gst/rtpmanager/gstrtpbin.c: fix for pad name change

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.

9 files changed, 665 insertions, 86 deletions

diff --git a/ChangeLog b/ChangeLog
index 5cae6e7e..d285d3d8 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,43 @@
+2007-04-25  Wim Taymans  <wim@fluendo.com>
+
+	* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
+	fix for pad name change
+
+	* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
+	(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
+	Fix for renamed methods.
+
+	* gst/rtpmanager/rtpsession.c: (rtp_session_init),
+	(rtp_session_finalize), (rtp_session_set_cname),
+	(rtp_session_get_cname), (rtp_session_set_name),
+	(rtp_session_get_name), (rtp_session_set_email),
+	(rtp_session_get_email), (rtp_session_set_phone),
+	(rtp_session_get_phone), (rtp_session_set_location),
+	(rtp_session_get_location), (rtp_session_set_tool),
+	(rtp_session_get_tool), (rtp_session_set_note),
+	(rtp_session_get_note), (source_push_rtp), (obtain_source),
+	(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
+	(rtp_session_create_source), (rtp_session_process_rtp),
+	(rtp_session_process_sr), (rtp_session_process_sdes),
+	(rtp_session_process_rtcp), (rtp_session_send_rtp),
+	(rtp_session_get_reporting_interval), (session_report_blocks),
+	(session_sdes), (rtp_session_perform_reporting):
+	* gst/rtpmanager/rtpsession.h:
+	Prepare for implementing SSRC sampling.
+	Create SSRC for the session. 
+	Add methods to set the SDES entries.
+	fix accounting of senders/receivers.
+	Implement SR/RR/SDES RTCP reporting.
+
+	* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
+	(rtp_source_process_rtp), (rtp_source_process_sr):
+	* gst/rtpmanager/rtpsource.h:
+	Implement extended sequence number.
+
+	* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
+	* gst/rtpmanager/rtpstats.h:
+	Rename some fields.
+
 2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>
 
 	* gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 9162d76c..215dbc73 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -959,7 +959,7 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
 
   /* get rtcp_src pad and store */
   session->rtcp_src =
-      gst_element_get_request_pad (session->session, "rtcp_src");
+      gst_element_get_request_pad (session->session, "send_rtcp_src");
   if (session->rtcp_src == NULL)
     goto pad_failed;
 
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 03b0802b..b11dbbba 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -305,7 +305,7 @@ rtcp_thread (GstRTPSession * rtpsession)
     gdouble timeout;
     GstClockTime target;
 
-    timeout = rtp_session_get_rtcp_interval (rtpsession->priv->session);
+    timeout = rtp_session_get_reporting_interval (rtpsession->priv->session);
     GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
 
     target = gst_clock_get_time (clock);
@@ -318,7 +318,7 @@ rtcp_thread (GstRTPSession * rtpsession)
     GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
 
     /* make the session manager produce RTCP, we ignore the result. */
-    rtp_session_produce_rtcp (rtpsession->priv->session);
+    rtp_session_perform_reporting (rtpsession->priv->session);
 
     GST_RTP_SESSION_LOCK (rtpsession);
     gst_clock_id_unref (id);
@@ -472,6 +472,8 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
   rtpsession = GST_RTP_SESSION (user_data);
   priv = rtpsession->priv;
 
+  GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
+
   if (rtpsession->send_rtcp_src) {
     result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
   } else {
@@ -515,6 +517,8 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
   if (!gst_structure_get_int (caps_struct, "clock-rate", &result))
     goto no_clock_rate;
 
+  GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
+
   return result;
 
   /* ERRORS */
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index e13f7d6c..e4925a2a 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -57,6 +57,9 @@ 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 void
 rtp_session_class_init (RTPSessionClass * klass)
 {
@@ -123,18 +126,35 @@ rtp_session_class_init (RTPSessionClass * klass)
 static void
 rtp_session_init (RTPSession * sess)
 {
+  gint i;
+
   sess->lock = g_mutex_new ();
-  sess->ssrcs =
-      g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) g_object_unref);
+  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);
 
-  /* create an SSRC for this session manager */
-  sess->source = rtp_session_create_source (sess);
-
   rtp_stats_init_defaults (&sess->stats);
 
