diff --git a/plugins/sip/calls-sip-media-pipeline.c b/plugins/sip/calls-sip-media-pipeline.c
index ca9afc6..0d7b762 100644
--- a/plugins/sip/calls-sip-media-pipeline.c
+++ b/plugins/sip/calls-sip-media-pipeline.c
@@ -24,6 +24,7 @@
 
 #define G_LOG_DOMAIN "CallsSipMediaPipeline"
 
+#include "calls-media-pipeline-enums.h"
 #include "calls-sip-media-pipeline.h"
 #include "util.h"
 
@@ -57,6 +58,39 @@
  * Both pipelines are using RTCP.
  */
 
+
+/* 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_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
+
+/* leave some room for more elements to be added later */
+
+#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_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
+
+
 enum {
   PROP_0,
   PROP_CODEC,
@@ -66,15 +100,30 @@ enum {
   PROP_LPORT_RTCP,
   PROP_RPORT_RTCP,
   PROP_DEBUG,
+  PROP_STATE,
   PROP_LAST_PROP,
 };
+
+enum {
+  SENDING_STARTED,
+  N_SIGNALS
+};
+
 static GParamSpec *props[PROP_LAST_PROP];
+static uint signals[N_SIGNALS];
+
 
 struct _CallsSipMediaPipeline {
   GObject parent;
 
   MediaCodecInfo *codec;
   gboolean debug;
+
+  CallsMediaPipelineState state;
+  uint element_map_playing;
+  uint element_map_paused;
+  uint element_map_stopped;
+  gboolean emitted_sending_signal;
   /* Connection details */
   char *remote;
 
@@ -84,8 +133,6 @@ struct _CallsSipMediaPipeline {
   gint rport_rtcp;
   gint lport_rtcp;
 
-  gboolean is_running;
-
   /* Gstreamer Elements (sending) */
   GstElement *send_pipeline;
   GstElement *audiosrc;
@@ -119,6 +166,56 @@ static void initable_iface_init (GInitableIface *iface);
 G_DEFINE_TYPE_WITH_CODE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT,
                          G_IMPLEMENT_INTERFACE (G_TYPE_INITABLE, initable_iface_init));
 
+
+static void
+set_state (CallsSipMediaPipeline  *self,
+           CallsMediaPipelineState state)
+{
+  g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
+
+  if (self->state == state)
+    return;
+
+  self->state = state;
+  g_object_notify_by_pspec (G_OBJECT (self), props[PROP_STATE]);
+
+  self->emitted_sending_signal = FALSE;
+}
+
+
+static void
+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)) {
+    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)) {
+    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)) {
+    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) &&
+      !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);
+    g_signal_emit (self, signals[SENDING_STARTED], 0);
+    self->emitted_sending_signal = TRUE;
+  }
+}
+
+
 /* rtpbin adds a pad once the payload is verified */
 static void
 on_pad_added (GstElement *rtpbin,
@@ -144,7 +241,7 @@ on_bus_message (GstBus     *bus,
                 GstMessage *message,
                 gpointer    data)
 {
-  CallsSipMediaPipeline *pipeline = CALLS_SIP_MEDIA_PIPELINE (data);
+  CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (data);
 
   switch (GST_MESSAGE_TYPE (message)) {
   case GST_MESSAGE_ERROR:
@@ -169,24 +266,79 @@ on_bus_message (GstBus     *bus,
 
   case GST_MESSAGE_EOS:
     g_debug ("Received end of stream");
-    calls_sip_media_pipeline_stop (pipeline);
+    calls_sip_media_pipeline_stop (self);
     break;
 
   case GST_MESSAGE_STATE_CHANGED:
     {
       GstState oldstate;
       GstState newstate;
+      uint element_id = 0;
+      uint unset_element_id;
 
