rtpmanagerbad: add RTP streaming elements

This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.

https://bugzilla.gnome.org/show_bug.cgi?id=703111

The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.

The code can be used as follows

```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234

gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink

gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```

rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate

GStreamer 1.16 does not yet support the newer GLib templates, so revert.

rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources

for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.

rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches

beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.

rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even

According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.

rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters

Locking is added because the URI allows to access the properties too.

rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction

In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.

9 files changed, 1567 insertions, 0 deletions

diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am
new file mode 100644
index 000000000..2d8dd1bdd
--- /dev/null
+++ b/gst/rtp/Makefile.am
@@ -0,0 +1,17 @@
+plugin_LTLIBRARIES = libgstrtpmanagerbad.la
+
+libgstrtpmanagerbad_la_SOURCES = \
+	gstrtp-utils.c \
+	gstrtpsink.c \
+	gstrtpsrc.c \
+	plugin.c
+
+libgstrtpmanagerbad_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+libgstrtpmanagerbad_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS)
+libgstrtpmanagerbad_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = \
+	gstrtp-utils.h \
+	gstrtpcaps.h \
+	gstrtpsink.h \
+	gstrtpsrc.h
diff --git a/gst/rtp/gstrtp-utils.c b/gst/rtp/gstrtp-utils.c
new file mode 100644
index 000000000..fc06bba79
--- /dev/null
+++ b/gst/rtp/gstrtp-utils.c
@@ -0,0 +1,39 @@
+/*
+ * See: https://bugzilla.gnome.org/show_bug.cgi?id=779765
+ */
+
+#include "gstrtp-utils.h"
+
+static void
+gst_rtp_utils_uri_query_foreach (const gchar * key, const gchar * value,
+    GObject * src)
+{
+  if (key == NULL) {
+    GST_WARNING_OBJECT (src, "Refusing to use empty key.");
+    return;
+  }
+
+  if (value == NULL) {
+    GST_WARNING_OBJECT (src, "Refusing to use NULL for key %s.", key);
+    return;
+  }
+
+  GST_DEBUG_OBJECT (src, "Setting property '%s' to '%s'", key, value);
+  gst_util_set_object_arg (src, key, value);
+}
+
+void
+gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri)
+{
+  GHashTable *hash_table;
+
+  g_return_if_fail (uri != NULL);
+  hash_table = gst_uri_get_query_table (uri);
+
+  if (hash_table) {
+    g_hash_table_foreach (hash_table,
+        (GHFunc) gst_rtp_utils_uri_query_foreach, obj);
+
+    g_hash_table_unref (hash_table);
+  }
+}
diff --git a/gst/rtp/gstrtp-utils.h b/gst/rtp/gstrtp-utils.h
new file mode 100644
index 000000000..62ec2aafa
--- /dev/null
+++ b/gst/rtp/gstrtp-utils.h
@@ -0,0 +1,8 @@
+#ifndef __GST_RTP_UTILS_H__
+#define __GST_RTP_UTILS_H__
+
+#include <gst/gst.h>
+
+void gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri);
+
+#endif
diff --git a/gst/rtp/gstrtpsink.c b/gst/rtp/gstrtpsink.c
new file mode 100644
index 000000000..23b6df959
--- /dev/null
+++ b/gst/rtp/gstrtpsink.c
@@ -0,0 +1,581 @@
+/* GStreamer
+ * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION: gstrtsinkp
+ * @title: GstRtpSink
+ * @short description: element with Uri interface to stream RTP data to
+ * the network.
+ *
+ * RTP (RFC 3550) is a protocol to stream media over the network while
+ * retaining the timing information and providing enough information to
+ * reconstruct the correct timing domain by the receiver.
+ *
+ * The RTP data port should be even, while the RTCP port should be
+ * odd. The URI that is entered defines the data port, the RTCP port will
+ * be allocated to the next port.
+ *
+ * This element hooks up the correct sockets to support both RTP as the
+ * accompanying RTCP layer.
+ *
+ * This Bin handles streaming RTP payloaded data on the network.
+ *
+ * This element also implements the URI scheme `rtp://` allowing to send
+ * data on the network by bins that allow use the URI to determine the sink.
+ * The RTP URI handler also allows setting properties through the URI query.
