diff --git a/plugins/sip/calls-sip-media-pipeline.c b/plugins/sip/calls-sip-media-pipeline.c
index fa7050a..bcb3387 100644
--- a/plugins/sip/calls-sip-media-pipeline.c
+++ b/plugins/sip/calls-sip-media-pipeline.c
@@ -60,35 +60,36 @@
 
 
 /* The following defines are used to set/reset bitmaps of playing/paused/stop state */
-#define EL_SEND_PIPELINE (1<<0)
-#define EL_SEND_AUDIO_SRC (1<<1)
-#define EL_SEND_RTPBIN (1<<2)
-#define EL_SEND_RTP_SINK (1<<3)
-#define EL_SEND_RTCP_SINK (1<<4)
-#define EL_SEND_RTCP_SRC (1<<5)
-#define EL_SEND_PAYLOADER (1<<6)
-#define EL_SEND_ENCODER (1<<7)
+#define EL_PIPELINE (1<<0)
+#define EL_RTPBIN (1<<1)
+#define EL_RTP_SRC (1<<2)
+#define EL_RTP_SINK (1<<3)
+#define EL_RTCP_SRC (1<<4)
+#define EL_RTCP_SINK (1<<5)
 
-#define EL_SEND_ALL_RTP EL_SEND_PIPELINE | EL_SEND_AUDIO_SRC |                  \
-  EL_SEND_RTPBIN | EL_SEND_RTP_SINK | EL_SEND_RTCP_SRC | EL_SEND_RTCP_SINK |    \
-  EL_SEND_PAYLOADER | EL_SEND_ENCODER
-#define EL_SEND_SENDING EL_SEND_AUDIO_SRC | EL_SEND_RTPBIN | EL_SEND_RTP_SINK | \
-  EL_SEND_PAYLOADER | EL_SEND_ENCODER
+#define EL_SRTP_ENCODER (1<<6)
+#define EL_SRTP_DECODER (1<<7)
 
-/* leave some room for more elements to be added later */
+#define EL_AUDIO_SRC (1<<8)
+#define EL_AUDIO_SINK (1<<9)
 
-#define EL_RECV_PIPELINE (1<<16)
-#define EL_RECV_AUDIO_SINK (1<<17)
-#define EL_RECV_RTPBIN (1<<18)
-#define EL_RECV_RTP_SRC (1<<19)
-#define EL_RECV_RTCP_SINK (1<<20)
-#define EL_RECV_RTCP_SRC (1<<21)
-#define EL_RECV_DEPAYLOADER (1<<22)
-#define EL_RECV_DECODER (1<<23)
+#define EL_PAYLOADER (1<<10)
+#define EL_DEPAYLOADER (1<<11)
 
-#define EL_RECV_ALL_RTP EL_RECV_PIPELINE | EL_RECV_AUDIO_SINK |                 \
-  EL_RECV_RTPBIN | EL_RECV_RTP_SRC | EL_RECV_RTCP_SRC | EL_RECV_RTCP_SINK |     \
-  EL_RECV_DEPAYLOADER | EL_RECV_DECODER
+#define EL_ENCODER (1<<12)
+#define EL_DECODER (1<<13)
+
+#define EL_SENDING                                                \
+  (EL_AUDIO_SRC | EL_ENCODER | EL_PAYLOADER |                     \
+   EL_RTPBIN | EL_RTP_SINK | EL_RTCP_SINK)
+
+#define EL_ALL_RTP                                                \
+  (EL_PIPELINE | EL_RTPBIN |                                      \
+   EL_RTP_SRC | EL_RTP_SINK | EL_RTCP_SRC | EL_RTCP_SINK |        \
+   EL_AUDIO_SRC | EL_AUDIO_SINK |                                 \
+   EL_ENCODER | EL_DECODER | EL_PAYLOADER | EL_DEPAYLOADER)
+
+#define EL_ALL_SRTP (EL_ALL_RTP | EL_SRTP_ENCODER | EL_SRTP_DECODER)
 
 
 enum {
@@ -122,37 +123,31 @@ struct _CallsSipMediaPipeline {
   uint element_map_paused;
   uint element_map_stopped;
   gboolean emitted_sending_signal;
+
   /* Connection details */
   char *remote;
-
   gint rport_rtp;
-
   gint rport_rtcp;
 
-  /* Gstreamer Elements (sending) */
-  GstElement *send_pipeline;
-  GstElement *audiosrc;
-  GstElement *send_rtpbin;
-  GstElement *rtp_sink; /* UDP out */
+  GstElement *pipeline;
+  GstElement *rtpbin;
+
+  GstElement *rtp_src;
+  GstElement *rtp_sink;
+  GstElement *rtcp_sink;
+  GstElement *rtcp_src;
+
+  GstElement *audio_src;
   GstElement *payloader;
   GstElement *encoder;
-  GstElement *rtcp_send_sink;
-  GstElement *rtcp_send_src;
-  /* Gstreamer elements (receiving) */
-  GstElement *recv_pipeline;
-  GstElement *audiosink;
-  GstElement *recv_rtpbin;
-  GstElement *rtp_src; /* UDP in */
+
+  GstElement *audio_sink;
   GstElement *depayloader;
   GstElement *decoder;
-  GstElement *rtcp_recv_sink;
-  GstElement *rtcp_recv_src;
 