+  /* create an active SSRC for this session manager */
+  sess->source = rtp_session_create_source (sess);
+  sess->stats.active_sources++;
+
   /* default UDP header length */
   sess->header_len = 28;
+  sess->mtu = 1400;
+
+  /* some default SDES entries */
+  //sess->cname = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
+  sess->cname = g_strdup_printf ("foo@%s", g_get_host_name ());
+  sess->name = g_strdup (g_get_real_name ());
+  sess->tool = g_strdup ("GStreamer");
 
   GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
 }
@@ -143,14 +163,20 @@ static void
 rtp_session_finalize (GObject * object)
 {
   RTPSession *sess;
+  gint i;
 
   sess = RTP_SESSION_CAST (object);
 
   g_mutex_free (sess->lock);
-  g_hash_table_destroy (sess->ssrcs);
+  for (i = 0; i < 32; i++)
+    g_hash_table_destroy (sess->ssrcs[i]);
+
   g_hash_table_destroy (sess->cnames);
   g_object_unref (sess->source);
 
+  g_free (sess->cname);
+  g_free (sess->tool);
+
   G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
 }
 
@@ -312,6 +338,230 @@ rtp_session_get_rtcp_bandwidth (RTPSession * sess)
   return sess->stats.rtcp_bandwidth;
 }
 
+/**
+ * rtp_session_set_cname:
+ * @sess: an #RTPSession
+ * @cname: a CNAME for the session
+ *
+ * Set the CNAME for the session. 
+ */
+void
+rtp_session_set_cname (RTPSession * sess, const gchar * cname)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->cname);
+  sess->cname = g_strdup (cname);
+}
+
+/**
+ * rtp_session_get_cname:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured CNAME for the session.
+ *
+ * Returns: The CNAME. g_free after usage.
+ */
+gchar *
+rtp_session_get_cname (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->cname);
+}
+
+/**
+ * rtp_session_set_name:
+ * @sess: an #RTPSession
+ * @name: a NAME for the session
+ *
+ * Set the NAME for the session. 
+ */
+void
+rtp_session_set_name (RTPSession * sess, const gchar * name)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->name);
+  sess->name = g_strdup (name);
+}
+
+/**
+ * rtp_session_get_name:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured NAME for the session.
+ *
+ * Returns: The NAME. g_free after usage.
+ */
+gchar *
+rtp_session_get_name (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->name);
+}
+
+/**
+ * rtp_session_set_email:
+ * @sess: an #RTPSession
+ * @email: an EMAIL for the session
+ *
+ * Set the EMAIL the session. 
+ */
+void
+rtp_session_set_email (RTPSession * sess, const gchar * email)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->email);
+  sess->email = g_strdup (email);
+}
+
+/**
+ * rtp_session_get_email:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured EMAIL of the session.
+ *
+ * Returns: The EMAIL. g_free after usage.
+ */
+gchar *
+rtp_session_get_email (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->email);
+}
+
+/**
+ * rtp_session_set_phone:
+ * @sess: an #RTPSession
+ * @phone: a PHONE for the session
+ *
+ * Set the PHONE the session. 
+ */
+void
+rtp_session_set_phone (RTPSession * sess, const gchar * phone)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->phone);
+  sess->phone = g_strdup (phone);
+}
+
+/**
+ * rtp_session_get_location:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured PHONE of the session.
+ *
+ * Returns: The PHONE. g_free after usage.
+ */
+gchar *
+rtp_session_get_phone (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->phone);
+}
+
+/**
+ * rtp_session_set_location:
+ * @sess: an #RTPSession
+ * @location: a LOCATION for the session
+ *
+ * Set the LOCATION the session. 
+ */
+void
+rtp_session_set_location (RTPSession * sess, const gchar * location)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->location);
+  sess->location = g_strdup (location);
+}
+
+/**
+ * rtp_session_get_location:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured LOCATION of the session.
+ *
+ * Returns: The LOCATION. g_free after usage.
+ */
+gchar *
+rtp_session_get_location (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->location);
+}
+
+/**
+ * rtp_session_set_tool:
+ * @sess: an #RTPSession
+ * @tool: a TOOL for the session
+ *
+ * Set the TOOL the session. 
+ */
+void
+rtp_session_set_tool (RTPSession * sess, const gchar * tool)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->tool);
+  sess->tool = g_strdup (tool);
+}
+
+/**
+ * rtp_session_get_tool:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured TOOL of the session.
+ *
+ * Returns: The TOOL. g_free after usage.
+ */
+gchar *
+rtp_session_get_tool (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->tool);
+}
+
+/**
+ * rtp_session_set_note:
+ * @sess: an #RTPSession
+ * @note: a NOTE for the session
+ *
+ * Set the NOTE the session. 
+ */
+void
+rtp_session_set_note (RTPSession * sess, const gchar * note)
+{
+  g_return_if_fail (RTP_IS_SESSION (sess));
+
+  g_free (sess->note);
+  sess->note = g_strdup (note);
+}
+
+/**
+ * rtp_session_get_note:
+ * @sess: an #RTPSession
+ *
+ * Get the currently configured NOTE of the session.
+ *
+ * Returns: The NOTE. g_free after usage.
+ */
+gchar *
+rtp_session_get_note (RTPSession * sess)
+{
+  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
+
+  return g_strdup (sess->note);
+}
+
 static GstFlowReturn
 source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
 {
@@ -319,6 +569,8 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
 
