diff --git a/docs/libs/gst-plugins-base-libs-docs.sgml b/docs/libs/gst-plugins-base-libs-docs.sgml
index 6b6c9966ff..8149d29ab9 100644
--- a/docs/libs/gst-plugins-base-libs-docs.sgml
+++ b/docs/libs/gst-plugins-base-libs-docs.sgml
@@ -110,6 +110,7 @@
         <filename>gstreamer-plugins-base-&GST_API_VERSION;.pc</filename> and adding
         <filename>-lgstrtp-&GST_API_VERSION;</filename> to the library flags.
       </para>
+      <xi:include href="xml/gstrtpmeta.xml" />
       <xi:include href="xml/gstrtpbaseaudiopayload.xml" />
       <xi:include href="xml/gstrtpbasedepayload.xml" />
       <xi:include href="xml/gstrtpbasepayload.xml" />
diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt
index 5256114fff..687c1e67d3 100644
--- a/docs/libs/gst-plugins-base-libs-sections.txt
+++ b/docs/libs/gst-plugins-base-libs-sections.txt
@@ -1374,6 +1374,9 @@ GST_RTP_BASE_DEPAYLOAD_SRCPAD
 gst_rtp_base_depayload_push
 gst_rtp_base_depayload_push_list
 
+gst_rtp_base_depayload_is_source_info_enabled
+gst_rtp_base_depayload_set_source_info_enabled
+
 <SUBSECTION Standard>
 GstRTPBaseDepayloadPrivate
 GST_TYPE_RTP_BASE_DEPAYLOAD
@@ -1404,6 +1407,11 @@ gst_rtp_base_payload_push
 gst_rtp_base_payload_push_list
 gst_rtp_base_payload_set_options
 gst_rtp_base_payload_set_outcaps
+
+gst_rtp_base_payload_allocate_output_buffer
+gst_rtp_base_payload_get_source_count
+gst_rtp_base_payload_is_source_info_enabled
+gst_rtp_base_payload_set_source_info_enabled
 <SUBSECTION Standard>
 GST_TYPE_RTP_BASE_PAYLOAD
 GST_RTP_BASE_PAYLOAD
@@ -1414,6 +1422,23 @@ GST_IS_RTP_BASE_PAYLOAD_CLASS
 gst_rtp_base_payload_get_type
 </SECTION>
 
+<SECTION>
+<FILE>gstrtpmeta</FILE>
+<INCLUDE>gst/rtp/rtp.h</INCLUDE>
+GstRTPSourceMeta
+gst_buffer_add_rtp_source_meta
+gst_buffer_get_rtp_source_meta
+gst_rtp_source_meta_append_csrc
+gst_rtp_source_meta_get_info
+gst_rtp_source_meta_get_source_count
+gst_rtp_source_meta_set_ssrc
+GST_RTP_SOURCE_META_MAX_CSRC_COUNT
+<SUBSECTION Standard>
+gst_rtp_source_meta_api_get_type
+GST_RTP_SOURCE_META_API_TYPE
+GST_RTP_SOURCE_META_INFO
+</SECTION>
+
 <SECTION>
 <FILE>gstrtcpbuffer</FILE>
 <INCLUDE>gst/rtp/rtp.h</INCLUDE>
diff --git a/gst-libs/gst/rtp/Makefile.am b/gst-libs/gst/rtp/Makefile.am
index 4dce448945..f6b9963247 100644
--- a/gst-libs/gst/rtp/Makefile.am
+++ b/gst-libs/gst/rtp/Makefile.am
@@ -10,7 +10,8 @@ libgstrtpinclude_HEADERS = \
 			   gstrtphdrext.h \
 			gstrtpbaseaudiopayload.h \
 			gstrtpbasepayload.h \
-			gstrtpbasedepayload.h
+			gstrtpbasedepayload.h \
+			gstrtpmeta.h
 
 lib_LTLIBRARIES = libgstrtp-@GST_API_VERSION@.la
 
@@ -20,7 +21,8 @@ libgstrtp_@GST_API_VERSION@_la_SOURCES = gstrtpbuffer.c \
 			   	gstrtphdrext.c \
 			        gstrtpbaseaudiopayload.c \
 				gstrtpbasepayload.c \
-				gstrtpbasedepayload.c
+				gstrtpbasedepayload.c \
+				gstrtpmeta.c
 
 built_sources = gstrtp-enumtypes.c
 built_headers = gstrtp-enumtypes.h
diff --git a/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
index 52a4c095a3..3787448667 100644
--- a/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
+++ b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
@@ -110,6 +110,7 @@ struct _GstRTPBaseAudioPayloadPrivate
   guint cached_max_length;
   guint cached_ptime_multiple;
   guint cached_align;
+  guint cached_csrc_count;
 
   gboolean buffer_list;
 };
@@ -453,7 +454,8 @@ gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
       payload_len, GST_TIME_ARGS (timestamp));
 
   /* create buffer to hold the payload */
-  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
+      payload_len, 0, 0);
 
   /* copy payload */
   gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
@@ -519,7 +521,7 @@ gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
       payload_len, GST_TIME_ARGS (timestamp));
 
   /* create just the RTP header buffer */
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
   /* set metadata */
   gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
@@ -628,7 +630,7 @@ gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
 
 
     /* create buffer to hold the payload */
-    outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+    outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
     paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
 
@@ -653,8 +655,8 @@ gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
  * mtu and min/max_ptime values. We cache those so that we don't have to redo
  * all the calculations */
 static gboolean
-gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
-    basepayload, guint * min_payload_len, guint * max_payload_len,
+gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload * basepayload,
+    guint csrc_count, guint * min_payload_len, guint * max_payload_len,
     guint * align)
 {
   GstRTPBaseAudioPayload *payload;
@@ -672,13 +674,16 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
 
   mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
 
-  /* check cached values */
+  /* check cached values. Since csrc_count may vary for each packet, we only
+   * check whether the new value exceeds the cached value and thus result in
+   * smaller payload. */
   if (G_LIKELY (priv->cached_mtu == mtu
           && priv->cached_ptime_multiple ==
           basepayload->ptime_multiple
           && priv->cached_ptime == basepayload->ptime
           && priv->cached_max_ptime == basepayload->max_ptime
-          && priv->cached_min_ptime == basepayload->min_ptime)) {
+          && priv->cached_min_ptime == basepayload->min_ptime
+          && priv->cached_csrc_count >= csrc_count)) {
     /* if nothing changed, return cached values */
     *min_payload_len = priv->cached_min_length;
     *max_payload_len = priv->cached_max_length;
@@ -697,7 +702,7 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
     maxptime_octets = G_MAXUINT;
   }
   /* MTU max */
-  max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
+  max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, csrc_count);
   /* round down to alignment */
   max_mtu = ALIGN_DOWN (max_mtu, *align);
 