+ */
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gio/gio.h>
+
+#include "gstrtpsink.h"
+#include "gstrtp-utils.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
+#define GST_CAT_DEFAULT gst_rtp_sink_debug
+
+#define DEFAULT_PROP_URI              "rtp://0.0.0.0:5004"
+#define DEFAULT_PROP_TTL              64
+#define DEFAULT_PROP_TTL_MC           1
+
+enum
+{
+  PROP_0,
+
+  PROP_URI,
+  PROP_TTL,
+  PROP_TTL_MC,
+
+  PROP_LAST
+};
+
+static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
+    gpointer iface_data);
+
+#define gst_rtp_sink_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
+    G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
+    GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
+
+#define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
+#define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
+#define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtp"));
+
+static GstStateChangeReturn
+gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
+
+static void
+gst_rtp_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRtpSink *self = GST_RTP_SINK (object);
+
+  switch (prop_id) {
+    case PROP_URI:{
+      GstUri *uri = NULL;
+
+      GST_RTP_SINK_LOCK (object);
+      uri = gst_uri_from_string (g_value_get_string (value));
+      if (uri == NULL)
+        break;
+
+      if (self->uri)
+        gst_uri_unref (self->uri);
+      self->uri = uri;
+      /* RTP data ports should be even according to RFC 3550, while the
+       * RTCP is sent on odd ports. Just warn if there is a mismatch. */
+      if (gst_uri_get_port (self->uri) % 2)
+        GST_WARNING_OBJECT (self,
+            "Port %u is not even, this is not standard (see RFC 3550).",
+            gst_uri_get_port (self->uri));
+
+      gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
+      GST_RTP_SINK_UNLOCK (object);
+      break;
+    }
+    case PROP_TTL:
+      self->ttl = g_value_get_int (value);
+      break;
+    case PROP_TTL_MC:
+      self->ttl_mc = g_value_get_int (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRtpSink *self = GST_RTP_SINK (object);
+
+  switch (prop_id) {
+    case PROP_URI:
+      GST_RTP_SINK_LOCK (object);
+      if (self->uri)
+        g_value_take_string (value, gst_uri_to_string (self->uri));
+      else
+        g_value_set_string (value, NULL);
+      GST_RTP_SINK_UNLOCK (object);
+      break;
+    case PROP_TTL:
+      g_value_set_int (value, self->ttl);
+      break;
+    case PROP_TTL_MC:
+      g_value_set_int (value, self->ttl_mc);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_sink_finalize (GObject * gobject)
+{
+  GstRtpSink *self = GST_RTP_SINK (gobject);
+
+  if (self->uri)
+    gst_uri_unref (self->uri);
+
+  g_mutex_clear (&self->lock);
+  G_OBJECT_CLASS (parent_class)->finalize (gobject);
+}
+
+static gboolean
+gst_rtp_sink_setup_elements (GstRtpSink * self)
+{
+  /*GstPad *pad; */
+  GSocket *socket;
+  GInetAddress *addr;
+  gchar name[48];
+  GstCaps *caps;
+
+  /* Should not be NULL */
+  g_return_val_if_fail (self->uri != NULL, FALSE);
+
+  /* if not already configured */
+  if (self->funnel_rtp == NULL) {
+    self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
+    if (self->funnel_rtp == NULL) {
+      GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+          ("%s", "funnel_rtp element is not available"));
+      return FALSE;
+    }
+
+    self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
+    if (self->funnel_rtcp == NULL) {
+      GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+          ("%s", "funnel_rtcp element is not available"));
+      return FALSE;
+    }
+
+    self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
+    if (self->rtp_sink == NULL) {
+      GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+          ("%s", "rtp_sink element is not available"));
+      return FALSE;
+    }
+
+    self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
+    if (self->rtcp_src == NULL) {
+      GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+          ("%s", "rtcp_src element is not available"));
+      return FALSE;
+    }
+
+    self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
+    if (self->rtcp_sink == NULL) {
+      GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+          ("%s", "rtcp_sink element is not available"));
+      return FALSE;
+    }
+
+    gst_bin_add (GST_BIN (self), self->funnel_rtp);
+    gst_bin_add (GST_BIN (self), self->funnel_rtcp);
+
+    /* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
+     * not all at the same moment */
+    g_object_set (self->rtp_sink,
+        "host", gst_uri_get_host (self->uri),
+        "port", gst_uri_get_port (self->uri),
+        "ttl", self->ttl, "ttl-mc", self->ttl_mc, NULL);
+
+    gst_bin_add (GST_BIN (self), self->rtp_sink);
+
+    g_object_set (self->rtcp_sink,
+        "host", gst_uri_get_host (self->uri),
+        "port", gst_uri_get_port (self->uri) + 1,
+        "ttl", self->ttl, "ttl-mc", self->ttl_mc,
+        /* Set false since we're reusing a socket */
+        "auto-multicast", FALSE, NULL);
+
+    gst_bin_add (GST_BIN (self), self->rtcp_sink);
+
+    /* no need to set address if unicast */
+    caps = gst_caps_new_empty_simple ("application/x-rtcp");
+    g_object_set (self->rtcp_src,
+        "port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
+    gst_caps_unref (caps);
+
+    addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
+    if (g_inet_address_get_is_multicast (addr)) {
+      g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
+          NULL);
+    }
+    g_object_unref (addr);
+
+    gst_bin_add (GST_BIN (self), self->rtcp_src);
+
+    gst_element_link (self->funnel_rtp, self->rtp_sink);
+    gst_element_link (self->funnel_rtcp, self->rtcp_sink);
+
+    gst_element_sync_state_with_parent (self->funnel_rtp);
+    gst_element_sync_state_with_parent (self->funnel_rtcp);
+    gst_element_sync_state_with_parent (self->rtp_sink);
+    gst_element_sync_state_with_parent (self->rtcp_src);
+
+    g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
+    g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
+
+    gst_element_sync_state_with_parent (self->rtcp_sink);
+
+  }
+
+  /* pads are all named */
+  g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
+  gst_element_link_pads (self->rtpbin, name, self->funnel_rtp, "sink_%u");
+
+  g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
+  gst_element_link_pads (self->rtpbin, name, self->funnel_rtcp, "sink_%u");
+
+  g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
+  gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
+
+  return TRUE;
+}
+
+static GstPad *
+gst_rtp_sink_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
+{
+  GstRtpSink *self = GST_RTP_SINK (element);
+  GstPad *pad = NULL;
+
+  if (self->rtpbin == NULL) {
+    GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+        ("%s", "rtpbin element is not available"));
+    return NULL;
+  }
+
+  if (gst_rtp_sink_setup_elements (self) == FALSE)
+    return NULL;
+
+  GST_RTP_SINK_LOCK (self);
+
+  pad = gst_element_get_request_pad (self->rtpbin, "send_rtp_sink_%u");
+  g_return_val_if_fail (pad != NULL, NULL);
+
+  GST_RTP_SINK_UNLOCK (self);
+
+  return pad;
+}
+
+static void
+gst_rtp_sink_release_pad (GstElement * element, GstPad * pad)
+{
+  GstRtpSink *self = GST_RTP_SINK (element);
+  GstPad *rpad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
+
+  GST_RTP_SINK_LOCK (self);
+  gst_element_release_request_pad (self->rtpbin, rpad);
+  gst_object_unref (rpad);
+
+  gst_pad_set_active (pad, FALSE);
+  gst_element_remove_pad (GST_ELEMENT (self), pad);
+
+  GST_RTP_SINK_UNLOCK (self);
+}
+
+static void
+gst_rtp_sink_class_init (GstRtpSinkClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  gobject_class->set_property = gst_rtp_sink_set_property;
+  gobject_class->get_property = gst_rtp_sink_get_property;
+  gobject_class->finalize = gst_rtp_sink_finalize;
+  gstelement_class->change_state = gst_rtp_sink_change_state;
+
+  gstelement_class->request_new_pad =
+      GST_DEBUG_FUNCPTR (gst_rtp_sink_request_new_pad);
+  gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_sink_release_pad);
+
+  /**
+   * GstRtpSink:uri:
+   *
+   * uri to stream RTP to. All GStreamer parameters can be
+   * encoded in the URI, this URI format is RFC compliant.