       gst_message_parse_state_changed (message, &oldstate, &newstate, NULL);
+
       g_debug ("Element %s has changed state from %s to %s",
                GST_OBJECT_NAME (message->src),
                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;
+      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;
+      else if (message->src == GST_OBJECT (self->depayloader))
+        element_id = EL_RECV_DEPAYLOADER;
+      else if (message->src == GST_OBJECT (self->decoder))
+        element_id = EL_RECV_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;
+        self->element_map_stopped &= unset_element_id;
+      } else if (newstate == GST_STATE_PAUSED) {
+        self->element_map_paused |= element_id;
+        self->element_map_playing &= unset_element_id;
+        self->element_map_stopped &= unset_element_id;
+      } else if (newstate == GST_STATE_NULL) {
+        self->element_map_stopped |= element_id;
+        self->element_map_playing &= unset_element_id;
+        self->element_map_paused &= unset_element_id;
+      }
+
+      check_element_maps (self);
       break;
     }
 
   default:
-    if (pipeline->debug)
+    if (self->debug)
       g_debug ("Got unhandled %s message", GST_MESSAGE_TYPE_NAME (message));
     break;
   }
@@ -610,6 +762,10 @@ calls_sip_media_pipeline_get_property (GObject    *object,
     g_value_set_boolean (value, self->debug);
     break;
 
+  case PROP_STATE:
+    g_value_set_enum (value, calls_sip_media_pipeline_get_state (self));
+    break;
+
   default:
     G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
     break;
@@ -733,7 +889,21 @@ calls_sip_media_pipeline_class_init (CallsSipMediaPipelineClass *klass)
                                             FALSE,
                                             G_PARAM_READWRITE);
 
+  props[PROP_STATE] = g_param_spec_enum ("state",
+                                         "State",
+                                         "The state of the media pipeline",
+                                         CALLS_TYPE_MEDIA_PIPELINE_STATE,
+                                         CALLS_MEDIA_PIPELINE_STATE_UNKNOWN,
+                                         G_PARAM_READABLE);
+
   g_object_class_install_properties (object_class, PROP_LAST_PROP, props);
+
+  signals[SENDING_STARTED] =
+    g_signal_new ("sending-started",
+                  G_TYPE_FROM_CLASS (klass),
+                  G_SIGNAL_RUN_LAST,
+                  0, NULL, NULL, NULL,
+                  G_TYPE_NONE, 0);
 }
 
 
@@ -752,13 +922,20 @@ pipelines_initable_init (GInitable    *initable,
 {
   CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (initable);
 
+  set_state (self, CALLS_MEDIA_PIPELINE_STATE_INITIALIZING);
+
   if (!recv_pipeline_init (self, cancellable, error))
-    return FALSE;
+    goto err;
 
   if (!send_pipeline_init (self, cancellable, error))
-    return FALSE;
+    goto err;
 
+  set_state (self, CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC);
   return TRUE;
+
+ err:
+  set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
+  return FALSE;
 }
 
 
@@ -815,17 +992,21 @@ calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
   if (!recv_pipeline_setup_codecs (self, codec, &error)) {
     g_warning ("Error trying to setup codec for receive 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",
                error->message);
+    set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
     return;
   }
 
   self->codec = codec;
   g_object_notify_by_pspec (G_OBJECT (self), props[PROP_CODEC]);
+
+  set_state (self, CALLS_MEDIA_PIPELINE_STATE_READY);
 }
 
 static void
@@ -867,7 +1048,12 @@ diagnose_ports_in_use (CallsSipMediaPipeline *self)
   gboolean same_socket = FALSE;
 
   g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
-  g_assert (self->is_running);
+
+  if (self->state != CALLS_MEDIA_PIPELINE_STATE_PLAYING &&
+      self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSED) {
+    g_warning ("Cannot diagnose ports when pipeline is not active");
+    return;
+  }
 
   g_object_get (self->rtp_src, "used-socket", &socket_in, NULL);
   g_object_get (self->rtp_sink, "used-socket", &socket_out, NULL);
@@ -897,13 +1083,12 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
   GSocket *socket;
   g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
 
-  if (!self->codec) {
-    g_warning ("Codec not set for this pipeline. Cannot start");
+  if (self->state != CALLS_MEDIA_PIPELINE_STATE_READY) {
+    g_warning ("Cannot start pipeline because it's not ready");
     return;
   }
 
   g_debug ("Starting media pipeline");
-  self->is_running = TRUE;
 