@@ -733,6 +738,7 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
   priv->cached_min_length = *min_payload_len;
   priv->cached_max_length = *max_payload_len;
   priv->cached_align = *align;
+  priv->cached_csrc_count = csrc_count;
 
   return TRUE;
 }
@@ -875,8 +881,9 @@ gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
     }
   }
 
-  if (!gst_rtp_base_audio_payload_get_lengths (basepayload, &min_payload_len,
-          &max_payload_len, &align))
+  if (!gst_rtp_base_audio_payload_get_lengths (basepayload,
+          gst_rtp_base_payload_get_source_count (basepayload, buffer),
+          &min_payload_len, &max_payload_len, &align))
     goto config_error;
 
   GST_DEBUG_OBJECT (payload,
diff --git a/gst-libs/gst/rtp/gstrtpbasedepayload.c b/gst-libs/gst/rtp/gstrtpbasedepayload.c
index 9047a011b3..c9c5204e1c 100644
--- a/gst-libs/gst/rtp/gstrtpbasedepayload.c
+++ b/gst-libs/gst/rtp/gstrtpbasedepayload.c
@@ -30,6 +30,7 @@
 #endif
 
 #include "gstrtpbasedepayload.h"
+#include "gstrtpmeta.h"
 
 GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
 #define GST_CAT_DEFAULT (rtpbasedepayload_debug)
@@ -57,6 +58,9 @@ struct _GstRTPBaseDepayloadPrivate
   GstCaps *last_caps;
   GstEvent *segment_event;
   guint32 segment_seqnum;       /* Note: this is a GstEvent seqnum */
+
+  gboolean source_info;
+  GstBuffer *input_buffer;
 };
 
 /* Filter signals and args */
@@ -66,10 +70,13 @@ enum
   LAST_SIGNAL
 };
 
+#define DEFAULT_SOURCE_INFO FALSE
+
 enum
 {
   PROP_0,
   PROP_STATS,
+  PROP_SOURCE_INFO,
   PROP_LAST
 };
 
@@ -183,6 +190,18 @@ gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass)
       g_param_spec_boxed ("stats", "Statistics", "Various statistics",
           GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstRTPBaseDepayload:source-info:
+   *
+   * Add RTP source information found in RTP header as meta to output buffer.
+   *
+   * Since: 1.16
+   **/
+  g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
+      g_param_spec_boolean ("source-info", "RTP source information",
+          "Add RTP source information as buffer meta",
+          DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
+
   gstelement_class->change_state = gst_rtp_base_depayload_change_state;
 
   klass->packet_lost = gst_rtp_base_depayload_packet_lost;
@@ -231,6 +250,7 @@ gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
   priv->dts = -1;
   priv->pts = -1;
   priv->duration = -1;
+  priv->source_info = DEFAULT_SOURCE_INFO;
 
   gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
 }
@@ -442,6 +462,8 @@ gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
     filter->need_newsegment = FALSE;
   }
 
+  priv->input_buffer = in;
+
   if (process_rtp_packet_func != NULL) {
     out_buf = process_rtp_packet_func (filter, &rtp);
     gst_rtp_buffer_unmap (&rtp);
@@ -458,6 +480,7 @@ gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
   }
 
   gst_buffer_unref (in);
+  priv->input_buffer = NULL;
 
   return ret;
 
@@ -727,6 +750,30 @@ create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
   return event;
 }
 
+static void
+add_rtp_source_meta (GstBuffer * outbuf, GstBuffer * rtpbuf)
+{
+  GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+  GstRTPSourceMeta *meta;
+  guint32 ssrc;
+
+  if (!gst_rtp_buffer_map (rtpbuf, GST_MAP_READ, &rtp))
+    return;
+
+  ssrc = gst_rtp_buffer_get_ssrc (&rtp);
+  meta = gst_buffer_add_rtp_source_meta (outbuf, &ssrc, NULL, 0);
+  if (meta != NULL) {
+    gint i;
+    gint csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
+    for (i = 0; i < csrc_count; i++) {
+      guint32 csrc = gst_rtp_buffer_get_csrc (&rtp, i);
+      gst_rtp_source_meta_append_csrc (meta, &csrc, 1);
+    }
+  }
+
+  gst_rtp_buffer_unmap (&rtp);
+}
+
 static gboolean
 set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
 {
@@ -759,6 +806,9 @@ set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
   priv->dts = GST_CLOCK_TIME_NONE;
   priv->duration = GST_CLOCK_TIME_NONE;
 
+  if (priv->source_info && priv->input_buffer)
+    add_rtp_source_meta (*buffer, priv->input_buffer);
+
   return TRUE;
 }
 
@@ -959,7 +1009,15 @@ static void
 gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
+  GstRTPBaseDepayload *depayload;
+
+  depayload = GST_RTP_BASE_DEPAYLOAD (object);
+
   switch (prop_id) {
+    case PROP_SOURCE_INFO:
+      gst_rtp_base_depayload_set_source_info_enabled (depayload,
+          g_value_get_boolean (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -979,8 +1037,44 @@ gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
       g_value_take_boxed (value,
           gst_rtp_base_depayload_create_stats (depayload));
       break;
+    case PROP_SOURCE_INFO:
+      g_value_set_boolean (value,
+          gst_rtp_base_depayload_is_source_info_enabled (depayload));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
 }
+
+/**
+ * gst_rtp_base_depayload_set_source_info_enabled:
+ * @depayload: a #GstRTPBaseDepayload
+ * @enable: whether to add meta about RTP sources to buffer
+ *
+ * Enable or disable adding #GstRTPSourceMeta to depayloaded buffers.
+ *
+ * Since: 1.16
+ **/
+void
+gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
+    gboolean enable)
+{
+  depayload->priv->source_info = enable;
+}
+
+/**
+ * gst_rtp_base_depayload_is_source_info_enabled:
+ * @depayload: a #GstRTPBaseDepayload
+ *
+ * Queries whether #GstRTPSourceMeta will be added to depayloaded buffers.
+ *
+ * Returns: %TRUE if source-info is enabled.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_base_depayload_is_source_info_enabled (GstRTPBaseDepayload * depayload)
+{
+  return depayload->priv->source_info;
+}
diff --git a/gst-libs/gst/rtp/gstrtpbasedepayload.h b/gst-libs/gst/rtp/gstrtpbasedepayload.h
index f409ff9743..42af745071 100644
--- a/gst-libs/gst/rtp/gstrtpbasedepayload.h
+++ b/gst-libs/gst/rtp/gstrtpbasedepayload.h
@@ -120,6 +120,13 @@ GstFlowReturn   gst_rtp_base_depayload_push       (GstRTPBaseDepayload *filter,
 GST_RTP_API
 GstFlowReturn   gst_rtp_base_depayload_push_list  (GstRTPBaseDepayload *filter, GstBufferList *out_list);
 