   if (source == session->source) {
     GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
+
+
     if (session->callbacks.send_rtp)
       result =
           session->callbacks.send_rtp (session, source, buffer,
@@ -371,7 +623,8 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
 {
   RTPSource *source;
 
-  source = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc));
+  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);
@@ -392,7 +645,8 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
     /* configure a callback on the source */
     rtp_source_set_callbacks (source, &callbacks, sess);
 
-    g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (ssrc), source);
+    g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
+        source);
 
     /* we have one more source now */
     sess->total_sources++;
@@ -426,9 +680,12 @@ rtp_session_add_source (RTPSession * sess, RTPSource * src)
   g_return_val_if_fail (src != NULL, FALSE);
 
   RTP_SESSION_LOCK (sess);
-  find = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (src->ssrc));
+  find =
+      g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
+      GINT_TO_POINTER (src->ssrc));
   if (find == NULL) {
-    g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (src->ssrc), src);
+    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;
@@ -501,7 +758,8 @@ rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
   g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
 
   RTP_SESSION_LOCK (sess);
-  result = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc));
+  result =
+      g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
   if (result)
     g_object_ref (result);
   RTP_SESSION_UNLOCK (sess);
@@ -556,11 +814,13 @@ rtp_session_create_source (RTPSession * sess)
     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, GINT_TO_POINTER (ssrc)) == NULL)
+    if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
+            GINT_TO_POINTER (ssrc)) == NULL)
       break;
   }
   source = rtp_source_new (ssrc);
-  g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (ssrc), 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);
@@ -633,6 +893,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
   prevsender = RTP_SOURCE_IS_SENDER (source);
   prevactive = RTP_SOURCE_IS_ACTIVE (source);
 
+  gst_buffer_ref (buffer);
+
   /* let source process the packet */
   result = rtp_source_process_rtp (source, buffer, &arrival);
 
@@ -652,10 +914,11 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
   if (created)
     on_new_ssrc (sess, source);
 
-  /* for validated sources, we add the CSRCs as well */
   if (source->validated) {
     guint8 i, count;
+    gboolean created;
 
+    /* for validated sources, we add the CSRCs as well */
     count = gst_rtp_buffer_get_csrc_count (buffer);
 
     for (i = 0; i < count; i++) {
@@ -675,6 +938,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
       }
     }
   }
+  gst_buffer_unref (buffer);
+
   RTP_SESSION_UNLOCK (sess);
 
   return result;
@@ -704,17 +969,27 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
   guint64 ntptime;
   guint count, i;
   RTPSource *source;
-  gboolean created;
+  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", senderssrc);
+
   RTP_SESSION_LOCK (sess);
   source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
 
+  prevsender = RTP_SOURCE_IS_SENDER (source);
+
   /* first update the source */
   rtp_source_process_sr (source, 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);
 