+   */
+  g_object_class_install_property (gobject_class, PROP_URI,
+      g_param_spec_string ("uri", "URI",
+          "URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpSink:ttl:
+   *
+   * Set the unicast TTL parameter.
+   */
+  g_object_class_install_property (gobject_class, PROP_TTL,
+      g_param_spec_int ("ttl", "Unicast TTL",
+          "Used for setting the unicast TTL parameter",
+          0, 255, DEFAULT_PROP_TTL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpSink:ttl-mc:
+   *
+   * Set the multicast TTL parameter.
+   */
+  g_object_class_install_property (gobject_class, PROP_TTL_MC,
+      g_param_spec_int ("ttl-mc", "Multicast TTL",
+          "Used for setting the multicast TTL parameter", 0, 255,
+          DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&sink_template));
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTP Sink element",
+      "Generic/Bin/Sink",
+      "Simple RTP sink", "Marc Leeman <marc.leeman@gmail.com>");
+}
+
+static void
+gst_rtp_sink_rtpbin_element_added_cb (GstBin * element,
+    GstElement * new_element, gpointer data)
+{
+  GstRtpSink *self = GST_RTP_SINK (data);
+  GST_INFO_OBJECT (self,
+      "Element %" GST_PTR_FORMAT " added element %" GST_PTR_FORMAT ".", element,
+      new_element);
+}
+
+static void
+gst_rtp_sink_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
+    gpointer data)
+{
+  GstRtpSink *self = GST_RTP_SINK (data);
+  GstCaps *caps = gst_pad_query_caps (pad, NULL);
+  GstPad *upad;
+
+  /* Expose RTP data pad only */
+  GST_INFO_OBJECT (self,
+      "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
+      GST_PTR_FORMAT ".", element, pad, caps);
+
+  /* Sanity checks */
+  if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
+    /* Src pad, do not expose */
+    gst_caps_unref (caps);
+    return;
+  }
+
+  if (G_LIKELY (caps)) {
+    GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
+
+    if (gst_caps_can_intersect (caps, ref_caps)) {
+      /* SRC RTCP caps, do not expose */
+      gst_caps_unref (ref_caps);
+      gst_caps_unref (caps);
+
+      return;
+    }
+    gst_caps_unref (ref_caps);
+  } else {
+    GST_ERROR_OBJECT (self, "Pad with no caps detected.");
+    gst_caps_unref (caps);
+
+    return;
+  }
+  gst_caps_unref (caps);
+
+  upad = gst_element_get_compatible_pad (self->funnel_rtp, pad, NULL);
+  if (upad == NULL) {
+    GST_ERROR_OBJECT (self, "No compatible pad found to link pad.");
+    gst_caps_unref (caps);
+
+    return;
+  }
+  GST_INFO_OBJECT (self, "Linking with pad %" GST_PTR_FORMAT ".", upad);
+  gst_pad_link (pad, upad);
+  gst_object_unref (upad);
+}
+
+static void
+gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
+    gpointer data)
+{
+  GstRtpSink *self = GST_RTP_SINK (data);
+  GST_INFO_OBJECT (self,
+      "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
+      pad);
+}
+
+static gboolean
+gst_rtp_sink_setup_rtpbin (GstRtpSink * self)
+{
+  self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
+  if (self->rtpbin == NULL) {
+    GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+        ("%s", "rtpbin element is not available"));
+    return FALSE;
+  }
+
+  /* Add rtpbin callbacks to monitor the operation of rtpbin */
+  g_signal_connect (self->rtpbin, "element-added",
+      G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
+  g_signal_connect (self->rtpbin, "pad-added",
+      G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
+  g_signal_connect (self->rtpbin, "pad-removed",
+      G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
+
+  gst_bin_add (GST_BIN (self), self->rtpbin);
+
+  gst_element_sync_state_with_parent (self->rtpbin);
+
+  return TRUE;
+}
+
+static GstStateChangeReturn
+gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRtpSink *self = GST_RTP_SINK (element);
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+  GST_DEBUG_OBJECT (self, "changing state: %s => %s",
+      gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
+      gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    return ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+
+static void
+gst_rtp_sink_init (GstRtpSink * self)
+{
+  self->rtpbin = NULL;
+  self->funnel_rtp = NULL;
+  self->funnel_rtcp = NULL;
+  self->rtp_sink = NULL;
+  self->rtcp_src = NULL;
+  self->rtcp_sink = NULL;
+
+  self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
+  self->ttl = DEFAULT_PROP_TTL;
+  self->ttl_mc = DEFAULT_PROP_TTL_MC;
+
+  if (gst_rtp_sink_setup_rtpbin (self) == FALSE)
+    return;
+
+  GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SINK);
+  gst_bin_set_suppressed_flags (GST_BIN (self),
+      GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
+
+  g_mutex_init (&self->lock);
+}
+
+static guint
+gst_rtp_sink_uri_get_type (GType type)
+{
+  return GST_URI_SINK;
+}
+
+static const gchar *const *
+gst_rtp_sink_uri_get_protocols (GType type)
+{