+GST_RTP_API
+gboolean        gst_rtp_base_depayload_is_source_info_enabled  (GstRTPBaseDepayload * depayload);
+
+GST_RTP_API
+void            gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
+                                                                gboolean enable);
+
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseDepayload, gst_object_unref)
diff --git a/gst-libs/gst/rtp/gstrtpbasepayload.c b/gst-libs/gst/rtp/gstrtpbasepayload.c
index b518ac735f..23129d9ea8 100644
--- a/gst-libs/gst/rtp/gstrtpbasepayload.c
+++ b/gst-libs/gst/rtp/gstrtpbasepayload.c
@@ -29,6 +29,7 @@
 #include <gst/rtp/gstrtpbuffer.h>
 
 #include "gstrtpbasepayload.h"
+#include "gstrtpmeta.h"
 
 GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug);
 #define GST_CAT_DEFAULT (rtpbasepayload_debug)
@@ -44,6 +45,9 @@ struct _GstRTPBasePayloadPrivate
 
   gboolean pt_set;
 
+  gboolean source_info;
+  GstBuffer *input_meta_buffer;
+
   guint64 base_offset;
   gint64 base_rtime;
   guint64 base_rtime_hz;
@@ -84,6 +88,7 @@ enum
 #define DEFAULT_PERFECT_RTPTIME         TRUE
 #define DEFAULT_PTIME_MULTIPLE          0
 #define DEFAULT_RUNNING_TIME            GST_CLOCK_TIME_NONE
+#define DEFAULT_SOURCE_INFO             FALSE
 
 enum
 {
@@ -100,6 +105,7 @@ enum
   PROP_PERFECT_RTPTIME,
   PROP_PTIME_MULTIPLE,
   PROP_STATS,
+  PROP_SOURCE_INFO,
   PROP_LAST
 };
 
@@ -299,6 +305,19 @@ gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
       g_param_spec_boxed ("stats", "Statistics", "Various statistics",
           GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstRTPBasePayload:source-info:
+   *
+   * Enable writing the CSRC field in allocated RTP header based on RTP source
+   * information found in the input buffer's #GstRTPSourceMeta.
+   *
+   * Since: 1.16
+   **/
+  g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
+      g_param_spec_boolean ("source-info", "RTP source information",
+          "Write CSRC based on buffer meta RTP source information",
+          DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
+
   gstelement_class->change_state = gst_rtp_base_payload_change_state;
 
   klass->get_caps = gst_rtp_base_payload_getcaps_default;
@@ -351,6 +370,7 @@ gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class)
   priv->ts_offset_random = (rtpbasepayload->ts_offset == -1);
   priv->ssrc_random = (rtpbasepayload->ssrc == -1);
   priv->pt_set = FALSE;
+  priv->source_info = DEFAULT_SOURCE_INFO;
 
   rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
   rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME;
@@ -633,6 +653,15 @@ gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
   if (!rtpbasepayload->priv->negotiated)
     goto not_negotiated;
 
+  if (rtpbasepayload->priv->source_info) {
+    /* Save a copy of meta (instead of taking an extra reference before
+     * handle_buffer) to make the meta available when allocating a output
+     * buffer. */
+    rtpbasepayload->priv->input_meta_buffer = gst_buffer_new ();
+    gst_buffer_copy_into (rtpbasepayload->priv->input_meta_buffer, buffer,
+        GST_BUFFER_COPY_META, 0, -1);
+  }
+
   if (gst_pad_check_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
     if (!gst_rtp_base_payload_negotiate (rtpbasepayload)) {
       gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload));
@@ -646,6 +675,8 @@ gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
 
   ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
 
+  gst_buffer_replace (&rtpbasepayload->priv->input_meta_buffer, NULL);
+
   return ret;
 
   /* ERRORS */
@@ -1156,6 +1187,25 @@ map_failed:
   }
 }
 
+static gboolean
+foreach_metadata_drop (GstBuffer * buffer, GstMeta ** meta, gpointer user_data)
+{
+  GType drop_api_type = (GType) GPOINTER_TO_INT (user_data);
+  const GstMetaInfo *info = (*meta)->info;
+
+  if (info->api == drop_api_type)
+    *meta = NULL;
+
+  return TRUE;
+}
+
+static gboolean
+filter_meta (GstBuffer ** buffer, guint idx, gpointer user_data)
+{
+  return gst_buffer_foreach_meta (*buffer, foreach_metadata_drop,
+      GINT_TO_POINTER (GST_RTP_SOURCE_META_API_TYPE));
+}
+
 /* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer
  * before the buffer is pushed. */
 static GstFlowReturn
@@ -1252,11 +1302,14 @@ gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
   }
 
   /* set ssrc, payload type, seq number, caps and rtptime */
+  /* remove unwanted meta */
   if (is_list) {
     gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), set_headers, &data);
+    gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), filter_meta, NULL);
   } else {
     GstBuffer *buf = GST_BUFFER_CAST (obj);
     set_headers (&buf, 0, &data);
+    filter_meta (&buf, 0, NULL);
   }
 
   priv->next_seqnum = data.seqnum;
@@ -1267,8 +1320,8 @@ gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
       (is_list) ? -1 : gst_buffer_get_size (GST_BUFFER (obj)),
       payload->seqnum, data.rtptime, GST_TIME_ARGS (data.pts));
 
-  if (g_atomic_int_compare_and_exchange (&payload->
-          priv->notified_first_timestamp, 1, 0)) {
+  if (g_atomic_int_compare_and_exchange (&payload->priv->
+          notified_first_timestamp, 1, 0)) {
     g_object_notify (G_OBJECT (payload), "timestamp");
     g_object_notify (G_OBJECT (payload), "seqnum");
   }
@@ -1351,6 +1404,62 @@ gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer)
   return res;
 }
 