@@ -785,36 +1060,36 @@ static void
 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
     RTPArrivalStats * arrival)
 {
-  guint chunks, i, j;
-  gboolean more_chunks, more_items;
+  guint items, i, j;
+  gboolean more_items, more_entries;
 
-  chunks = gst_rtcp_packet_sdes_get_chunk_count (packet);
-  GST_DEBUG ("got SDES packet with %d chunks", chunks);
+  items = gst_rtcp_packet_sdes_get_item_count (packet);
+  GST_DEBUG ("got SDES packet with %d items", items);
 
-  more_chunks = gst_rtcp_packet_sdes_first_chunk (packet);
+  more_items = gst_rtcp_packet_sdes_first_item (packet);
   i = 0;
-  while (more_chunks) {
+  while (more_items) {
     guint32 ssrc;
 
     ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
 
-    GST_DEBUG ("chunk %d, SSRC %08x", i, ssrc);
+    GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
 
-    more_items = gst_rtcp_packet_sdes_first_item (packet);
+    more_entries = gst_rtcp_packet_sdes_first_entry (packet);
     j = 0;
-    while (more_items) {
+    while (more_entries) {
       GstRTCPSDESType type;
       guint8 len;
-      gchar *data;
+      guint8 *data;
 
-      gst_rtcp_packet_sdes_get_item (packet, &type, &len, &data);
+      gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
 
-      GST_DEBUG ("item %d, type %d, len %d, data %s", j, type, len, data);
+      GST_DEBUG ("entry %d, type %d, len %d, data %s", j, type, len, data);
 
-      more_items = gst_rtcp_packet_sdes_next_item (packet);
+      more_entries = gst_rtcp_packet_sdes_next_entry (packet);
       j++;
     }
-    more_chunks = gst_rtcp_packet_sdes_next_chunk (packet);
+    more_items = gst_rtcp_packet_sdes_next_item (packet);
     i++;
   }
 }
@@ -906,6 +1181,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
   GST_DEBUG ("received RTCP packet");
 
   /* get packet size including header overhead */
+  RTP_SESSION_LOCK (sess);
   size = GST_BUFFER_SIZE (buffer) + sess->header_len;
 
   /* update average RTCP packet size */
@@ -914,6 +1190,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
   else
     sess->stats.avg_rtcp_packet_size =
         (size + (15 * sess->stats.avg_rtcp_packet_size)) >> 4;
+  RTP_SESSION_UNLOCK (sess);
 
   /* start processing the compound packet */
   more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
@@ -972,6 +1249,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
   g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
   g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
 
+  RTP_SESSION_LOCK (sess);
   source = sess->source;
 
   prevsender = RTP_SOURCE_IS_SENDER (source);
@@ -981,20 +1259,22 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
 
   if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
     sess->stats.sender_sources++;
+  RTP_SESSION_UNLOCK (sess);
 
   return result;
 }
 
 /**
- * rtp_session_get_rtcp_interval:
+ * rtp_session_get_reporting_interval:
  * @sess: an #RTPSession
  *
- * Get the interval for sending out the next RTCP packet
+ * Get the interval for sending out the next RTCP packet and doing general
+ * maintenance tasks.
  *
  * Returns: an interval in seconds.
  */
 gdouble
-rtp_session_get_rtcp_interval (RTPSession * sess)
+rtp_session_get_reporting_interval (RTPSession * sess)
 {
   gdouble result;
 
@@ -1008,8 +1288,112 @@ rtp_session_get_rtcp_interval (RTPSession * sess)
   return result;
 }
 