   /* First start the receiver pipeline so that
      we may reuse the socket in the sender pipeline */
@@ -924,9 +1109,10 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
   /* Now start the sender pipeline */
   gst_element_set_state (self->send_pipeline, GST_STATE_PLAYING);
 
+  set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
+
   if (self->debug)
     diagnose_ports_in_use (self);
-
 }
 
 
@@ -936,11 +1122,12 @@ calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self)
   g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
 
   g_debug ("Stopping media pipeline");
-  self->is_running = FALSE;
 
   /* 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);
+
+  set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING);
 }
 
 
@@ -950,20 +1137,38 @@ calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
 {
   g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
 
-  if (self->is_running != pause)
+  if (pause &&
+      (self->state == CALLS_MEDIA_PIPELINE_STATE_PAUSED ||
+       self->state == CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING))
     return;
 
-  g_debug ("%s media pipeline", self->is_running ?
+  if (!pause &&
+      (self->state == CALLS_MEDIA_PIPELINE_STATE_PLAYING ||
+       self->state == CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING))
+    return;
+
+  if (self->state != CALLS_MEDIA_PIPELINE_STATE_PLAYING &&
+      self->state != CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING &&
+      self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSED &&
+      self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING) {
+    g_warning ("Cannot pause or unpause pipeline because it's not currently active");
+    return;
+  }
+
+  g_debug ("%s media pipeline", pause ?
            "Pausing" :
            "Unpausing");
-  gst_element_set_state (self->recv_pipeline, self->is_running ?
+
+  gst_element_set_state (self->recv_pipeline, pause ?
                          GST_STATE_PAUSED :
                          GST_STATE_PLAYING);
-  gst_element_set_state (self->send_pipeline, self->is_running ?
+  gst_element_set_state (self->send_pipeline, pause ?
                          GST_STATE_PAUSED :
                          GST_STATE_PLAYING);
 
-  self->is_running = !self->is_running;
+  set_state (self, pause ?
+             CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING :
+             CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
 }
 
 
@@ -992,4 +1197,38 @@ calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self)
   return port;
 }
 
+
+CallsMediaPipelineState
+calls_sip_media_pipeline_get_state (CallsSipMediaPipeline *self)
+{
+  g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self),
+                        CALLS_MEDIA_PIPELINE_STATE_UNKNOWN);
+
+  return self->state;
+}
+
 #undef MAKE_ELEMENT
+
+#undef EL_SEND_PIPELINE
+#undef EL_SEND_AUDIO_SRC
+#undef EL_SEND_RTPBIN
+#undef EL_SEND_RTP_SINK
+#undef EL_SEND_RTCP_SINK
+#undef EL_SEND_RTCP_SRC
+#undef EL_SEND_PAYLOADER
+#undef EL_SEND_ENCODER
+
+#undef EL_SEND_ALL_RTP
+#undef EL_SEND_SENDING
+
+#undef EL_RECV_PIPELINE
+#undef EL_RECV_AUDIO_SINK
+#undef EL_RECV_RTPBIN
+#undef EL_RECV_RTP_SRC
+#undef EL_RECV_RTCP_SINK
+#undef EL_RECV_RTCP_SRC
+#undef EL_RECV_DEPAYLOADER
+#undef EL_RECV_DECODER
+
+#undef EL_RECV_ALL_RTP
+
diff --git a/plugins/sip/calls-sip-media-pipeline.h b/plugins/sip/calls-sip-media-pipeline.h
index 4cc0115..951cc92 100644
--- a/plugins/sip/calls-sip-media-pipeline.h
+++ b/plugins/sip/calls-sip-media-pipeline.h
@@ -30,19 +30,49 @@
 