+/**
+ * gst_rtp_base_payload_allocate_output_buffer:
+ * @payload: a #GstRTPBasePayload
+ * @payload_len: the length of the payload
+ * @pad_len: the amount of padding
+ * @csrc_count: the minimum number of CSRC entries
+ *
+ * Allocate a new #GstBuffer with enough data to hold an RTP packet with
+ * minimum @csrc_count CSRCs, a payload length of @payload_len and padding of
+ * @pad_len. If @payload has #GstRTPBasePayload:source-info %TRUE additional
+ * CSRCs may be allocated and filled with RTP source information.
+ *
+ * Returns: A newly allocated buffer that can hold an RTP packet with given
+ * parameters.
+ *
+ * Since: 1.16
+ */
+GstBuffer *
+gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
+    guint payload_len, guint8 pad_len, guint8 csrc_count)
+{
+  GstBuffer *buffer = NULL;
+
+  if (payload->priv->input_meta_buffer != NULL) {
+    GstRTPSourceMeta *meta =
+        gst_buffer_get_rtp_source_meta (payload->priv->input_meta_buffer);
+    if (meta != NULL) {
+      guint total_csrc_count, idx, i;
+      GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+      total_csrc_count = csrc_count + meta->csrc_count +
+          (meta->ssrc_valid ? 1 : 0);
+      total_csrc_count = MIN (total_csrc_count, 15);
+      buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len,
+          total_csrc_count);
+
+      gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtp);
+
+      /* Skip CSRC fields requested by derived class and fill CSRCs from meta.
+       * Finally append the SSRC as a new CSRC. */
+      idx = csrc_count;
+      for (i = 0; i < meta->csrc_count && idx < 15; i++, idx++)
+        gst_rtp_buffer_set_csrc (&rtp, idx, meta->csrc[i]);
+      if (meta->ssrc_valid && idx < 15)
+        gst_rtp_buffer_set_csrc (&rtp, idx, meta->ssrc);
+
+      gst_rtp_buffer_unmap (&rtp);
+    }
+  }
+
+  if (buffer == NULL)
+    buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len, csrc_count);
+
+  return buffer;
+}
+
 static GstStructure *
 gst_rtp_base_payload_create_stats (GstRTPBasePayload * rtpbasepayload)
 {
@@ -1421,6 +1530,10 @@ gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
     case PROP_PTIME_MULTIPLE:
       rtpbasepayload->ptime_multiple = g_value_get_int64 (value);
       break;
+    case PROP_SOURCE_INFO:
+      gst_rtp_base_payload_set_source_info_enabled (rtpbasepayload,
+          g_value_get_boolean (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -1484,6 +1597,10 @@ gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
       g_value_take_boxed (value,
           gst_rtp_base_payload_create_stats (rtpbasepayload));
       break;
+    case PROP_SOURCE_INFO:
+      g_value_set_boolean (value,
+          gst_rtp_base_payload_is_source_info_enabled (rtpbasepayload));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -1551,3 +1668,68 @@ gst_rtp_base_payload_change_state (GstElement * element,
   }
   return ret;
 }
+
+/**
+ * gst_rtp_base_payload_set_source_info_enabled:
+ * @payload: a #GstRTPBasePayload
+ * @enable: whether to add contributing sources to RTP packets
+ *
+ * Enable or disable adding contributing sources to RTP packets from
+ * #GstRTPSourceMeta.
+ *
+ * Since: 1.16
+ **/
+void
+gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
+    gboolean enable)
+{
+  payload->priv->source_info = enable;
+}
+
+/**
+ * gst_rtp_base_payload_is_source_info_enabled:
+ * @payload: a #GstRTPBasePayload
+ *
+ * Queries whether the payloader will add contributing sources (CSRCs) to the
+ * RTP header from #GstRTPSourceMeta.
+ *
+ * Returns: %TRUE if source-info is enabled.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload)
+{
+  return payload->priv->source_info;
+}
+
+
+/**
+ * gst_rtp_base_payload_get_source_count:
+ * @payload: a #GstRTPBasePayload
+ * @buffer: (transfer none): a #GstBuffer, typically the buffer to payload
+ *
+ * Count the total number of RTP sources found in the meta of @buffer, which
+ * will be automically added by gst_rtp_base_payload_allocate_output_buffer().
+ * If #GstRTPBasePayload:source-info is %FALSE the count will be 0.
+ *
+ * Returns: The number of sources.
+ *
+ * Since: 1.16
+ **/
+guint
+gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
+    GstBuffer * buffer)
+{
+  guint count = 0;
+
+  g_return_val_if_fail (buffer != NULL, 0);
+
+  if (gst_rtp_base_payload_is_source_info_enabled (payload)) {
+    GstRTPSourceMeta *meta = gst_buffer_get_rtp_source_meta (buffer);
+    if (meta != NULL)
+      count = gst_rtp_source_meta_get_source_count (meta);
+  }
+
+  return count;
+}
diff --git a/gst-libs/gst/rtp/gstrtpbasepayload.h b/gst-libs/gst/rtp/gstrtpbasepayload.h
index b477e8a216..8b9e984aeb 100644
--- a/gst-libs/gst/rtp/gstrtpbasepayload.h
+++ b/gst-libs/gst/rtp/gstrtpbasepayload.h
@@ -172,6 +172,22 @@ GST_RTP_API
 GstFlowReturn   gst_rtp_base_payload_push_list          (GstRTPBasePayload *payload,
                                                          GstBufferList *list);
 
