST_DEBUG ("first RTCP packet");
      /* we are called for the first time */
      interval = calculate_rtcp_interval (sess, FALSE, TRUE);
    } else if (sess->next_rtcp_check_time < current_time) {
      GST_DEBUG ("old check time expired, getting new timeout");
      /* get a new timeout when we need to */
      interval = calculate_rtcp_interval (sess, FALSE, FALSE);
    }
  }

  if (interval != GST_CLOCK_TIME_NONE)
    result += interval;
  else
    result = GST_CLOCK_TIME_NONE;

  sess->next_rtcp_check_time = result;

early_exit:

  GST_DEBUG ("current time: %" GST_TIME_FORMAT
      ", next time: %" GST_TIME_FORMAT,
      GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
  RTP_SESSION_UNLOCK (sess);

  return result;
}

typedef struct
{
  RTPSession *sess;
  GstBuffer *rtcp;
  GstClockTime current_time;
  guint64 ntpnstime;
  GstClockTime running_time;
  GstClockTime interval;
  GstRTCPPacket packet;
  gboolean is_bye;
  gboolean has_sdes;
  gboolean is_early;
  gboolean may_suppress;
} ReportData;

static void
session_start_rtcp (RTPSession * sess, ReportData * data)
{
  GstRTCPPacket *packet = &data->packet;
  RTPSource *own = sess->source;

  data->rtcp = gst_rtcp_buffer_new (sess->mtu);

  if (RTP_SOURCE_IS_SENDER (own)) {
    guint64 ntptime;
    guint32 rtptime;
    guint32 packet_count, octet_count;

    /* we are a sender, create SR */
    GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
    gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);

    /* get latest stats */
    rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
        &ntptime, &rtptime, &packet_count, &octet_count);
    /* store stats */
    rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
        packet_count, octet_count);

    /* fill in sender report info */
    gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
        ntptime, rtptime, packet_count, octet_count);
  } else {
    /* we are only receiver, create RR */
    GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
    gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
    gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
  }
}

/* construct a Sender or Receiver Report */
static void
session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
{
  RTPSession *sess = data->sess;
  GstRTCPPacket *packet = &data->packet;

  /* create a new buffer if needed */
  if (data->rtcp == NULL) {
    session_start_rtcp (sess, data);
  } else if (data->is_early) {
    /* Put a single RR or SR in minimal compound packets */
    return;
  }
  if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
    /* only report about other sender sources */
    if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
      guint8 fractionlost;
      gint32 packetslost;
      guint32 exthighestseq, jitter;
      guint32 lsr, dlsr;

      /* get new stats */
      rtp_source_get_new_rb (source, data->current_time, &fractionlost,
          &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);

      /* store last generated RR packet */
      source->last_rr.is_valid = TRUE;
      source->last_rr.fractionlost = fractionlost;
      source->last_rr.packetslost = packetslost;
      source->last_rr.exthighestseq = exthighestseq;
      source->last_rr.jitter = jitter;
      source->last_rr.lsr = lsr;
      source->last_rr.dlsr = dlsr;

      /* packet is not yet filled, add report block for this source. */
      gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
          exthighestseq, jitter, lsr, dlsr);
    }
  }
}

/* perform cleanup of sources that timed out */
static void
session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
{
  gboolean remove = FALSE;
  gboolean byetimeout = FALSE;
  gboolean sendertimeout = FALSE;
  gboolean is_sender, is_active;
  RTPSession *sess = data->sess;
  GstClockTime interval;

  is_sender = RTP_SOURCE_IS_SENDER (source);
  is_active = RTP_SOURCE_IS_ACTIVE (source);

  /* check for our own source, we don't want to delete our own source. */
  if (!(source == sess->source)) {
    if (source->received_bye) {
      /* if we received a BYE from the source, remove the source after some
       * time. */
      if (data->current_time > source->bye_time &&
          data->current_time - source->bye_time > sess->stats.bye_timeout) {
        GST_DEBUG ("removing BYE source %08x", source->ssrc);
        remove = TRUE;
        byetimeout = TRUE;
      }
    }
    /* sources that were inactive for more than 5 times the deterministic reporting
     * interval get timed out. the min timeout is 5 seconds. */
    if (data->current_time > source->last_activity) {
      interval = MAX (data->interval * 5, 5 * GST_SECOND);
      if (data->current_time - source->last_activity > interval) {
        GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
            source->ssrc, GST_TIME_ARGS (source->last_activity));
        remove = TRUE;
      }
    }
  }