+typedef struct
+{
+  RTPSession *sess;
+  GstBuffer *rtcp;
+  GstRTCPPacket packet;
+} ReportData;
+
+static void
+session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
+{
+  RTPSession *sess = data->sess;
+  RTPSource *own = sess->source;
+  GstRTCPPacket *packet = &data->packet;
+
+  /* create a new buffer if needed */
+  if (data->rtcp == NULL) {
+    data->rtcp = gst_rtcp_buffer_new (sess->mtu);
+
+    if (RTP_SOURCE_IS_SENDER (own)) {
+      /* 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);
+
+      /* fill in sender report info */
+      gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
+          0, 0, own->stats.packets_sent, own->stats.octets_sent);
+    } 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);
+    }
+  }
+  if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
+    /* only report about other sources */
+    if (source != sess->source) {
+      RTPSourceStats *stats;
+      guint32 extended_max, expected;
+      guint32 expected_interval, received_interval;
+      guint32 lost, lost_interval, fraction;
+
+      stats = &source->stats;
+
+      extended_max = (stats->cycles << 16) + stats->max_seq;
+      expected = extended_max - stats->base_seq + 1;
+
+      if (expected > stats->packets_received) {
+        lost = expected - stats->packets_received;
+        if (lost > 0x7fffff)
+          lost = 0x7fffff;
+      } else {
+        lost = stats->packets_received - expected;
+        if (lost > 0x800000)
+          lost = 0x800000;
+        else
+          lost = -lost;
+      }
+
+      expected_interval = expected - stats->prev_expected;
+      stats->prev_expected = expected;
+      received_interval = stats->packets_received - stats->prev_received;
+      stats->prev_received = stats->packets_received;
+
+      lost_interval = expected_interval - received_interval;
+
+      if (expected_interval == 0 || lost_interval <= 0)
+        fraction = 0;
+      else
+        fraction = (lost_interval << 8) / expected_interval;
+
+      GST_DEBUG ("add RR for SSRC %08x", source->ssrc);
+      /* we scaled the jitter up for additional precision */
+      GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
+          extended_max, stats->jitter >> 4);
+
+      /* packet is not yet filled, add report block for this source. */
+      gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
+          extended_max, stats->jitter >> 4, 0, 0);
+    }
+  }
+}
+
+static void
+session_sdes (RTPSession * sess, GstBuffer * buffer)
+{
+  GstRTCPPacket packet;
+
+  /* add SDES packet */
+  gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_SDES, &packet);
+
+  gst_rtcp_packet_sdes_add_item (&packet, sess->source->ssrc);
+  gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_CNAME,
+      strlen (sess->cname), (guint8 *) sess->cname);
+
+  /* other SDES items must only be added at regular intervals and only when the
+   * user requests to since it might be a privacy problem */
+#if 0
+  gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
+      strlen (sess->name), (guint8 *) sess->name);
+  gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
+      strlen (sess->tool), (guint8 *) sess->tool);
+#endif
+}
+
 /**
- * rtp_session_produce_rtcp:
+ * rtp_session_perform_reporting:
  * @sess: an #RTPSession
  *
  * Instruct the session manager to generate RTCP packets with current stats.
@@ -1019,8 +1403,39 @@ rtp_session_get_rtcp_interval (RTPSession * sess)
  * Returns: a #GstFlowReturn.
  */
 GstFlowReturn
-rtp_session_produce_rtcp (RTPSession * sess)
+rtp_session_perform_reporting (RTPSession * sess)
 {
-  /* FIXME: implement me */
-  return GST_FLOW_NOT_SUPPORTED;
+  GstFlowReturn result = GST_FLOW_OK;
+  ReportData data;
+
+  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
+
+  data.sess = sess;
+  data.rtcp = NULL;
+
+  RTP_SESSION_LOCK (sess);
+  /* loop over all known sources and do something */
+  g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
+      (GHFunc) session_report_blocks, &data);
+
+  /* add SDES for this source */
+  if (data.rtcp) {
+    session_sdes (sess, data.rtcp);
+    sess->stats.sent_rtcp = TRUE;
+  }
+
+  RTP_SESSION_UNLOCK (sess);
+
+  /* push out the RTCP packet */
+  if (data.rtcp) {
+    /* close the RTCP packet */
+    gst_rtcp_buffer_end (data.rtcp);
+
+    if (sess->callbacks.send_rtcp)
+      result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
+          sess->user_data);
+    else
+      gst_buffer_unref (data.rtcp);
+  }
+  return result;
 }
diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h
index 46062c99..493387fa 100644
--- a/gst/rtpmanager/rtpsession.h
+++ b/gst/rtpmanager/rtpsession.h
@@ -143,9 +143,24 @@ struct _RTPSession {
   GMutex       *lock;
 
   guint         header_len;
+  guint         mtu;
 