 G_BEGIN_DECLS
 
+/**
+ * CallsMediaPipelineState:
+ * @CALLS_MEDIA_PIPELINE_STATE_UNKNOWN: Default state for new pipelines
+ * @CALLS_MEDIA_PIPELINE_STATE_ERROR: Pipeline is in an error state
+ * @CALLS_MEDIA_PIPELINE_STATE_INITIALIZING: Pipeline is initializing
+ * @CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC: Pipeline was initialized and needs a codec set
+ * @CALLS_MEDIA_PIPELINE_STATE_READY: Pipeline is ready to be set into playing state
+ * @CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING: Request to start pipeline pending
+ * @CALLS_MEDIA_PIPELINE_STATE_PLAYING: Pipeline is currently playing
+ * @CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING: Request to pause pipeline pending
+ * @CALLS_MEDIA_PIPELINE_STATE_PAUSED: Pipeline is currently paused
+ * @CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING: Request to stop pipeline pending
+ * @CALLS_MEDIA_PIPELINE_STATE_STOPPED: Pipeline has stopped playing (f.e. received BYE packet)
+ */
+typedef enum {
+  CALLS_MEDIA_PIPELINE_STATE_UNKNOWN = 0,
+  CALLS_MEDIA_PIPELINE_STATE_ERROR,
+  CALLS_MEDIA_PIPELINE_STATE_INITIALIZING,
+  CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC,
+  CALLS_MEDIA_PIPELINE_STATE_READY,
+  CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING,
+  CALLS_MEDIA_PIPELINE_STATE_PLAYING,
+  CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING,
+  CALLS_MEDIA_PIPELINE_STATE_PAUSED,
+  CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING,
+  CALLS_MEDIA_PIPELINE_STATE_STOPPED
+} CallsMediaPipelineState;
+
+
 #define CALLS_TYPE_SIP_MEDIA_PIPELINE (calls_sip_media_pipeline_get_type ())
 
 G_DECLARE_FINAL_TYPE (CallsSipMediaPipeline, calls_sip_media_pipeline, CALLS, SIP_MEDIA_PIPELINE, GObject)
 
 
-CallsSipMediaPipeline* calls_sip_media_pipeline_new                    (MediaCodecInfo *codec);
-void                   calls_sip_media_pipeline_set_codec              (CallsSipMediaPipeline *self,
-                                                                        MediaCodecInfo        *info);
-void                   calls_sip_media_pipeline_start                  (CallsSipMediaPipeline *self);
-void                   calls_sip_media_pipeline_stop                   (CallsSipMediaPipeline *self);
-void                   calls_sip_media_pipeline_pause                  (CallsSipMediaPipeline *self,
-                                                                        gboolean               pause);
-int                    calls_sip_media_pipeline_get_rtp_port           (CallsSipMediaPipeline *self);
-int                    calls_sip_media_pipeline_get_rtcp_port          (CallsSipMediaPipeline *self);
+CallsSipMediaPipeline*  calls_sip_media_pipeline_new              (MediaCodecInfo *codec);
+void                    calls_sip_media_pipeline_set_codec        (CallsSipMediaPipeline *self,
+                                                                   MediaCodecInfo        *info);
+void                    calls_sip_media_pipeline_start            (CallsSipMediaPipeline *self);
+void                    calls_sip_media_pipeline_stop             (CallsSipMediaPipeline *self);
+void                    calls_sip_media_pipeline_pause            (CallsSipMediaPipeline *self,
+                                                                   gboolean               pause);
+int                     calls_sip_media_pipeline_get_rtp_port     (CallsSipMediaPipeline *self);
+int                     calls_sip_media_pipeline_get_rtcp_port    (CallsSipMediaPipeline *self);
+CallsMediaPipelineState calls_sip_media_pipeline_get_state        (CallsSipMediaPipeline *self);
 
 G_END_DECLS
diff --git a/plugins/sip/meson.build b/plugins/sip/meson.build
index f092d15..4e97c84 100644
--- a/plugins/sip/meson.build
+++ b/plugins/sip/meson.build
@@ -55,6 +55,14 @@ sip_sources = files(
   ]
 )
 
+pipeline_enum_headers = [
+  'calls-sip-media-pipeline.h',
+]
+
+pipeline_enums = gnome.mkenums_simple('calls-media-pipeline-enums',
+                                      sources: pipeline_enum_headers)
+sip_sources += pipeline_enums
+
 sip_enum_headers = [
   'calls-sip-util.h',
 ]