+  static const gchar *protocols[] = { (char *) "rtp", NULL };
+
+  return protocols;
+}
+
+static gchar *
+gst_rtp_sink_uri_get_uri (GstURIHandler * handler)
+{
+  GstRtpSink *self = (GstRtpSink *) handler;
+
+  return gst_uri_to_string (self->uri);
+}
+
+static gboolean
+gst_rtp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
+    GError ** error)
+{
+  GstRtpSink *self = (GstRtpSink *) handler;
+
+  g_object_set (G_OBJECT (self), "uri", uri, NULL);
+
+  return TRUE;
+}
+
+static void
+gst_rtp_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
+{
+  GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
+
+  iface->get_type = gst_rtp_sink_uri_get_type;
+  iface->get_protocols = gst_rtp_sink_uri_get_protocols;
+  iface->get_uri = gst_rtp_sink_uri_get_uri;
+  iface->set_uri = gst_rtp_sink_uri_set_uri;
+}
+
+/* ex: set tabstop=2 shiftwidth=2 expandtab: */
diff --git a/gst/rtp/gstrtpsink.h b/gst/rtp/gstrtpsink.h
new file mode 100644
index 000000000..6f3fec0ac
--- /dev/null
+++ b/gst/rtp/gstrtpsink.h
@@ -0,0 +1,72 @@
+/* GStreamer
+ * Copyright (C) 2019 Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_SINK_H__
+#define __GST_RTP_SINK_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_RTP_SINK \
+  (gst_rtp_sink_get_type())
+#define GST_RTP_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTP_SINK, GstRtpSink))
+#define GST_RTP_SINK_CAST(obj) \
+  ((GstRtpSink *) obj)
+#define GST_RTP_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTP_SINK, GstRtpSinkClass))
+#define GST_IS_RTP_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTP_SINK))
+#define GST_IS_RTP_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTP_SINK))
+
+typedef struct _GstRtpSink GstRtpSink;
+typedef struct _GstRtpSinkClass GstRtpSinkClass;
+
+struct _GstRtpSink
+{
+  GstBin parent;
+
+  GstBin parent_instance;
+
+  /* Properties */
+  GstUri *uri;
+  gint ttl;
+  gint ttl_mc;
+
+  /* Internal elements */
+  GstElement *rtpbin;
+  GstElement *funnel_rtp;
+  GstElement *funnel_rtcp;
+  GstElement *rtp_sink;
+  GstElement *rtcp_src;
+  GstElement *rtcp_sink;
+
+  GMutex lock;
+};
+
+struct _GstRtpSinkClass
+{
+  GstBinClass parent;
+};
+
+GType gst_rtp_sink_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_RTP_SINK_H__ */
diff --git a/gst/rtp/gstrtpsrc.c b/gst/rtp/gstrtpsrc.c
new file mode 100644
index 000000000..bf958603b
--- /dev/null
+++ b/gst/rtp/gstrtpsrc.c
@@ -0,0 +1,731 @@
+/* GStreamer
+ * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION: gstrtpsrc
+ * @title: GstRtpSrc
+ * @short description: element with Uri interface to get RTP data from
+ * the network.
+ *
+ * RTP (RFC 3550) is a protocol to stream media over the network while
+ * retaining the timing information and providing enough information to
+ * reconstruct the correct timing domain by the receiver.
+ *
+ * The RTP data port should be even, while the RTCP port should be
+ * odd. The URI that is entered defines the data port, the RTCP port will
+ * be allocated to the next port.
+ *
+ * This element hooks up the correct sockets to support both RTP as the
+ * accompanying RTCP layer.
+ *
+ * This Bin handles taking in of data from the network and provides the
+ * RTP payloaded data.
+ *
+ * This element also implements the URI scheme `rtp://` allowing to render
+ * RTP streams in GStreamer based media players. The RTP URI handler also
+ * allows setting properties through the URI query.
+ */
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/net/net.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#include "gstrtpsrc.h"
+#include "gstrtp-utils.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug);
+#define GST_CAT_DEFAULT gst_rtp_src_debug
+
+#define DEFAULT_PROP_TTL              64
+#define DEFAULT_PROP_TTL_MC           1
+#define DEFAULT_PROP_ENCODING_NAME    NULL
+#define DEFAULT_PROP_LATENCY          200
+
+#define DEFAULT_PROP_URI              "rtp://0.0.0.0:5004"
+
+enum
+{
+  PROP_0,
+
+  PROP_URI,
+  PROP_TTL,
+  PROP_TTL_MC,
+  PROP_ENCODING_NAME,
+  PROP_LATENCY,
+
+  PROP_LAST
+};
+
+static void gst_rtp_src_uri_handler_init (gpointer g_iface,
+    gpointer iface_data);
+
+#define gst_rtp_src_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstRtpSrc, gst_rtp_src, GST_TYPE_BIN,
+    G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_src_uri_handler_init);
+    GST_DEBUG_CATEGORY_INIT (gst_rtp_src_debug, "rtpsrc", 0, "RTP Source"));
+
+#define GST_RTP_SRC_GET_LOCK(obj) (&((GstRtpSrc*)(obj))->lock)
+#define GST_RTP_SRC_LOCK(obj) (g_mutex_lock (GST_RTP_SRC_GET_LOCK(obj)))
+#define GST_RTP_SRC_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SRC_GET_LOCK(obj)))
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtp"));
+
+static GstStateChangeReturn
+gst_rtp_src_change_state (GstElement * element, GstStateChange transition);
+
+/**
+ * gst_rtp_src_rtpbin_erquest_pt_map_cb:
+ * @self: The current #GstRtpSrc object
+ *
+ * #GstRtpBin callback to map a pt on RTP caps.