   RTPSource    *source;
-  GHashTable   *ssrcs;
+
+  /* info for creating reports */
+  gchar        *cname;
+  gchar        *name;
+  gchar        *email;
+  gchar        *phone;
+  gchar        *location;
+  gchar        *tool;
+  gchar        *note;
+
+  /* for sender/receiver counting */
+  guint32       key;
+  guint32       mask_idx;
+  guint32       mask;
+  GHashTable   *ssrcs[32];
   GHashTable   *cnames;
   guint         total_sources;
 
@@ -184,6 +199,21 @@ gdouble         rtp_session_get_bandwidth          (RTPSession *sess);
 void            rtp_session_set_rtcp_fraction      (RTPSession *sess, gdouble fraction);
 gdouble         rtp_session_get_rtcp_fraction      (RTPSession *sess);
 
+void            rtp_session_set_cname              (RTPSession *sess, const gchar *cname);
+gchar*          rtp_session_get_cname              (RTPSession *sess);
+void            rtp_session_set_name               (RTPSession *sess, const gchar *name);
+gchar*          rtp_session_get_name               (RTPSession *sess);
+void            rtp_session_set_email              (RTPSession *sess, const gchar *email);
+gchar*          rtp_session_get_email              (RTPSession *sess);
+void            rtp_session_set_phone              (RTPSession *sess, const gchar *phone);
+gchar*          rtp_session_get_phone              (RTPSession *sess);
+void            rtp_session_set_location           (RTPSession *sess, const gchar *location);
+gchar*          rtp_session_get_location           (RTPSession *sess);
+void            rtp_session_set_tool               (RTPSession *sess, const gchar *tool);
+gchar*          rtp_session_get_tool               (RTPSession *sess);
+void            rtp_session_set_note               (RTPSession *sess, const gchar *note);
+gchar*          rtp_session_get_note               (RTPSession *sess);
+
 /* handling sources */
 gboolean        rtp_session_add_source             (RTPSession *sess, RTPSource *src);
 gint            rtp_session_get_num_sources        (RTPSession *sess);
@@ -200,7 +230,7 @@ GstFlowReturn   rtp_session_process_rtcp           (RTPSession *sess, GstBuffer
 GstFlowReturn   rtp_session_send_rtp               (RTPSession *sess, GstBuffer *buffer);
 
 /* get interval for next RTCP interval */
-gdouble         rtp_session_get_rtcp_interval      (RTPSession *sess);
-GstFlowReturn   rtp_session_produce_rtcp           (RTPSession *sess);
+gdouble         rtp_session_get_reporting_interval (RTPSession *sess);
+GstFlowReturn   rtp_session_perform_reporting      (RTPSession *sess);
 
 #endif /* __RTP_SESSION_H__ */
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 36f54381..43acb084 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -70,6 +70,7 @@ rtp_source_init (RTPSource * src)
   src->clock_rate = -1;
   src->packets = g_queue_new ();
 
+  src->stats.cycles = -1;
   src->stats.jitter = 0;
   src->stats.transit = -1;
   src->stats.curr_sr = 0;
@@ -279,6 +280,20 @@ no_clock_rate:
   }
 }
 
+static void
+init_seq (RTPSource * src, guint16 seq)
+{
+  src->stats.base_seq = seq;
+  src->stats.max_seq = seq;
+  src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
+  src->stats.cycles = 0;
+  src->stats.packets_received = 0;
+  src->stats.octets_received = 0;
+  src->stats.bytes_received = 0;
+  src->stats.prev_received = 0;
+  src->stats.prev_expected = 0;
+}
+
 /**
  * rtp_source_process_rtp:
  * @src: an #RTPSource
@@ -293,58 +308,108 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
     RTPArrivalStats * arrival)
 {
   GstFlowReturn result = GST_FLOW_OK;
+  guint16 seqnr, udelta;
+  RTPSourceStats *stats;
 
   g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
   g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
 