  /* senders that did not send for a long time become a receiver, this also
   * holds for our own source. */
  if (is_sender) {
    if (data->current_time > source->last_rtp_activity) {
      interval = MAX (data->interval * 2, 5 * GST_SECOND);
      if (data->current_time - source->last_rtp_activity > interval) {
        GST_DEBUG ("sender source %08x timed out and became receiver, last %"
            GST_TIME_FORMAT, source->ssrc,
            GST_TIME_ARGS (source->last_rtp_activity));
        source->is_sender = FALSE;
        sess->stats.sender_sources--;
        sendertimeout = TRUE;
      }
    }
  }

  if (remove) {
    sess->total_sources--;
    if (is_sender)
      sess->stats.sender_sources--;
    if (is_active)
      sess->stats.active_sources--;

    if (byetimeout)
      on_bye_timeout (sess, source);
    else
      on_timeout (sess, source);
  } else {
    if (sendertimeout)
      on_sender_timeout (sess, source);
  }

  source->closing = remove;
}

static void
session_sdes (RTPSession * sess, ReportData * data)
{
  GstRTCPPacket *packet = &data->packet;
  const GstStructure *sdes;
  gint i, n_fields;

  /* add SDES packet */
  gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);

  gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);

  sdes = rtp_source_get_sdes_struct (sess->source);

  /* add all fields in the structure, the order is not important. */
  n_fields = gst_structure_n_fields (sdes);
  for (i = 0; i < n_fields; ++i) {
    const gchar *field;
    const gchar *value;
    GstRTCPSDESType type;

    field = gst_structure_nth_field_name (sdes, i);
    if (field == NULL)
      continue;
    value = gst_structure_get_string (sdes, field);
    if (value == NULL)
      continue;
    type = gst_rtcp_sdes_name_to_type (field);

    /* Early packets are minimal and only include the CNAME */
    if (data->is_early && type != GST_RTCP_SDES_CNAME)
      continue;

    if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
      gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
          (const guint8 *) value);
    } else if (type == GST_RTCP_SDES_PRIV) {
      gsize prefix_len;
      gsize value_len;
      gsize data_len;
      guint8 data[256];

      /* don't accept entries that are too big */
      prefix_len = strlen (field);
      if (prefix_len > 255)
        continue;
      value_len = strlen (value);
      if (value_len > 255)
        continue;
      data_len = 1 + prefix_len + value_len;
      if (data_len > 255)
        continue;

      data[0] = prefix_len;
      memcpy (&data[1], field, prefix_len);
      memcpy (&data[1 + prefix_len], value, value_len);

      gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
    }
  }

  data->has_sdes = TRUE;
}

/* schedule a BYE packet */
static void
session_bye (RTPSession * sess, ReportData * data)
{
  GstRTCPPacket *packet = &data->packet;

  /* open packet */
  session_start_rtcp (sess, data);

  /* add SDES */
  session_sdes (sess, data);

  /* add a BYE packet */
  gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
  gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
  if (sess->bye_reason)
    gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);

  /* we have a BYE packet now */
  data->is_bye = TRUE;
}

static gboolean
is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
{
  GstClockTime new_send_time, elapsed;

  if (data->is_early && sess->next_early_rtcp_time < current_time)
    goto early;

  /* no need to check yet */
  if (sess->next_rtcp_check_time > current_time) {
    GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
        GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
        GST_TIME_ARGS (current_time));
    return FALSE;
  }

  /* get elapsed time since we last reported */
  elapsed = current_time - sess->last_rtcp_send_time;

  /* perform forward reconsideration */
  new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);

  GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
      GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));

  new_send_time += sess->last_rtcp_send_time;

  /* check if reconsideration */
  if (current_time < new_send_time) {
    GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
        GST_TIME_ARGS (new_send_time));
    /* store new check time */
    sess->next_rtcp_check_time = new_send_time;
    return FALSE;
  }

early:

  new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);

  GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
      GST_TIME_ARGS (new_send_time));
  sess->next_rtcp_check_time = current_time + new_send_time;

  /* Apply the rules from RFC 4585 section 3.5.3 */
  if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
    GstClockTimeDiff T_rr_current_interval = g_random_double_range (0.5, 1.5) *
        sess->stats.min_interval;