+ *
+ * Returns: (transfer none): the guess on the RTP caps based on the PT
+ * and caps.
+ */
+static GstCaps *
+gst_rtp_src_rtpbin_request_pt_map_cb (GstElement * rtpbin, guint session_id,
+    guint pt, gpointer data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (data);
+  const GstRTPPayloadInfo *p = NULL;
+
+  GST_DEBUG_OBJECT (self,
+      "Requesting caps for session-id 0x%x and pt %u.", session_id, pt);
+
+  /* the encoding-name has more relevant information */
+  if (self->encoding_name != NULL) {
+    /* Unfortunately, the media needs to be passed in the function. Since
+     * it is not known, try for video if video not found. */
+    p = gst_rtp_payload_info_for_name ("video", self->encoding_name);
+    if (p == NULL)
+      p = gst_rtp_payload_info_for_name ("audio", self->encoding_name);
+
+  }
+
+  /* Static payload types, this is a simple lookup */
+  if (!GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
+    p = gst_rtp_payload_info_for_pt (pt);
+  }
+
+  if (p != NULL) {
+    GstCaps *ret = gst_caps_new_simple ("application/x-rtp",
+        "encoding-name", G_TYPE_STRING, p->encoding_name,
+        "clock-rate", G_TYPE_INT, p->clock_rate,
+        "media", G_TYPE_STRING, p->media, NULL);
+
+    GST_DEBUG_OBJECT (self, "Decided on caps %" GST_PTR_FORMAT, ret);
+
+    return ret;
+  }
+
+  GST_DEBUG_OBJECT (self, "Could not determine caps based on pt and"
+      " the encoding-name was not set.");
+  return NULL;
+}
+
+static void
+gst_rtp_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRtpSrc *self = GST_RTP_SRC (object);
+  GstCaps *caps;
+
+  switch (prop_id) {
+    case PROP_URI:{
+      GstUri *uri = NULL;
+
+      GST_RTP_SRC_LOCK (object);
+      uri = gst_uri_from_string (g_value_get_string (value));
+      if (uri == NULL)
+        break;
+
+      if (self->uri)
+        gst_uri_unref (self->uri);
+      self->uri = uri;
+      if (gst_uri_get_port (self->uri) % 2)
+        GST_WARNING_OBJECT (self,
+            "Port %u is not even, this is not standard (see RFC 3550).",
+            gst_uri_get_port (self->uri));
+      gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
+      GST_RTP_SRC_UNLOCK (object);
+      break;
+    }
+    case PROP_TTL:
+      self->ttl = g_value_get_int (value);
+      break;
+    case PROP_TTL_MC:
+      self->ttl_mc = g_value_get_int (value);
+      break;
+    case PROP_ENCODING_NAME:
+      g_free (self->encoding_name);
+      self->encoding_name = g_value_dup_string (value);
+      if (self->rtp_src) {
+        caps = gst_rtp_src_rtpbin_request_pt_map_cb (NULL, 0, 96, self);
+        g_object_set (G_OBJECT (self->rtp_src), "caps", caps, NULL);
+        gst_caps_unref (caps);
+      }
+      break;
+    case PROP_LATENCY:
+      self->latency = g_value_get_uint (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_src_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRtpSrc *self = GST_RTP_SRC (object);
+
+  switch (prop_id) {
+    case PROP_URI:
+      GST_RTP_SRC_LOCK (object);
+      if (self->uri)
+        g_value_take_string (value, gst_uri_to_string (self->uri));
+      else
+        g_value_set_string (value, NULL);
+      GST_RTP_SRC_UNLOCK (object);
+      break;
+    case PROP_TTL:
+      g_value_set_int (value, self->ttl);
+      break;
+    case PROP_TTL_MC:
+      g_value_set_int (value, self->ttl_mc);
+      break;
+    case PROP_ENCODING_NAME:
+      g_value_set_string (value, self->encoding_name);
+      break;
+    case PROP_LATENCY:
+      g_value_set_uint (value, self->latency);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_src_finalize (GObject * gobject)
+{
+  GstRtpSrc *self = GST_RTP_SRC (gobject);
+
+  if (self->uri)
+    gst_uri_unref (self->uri);
+  g_free (self->encoding_name);
+
+  g_mutex_clear (&self->lock);
+  G_OBJECT_CLASS (parent_class)->finalize (gobject);
+}
+
+static void
+gst_rtp_src_class_init (GstRtpSrcClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  gobject_class->set_property = gst_rtp_src_set_property;
+  gobject_class->get_property = gst_rtp_src_get_property;
+  gobject_class->finalize = gst_rtp_src_finalize;
+  gstelement_class->change_state = gst_rtp_src_change_state;
+
+  /**
+   * GstRtpSrc:uri:
+   *
+   * uri to an RTP from. All GStreamer parameters can be
+   * encoded in the URI, this URI format is RFC compliant.
+   */
+  g_object_class_install_property (gobject_class, PROP_URI,
+      g_param_spec_string ("uri", "URI",
+          "URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpSrc:ttl:
+   *
+   * Set the unicast TTL parameter. In RTP this of importance for RTCP.
+   */
+  g_object_class_install_property (gobject_class, PROP_TTL,
+      g_param_spec_int ("ttl", "Unicast TTL",
+          "Used for setting the unicast TTL parameter",
+          0, 255, DEFAULT_PROP_TTL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpSrc:ttl-mc:
+   *
+   * Set the multicast TTL parameter. In RTP this of importance for RTCP.