+  stats = &src->stats;
+
+  seqnr = gst_rtp_buffer_get_seq (buffer);
+
+  if (stats->cycles == -1) {
+    GST_DEBUG ("first buffer");
+    /* first time we heard of this source */
+    init_seq (src, seqnr);
+    src->stats.max_seq = seqnr - 1;
+    src->probation = RTP_DEFAULT_PROBATION;
+  }
+
+  udelta = seqnr - stats->max_seq;
+
   /* if we are still on probation, check seqnum */
   if (src->probation) {
-    guint16 seqnr, expected;
+    guint16 expected;
 
-    expected = src->stats.max_seqnr + 1;
+    expected = src->stats.max_seq + 1;
 
     /* when in probation, we require consecutive seqnums */
-    seqnr = gst_rtp_buffer_get_seq (buffer);
     if (seqnr == expected) {
       /* expected packet */
-      src->probation--;
-      src->stats.max_seqnr = seqnr;
-
       GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
+      src->probation--;
+      src->stats.max_seq = seqnr;
+      if (src->probation == 0) {
+        GST_DEBUG ("probation done!", src->probation);
+        init_seq (src, seqnr);
+      } else {
+        GstBuffer *q;
+
+        GST_DEBUG ("probation %d: queue buffer", src->probation);
+        /* when still in probation, keep packets in a list. */
+        g_queue_push_tail (src->packets, buffer);
+        /* remove packets from queue if there are too many */
+        while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
+          q = g_queue_pop_head (src->packets);
+          gst_object_unref (q);
+        }
+        goto done;
+      }
     } else {
       GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
       src->probation = RTP_DEFAULT_PROBATION;
-      src->stats.max_seqnr = seqnr;
+      src->stats.max_seq = seqnr;
+      goto done;
     }
-  }
-  if (src->probation) {
-    GstBuffer *q;
-
-    GST_DEBUG ("probation %d: queue buffer", src->probation);
-    /* when still in probation, keep packets in a list. */
-    g_queue_push_tail (src->packets, buffer);
-    /* remove packets from queue if there are too many */
-    while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
-      q = g_queue_pop_head (src->packets);
-      gst_object_unref (q);
+  } else if (udelta < RTP_MAX_DROPOUT) {
+    /* in order, with permissible gap */
+    if (seqnr < stats->max_seq) {
+      /* sequence number wrapped - count another 64K cycle. */
+      stats->cycles++;
+    }
+    stats->max_seq = seqnr;
+  } else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
+    /* the sequence number made a very large jump */
+    if (seqnr == stats->bad_seq) {
+      /* two sequential packets -- assume that the other side
+       * restarted without telling us so just re-sync
+       * (i.e., pretend this was the first packet).  */
+      init_seq (src, seqnr);
+    } else {
+      /* unacceptable jump */
+      stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
+      goto bad_sequence;
     }
   } else {
-    /* we are not in probation */
-    src->stats.octetsreceived += arrival->payload_len;
-    src->stats.bytesreceived += arrival->bytes;
-    src->stats.packetsreceived++;
-    src->is_sender = TRUE;
+    /* duplicate or reordered packet, will be filtered by jitterbuffer. */
+  }
 
-    GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
-        src->stats.packetsreceived, src->stats.octetsreceived);
+  src->stats.octets_received += arrival->payload_len;
+  src->stats.bytes_received += arrival->bytes;
+  src->stats.packets_received++;
+  /* the source that sent the packet must be a sender */
+  src->is_sender = TRUE;
+  src->validated = TRUE;
 
-    /* calculate jitter */
-    calculate_jitter (src, buffer, arrival);
+  GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
+      src->stats.packets_received, src->stats.octets_received);
 
-    /* we're ready to push the RTP packet now */
-    result = push_packet (src, buffer);
-  }
+  /* calculate jitter */
+  calculate_jitter (src, buffer, arrival);
+
+  /* we're ready to push the RTP packet now */
+  result = push_packet (src, buffer);
+
+done:
   return result;
+
+  /* ERRORS */
+bad_sequence:
+  {
+    GST_WARNING ("unacceptable seqnum received");
+    return GST_FLOW_OK;
+  }
 }
 
 /**
@@ -424,6 +489,9 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
   curridx = src->stats.curr_sr ^ 1;
   curr = &src->stats.sr[curridx];
 