   /* Gstreamer busses */
-  GstBus *bus_send;
-  GstBus *bus_recv;
-  guint bus_watch_send;
-  guint bus_watch_recv;
+  GstBus *bus;
+  guint bus_watch_id;
 };
 
 #if GLIB_CHECK_VERSION(2, 70, 0)
@@ -170,8 +165,7 @@ set_state (CallsSipMediaPipeline  *self,
   g_autoptr (GEnumClass) enum_class = NULL;
   GEnumValue *enum_val;
 
-  g_autofree char *recv_fname = NULL;
-  g_autofree char *send_fname = NULL;
+  g_autofree char *fname = NULL;
 
   g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
 
@@ -189,15 +183,11 @@ set_state (CallsSipMediaPipeline  *self,
   enum_class = g_type_class_ref (CALLS_TYPE_MEDIA_PIPELINE_STATE);
   enum_val = g_enum_get_value (enum_class, state);
 
-  recv_fname = g_strdup_printf ("recv-%s", enum_val->value_nick);
-  send_fname = g_strdup_printf ("send-%s", enum_val->value_nick);
+  fname = g_strdup_printf ("calls-%s", enum_val->value_nick);
 
-  GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (self->recv_pipeline),
+  GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (self->pipeline),
                                      GST_DEBUG_GRAPH_SHOW_ALL,
-                                     recv_fname);
-  GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (self->send_pipeline),
-                                     GST_DEBUG_GRAPH_SHOW_ALL,
-                                     send_fname);
+                                     fname);
 }
 
 
@@ -206,25 +196,26 @@ check_element_maps (CallsSipMediaPipeline *self)
 {
   g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
 
-  if (self->element_map_playing == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
+  /* TODO take encryption into account */
+  if (self->element_map_playing == EL_ALL_RTP) {
     g_debug ("All pipeline elements are playing");
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
     return;
   }
 
-  if (self->element_map_paused == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
+  if (self->element_map_paused == EL_ALL_RTP) {
     g_debug ("All pipeline elements are paused");
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
     return;
   }
 
-  if (self->element_map_stopped == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
+  if (self->element_map_stopped == EL_ALL_RTP) {
     g_debug ("All pipeline elements are stopped");
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
     return;
   }
 
-  if ((self->element_map_playing & (EL_SEND_SENDING)) == (EL_SEND_SENDING) &&
+  if ((self->element_map_playing & (EL_SENDING)) == (EL_SENDING) &&
       !self->emitted_sending_signal) {
     g_debug ("Sender pipeline is sending data to %s RTP/RTCP %d/%d",
              self->remote, self->rport_rtp, self->rport_rtcp);
@@ -301,42 +292,46 @@ on_bus_message (GstBus     *bus,
                gst_element_state_get_name (oldstate),
                gst_element_state_get_name (newstate));
 