    /* This will caused the RTCP to be suppressed if no FB packets are added */
    if (sess->last_rtcp_send_time + T_rr_current_interval >
        sess->next_rtcp_check_time) {
      GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
          " last: %" GST_TIME_FORMAT
          " + T_rr_current_interval: %" GST_TIME_FORMAT
          " >  sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
          GST_TIME_ARGS (sess->stats.min_interval),
          GST_TIME_ARGS (sess->last_rtcp_send_time),
          GST_TIME_ARGS (T_rr_current_interval),
          GST_TIME_ARGS (sess->next_rtcp_check_time));
      data->may_suppress = TRUE;
    }
  }

  return TRUE;
}

static void
clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
{
  g_hash_table_insert (hash_table, key, g_object_ref (source));
}

static gboolean
remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
{
  return source->closing;
}

/**
 * rtp_session_on_timeout:
 * @sess: an #RTPSession
 * @current_time: the current system time
 * @ntpnstime: the current NTP time in nanoseconds
 * @running_time: the current running_time of the pipeline
 *
 * Perform maintenance actions after the timeout obtained with
 * rtp_session_next_timeout() expired.
 *
 * This function will perform timeouts of receivers and senders, send a BYE
 * packet or generate RTCP packets with current session stats.
 *
 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
 * times, for each packet that should be processed.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
    guint64 ntpnstime, GstClockTime running_time)
{
  GstFlowReturn result = GST_FLOW_OK;
  ReportData data;
  RTPSource *own;
  GHashTable *table_copy;
  gboolean notify = FALSE;

  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);

  GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
      GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));

  data.sess = sess;
  data.rtcp = NULL;
  data.current_time = current_time;
  data.ntpnstime = ntpnstime;
  data.is_bye = FALSE;
  data.has_sdes = FALSE;
  data.may_suppress = FALSE;
  data.running_time = running_time;

  own = sess->source;

  RTP_SESSION_LOCK (sess);
  /* get a new interval, we need this for various cleanups etc */
  data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);

  /* Make a local copy of the hashtable. We need to do this because the
   * cleanup stage below releases the session lock. */
  table_copy = g_hash_table_new_full (NULL, NULL, NULL,
      (GDestroyNotify) g_object_unref);
  g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
      (GHFunc) clone_ssrcs_hashtable, table_copy);

  /* Clean up the session, mark the source for removing, this might release the
   * session lock. */
  g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
  g_hash_table_destroy (table_copy);

  /* Now remove the marked sources */
  g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
      (GHRFunc) remove_closing_sources, NULL);

  if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
    data.is_early = TRUE;
  else
    data.is_early = FALSE;

  /* see if we need to generate SR or RR packets */
  if (is_rtcp_time (sess, current_time, &data)) {
    if (own->received_bye) {
      /* generate BYE instead */
      GST_DEBUG ("generating BYE message");
      session_bye (sess, &data);
      sess->sent_bye = TRUE;
    } else {
      /* loop over all known sources and do something */
      g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
          (GHFunc) session_report_blocks, &data);
    }
  }

  if (data.rtcp) {
    /* we keep track of the last report time in order to timeout inactive
     * receivers or senders */
    if (!data.is_early && !data.may_suppress)
      sess->last_rtcp_send_time = data.current_time;
    sess->first_rtcp = FALSE;
    sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;

    /* add SDES for this source when not already added */
    if (!data.has_sdes)
      session_sdes (sess, &data);
  }

  /* check for outdated collisions */
  GST_DEBUG ("Timing out collisions");
  rtp_source_timeout (sess->source, current_time,
      data.interval * RTCP_INTERVAL_COLLISION_TIMEOUT,
      running_time - sess->rtcp_feedback_retention_window);

  if (sess->change_ssrc) {
    GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
    g_hash_table_steal (sess->ssrcs[sess->mask_idx],
        GINT_TO_POINTER (own->ssrc));

    own->ssrc = rtp_session_create_new_ssrc (sess);
    rtp_source_reset (own);

    g_hash_table_insert (sess->ssrcs[sess->mask_idx],
        GINT_TO_POINTER (own->ssrc), own);

    g_free (sess->bye_reason);
    sess->bye_reason = NULL;
    sess->sent_bye = FALSE;
    sess->change_ssrc = FALSE;
    notify = TRUE;
    GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
  }

  sess->allow_early = TRUE;