+   */
+  g_object_class_install_property (gobject_class, PROP_TTL_MC,
+      g_param_spec_int ("ttl-mc", "Multicast TTL",
+          "Used for setting the multicast TTL parameter", 0, 255,
+          DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpSrc:encoding-name:
+   *
+   * Set the encoding name of the stream to use. This is a short-hand for
+   * the full caps and maps typically to the encoding-name in the RTP caps.
+   */
+  g_object_class_install_property (gobject_class, PROP_ENCODING_NAME,
+      g_param_spec_string ("encoding-name", "Caps encoding name",
+          "Encoding name use to determine caps parameters",
+          DEFAULT_PROP_ENCODING_NAME,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpSrc:latency:
+   *
+   * Set the size of the latency buffer in the
+   * GstRtpBin/GstRtpJitterBuffer to compensate for network jitter.
+   */
+  g_object_class_install_property (gobject_class, PROP_LATENCY,
+      g_param_spec_uint ("latency", "Buffer latency in ms",
+          "Default amount of ms to buffer in the jitterbuffers", 0,
+          G_MAXUINT, DEFAULT_PROP_LATENCY,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&src_template));
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTP Source element",
+      "Generic/Bin/Src",
+      "Simple RTP src", "Marc Leeman <marc.leeman@gmail.com>");
+}
+
+static void
+gst_rtp_src_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
+    gpointer data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (data);
+  GstCaps *caps = gst_pad_query_caps (pad, NULL);
+  GstPad *upad;
+  gchar name[48];
+
+  /* Expose RTP data pad only */
+  GST_INFO_OBJECT (self,
+      "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
+      GST_PTR_FORMAT ".", element, pad, caps);
+
+  /* Sanity checks */
+  if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
+    /* Sink pad, do not expose */
+    gst_caps_unref (caps);
+    return;
+  }
+
+  if (G_LIKELY (caps)) {
+    GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
+
+    if (gst_caps_can_intersect (caps, ref_caps)) {
+      /* SRC RTCP caps, do not expose */
+      gst_caps_unref (ref_caps);
+      gst_caps_unref (caps);
+
+      return;
+    }
+    gst_caps_unref (ref_caps);
+  } else {
+    GST_ERROR_OBJECT (self, "Pad with no caps detected.");
+    gst_caps_unref (caps);
+
+    return;
+  }
+  gst_caps_unref (caps);
+
+  GST_RTP_SRC_LOCK (self);
+  g_snprintf (name, 48, "src_%u", GST_ELEMENT (self)->numpads);
+  upad = gst_ghost_pad_new (name, pad);
+
+  gst_pad_set_active (upad, TRUE);
+  gst_element_add_pad (GST_ELEMENT (self), upad);
+
+  GST_RTP_SRC_UNLOCK (self);
+}
+
+static void
+gst_rtp_src_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
+    gpointer data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (data);
+  GST_INFO_OBJECT (self,
+      "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
+      pad);
+}
+
+static void
+gst_rtp_src_rtpbin_on_ssrc_collision_cb (GstElement * rtpbin, guint session_id,
+    guint ssrc, gpointer data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (data);
+
+  GST_INFO_OBJECT (self,
+      "Dectected an SSRC collision: session-id 0x%x, ssrc 0x%x.", session_id,
+      ssrc);
+}
+
+static void
+gst_rtp_src_rtpbin_on_new_ssrc_cb (GstElement * rtpbin, guint session_id,
+    guint ssrc, gpointer data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (data);
+
+  GST_INFO_OBJECT (self, "Dectected a new SSRC: session-id 0x%x, ssrc 0x%x.",
+      session_id, ssrc);
+}
+
+static GstPadProbeReturn
+gst_rtp_src_on_recv_rtcp (GstPad * pad, GstPadProbeInfo * info,
+    gpointer user_data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (user_data);
+  GstBuffer *buffer;
+  GstNetAddressMeta *meta;
+
+  if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
+    GstBufferList *buffer_list = info->data;
+    buffer = gst_buffer_list_get (buffer_list, 0);
+  } else {
+    buffer = info->data;
+  }
+
+  meta = gst_buffer_get_net_address_meta (buffer);
+
+  GST_OBJECT_LOCK (self);
+  g_clear_object (&self->rtcp_send_addr);
+  self->rtcp_send_addr = g_object_ref (meta->addr);
+  GST_OBJECT_UNLOCK (self);
+
+  return GST_PAD_PROBE_OK;
+}
+
+static inline void
+gst_rtp_src_attach_net_address_meta (GstRtpSrc * self, GstBuffer * buffer)
+{
+  GST_OBJECT_LOCK (self);
+  if (self->rtcp_send_addr)
+    gst_buffer_add_net_address_meta (buffer, self->rtcp_send_addr);
+  GST_OBJECT_UNLOCK (self);
+}
+
+static GstPadProbeReturn
+gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info,
+    gpointer user_data)
+{
+  GstRtpSrc *self = GST_RTP_SRC (user_data);
+
+  if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
+    GstBufferList *buffer_list = info->data;
+    GstBuffer *buffer;
+    gint i;
+
+    info->data = buffer_list = gst_buffer_list_make_writable (buffer_list);
+    for (i = 0; i < gst_buffer_list_length (buffer_list); i++) {
+      buffer = gst_buffer_list_get (buffer_list, i);
+      gst_rtp_src_attach_net_address_meta (self, buffer);
+    }
+  } else {
+    GstBuffer *buffer = info->data;
+    info->data = buffer = gst_buffer_make_writable (buffer);
+    gst_rtp_src_attach_net_address_meta (self, buffer);
+  }
+
+  return GST_PAD_PROBE_OK;
+}
+
+static gboolean
+gst_rtp_src_setup_elements (GstRtpSrc * self)
+{
+  GstPad *pad;
+  GSocket *socket;
+  GInetAddress *addr;
+  gchar name[48];
+  GstCaps *caps;
+  gchar *address;
+  guint rtcp_port;
+
+  /* Construct the RTP receiver pipeline.