-      /* Sender pipeline elements */
-      if (message->src == GST_OBJECT (self->send_pipeline))
-        element_id = EL_SEND_PIPELINE;
-      else if (message->src == GST_OBJECT (self->audiosrc))
-        element_id = EL_SEND_AUDIO_SRC;
-      else if (message->src == GST_OBJECT (self->send_rtpbin))
-        element_id = EL_SEND_RTPBIN;
-      else if (message->src == GST_OBJECT (self->rtp_sink))
-        element_id = EL_SEND_RTP_SINK;
-      else if (message->src == GST_OBJECT (self->rtcp_send_sink))
-        element_id = EL_SEND_RTCP_SINK;
-      else if (message->src == GST_OBJECT (self->rtcp_send_src))
-        element_id = EL_SEND_RTCP_SRC;
-      else if (message->src == GST_OBJECT (self->payloader))
-        element_id = EL_SEND_PAYLOADER;
-      else if (message->src == GST_OBJECT (self->encoder))
-        element_id = EL_SEND_ENCODER;
-      /* Receiver pipeline elements */
-      else if (message->src == GST_OBJECT (self->recv_pipeline))
-        element_id = EL_RECV_PIPELINE;
-      else if (message->src == GST_OBJECT (self->audiosink))
-        element_id = EL_RECV_AUDIO_SINK;
-      else if (message->src == GST_OBJECT (self->recv_rtpbin))
-        element_id = EL_RECV_RTPBIN;
+      if (message->src == GST_OBJECT (self->pipeline))
+        element_id = EL_PIPELINE;
+      else if (message->src == GST_OBJECT (self->rtpbin))
+        element_id = EL_RTPBIN;
+
       else if (message->src == GST_OBJECT (self->rtp_src))
-        element_id = EL_RECV_RTP_SRC;
-      else if (message->src == GST_OBJECT (self->rtcp_recv_sink))
-        element_id = EL_RECV_RTCP_SINK;
-      else if (message->src == GST_OBJECT (self->rtcp_recv_src))
-        element_id = EL_RECV_RTCP_SRC;
+        element_id = EL_RTP_SRC;
+      else if (message->src == GST_OBJECT (self->rtp_sink))
+        element_id = EL_RTP_SINK;
+
+      else if (message->src == GST_OBJECT (self->rtcp_src))
+        element_id = EL_RTCP_SRC;
+      else if (message->src == GST_OBJECT (self->rtcp_sink))
+        element_id = EL_RTCP_SINK;
+
+      /* TODO srtp encryption
+      else if (message->src == GST_OBJECT (self->srtpenc))
+        element_id = EL_SRTP_ENCODER;
+      else if (message->src == GST_OBJECT (self->srtpdec))
+        element_id = EL_SRTP_DECODER;
+      */
+
+
+      else if (message->src == GST_OBJECT (self->audio_src))
+        element_id = EL_AUDIO_SRC;
+      else if (message->src == GST_OBJECT (self->audio_sink))
+        element_id = EL_AUDIO_SINK;
+
+      else if (message->src == GST_OBJECT (self->payloader))
+        element_id = EL_PAYLOADER;
       else if (message->src == GST_OBJECT (self->depayloader))
-        element_id = EL_RECV_DEPAYLOADER;
+        element_id = EL_DEPAYLOADER;
+
+      else if (message->src == GST_OBJECT (self->encoder))
+        element_id = EL_ENCODER;
       else if (message->src == GST_OBJECT (self->decoder))
-        element_id = EL_RECV_DECODER;
+        element_id = EL_DECODER;
 
       unset_element_id = G_MAXUINT ^ element_id;
+
       if (newstate == GST_STATE_PLAYING) {
         self->element_map_playing |= element_id;
         self->element_map_paused &= unset_element_id;
@@ -366,118 +361,52 @@ on_bus_message (GstBus     *bus,
 }
 
 
-/* Setting up pipelines */
+/* Pipeline setup */
 
 static gboolean
-send_pipeline_link_elements (CallsSipMediaPipeline *self,
-                             GError               **error)
+setup_socket_reuse (CallsSipMediaPipeline *self,
+                    GError               **error)
 {
-  g_autoptr (GstPad) srcpad = NULL;
-  g_autoptr (GstPad) sinkpad = NULL;
+  g_autoptr (GSocket) rtp_sock = NULL;
+  g_autoptr (GSocket) rtcp_sock = NULL;
 
-  g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
+  /* Set udp sources to ready to get ports allocated */
+  gst_element_set_state (self->pipeline, GST_STATE_READY);
 
-#if GST_CHECK_VERSION (1, 20, 0)
-  sinkpad = gst_element_request_pad_simple (self->send_rtpbin, "send_rtp_sink_0");
-#else
-  sinkpad = gst_element_get_request_pad (self->send_rtpbin, "send_rtp_sink_0");
-#endif
-  srcpad = gst_element_get_static_pad (self->payloader, "src");
-  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
+  g_object_get (self->rtp_src, "used-socket", &rtp_sock, NULL);
+  if (!rtp_sock) {
     if (error)
       g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Failed to link payloader to rtpbin");
+                   "Could not retrieve used socket from RTP udpsrc element");
     return FALSE;
   }
 
-  gst_object_unref (srcpad);
-  gst_object_unref (sinkpad);
-
-  /* link RTP srcpad to udpsink */
-  srcpad = gst_element_get_static_pad (self->send_rtpbin, "send_rtp_src_0");
-  sinkpad = gst_element_get_static_pad (self->rtp_sink, "sink");
-  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
+  g_object_get (self->rtcp_src, "used-socket", &rtcp_sock, NULL);
+  if (!rtcp_sock) {
     if (error)
       g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Failed to link rtpbin to rtpsink");
+                   "Could not retrieve used socket from RTCP udpsrc element");
     return FALSE;
   }
 
-  gst_object_unref (srcpad);
-  gst_object_unref (sinkpad);
+  /* Ports are allocated. Let's reuse the socket for rtcp source in the sink for NAT traversal*/
+  g_object_set (self->rtp_sink,
+                "socket", rtp_sock,
+                "close-socket", FALSE,
+                NULL);
 