  RTP_SESSION_UNLOCK (sess);

  if (notify)
    g_object_notify (G_OBJECT (sess), "internal-ssrc");

  /* push out the RTCP packet */
  if (data.rtcp) {
    gboolean do_not_suppress;

    /* Give the user a change to add its own packet */
    g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
        data.rtcp, data.is_early, &do_not_suppress);

    if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
      guint packet_size;

      /* close the RTCP packet */
      gst_rtcp_buffer_end (data.rtcp);

      packet_size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;

      UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
      GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
          sess->stats.avg_rtcp_packet_size, packet_size);
      result =
          sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye,
          sess->send_rtcp_user_data);
    } else {
      GST_DEBUG ("freeing packet callback: %p"
          " do_not_suppress: %d may_suppress: %d",
          sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
      gst_buffer_unref (data.rtcp);
    }
  }

  return result;
}

void
rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
    GstClockTimeDiff max_delay)
{
  GstClockTime T_dither_max;

  /* Implements the algorithm described in RFC 4585 section 3.5.2 */

  RTP_SESSION_LOCK (sess);

  /* Check if already requested */
  /*  RFC 4585 section 3.5.2 step 2 */
  if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
    goto dont_send;

  /* Ignore the request a scheduled packet will be in time anyway */
  if (current_time + max_delay > sess->next_rtcp_check_time)
    goto dont_send;

  /*  RFC 4585 section 3.5.2 step 2b */
  /* If the total sources is <=2, then there is only us and one peer */
  if (sess->total_sources <= 2) {
    T_dither_max = 0;
  } else {
    /* Divide by 2 because l = 0.5 */
    T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
    T_dither_max /= 2;
  }

  /*  RFC 4585 section 3.5.2 step 3 */
  if (current_time + T_dither_max > sess->next_rtcp_check_time)
    goto dont_send;

  /*  RFC 4585 section 3.5.2 step 4 */
  if (sess->allow_early == FALSE)
    goto dont_send;

  if (T_dither_max) {
    /* Schedule an early transmission later */
    sess->next_early_rtcp_time = g_random_double () * T_dither_max +
        current_time;
  } else {
    /* If no dithering, schedule it for NOW */
    sess->next_early_rtcp_time = current_time;
  }

  RTP_SESSION_UNLOCK (sess);

  /* notify app of need to send packet early
   * and therefore of timeout change */
  if (sess->callbacks.reconsider)
    sess->callbacks.reconsider (sess, sess->reconsider_user_data);

  return;

dont_send:

  RTP_SESSION_UNLOCK (sess);

}

void
rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean fir)
{
  guint i;

  if (fir)
    return;

  for (i = 0; i < sess->rtcp_pli_requests->len; i++)
    if (ssrc == g_array_index (sess->rtcp_pli_requests, guint32, i))
      return;

  g_array_append_val (sess->rtcp_pli_requests, ssrc);
}

static gboolean
has_pli_compare_func (gconstpointer a, gconstpointer ignored)
{
  GstRTCPPacket packet;

  packet.buffer = (GstBuffer *) a;
  packet.offset = 0;

  if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
      gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
    return TRUE;
  else
    return FALSE;
}

static gboolean
rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
    gboolean early)
{
  gboolean ret = FALSE;

  RTP_SESSION_LOCK (sess);

  while (sess->rtcp_pli_requests->len) {
    GstRTCPPacket rtcppacket;
    guint media_ssrc = g_array_index (sess->rtcp_pli_requests, guint32, 0);
    RTPSource *media_src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
        GUINT_TO_POINTER (media_ssrc));

    if (media_src && !rtp_source_has_retained (media_src,
            has_pli_compare_func, NULL)) {
      if (gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_PSFB, &rtcppacket)) {
        gst_rtcp_packet_fb_set_type (&rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
        gst_rtcp_packet_fb_set_sender_ssrc (&rtcppacket,
            rtp_source_get_ssrc (sess->source));
        gst_rtcp_packet_fb_set_media_ssrc (&rtcppacket, media_ssrc);
        ret = TRUE;
      } else {
        /* Break because the packet is full, will put next request in a
         * further packet
         */
        break;
      }
    }

    g_array_remove_index (sess->rtcp_pli_requests, 0);
  }

  RTP_SESSION_UNLOCK (sess);

  return ret;
}