+  /* this is a sender now */
+  src->is_sender = TRUE;
+
   /* update current */
   curr->is_valid = TRUE;
   curr->ntptime = ntptime;
diff --git a/gst/rtpmanager/rtpsource.h b/gst/rtpmanager/rtpsource.h
index d4ae6f55..2f997fb5 100644
--- a/gst/rtpmanager/rtpsource.h
+++ b/gst/rtpmanager/rtpsource.h
@@ -31,6 +31,10 @@
 #define RTP_NO_PROBATION        0
 #define RTP_DEFAULT_PROBATION   2
 
+#define RTP_SEQ_MOD          (1 << 16)
+#define RTP_MAX_DROPOUT      3000
+#define RTP_MAX_MISORDER     100
+
 typedef struct _RTPSource RTPSource;
 typedef struct _RTPSourceClass RTPSourceClass;
 
@@ -69,7 +73,8 @@ typedef struct _RTPSourceClass RTPSourceClass;
  *
  * Returns: a #GstFlowReturn.
  */
-typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, gpointer user_data);
+typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, 
+	gpointer user_data);
 
 /**
  * RTPSourceClockRate:
@@ -106,19 +111,23 @@ struct _RTPSource {
   GObject       object;
 
   /*< private >*/
-  RTPSourceCallbacks callbacks;
-  gpointer           user_data;
-
   guint32       ssrc;
-  gchar        *cname;
+
   gint          probation;
   gboolean      validated;
-  gboolean      received_bye;
-  gchar        *bye_reason;
-
   gboolean      is_csrc;
   gboolean      is_sender;
 
+  gchar        *cname;
+  gchar        *name;
+  gchar        *email;
+  gchar        *phone;
+  gchar        *location;
+  gchar        *tool;
+  gchar        *note;
+  gboolean      received_bye;
+  gchar        *bye_reason;
+
   gboolean      have_rtp_from;
   GstNetAddress rtp_from;
   gboolean      have_rtcp_from;
@@ -129,6 +138,9 @@ struct _RTPSource {
 
   GQueue       *packets;
 
+  RTPSourceCallbacks callbacks;
+  gpointer           user_data;
+
   RTPSourceStats stats;
 };
 
@@ -147,6 +159,7 @@ void            rtp_source_set_as_csrc    (RTPSource *src);
 void            rtp_source_set_rtp_from   (RTPSource *src, GstNetAddress *address);
 void            rtp_source_set_rtcp_from  (RTPSource *src, GstNetAddress *address);
 
+/* handling RTP */
 GstFlowReturn   rtp_source_process_rtp    (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
 
 GstFlowReturn   rtp_source_send_rtp       (RTPSource *src, GstBuffer *buffer);
diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c
index b9076eac..456ed15f 100644
--- a/gst/rtpmanager/rtpstats.c
+++ b/gst/rtpmanager/rtpstats.c
@@ -65,7 +65,7 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
   GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
       senders, receivers, avg_rtcp, sfraction);
 
-  if (sfraction <= stats->sender_fraction) {
+  if (senders > 0 && sfraction <= stats->sender_fraction) {
     if (sender) {
       interval =
           (avg_rtcp * senders) / (stats->sender_fraction *
diff --git a/gst/rtpmanager/rtpstats.h b/gst/rtpmanager/rtpstats.h
index 66aa7bf7..64321427 100644
--- a/gst/rtpmanager/rtpstats.h
+++ b/gst/rtpmanager/rtpstats.h
@@ -92,14 +92,23 @@ typedef struct {
  * Stats about a source.
  */
 typedef struct {
-  guint64      packetsreceived;
-  guint64      prevpacketsreceived;
-  guint64      octetsreceived;
-  guint64      bytesreceived;
-  guint16      max_seqnr;
+  guint64      packets_received;
+  guint64      octets_received;
+  guint64      bytes_received;
+
+  guint32      prev_expected;
+  guint32      prev_received;
+
+  guint16      max_seq;
+  guint32      cycles;
+  guint32      base_seq;
+  guint32      bad_seq;
   guint32      transit;
   guint32      jitter;
 
+  guint64      packets_sent;
+  guint64      octets_sent;
+
   /* when we received stuff */
   GstClockTime prev_rtptime;
   GstClockTime prev_rtcptime;