-  /* RTCP srcpad to udpsink */
-#if GST_CHECK_VERSION (1, 20, 0)
-  srcpad = gst_element_request_pad_simple (self->send_rtpbin, "send_rtcp_src_0");
-#else
-  srcpad = gst_element_get_request_pad (self->send_rtpbin, "send_rtcp_src_0");
-#endif
-  sinkpad = gst_element_get_static_pad (self->rtcp_send_sink, "sink");
-  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Failed to link rtpbin to rtcpsink");
-    return FALSE;
-  }
-
-  gst_object_unref (srcpad);
-  gst_object_unref (sinkpad);
-
-  /* receive RTCP */
-  srcpad = gst_element_get_static_pad (self->rtcp_send_src, "src");
-#if GST_CHECK_VERSION (1, 20, 0)
-  sinkpad = gst_element_request_pad_simple (self->send_rtpbin, "recv_rtcp_sink_0");
-#else
-  sinkpad = gst_element_get_request_pad (self->send_rtpbin, "recv_rtcp_sink_0");
-#endif
-  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
-    if (error)
-    g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                 "Failed to link rtcpsrc to rtpbin");
-    return FALSE;
-  }
+  g_object_set (self->rtcp_sink,
+                "socket", rtcp_sock,
+                "close-socket", FALSE,
+                NULL);
 
   return TRUE;
 }
 
 
 static gboolean
-send_pipeline_setup_codecs (CallsSipMediaPipeline *self,
-                            MediaCodecInfo        *codec,
-                            GError               **error)
-{
-  g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
-  g_assert (codec);
-
-  MAKE_ELEMENT (encoder, codec->gst_encoder_name, "encoder");
-  MAKE_ELEMENT (payloader, codec->gst_payloader_name, "payloader");
-
-  gst_bin_add_many (GST_BIN (self->send_pipeline), self->payloader, self->encoder,
-                    self->audiosrc, NULL);
-
-  if (!gst_element_link_many (self->audiosrc, self->encoder, self->payloader, NULL)) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Failed to link audiosrc encoder and payloader");
-    return FALSE;
-  }
-
-  return send_pipeline_link_elements (self, error);
-}
-
-/**
- * Prepare a skeleton send pipeline where we can later
- * plug the codec specific elements into.
- *
- * In contrast to the receiver pipeline there is no need to start the
- * pipeline until we actually want to establish a media session.
- *
- * The receiver pipeline should have been initialized at this point
- * allowing us to reuse GSockets.
- */
-static gboolean
-send_pipeline_init (CallsSipMediaPipeline *self,
-                    GError               **error)
+pipeline_init (CallsSipMediaPipeline *self,
+               GError               **error)
 {
   g_autoptr (GSocket) rtp_sock = NULL;
   g_autoptr (GSocket) rtcp_sock = NULL;
@@ -485,24 +414,25 @@ send_pipeline_init (CallsSipMediaPipeline *self,
 
   g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
 
-  self->send_pipeline = gst_pipeline_new ("rtp-send-pipeline");
+  self->pipeline = gst_pipeline_new ("media-pipeline");
 
-  if (!self->send_pipeline) {
+  if (!self->pipeline) {
     if (error)
       g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Could not create send pipeline");
+                   "Could not create media pipeline");
     return FALSE;
   }
 
-  gst_object_ref_sink (self->send_pipeline);
+  gst_object_ref_sink (self->pipeline);
 
+  /* Audio source*/
   env_var = g_getenv ("CALLS_AUDIOSRC");
   if (!STR_IS_NULL_OR_EMPTY (env_var)) {
-    MAKE_ELEMENT (audiosrc, env_var, "audiosource");
+    MAKE_ELEMENT (audio_src, env_var, "audiosource");
   } else {
     g_autoptr (GstStructure) gst_props = NULL;
 
-    MAKE_ELEMENT (audiosrc, "pulsesrc", "audiosource");
+    MAKE_ELEMENT (audio_src, "pulsesrc", "audiosource");
 
     /* enable echo cancellation and set buffer size to 40ms */
     gst_props = gst_structure_new ("props",
@@ -510,22 +440,61 @@ send_pipeline_init (CallsSipMediaPipeline *self,
                                    "filter.want", G_TYPE_STRING, "echo-cancel",
                                    NULL);
 
-    g_object_set (self->audiosrc,
+    g_object_set (self->audio_src,
                   "buffer-time", (gint64) 40000,
                   "stream-properties", gst_props,
                   NULL);
   }
 
-  MAKE_ELEMENT (send_rtpbin, "rtpbin", "send-rtpbin");
-  MAKE_ELEMENT (rtp_sink, "udpsink", "rtp-udp-sink");
-  MAKE_ELEMENT (rtcp_send_src, "udpsrc", "rtcp-udp-send-src");
-  MAKE_ELEMENT (rtcp_send_sink, "udpsink", "rtcp-udp-send-sink");
+  /* Audio sink */
+  env_var = g_getenv ("CALLS_AUDIOSINK");
+  if (!STR_IS_NULL_OR_EMPTY (env_var)) {
+    MAKE_ELEMENT (audio_sink, env_var, "audiosink");
+  } else {
+    g_autoptr (GstStructure) gst_props = NULL;
 