+GST_RTP_API
+GstBuffer *     gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
+                                                             guint payload_len, guint8 pad_len,
+                                                             guint8 csrc_count);
+
+GST_RTP_API
+void            gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
+                                                              gboolean enable);
+
+GST_RTP_API
+gboolean        gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload);
+
+GST_RTP_API
+guint           gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
+                                                       GstBuffer * buffer);
+
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBasePayload, gst_object_unref)
 #endif
diff --git a/gst-libs/gst/rtp/gstrtpmeta.c b/gst-libs/gst/rtp/gstrtpmeta.c
new file mode 100644
index 0000000000..9028db82db
--- /dev/null
+++ b/gst-libs/gst/rtp/gstrtpmeta.c
@@ -0,0 +1,229 @@
+/* GStreamer
+ * Copyright (C) <2016> Stian Selnes <stian@pexip.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstrtpmeta.h"
+#include <string.h>
+
+/**
+ * SECTION:gstrtpmeta
+ * @title: GstMeta for RTP
+ * @short_description: RTP related GstMeta
+ *
+ */
+
+/**
+ * gst_buffer_add_rtp_source_meta:
+ * @buffer: a #GstBuffer
+ * @ssrc: (allow-none) (transfer none): pointer to the SSRC
+ * @csrc: (allow-none) (transfer none): pointer to the CSRCs
+ * @csrc_count: number of elements in @csrc
+ *
+ * Attaches RTP source information to @buffer.
+ *
+ * Returns: (transfer none): the #GstRTPSourceMeta on @buffer.
+ *
+ * Since: 1.16
+ */
+GstRTPSourceMeta *
+gst_buffer_add_rtp_source_meta (GstBuffer * buffer, const guint32 * ssrc,
+    const guint * csrc, guint csrc_count)
+{
+  gint i;
+  GstRTPSourceMeta *meta;
+
+  g_return_val_if_fail (buffer != NULL, NULL);
+  g_return_val_if_fail (csrc_count <= GST_RTP_SOURCE_META_MAX_CSRC_COUNT, NULL);
+  g_return_val_if_fail (csrc_count == 0 || csrc != NULL, NULL);
+
+  meta = (GstRTPSourceMeta *) gst_buffer_add_meta (buffer,
+      GST_RTP_SOURCE_META_INFO, NULL);
+  if (!meta)
+    return NULL;
+
+  if (ssrc != NULL) {
+    meta->ssrc = *ssrc;
+    meta->ssrc_valid = TRUE;
+  } else {
+    meta->ssrc_valid = FALSE;
+  }
+
+  meta->csrc_count = csrc_count;
+  for (i = 0; i < csrc_count; i++) {
+    meta->csrc[i] = csrc[i];
+  }
+
+  return meta;
+}
+
+/**
+ * gst_buffer_get_rtp_source_meta:
+ * @buffer: a #GstBuffer
+ *
+ * Find the #GstRTPSourceMeta on @buffer.
+ *
+ * Returns: (transfer none): the #GstRTPSourceMeta or %NULL when there
+ * is no such metadata on @buffer.
+ *
+ * Since: 1.16
+ */
+GstRTPSourceMeta *
+gst_buffer_get_rtp_source_meta (GstBuffer * buffer)
+{
+  return (GstRTPSourceMeta *) gst_buffer_get_meta (buffer,
+      gst_rtp_source_meta_api_get_type ());
+}
+
+static gboolean
+gst_rtp_source_meta_transform (GstBuffer * dst, GstMeta * meta,
+    GstBuffer * src, GQuark type, gpointer data)
+{
+  if (GST_META_TRANSFORM_IS_COPY (type)) {
+    GstRTPSourceMeta *smeta = (GstRTPSourceMeta *) meta;
+    GstRTPSourceMeta *dmeta;
+    guint32 *ssrc = smeta->ssrc_valid ? &smeta->ssrc : NULL;
+
+    dmeta = gst_buffer_add_rtp_source_meta (dst, ssrc, smeta->csrc,
+        smeta->csrc_count);
+    if (dmeta == NULL)
+      return FALSE;
+  } else {
+    /* return FALSE, if transform type is not supported */
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+/**
+ * gst_rtp_source_meta_get_source_count:
+ * @meta: a #GstRTPSourceMeta
+ *
+ * Count the total number of RTP sources found in @meta, both SSRC and CSRC.
+ *
+ * Returns: The number of RTP sources
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta)
+{
+  /* Never return more than a count of 15 so that the returned value
+   * conveniently can be used as argument 'csrc_count' in
+   * gst_rtp_buffer-functions. */
+  guint ssrc_count = meta->ssrc_valid ? 1 : 0;
+  return MIN (meta->csrc_count + ssrc_count, 15);
+}
+
+/**
+ * gst_rtp_source_meta_set_ssrc:
+ * @meta: a #GstRTPSourceMeta
+ * @ssrc: (allow-none) (transfer none): pointer to the SSRC
+ *
+ * Sets @ssrc in @meta. If @ssrc is %NULL the ssrc of @meta will be unset.
+ *
+ * Returns: %TRUE on success, %FALSE otherwise.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_source_meta_set_ssrc (GstRTPSourceMeta * meta, guint32 * ssrc)
+{
+  if (ssrc != NULL) {
+    meta->ssrc = *ssrc;
+    meta->ssrc_valid = TRUE;
+  } else {
+    meta->ssrc_valid = FALSE;
+  }
+
+  return TRUE;
+}
+
+/**
+ * gst_rtp_source_meta_append_csrc:
+ * @meta: a #GstRTPSourceMeta
+ * @csrc: the csrcs to append
+ * @csrc_count: number of elements in @csrc
+ *
+ * Appends @csrc to the list of contributing sources in @meta.
+ *
+ * Returns: %TRUE if all elements in @csrc was added, %FALSE otherwise.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_source_meta_append_csrc (GstRTPSourceMeta * meta, const guint32 * csrc,
+    guint csrc_count)
+{
+  gint i;
+  guint new_csrc_count = meta->csrc_count + csrc_count;
+
+  if (new_csrc_count > GST_RTP_SOURCE_META_MAX_CSRC_COUNT)
+    return FALSE;
+
+  for (i = 0; i < csrc_count; i++)
+    meta->csrc[meta->csrc_count + i] = csrc[i];
+  meta->csrc_count = new_csrc_count;
+
+  return TRUE;
+}
+
+GType
+gst_rtp_source_meta_api_get_type (void)
+{
+  static volatile GType type = 0;
+  static const gchar *tags[] = { NULL };
+
+  if (g_once_init_enter (&type)) {
+    GType _type = gst_meta_api_type_register ("GstRTPSourceMetaAPI", tags);
+    g_once_init_leave (&type, _type);
+  }
+  return type;
+}
+
+static gboolean
+gst_rtp_source_meta_init (GstMeta * meta, gpointer params, GstBuffer * buffer)
+{
+  GstRTPSourceMeta *dmeta = (GstRTPSourceMeta *) meta;
+
+  dmeta->ssrc_valid = FALSE;
+  dmeta->csrc_count = 0;
+
+  return TRUE;
+}
+
+const GstMetaInfo *
+gst_rtp_source_meta_get_info (void)
+{
+  static const GstMetaInfo *rtp_source_meta_info = NULL;
+
+  if (g_once_init_enter (&rtp_source_meta_info)) {
+    const GstMetaInfo *meta = gst_meta_register (GST_RTP_SOURCE_META_API_TYPE,
+        "GstRTPSourceMeta",
+        sizeof (GstRTPSourceMeta),
+        gst_rtp_source_meta_init,
+        (GstMetaFreeFunction) NULL,
+        gst_rtp_source_meta_transform);
+    g_once_init_leave (&rtp_source_meta_info, meta);
+  }
+  return rtp_source_meta_info;
+}
diff --git a/gst-libs/gst/rtp/gstrtpmeta.h b/gst-libs/gst/rtp/gstrtpmeta.h
new file mode 100644
index 0000000000..956cff7ef9
--- /dev/null
+++ b/gst-libs/gst/rtp/gstrtpmeta.h
@@ -0,0 +1,79 @@
+/* GStreamer
+ * Copyright (C) <2016> Stian Selnes <stian@pexip.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_META_H__
+#define __GST_RTP_META_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+G_BEGIN_DECLS
+
+#define GST_RTP_SOURCE_META_API_TYPE  (gst_rtp_source_meta_api_get_type())
+#define GST_RTP_SOURCE_META_INFO  (gst_rtp_source_meta_get_info())
+typedef struct _GstRTPSourceMeta GstRTPSourceMeta;
+
+#define GST_RTP_SOURCE_META_MAX_CSRC_COUNT 15
+
+/**
+ * GstRTPSourceMeta:
+ * @meta: parent #GstMeta
+ * @ssrc: the SSRC
+ * @ssrc_valid: whether @ssrc is set and valid
+ * @csrc: (allow-none): pointer to the CSRCs
+ * @csrc_count: number of elements in @csrc
+ *
+ * Meta describing the source(s) of the buffer.
+ *
+ * Since: 1.16
+ */
+struct _GstRTPSourceMeta
+{
+  GstMeta meta;
+
+  guint32 ssrc;
+  gboolean ssrc_valid;
+  guint32 csrc[GST_RTP_SOURCE_META_MAX_CSRC_COUNT];
+  guint csrc_count;
+};
+
+GST_RTP_API
+GType               gst_rtp_source_meta_api_get_type     (void);
+
+GST_RTP_API
+GstRTPSourceMeta *  gst_buffer_add_rtp_source_meta       (GstBuffer * buf, const guint32 * ssrc,
+                                                          const guint32 * csrc, guint csrc_count);
+GST_RTP_API
+GstRTPSourceMeta *  gst_buffer_get_rtp_source_meta       (GstBuffer * buf);
+
+GST_RTP_API
+guint               gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta);
+
+GST_RTP_API
+gboolean            gst_rtp_source_meta_set_ssrc         (GstRTPSourceMeta * meta, guint32 * ssrc);
+
+GST_RTP_API
+gboolean            gst_rtp_source_meta_append_csrc      (GstRTPSourceMeta * meta,
+                                                          const guint32 * csrc, guint csrc_count);
+GST_RTP_API
+const GstMetaInfo * gst_rtp_source_meta_get_info         (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_META_H__ */
diff --git a/gst-libs/gst/rtp/meson.build b/gst-libs/gst/rtp/meson.build
index 25d3900ddc..f47ec65924 100644
--- a/gst-libs/gst/rtp/meson.build
+++ b/gst-libs/gst/rtp/meson.build
@@ -3,6 +3,7 @@ rtp_sources = [
   'gstrtcpbuffer.c',
   'gstrtppayloads.c',
   'gstrtphdrext.c',
+  'gstrtpmeta.c',
   'gstrtpbaseaudiopayload.c',
   'gstrtpbasepayload.c',
   'gstrtpbasedepayload.c'
@@ -16,6 +17,7 @@ rtp_headers = [
   'gstrtpbuffer.h',
   'gstrtpdefs.h',
   'gstrtphdrext.h',
+  'gstrtpmeta.h',
   'gstrtppayloads.h',
   'rtp-prelude.h',
   'rtp.h',
diff --git a/gst-libs/gst/rtp/rtp.h b/gst-libs/gst/rtp/rtp.h
index 546d4aef38..0e6633bd81 100644
--- a/gst-libs/gst/rtp/rtp.h
+++ b/gst-libs/gst/rtp/rtp.h
@@ -30,6 +30,7 @@
 #include <gst/rtp/gstrtpbaseaudiopayload.h>
 #include <gst/rtp/gstrtpbasepayload.h>
 #include <gst/rtp/gstrtpbasedepayload.h>
+#include <gst/rtp/gstrtpmeta.h>
 #include <gst/rtp/gstrtp-enumtypes.h>
 
