diff --git a/src/main.c b/src/main.c
index 5694e1a801b495c41cd1840eb2aa03f60b2c9e8f..c0c06e9cc7ef59727cb74a12050f4f43c5ff06b6 100644
--- a/src/main.c
+++ b/src/main.c
@@ -48,35 +48,39 @@ get_webrtcbin_for_client (GstBin *pipeline, MssClientId client_id)
   return webrtcbin;
 }
 
+static void
+connect_webrtc_to_tee (GstElement *webrtcbin)
+{
+  GstElement *pipeline;
+  GstElement *tee;
+  GstPad *srcpad;
+  GstPad *sinkpad;
+  GstPadLinkReturn ret;
+
+  pipeline = GST_ELEMENT (gst_element_get_parent (webrtcbin));
+  if (pipeline == NULL)
+    return;
+  tee = gst_bin_get_by_name (GST_BIN (pipeline), "webrtctee");
+  srcpad = gst_element_get_request_pad (tee, "src_%u");
+  sinkpad = gst_element_get_request_pad (webrtcbin, "sink_0");
+  ret = gst_pad_link (srcpad, sinkpad);
+  g_assert (ret == GST_PAD_LINK_OK);
+  gst_object_unref (srcpad);
+  gst_object_unref (sinkpad);
+  gst_object_unref (tee);
+  gst_object_unref (pipeline);
+}
+
 static void
 on_offer_created (GstPromise *promise, GstElement *webrtcbin)
 {
   GstWebRTCSessionDescription *offer = NULL;
   gchar *sdp;
-  GstSDPMedia *media;
-  guint i;
 
   gst_structure_get (gst_promise_get_reply (promise),
       "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
   gst_promise_unref (promise);
 
-  media = (GstSDPMedia *) gst_sdp_message_get_media (offer->sdp, 0);
-
-  for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
-    const GstSDPAttribute *a = gst_sdp_media_get_attribute (media, i);
-
-    if (!strcmp (a->key, "sendrecv")) {
-      GstSDPAttribute newa;
-
-      newa.key = g_strdup ("sendonly");
-      newa.value = g_strdup ("");
-      gst_sdp_media_replace_attribute (media, i, &newa);
-      break;
-    }
-  }
-  gst_sdp_media_add_attribute (media, "fmtp",
-      "96 packetization-mode=1;profile-level-id=42e01f");
-
   g_signal_emit_by_name (webrtcbin, "set-local-description", offer, NULL);
 
   sdp = gst_sdp_message_as_text (offer->sdp);
@@ -86,6 +90,8 @@ on_offer_created (GstPromise *promise, GstElement *webrtcbin)
   g_free (sdp);
 
   gst_webrtc_session_description_free (offer);
+
+  connect_webrtc_to_tee (webrtcbin);
 }
 
 static void
@@ -102,10 +108,8 @@ webrtc_client_connected_cb (MssHttpServer *server, MssClientId client_id,
     GstBin *pipeline)
 {
   gchar *name;
-  GstElement *tee;
   GstElement *webrtcbin;
-  GstPad *srcpad;
-  GstPad *sinkpad;
+  GstCaps *caps;
 
   name = g_strdup_printf ("webrtcbin_%p", client_id);
 
@@ -117,13 +121,12 @@ webrtc_client_connected_cb (MssHttpServer *server, MssClientId client_id,
   g_signal_connect (webrtcbin, "on-ice-candidate",
       G_CALLBACK (webrtc_on_ice_candidate_cb), NULL);
 
-  tee = gst_bin_get_by_name (pipeline, "webrtctee");
-  srcpad = gst_element_get_request_pad (tee, "src_%u");
-  sinkpad = gst_element_get_request_pad (webrtcbin, "sink_%u");
-  gst_pad_link (srcpad, sinkpad);
-  gst_object_unref (srcpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (tee);
+  caps = gst_caps_from_string ("application/x-rtp, payload=96,encoding-name=H264,clock-rate=90000,media=video,packetization-mode=(string)1,profile-level-id=(string)42e01f");
+  g_signal_emit_by_name (webrtcbin, "add-transceiver",
+      GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY,
+      caps);
+
+  gst_caps_unref (caps);
 
 
   g_signal_emit_by_name (webrtcbin, "create-offer", NULL,
@@ -141,17 +144,8 @@ webrtc_sdp_answer_cb (MssHttpServer *server, MssClientId client_id,
 {
   GstSDPMessage *sdp_msg = NULL;
   GstWebRTCSessionDescription *desc = NULL;
-  gchar *sdp_tmp;
-  char **split;
-
-  // TODO needed?
-  split = g_strsplit(sdp, "a=recvonly", -1);
-  sdp_tmp = g_strjoinv("a=sendrecv", split);
-  g_strfreev(split);
-
-  g_debug ("ANSWER: %s", sdp_tmp);
 
-  if (gst_sdp_message_new_from_text (sdp_tmp, &sdp_msg) != GST_SDP_OK) {
+  if (gst_sdp_message_new_from_text (sdp, &sdp_msg) != GST_SDP_OK) {
     g_debug ("Error parsing SDP description");
     goto out;
   }
@@ -179,7 +173,6 @@ webrtc_sdp_answer_cb (MssHttpServer *server, MssClientId client_id,
   }
 
 out:
-  g_free (sdp_tmp);
   g_clear_pointer (&desc, gst_webrtc_session_description_free);
 }