-  g_object_set (self->rtcp_send_sink,
-                "async", FALSE,
-                "sync", FALSE,
+    MAKE_ELEMENT (audio_sink, "pulsesink", "audiosink");
+
+    /* enable echo cancellation and set buffer size to 40ms */
+    gst_props = gst_structure_new ("props",
+                                   "media.role", G_TYPE_STRING, "phone",
+                                   "filter.want", G_TYPE_STRING, "echo-cancel",
+                                   NULL);
+
+    g_object_set (self->audio_sink,
+                  "buffer-time", (gint64) 40000,
+                  "stream-properties", gst_props,
+                  NULL);
+
+  }
+
+
+  /* rtpbin */
+  MAKE_ELEMENT (rtpbin, "rtpbin", "rtpbin");
+
+  /* UDP sources and sinks for RTP and RTCP */
+  MAKE_ELEMENT (rtp_src, "udpsrc", "rtp-udp-src");
+  MAKE_ELEMENT (rtp_sink, "udpsink", "rtp-udp-sink");
+
+  MAKE_ELEMENT (rtcp_src, "udpsrc", "rtcp-udp-src");
+  MAKE_ELEMENT (rtcp_sink, "udpsink", "rtcp-udp-sink");
+
+  /* port 0 means letting the OS allocate */
+  g_object_set (self->rtp_src,
+                "port", 0,
+                "close-socket", FALSE,
+                "reuse", TRUE,
                 NULL);
 
+  g_object_set (self->rtcp_src,
+                "port", 0,
+                "close-socket", FALSE,
+                "reuse", TRUE,
+                NULL);
+
+  g_object_set (self->rtp_sink, "async", FALSE, "sync", FALSE, NULL);
+  g_object_set (self->rtcp_sink, "async", FALSE, "sync", FALSE, NULL);
+
   g_object_bind_property (self, "rport-rtp",
                           self->rtp_sink, "port",
                           G_BINDING_BIDIRECTIONAL);
@@ -535,69 +504,65 @@ send_pipeline_init (CallsSipMediaPipeline *self,
                           G_BINDING_BIDIRECTIONAL);
 
   g_object_bind_property (self, "rport-rtcp",
-                          self->rtcp_send_sink, "port",
+                          self->rtcp_sink, "port",
                           G_BINDING_BIDIRECTIONAL);
 
   g_object_bind_property (self, "remote",
-                          self->rtcp_send_sink, "host",
+                          self->rtcp_sink, "host",
                           G_BINDING_BIDIRECTIONAL);
 
-  /* bind sockets for hole punching scheme */
-  g_object_get (self->rtp_src, "used-socket", &rtp_sock, NULL);
-  if (!rtp_sock) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Could not retrieve used socket from RTP udpsrc element");
+
+  /* Add all elements to the pipeline */
+  gst_bin_add_many (GST_BIN (self->pipeline),
+                    self->audio_src, self->audio_sink,
+                    self->rtpbin,
+                    self->rtp_src, self->rtp_sink,
+                    self->rtcp_src, self->rtcp_sink,
+                    NULL);
+
+  /* Setup bus watch */
+  self->bus = gst_pipeline_get_bus (GST_PIPELINE (self->pipeline));
+  self->bus_watch_id = gst_bus_add_watch (self->bus, on_bus_message, self);
+
+  if (!setup_socket_reuse (self, error))
     return FALSE;
-  }
-
-  g_object_set (self->rtp_sink,
-                "socket", rtp_sock,
-                "close-socket", FALSE,
-                NULL);
-
-
-  g_object_get (self->rtcp_recv_src, "used-socket", &rtcp_sock, NULL);
-  if (!rtcp_sock) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Could not retrieve used socket from RTCP udpsrc element");
-    return FALSE;
-  }
-  g_object_set (self->rtcp_send_sink,
-                "socket", rtcp_sock,
-                "close-socket", FALSE,
-                NULL);
-  g_object_set (self->rtcp_send_src,
-                "socket", rtcp_sock,
-                "close-socket", FALSE,
-                NULL);
-
-  gst_bin_add (GST_BIN (self->send_pipeline), self->send_rtpbin);
-  gst_bin_add_many (GST_BIN (self->send_pipeline), self->rtp_sink,
-                    self->rtcp_send_src, self->rtcp_send_sink, NULL);
-
-  self->bus_send = gst_pipeline_get_bus (GST_PIPELINE (self->send_pipeline));
-  self->bus_watch_send = gst_bus_add_watch (self->bus_send, on_bus_message, self);
 
   return TRUE;
 }
 
 
 static gboolean
-recv_pipeline_link_elements (CallsSipMediaPipeline *self,
-                             GError               **error)
+pipeline_link_elements (CallsSipMediaPipeline *self,
+                        GError               **error)
 {
   g_autoptr (GstPad) srcpad = NULL;
   g_autoptr (GstPad) sinkpad = NULL;
 