 #endif /* __GST_RTP_H__ */
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index f1569a6b65..6223650e99 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -245,6 +245,7 @@ check_PROGRAMS = \
 	libs/rtp \
 	libs/rtpbasedepayload \
 	libs/rtpbasepayload \
+	libs/rtpmeta \
 	libs/rtsp \
 	libs/rtspconnection \
 	libs/sdp \
@@ -581,6 +582,13 @@ libs_rtpbasedepayload_LDADD = \
 	$(top_builddir)/gst-libs/gst/rtp/libgstrtp-@GST_API_VERSION@.la \
 	$(GST_BASE_LIBS) $(LDADD)
 
+libs_rtpmeta_CFLAGS = \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(AM_CFLAGS)
+libs_rtpmeta_LDADD = \
+	$(top_builddir)/gst-libs/gst/rtp/libgstrtp-@GST_API_VERSION@.la \
+	$(GST_BASE_LIBS) $(LDADD)
+
 libs_rtsp_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
 	$(AM_CFLAGS)
diff --git a/tests/check/libs/.gitignore b/tests/check/libs/.gitignore
index 64a1eb63e7..78cf13f11c 100644
--- a/tests/check/libs/.gitignore
+++ b/tests/check/libs/.gitignore
@@ -29,6 +29,7 @@ profile
 rtp
 rtpbasedepayload
 rtpbasepayload
+rtpmeta
 rtsp
 rtspconnection
 sdp
diff --git a/tests/check/libs/rtpbasedepayload.c b/tests/check/libs/rtpbasedepayload.c
index f24b0c46cb..01f1a372d2 100644
--- a/tests/check/libs/rtpbasedepayload.c
+++ b/tests/check/libs/rtpbasedepayload.c
@@ -23,8 +23,8 @@
 
 #include <gst/gst.h>
 #include <gst/check/gstcheck.h>
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtpbasedepayload.h>
+#include <gst/check/gstharness.h>
+#include <gst/rtp/rtp.h>
 
 #define DEFAULT_CLOCK_RATE (42)
 
@@ -317,22 +317,14 @@ validate_event (guint index, const gchar * name, const gchar * field, ...)
   va_end (var_args);
 }
 
-#define push_rtp_buffer(state, field, ...) \
-	push_rtp_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
-#define push_rtp_buffer_fails(state, error, field, ...) \
-        push_rtp_buffer_full ((state), (error), (field), __VA_ARGS__)
-
 static void
-push_rtp_buffer_full (State * state, GstFlowReturn expected,
-    const gchar * field, ...)
+rtp_buffer_set_valist (GstBuffer * buf, const gchar * field, va_list var_args,
+    gboolean * extra_ref_)
 {
-  GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
   GstRTPBuffer rtp = { NULL };
   gboolean mapped = FALSE;
   gboolean extra_ref = FALSE;
-  va_list var_args;
 