+   *
+   * udpsrc -> [recv_rtp_sink_%u]  --------  [recv_rtp_src_%u_%u_%u]
+   *                              | rtpbin |
+   * udpsrc -> [recv_rtcp_sink_%u] --------  [send_rtcp_src_%u] -> udpsink
+   *
+   * This pipeline is fixed for now, note that optionally an FEC stream could
+   * be added later.
+   */
+
+  /* Should not be NULL */
+  g_return_val_if_fail (self->uri != NULL, FALSE);
+
+  self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
+  if (self->rtpbin == NULL) {
+    GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+        ("%s", "rtpbin element is not available"));
+    return FALSE;
+  }
+
+  self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
+  if (self->rtp_src == NULL) {
+    GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+        ("%s", "rtp_src element is not available"));
+    return FALSE;
+  }
+
+  self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
+  if (self->rtcp_src == NULL) {
+    GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+        ("%s", "rtcp_src element is not available"));
+    return FALSE;
+  }
+
+  self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
+  if (self->rtcp_sink == NULL) {
+    GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+        ("%s", "rtcp_sink element is not available"));
+    return FALSE;
+  }
+
+  /* Add rtpbin callbacks to monitor the operation of rtpbin */
+  g_signal_connect (self->rtpbin, "pad-added",
+      G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
+  g_signal_connect (self->rtpbin, "pad-removed",
+      G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
+  g_signal_connect (self->rtpbin, "request-pt-map",
+      G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
+  g_signal_connect (self->rtpbin, "on-new-ssrc",
+      G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
+  g_signal_connect (self->rtpbin, "on-ssrc-collision",
+      G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
+
+  g_object_set (self->rtpbin, "latency", self->latency, NULL);
+
+  /* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
+   * not all at the same moment */
+  gst_bin_add (GST_BIN (self), self->rtpbin);
+  gst_bin_add (GST_BIN (self), self->rtp_src);
+
+  g_object_set (self->rtp_src,
+      "address", gst_uri_get_host (self->uri),
+      "port", gst_uri_get_port (self->uri), NULL);
+
+  gst_bin_add (GST_BIN (self), self->rtcp_sink);
+
+  /* no need to set address if unicast */
+  caps = gst_caps_new_empty_simple ("application/x-rtcp");
+  g_object_set (self->rtcp_src,
+      "port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
+  gst_caps_unref (caps);
+
+  addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
+  if (g_inet_address_get_is_multicast (addr)) {
+    g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
+        NULL);
+  }
+  g_object_unref (addr);
+
+  g_object_set (self->rtcp_sink,
+      "host", gst_uri_get_host (self->uri),
+      "port", gst_uri_get_port (self->uri) + 1,
+      "ttl", self->ttl, "ttl-mc", self->ttl_mc,
+      /* Set false since we're reusing a socket */
+      "auto-multicast", FALSE, NULL);
+
+  gst_bin_add (GST_BIN (self), self->rtcp_src);
+
+  /* share the socket created by the source */
+  g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket,
+      "address", &address, "port", &rtcp_port, NULL);
+
+  addr = g_inet_address_new_from_string (address);
+  g_free (address);
+
+  if (g_inet_address_get_is_multicast (addr)) {
+    /* mc-ttl is not supported by dynudpsink */
+    g_socket_set_multicast_ttl (socket, self->ttl_mc);
+    /* In multicast, send RTCP to the multicast group */
+    self->rtcp_send_addr = g_inet_socket_address_new (addr, rtcp_port);
+  } else {
+    /* In unicast, send RTCP to the detected sender address */
+    pad = gst_element_get_static_pad (self->rtcp_src, "src");
+    self->rtcp_recv_probe = gst_pad_add_probe (pad,
+        GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
+        gst_rtp_src_on_recv_rtcp, self, NULL);
+    gst_object_unref (pad);
+  }
+  g_object_unref (addr);
+
+  pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
+  self->rtcp_send_probe = gst_pad_add_probe (pad,
+      GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
+      gst_rtp_src_on_send_rtcp, self, NULL);
+  gst_object_unref (pad);
+
+  g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
+
+  /* pads are all named */
+  g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
+  gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
+
+  g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
+  gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
+
+  gst_element_sync_state_with_parent (self->rtpbin);
+  gst_element_sync_state_with_parent (self->rtp_src);
+  gst_element_sync_state_with_parent (self->rtcp_sink);
+
+  g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
+  gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
+
+  gst_element_sync_state_with_parent (self->rtcp_src);
+
+  return TRUE;
+}
+
+static void
+gst_rtp_src_stop (GstRtpSrc * self)
+{
+  GstPad *pad;
+
+  if (self->rtcp_recv_probe) {
+    pad = gst_element_get_static_pad (self->rtcp_src, "src");
+    gst_pad_remove_probe (pad, self->rtcp_recv_probe);
+    self->rtcp_recv_probe = 0;
+    gst_object_unref (pad);
+  }
+