   g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
 
+  /* link to payloader */
+
+#if GST_CHECK_VERSION (1, 20, 0)
+  sinkpad = gst_element_request_pad_simple (self->rtpbin, "send_rtp_sink_0");
+#else
+  sinkpad = gst_element_get_request_pad (self->rtpbin, "send_rtp_sink_0");
+#endif
+  srcpad = gst_element_get_static_pad (self->payloader, "src");
+  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
+    if (error)
+      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
+                   "Failed to link payloader to rtpbin");
+    return FALSE;
+  }
+
+
+  /* Transmitter pads */
+
   srcpad = gst_element_get_static_pad (self->rtp_src, "src");
 #if GST_CHECK_VERSION (1, 20, 0)
-  sinkpad = gst_element_request_pad_simple (self->recv_rtpbin, "recv_rtp_sink_0");
+  sinkpad = gst_element_request_pad_simple (self->rtpbin, "recv_rtp_sink_0");
 #else
-  sinkpad = gst_element_get_request_pad (self->recv_rtpbin, "recv_rtp_sink_0");
+  sinkpad = gst_element_get_request_pad (self->rtpbin, "recv_rtp_sink_0");
 #endif
   if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
     if (error)
@@ -609,11 +574,23 @@ recv_pipeline_link_elements (CallsSipMediaPipeline *self,
   gst_object_unref (srcpad);
   gst_object_unref (sinkpad);
 
-  srcpad = gst_element_get_static_pad (self->rtcp_recv_src, "src");
+  srcpad = gst_element_get_static_pad (self->rtpbin, "send_rtp_src_0");
+  sinkpad = gst_element_get_static_pad (self->rtp_sink, "sink");
+  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
+    if (error)
+      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
+                   "Failed to link rtpbin to rtpsink");
+    return FALSE;
+  }
+
+  gst_object_unref (srcpad);
+  gst_object_unref (sinkpad);
+
+  srcpad = gst_element_get_static_pad (self->rtcp_src, "src");
 #if GST_CHECK_VERSION (1, 20 , 0)
-  sinkpad = gst_element_request_pad_simple (self->recv_rtpbin, "recv_rtcp_sink_0");
+  sinkpad = gst_element_request_pad_simple (self->rtpbin, "recv_rtcp_sink_0");
 #else
-  sinkpad = gst_element_get_request_pad (self->recv_rtpbin, "recv_rtcp_sink_0");
+  sinkpad = gst_element_get_request_pad (self->rtpbin, "recv_rtcp_sink_0");
 #endif
   if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
     if (error)
@@ -622,32 +599,33 @@ recv_pipeline_link_elements (CallsSipMediaPipeline *self,
     return FALSE;
   }
 
-  gst_object_unref (srcpad);
-  gst_object_unref (sinkpad);
+    gst_object_unref (srcpad);
+    gst_object_unref (sinkpad);
 
-#if GST_CHECK_VERSION (1, 20, 0)
-  srcpad = gst_element_request_pad_simple (self->recv_rtpbin, "send_rtcp_src_0");
-#else
-  srcpad = gst_element_get_request_pad (self->recv_rtpbin, "send_rtcp_src_0");
-#endif
-  sinkpad = gst_element_get_static_pad (self->rtcp_recv_sink, "sink");
-  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Failed to link rtpbin to rtcpsink");
-    return FALSE;
-  }
+  #if GST_CHECK_VERSION (1, 20, 0)
+    srcpad = gst_element_request_pad_simple (self->rtpbin, "send_rtcp_src_0");
+  #else
+    srcpad = gst_element_get_request_pad (self->rtpbin, "send_rtcp_src_0");
+  #endif
+    sinkpad = gst_element_get_static_pad (self->rtcp_sink, "sink");
+    if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
+      if (error)
+        g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
+                     "Failed to link rtpbin to rtcpsink");
+      return FALSE;
+    }
 
-  g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
+  /* can only link to depayloader after RTP payload has been verified */
+  g_signal_connect (self->rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
 
   return TRUE;
 }
 
 
 static gboolean
-recv_pipeline_setup_codecs (CallsSipMediaPipeline *self,
-                            MediaCodecInfo        *codec,
-                            GError               **error)
+pipeline_setup_codecs (CallsSipMediaPipeline *self,
+                       MediaCodecInfo        *codec,
+                       GError               **error)
 {
   g_autoptr (GstCaps) caps = NULL;
   g_autofree char *caps_string = NULL;
@@ -658,10 +636,22 @@ recv_pipeline_setup_codecs (CallsSipMediaPipeline *self,
   MAKE_ELEMENT (decoder, codec->gst_decoder_name, "decoder");
   MAKE_ELEMENT (depayloader, codec->gst_depayloader_name, "depayloader");
 