-  va_start (var_args, field);
   while (field) {
     if (!g_strcmp0 (field, "pts")) {
       GstClockTime pts = va_arg (var_args, GstClockTime);
@@ -366,13 +358,18 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
         gst_rtp_buffer_set_ssrc (&rtp, ssrc);
       } else if (!g_strcmp0 (field, "extra-ref")) {
         extra_ref = va_arg (var_args, gboolean);
+        if (extra_ref_)
+          *extra_ref_ = extra_ref;
+      } else if (!g_strcmp0 (field, "csrc")) {
+        guint idx = va_arg (var_args, guint);
+        guint csrc = va_arg (var_args, guint);
+        gst_rtp_buffer_set_csrc (&rtp, idx, csrc);
       } else {
         fail ("test cannot set unknown buffer field '%s'", field);
       }
     }
     field = va_arg (var_args, const gchar *);
   }
-  va_end (var_args);
 
   if (mapped) {
     gst_rtp_buffer_unmap (&rtp);
@@ -380,6 +377,34 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
 
   if (extra_ref)
     gst_buffer_ref (buf);
+}
+
+static void
+rtp_buffer_set (GstBuffer * buf, const gchar * field, ...)
+{
+  va_list var_args;
+
+  va_start (var_args, field);
+  rtp_buffer_set_valist (buf, field, var_args, NULL);
+  va_end (var_args);
+}
+
+#define push_rtp_buffer(state, field, ...) \
+    push_rtp_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
+#define push_rtp_buffer_fails(state, error, field, ...) \
+    push_rtp_buffer_full ((state), (error), (field), __VA_ARGS__)
+
+static void
+push_rtp_buffer_full (State * state, GstFlowReturn expected,
+    const gchar * field, ...)
+{
+  GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  va_list var_args;
+  gboolean extra_ref = FALSE;
+
+  va_start (var_args, field);
+  rtp_buffer_set_valist (buf, field, var_args, &extra_ref);
+  va_end (var_args);
 
   fail_unless_equals_int (gst_pad_push (state->srcpad, buf), expected);
 
@@ -388,7 +413,7 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
 }
 
 #define push_buffer(state, field, ...) \
-	push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
+    push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
 
 static void
 push_buffer_full (State * state, GstFlowReturn expected,
@@ -1237,7 +1262,64 @@ GST_START_TEST (rtp_base_depayload_clock_base_test)
   destroy_depayloader (state);
 }
 
-GST_END_TEST static Suite *
+GST_END_TEST
+/* basedepayloader has a property source-info that will add
+ * GstRTPSourceMeta to the output buffer with RTP source information, such as
+ * SSRC and CSRCs. The is useful for letting downstream know about the origin
+ * of the stream. */
+GST_START_TEST (rtp_base_depayload_source_info_test)
+{
+  GstHarness *h;
+  GstRtpDummyDepay *depay;
+  GstBuffer *buffer;
+  GstRTPSourceMeta *meta;
+  guint seq = 0;
+
+  depay = rtp_dummy_depay_new ();
+  h = gst_harness_new_with_element (GST_ELEMENT_CAST (depay), "sink", "src");
+  gst_harness_set_src_caps_str (h, "application/x-rtp");
+
+  /* Property enabled should always add meta, also when there is only SSRC and
+   * no CSRC. */
+  g_object_set (depay, "source-info", TRUE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, NULL);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  fail_unless ((meta = gst_buffer_get_rtp_source_meta (buffer)));
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, 0x11);
+  fail_unless_equals_int (meta->csrc_count, 0);
+  gst_buffer_unref (buffer);
+
+  /* Both SSRC and CSRC should be added to the meta */
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 2);
+  rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, "csrc", 0, 0x22,
+      "csrc", 1, 0x33, NULL);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  fail_unless ((meta = gst_buffer_get_rtp_source_meta (buffer)));
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, 0x11);
+  fail_unless_equals_int (meta->csrc_count, 2);
+  fail_unless_equals_int (meta->csrc[0], 0x22);
+  fail_unless_equals_int (meta->csrc[1], 0x33);
+  gst_buffer_unref (buffer);
+
+  /* Property disabled should never add meta */
+  g_object_set (depay, "source-info", FALSE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, NULL);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  g_object_unref (depay);
+  gst_harness_teardown (h);
+}
+
+GST_END_TEST;
+
+
+static Suite *
 rtp_basepayloading_suite (void)
 {
   Suite *s = suite_create ("rtp_base_depayloading_test");
@@ -1265,6 +1347,8 @@ rtp_basepayloading_suite (void)
   tcase_add_test (tc_chain, rtp_base_depayload_play_speed_test);
   tcase_add_test (tc_chain, rtp_base_depayload_clock_base_test);
 
+  tcase_add_test (tc_chain, rtp_base_depayload_source_info_test);
+
   return s;
 }
 
diff --git a/tests/check/libs/rtpbasepayload.c b/tests/check/libs/rtpbasepayload.c
index 9385ce949c..9bda374079 100644
--- a/tests/check/libs/rtpbasepayload.c
+++ b/tests/check/libs/rtpbasepayload.c
@@ -23,8 +23,8 @@
 
 #include <gst/gst.h>
 #include <gst/check/gstcheck.h>
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtpbasepayload.h>
+#include <gst/check/gstharness.h>
+#include <gst/rtp/rtp.h>
 
 #define DEFAULT_CLOCK_RATE (42)
 #define BUFFER_BEFORE_LIST (10)
@@ -121,7 +121,9 @@ gst_rtp_dummy_pay_handle_buffer (GstRTPBasePayload * pay, GstBuffer * buffer)
     }
   }
 
-  paybuffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  paybuffer =
+      gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
+      0, 0, 0);
 
   GST_BUFFER_PTS (paybuffer) = GST_BUFFER_PTS (buffer);
   GST_BUFFER_OFFSET (paybuffer) = GST_BUFFER_OFFSET (buffer);
@@ -444,20 +446,11 @@ validate_buffers_received (guint received_buffers)
 }
 
 static void
-validate_buffer (guint index, const gchar * field, ...)
+validate_buffer_valist (GstBuffer * buf, const gchar * field, va_list var_args)
 {
-  GstBuffer *buf;
   GstRTPBuffer rtp = { NULL };
   gboolean mapped = FALSE;
-  va_list var_args;
 
-  fail_if (index >= g_list_length (buffers));
-  buf = GST_BUFFER (g_list_nth_data (buffers, index));
-  fail_if (buf == NULL);
-
-  GST_TRACE ("%" GST_PTR_FORMAT, buf);
-
-  va_start (var_args, field);
   while (field) {
     if (!g_strcmp0 (field, "pts")) {
       GstClockTime pts = va_arg (var_args, GstClockTime);
@@ -489,19 +482,53 @@ validate_buffer (guint index, const gchar * field, ...)
       } else if (!g_strcmp0 (field, "ssrc")) {
         guint32 ssrc = va_arg (var_args, guint);
         fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp), ssrc);
+      } else if (!g_strcmp0 (field, "csrc")) {
+        guint idx = va_arg (var_args, guint);
+        guint csrc = va_arg (var_args, guint);
+        fail_unless_equals_int (gst_rtp_buffer_get_csrc (&rtp, idx), csrc);
+      } else if (!g_strcmp0 (field, "csrc-count")) {
+        guint csrc_count = va_arg (var_args, guint);
+        fail_unless_equals_int (gst_rtp_buffer_get_csrc_count (&rtp),
+            csrc_count);
       } else {
         fail ("test cannot validate unknown buffer field '%s'", field);
       }
     }
     field = va_arg (var_args, const gchar *);
   }
-  va_end (var_args);
 