+  pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
+  gst_pad_remove_probe (pad, self->rtcp_send_probe);
+  self->rtcp_send_probe = 0;
+  gst_object_unref (pad);
+}
+
+static GstStateChangeReturn
+gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRtpSrc *self = GST_RTP_SRC (element);
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+  GST_DEBUG_OBJECT (self, "Changing state: %s => %s",
+      gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
+      gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (gst_rtp_src_setup_elements (self) == FALSE)
+        return GST_STATE_CHANGE_FAILURE;
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    return ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      ret = GST_STATE_CHANGE_NO_PREROLL;
+      break;
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      ret = GST_STATE_CHANGE_NO_PREROLL;
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      gst_rtp_src_stop (self);
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+static void
+gst_rtp_src_init (GstRtpSrc * self)
+{
+  self->rtpbin = NULL;
+  self->rtp_src = NULL;
+  self->rtcp_src = NULL;
+  self->rtcp_sink = NULL;
+
+  self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
+  self->ttl = DEFAULT_PROP_TTL;
+  self->ttl_mc = DEFAULT_PROP_TTL_MC;
+  self->encoding_name = DEFAULT_PROP_ENCODING_NAME;
+  self->latency = DEFAULT_PROP_LATENCY;
+
+  GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SOURCE);
+  gst_bin_set_suppressed_flags (GST_BIN (self),
+      GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
+
+  g_mutex_init (&self->lock);
+}
+
+static guint
+gst_rtp_src_uri_get_type (GType type)
+{
+  return GST_URI_SRC;
+}
+
+static const gchar *const *
+gst_rtp_src_uri_get_protocols (GType type)
+{
+  static const gchar *protocols[] = { (char *) "rtp", NULL };
+
+  return protocols;
+}
+
+static gchar *
+gst_rtp_src_uri_get_uri (GstURIHandler * handler)
+{
+  GstRtpSrc *self = (GstRtpSrc *) handler;
+
+  return gst_uri_to_string (self->uri);
+}
+
+static gboolean
+gst_rtp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
+    GError ** error)
+{
+  GstRtpSrc *self = (GstRtpSrc *) handler;
+
+  g_object_set (G_OBJECT (self), "uri", uri, NULL);
+
+  return TRUE;
+}
+
+static void
+gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
+{
+  GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
+
+  iface->get_type = gst_rtp_src_uri_get_type;
+  iface->get_protocols = gst_rtp_src_uri_get_protocols;
+  iface->get_uri = gst_rtp_src_uri_get_uri;
+  iface->set_uri = gst_rtp_src_uri_set_uri;
+}
+
+/* ex: set tabstop=2 shiftwidth=2 expandtab: */
diff --git a/gst/rtp/gstrtpsrc.h b/gst/rtp/gstrtpsrc.h
new file mode 100644
index 000000000..4bc3535ef
--- /dev/null
+++ b/gst/rtp/gstrtpsrc.h
@@ -0,0 +1,76 @@
+/* GStreamer
+ * Copyright (C) 2019 Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_SRC_H__
+#define __GST_RTP_SRC_H__
+
+#include <gio/gio.h>
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_RTP_SRC \
+  (gst_rtp_src_get_type())
+#define GST_RTP_SRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTP_SRC, GstRtpSrc))
+#define GST_RTP_SRC_CAST(obj) \
+  ((GstRtpSrc *) obj)
+#define GST_RTP_SRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTP_SRC, GstRtpSrcClass))
+#define GST_IS_RTP_SRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTP_SRC))
+#define GST_IS_RTP_SRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTP_SRC))
+
+typedef struct _GstRtpSrc GstRtpSrc;
+typedef struct _GstRtpSrcClass GstRtpSrcClass;
+
+struct _GstRtpSrc
+{
+  GstBin parent;
+
+  /* Properties */
+  GstUri *uri;
+  gint ttl;
+  gint ttl_mc;
+  gint latency;
+  gchar *encoding_name;
+  guint latency_ms;
+
+  /* Internal elements */
+  GstElement *rtpbin;
+  GstElement *rtp_src;
+  GstElement *rtcp_src;
+  GstElement *rtcp_sink;
+
+  gulong rtcp_recv_probe;
+  gulong rtcp_send_probe;
+  GSocketAddress *rtcp_send_addr;
+
+  GMutex lock;
+};
+
+struct _GstRtpSrcClass
+{
+  GstBinClass parent;
+};
+
+GType gst_rtp_src_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_RTP_SRC_H__ */
diff --git a/gst/rtp/meson.build b/gst/rtp/meson.build
new file mode 100644
index 000000000..bb21dae5b
--- /dev/null
+++ b/gst/rtp/meson.build
@@ -0,0 +1,15 @@
+gst_plugins_rtp_sources = [
+  'plugin.c',
+  'gstrtpsink.c',
+  'gstrtpsrc.c',
+  'gstrtp-utils.c',
+]
+
+gstrtp = library('gstrtpmanagerbad',
+  gst_plugins_rtp_sources,
+  dependencies: [gio_dep, gst_dep, gstbase_dep, gstrtp_dep, gstnet_dep, gstcontroller_dep],
+  include_directories: [configinc],
+  install: true,
+  c_args: gst_plugins_bad_args,
+  install_dir: plugins_install_dir,
+)
diff --git a/gst/rtp/plugin.c b/gst/rtp/plugin.c
new file mode 100644
index 000000000..8c1d71f8b
--- /dev/null
+++ b/gst/rtp/plugin.c
@@ -0,0 +1,28 @@
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstrtpsink.h"
+#include "gstrtpsrc.h"
+
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+
+  gboolean ret = FALSE;
+
+  ret |= gst_element_register (plugin, "rtpsrc",
+      GST_RANK_PRIMARY + 1, GST_TYPE_RTP_SRC);
+
+  ret |= gst_element_register (plugin, "rtpsink",
+      GST_RANK_PRIMARY + 1, GST_TYPE_RTP_SINK);
+
+  return ret;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    rtpmanagerbad,
+    "GStreamer RTP Plugins",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);