-  gst_bin_add_many (GST_BIN (self->recv_pipeline), self->depayloader, self->decoder,
-                    self->audiosink, NULL);
+  MAKE_ELEMENT (encoder, codec->gst_encoder_name, "encoder");
+  MAKE_ELEMENT (payloader, codec->gst_payloader_name, "payloader");
 
-  if (!gst_element_link_many (self->depayloader, self->decoder, self->audiosink, NULL)) {
+  gst_bin_add_many (GST_BIN (self->pipeline),
+                    self->depayloader, self->decoder,
+                    self->payloader, self->encoder,
+                    NULL);
+
+  if (!gst_element_link_many (self->audio_src, self->encoder, self->payloader, NULL)) {
+    if (error)
+      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
+                   "Failed to link audiosrc encoder and payloader");
+    return FALSE;
+  }
+
+  if (!gst_element_link_many (self->depayloader, self->decoder, self->audio_sink, NULL)) {
     if (error)
       g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
                    "Failed to link depayloader decoder and audiosink");
@@ -679,119 +669,6 @@ recv_pipeline_setup_codecs (CallsSipMediaPipeline *self,
                 "caps", caps,
                 NULL);
 
-  return recv_pipeline_link_elements (self, error);
-}
-
-
-/**
- * Prepares a skeleton receiver pipeline which can later be
- * used to plug codec specific element in.
- * This pipeline just consists of (minimally linked) rtpbin
- * audio sink and two udpsrc elements, one for RTP and one for RTCP.
- *
- * The pipeline will be set ready to let the OS allocate sockets
- * for us instead of building and providing GSockets ourselves
- * by hand. These GSockets are reused for any outgoing traffic in our
- * hole punching scheme as a simple NAT traversal technique.
- */
-static gboolean
-recv_pipeline_init (CallsSipMediaPipeline *self,
-                    GError               **error)
-{
-  g_autoptr (GSocket) rtcp_sock = NULL;
-  const char *env_var;
-
-  g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
-
-  self->recv_pipeline = gst_pipeline_new ("rtp-recv-pipeline");
-
-  if (!self->recv_pipeline) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Could not create receiver pipeline");
-    return FALSE;
-  }
-
-  gst_object_ref_sink (self->recv_pipeline);
-
-  env_var = g_getenv ("CALLS_AUDIOSINK");
-  if (!STR_IS_NULL_OR_EMPTY (env_var)) {
-    MAKE_ELEMENT (audiosink, env_var, "audiosink");
-  } else {
-      g_autoptr (GstStructure) gst_props = NULL;
-
-      MAKE_ELEMENT (audiosink, "pulsesink", "audiosink");
-
-      /* enable echo cancellation and set buffer size to 40ms */
-      gst_props = gst_structure_new ("props",
-                                     "media.role", G_TYPE_STRING, "phone",
-                                     "filter.want", G_TYPE_STRING, "echo-cancel",
-                                     NULL);
-
-      g_object_set (self->audiosink,
-                    "buffer-time", (gint64) 40000,
-                    "stream-properties", gst_props,
-                    NULL);
-
-  }
-
-  MAKE_ELEMENT (recv_rtpbin, "rtpbin", "recv-rtpbin")
-  MAKE_ELEMENT (rtp_src, "udpsrc", "rtp-udp-src");
-  MAKE_ELEMENT (rtcp_recv_src, "udpsrc", "rtcp-udp-recv-src");
-  MAKE_ELEMENT (rtcp_recv_sink, "udpsink", "rtcp-udp-recv-sink");
-
-
-  g_object_set (self->rtcp_recv_sink,
-                "async", FALSE,
-                "sync", FALSE,
-                NULL);
-
-
-  /* port 0 means allocate */
-  g_object_set (self->rtp_src, "port", 0, NULL);
-  g_object_set (self->rtcp_recv_src, "port", 0, NULL);
-
-  g_object_bind_property (self, "rport-rtcp",
-                          self->rtcp_recv_sink, "port",
-                          G_BINDING_BIDIRECTIONAL);
-
-  g_object_bind_property (self, "remote",
-                          self->rtcp_recv_sink, "host",
-                          G_BINDING_BIDIRECTIONAL);
-
-  gst_bin_add (GST_BIN (self->recv_pipeline), self->recv_rtpbin);
-  gst_bin_add_many (GST_BIN (self->recv_pipeline), self->rtp_src,
-                    self->rtcp_recv_src, self->rtcp_recv_sink, NULL);
-
-  self->bus_recv = gst_pipeline_get_bus (GST_PIPELINE (self->recv_pipeline));
-  self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, on_bus_message, self);
-
-  g_object_set (self->rtp_src,
-                "close-socket", FALSE,
-                "reuse", TRUE,
-                NULL);
-  g_object_set (self->rtcp_recv_src,
-                "close-socket", FALSE,
-                "reuse", TRUE,
-                NULL);
-
-  /* Set pipeline to ready to get ports allocated */
-  gst_element_set_state (self->recv_pipeline, GST_STATE_READY);
-
-  g_object_get (self->rtcp_recv_src, "used-socket", &rtcp_sock, NULL);
-  if (!rtcp_sock) {
-    if (error)
-      g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
-                   "Could not retrieve used socket from RTCP udpsrc element");
-    return FALSE;
-  }
-
-  /* Ports are allocated. Let's reuse the socket for rtcp source in the sink for NAT traversal*/
-  g_object_set (self->rtcp_recv_sink,
-                "socket", rtcp_sock,
-                "close-socket", FALSE,
-                NULL);
-
   return TRUE;
 }
 