   if (mapped) {
     gst_rtp_buffer_unmap (&rtp);
   }
 }
 
+static void
+validate_buffer1 (GstBuffer * buf, const gchar * field, ...)
+{
+  va_list var_args;
+
+  va_start (var_args, field);
+  validate_buffer_valist (buf, field, var_args);
+  va_end (var_args);
+}
+
+static void
+validate_buffer (guint index, const gchar * field, ...)
+{
+  GstBuffer *buf;
+  va_list var_args;
+
+  fail_if (index >= g_list_length (buffers));
+  buf = GST_BUFFER (g_list_nth_data (buffers, index));
+  fail_if (buf == NULL);
+
+  GST_TRACE ("%" GST_PTR_FORMAT, buf);
+
+  va_start (var_args, field);
+  validate_buffer_valist (buf, field, var_args);
+  va_end (var_args);
+}
+
 static void
 get_buffer_field (guint index, const gchar * field, ...)
 {
@@ -1775,6 +1802,59 @@ GST_START_TEST (rtp_base_payload_property_stats_test)
 
 GST_END_TEST;
 
+/* basepayloader has a property source-info that makes it aware of RTP
+ * source information passed as GstRTPSourceMeta on the input buffers. All
+ * sources found in the meta will be added to the list of CSRCs in the RTP
+ * header. A useful scenario for this is, for instance, to signal which
+ * sources contributed to a mixed audio stream. */
+GST_START_TEST (rtp_base_payload_property_source_info_test)
+{
+  GstHarness *h;
+  GstRtpDummyPay *pay;
+  GstBuffer *buffer;
+  guint csrc_count = 2;
+  const guint32 csrc[] = { 0x11, 0x22 };
+  const guint32 ssrc = 0x33;
+
+  pay = rtp_dummy_pay_new ();
+  h = gst_harness_new_with_element (GST_ELEMENT_CAST (pay), "sink", "src");
+  gst_harness_set_src_caps_str (h, "application/x-rtp");
+
+  /* Input buffer has no meta, payloader should not add CSRC */
+  g_object_set (pay, "source-info", TRUE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  validate_buffer1 (buffer, "csrc-count", 0, NULL);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  /* Input buffer has meta, payloader should add CSRC */
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  fail_unless (gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc,
+          csrc_count));
+  buffer = gst_harness_push_and_pull (h, buffer);
+  /* The meta SSRC should be added as the last contributing source */
+  validate_buffer1 (buffer, "csrc-count", 3, "csrc", 0, csrc[0],
+      "csrc", 1, csrc[1], "csrc", 2, ssrc, NULL);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  /* When property is disabled, the meta should be ignored and no CSRC
+   * added. */
+  g_object_set (pay, "source-info", FALSE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  fail_unless (gst_buffer_add_rtp_source_meta (buffer, NULL, csrc, csrc_count));
+  buffer = gst_harness_push_and_pull (h, buffer);
+  validate_buffer1 (buffer, "csrc-count", 0, NULL);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  g_object_unref (pay);
+  gst_harness_teardown (h);
+}
+
+GST_END_TEST;
+
 /* push a single buffer to the payloader which should successfully payload it
  * into an RTP packet. besides the payloaded RTP packet there should be the
  * three events initial events: stream-start, caps and segment. because of that
@@ -1878,6 +1958,7 @@ rtp_basepayloading_suite (void)
   tcase_add_test (tc_chain, rtp_base_payload_property_perfect_rtptime_test);
   tcase_add_test (tc_chain, rtp_base_payload_property_ptime_multiple_test);
   tcase_add_test (tc_chain, rtp_base_payload_property_stats_test);
+  tcase_add_test (tc_chain, rtp_base_payload_property_source_info_test);
 
   tcase_add_test (tc_chain, rtp_base_payload_framerate_attribute);
   tcase_add_test (tc_chain, rtp_base_payload_max_framerate_attribute);
diff --git a/tests/check/libs/rtpmeta.c b/tests/check/libs/rtpmeta.c
new file mode 100644
index 0000000000..312f3916eb
--- /dev/null
+++ b/tests/check/libs/rtpmeta.c
@@ -0,0 +1,110 @@
+/* GStreamer RTP meta unit tests
+ * Copyright (C) 2016  Stian Selnes <stian@pexip.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General
+ * Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/check/gstcheck.h>
+#include <gst/rtp/rtp.h>
+
+GST_START_TEST (test_rtp_source_meta_set_get_sources)
+{
+  GstBuffer *buffer;
+  GstRTPSourceMeta *meta;
+  guint32 ssrc = 1000, ssrc2 = 2000;
+  const guint32 csrc[] = {
+    0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14
+  };
+
+  buffer = gst_buffer_new ();
+  meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 12);
+
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, ssrc);
+  for (gint i = 0; i < 12; i++)
+    fail_unless_equals_int (meta->csrc[i], csrc[i]);
+
+  /* Unset the ssrc */
+  fail_unless (gst_rtp_source_meta_set_ssrc (meta, NULL));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12);
+  fail_if (meta->ssrc_valid);
+
+  /* Set the ssrc again */
+  fail_unless (gst_rtp_source_meta_set_ssrc (meta, &ssrc2));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, ssrc2);
+
+  /* Append multiple csrcs */
+  fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[12], 2));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
+  for (gint i = 0; i < 14; i++)
+    fail_unless_equals_int (meta->csrc[i], csrc[i]);
+
+  gst_buffer_unref (buffer);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rtp_source_meta_set_get_max_sources)
+{
+  GstBuffer *buffer;
+  GstRTPSourceMeta *meta;
+  guint32 ssrc = 1000;
+  const guint32 csrc[16] = { 0, };
+
+  buffer = gst_buffer_new ();
+  meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 14);
+
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
+  fail_unless_equals_int (meta->csrc_count, 14);
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, ssrc);
+
+  /* Append one more csrc */
+  /* The source count should cap at 15 for convenient use with
+   * gst_rtp_buffer-functions! */
+  fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[14], 1));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
+  fail_unless_equals_int (meta->csrc_count, 15);
+
+  /* Try to append one more csrc, but we've reached max */
+  fail_if (gst_rtp_source_meta_append_csrc (meta, &csrc[15], 1));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
+  fail_unless_equals_int (meta->csrc_count, 15);
+
+  gst_buffer_unref (buffer);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtp_meta_suite (void)
+{
+  Suite *s = suite_create ("rtp_meta_tests");
+  TCase *tc_chain;
+
+  suite_add_tcase (s, (tc_chain = tcase_create ("GstRTPSourceMeta")));
+  tcase_add_test (tc_chain, test_rtp_source_meta_set_get_sources);
+  tcase_add_test (tc_chain, test_rtp_source_meta_set_get_max_sources);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtp_meta)