@@ -883,14 +760,8 @@ calls_sip_media_pipeline_constructed (GObject *object)
 
   set_state (self, CALLS_MEDIA_PIPELINE_STATE_INITIALIZING);
 
-  if (!recv_pipeline_init (self, &error)) {
-    g_warning ("Could not create receive pipeline: %s", error->message);
-    set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
-    return;
-  }
-
-  if (!send_pipeline_init (self, &error)) {
-    g_warning ("Could not create send pipeline: %s", error->message);
+  if (!pipeline_init (self, &error)) {
+    g_warning ("Could not create pipeline: %s", error->message);
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
     return;
   }
@@ -906,12 +777,9 @@ calls_sip_media_pipeline_finalize (GObject *object)
 
   calls_sip_media_pipeline_stop (self);
 
-  gst_object_unref (self->send_pipeline);
-  gst_object_unref (self->recv_pipeline);
-  gst_bus_remove_watch (self->bus_send);
-  gst_object_unref (self->bus_send);
-  gst_bus_remove_watch (self->bus_recv);
-  gst_object_unref (self->bus_recv);
+  gst_object_unref (self->pipeline);
+  gst_bus_remove_watch (self->bus);
+  gst_object_unref (self->bus);
 
   g_free (self->remote);
 
@@ -1022,15 +890,15 @@ calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
     return;
   }
 
-  if (!recv_pipeline_setup_codecs (self, codec, &error)) {
-    g_warning ("Error trying to setup codec for receive pipeline: %s",
+  if (!pipeline_setup_codecs (self, codec, &error)) {
+    g_warning ("Error trying to setup codecs for pipeline: %s",
                error->message);
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
     return;
   }
 
-  if (!send_pipeline_setup_codecs (self, codec, &error)) {
-    g_warning ("Error trying to setup codec for send pipeline: %s",
+  if (!pipeline_link_elements (self, &error)) {
+    g_warning ("Not all pads could be linked: %s",
                error->message);
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
     return;
@@ -1042,6 +910,7 @@ calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
   set_state (self, CALLS_MEDIA_PIPELINE_STATE_READY);
 }
 
+
 static void
 diagnose_used_ports_in_socket (GSocket *socket)
 {
@@ -1122,8 +991,11 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
 
   g_debug ("Starting media pipeline");
 
-  gst_element_set_state (self->recv_pipeline, GST_STATE_PLAYING);
-  gst_element_set_state (self->send_pipeline, GST_STATE_PLAYING);
+  g_debug ("RTP/RTCP port before starting pipeline: %d/%d",
+           calls_sip_media_pipeline_get_rtp_port (self),
+           calls_sip_media_pipeline_get_rtcp_port (self));
+
+  gst_element_set_state (self->pipeline, GST_STATE_PLAYING);
 
   g_debug ("RTP/RTCP port after starting pipeline: %d/%d",
            calls_sip_media_pipeline_get_rtp_port (self),
@@ -1143,9 +1015,7 @@ calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self)
 
   g_debug ("Stopping media pipeline");
 
-  /* Stop the pipelines in reverse order (compared to the starting) */
-  gst_element_set_state (self->send_pipeline, GST_STATE_NULL);
-  gst_element_set_state (self->recv_pipeline, GST_STATE_NULL);
+  gst_element_set_state (self->pipeline, GST_STATE_NULL);
 
   set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING);
 }
@@ -1179,10 +1049,8 @@ calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
            "Pausing" :
            "Unpausing");
 
-  gst_element_set_state (self->recv_pipeline, pause ?
-                         GST_STATE_PAUSED :
-                         GST_STATE_PLAYING);
-  gst_element_set_state (self->send_pipeline, pause ?
+
+  gst_element_set_state (self->pipeline, pause ?
                          GST_STATE_PAUSED :
                          GST_STATE_PLAYING);
 
@@ -1212,7 +1080,7 @@ calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self)
 
   g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self), 0);
 
-  g_object_get (self->rtcp_recv_src, "port", &port, NULL);
+  g_object_get (self->rtcp_src, "port", &port, NULL);
 
   return port;
 }