diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index bf032dcf92..3084087663 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -296,6 +296,12 @@ static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
     GstRTSPStream * stream, GstEvent * event);
 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
 
+typedef struct
+{
+  guint8 pt;
+  GstCaps *caps;
+} PtMapItem;
+
 /* commands we send to out loop to notify it of events */
 #define CMD_OPEN	(1 << 0)
 #define CMD_PLAY	(1 << 1)
@@ -1147,15 +1153,6 @@ find_stream_by_channel (GstRTSPStream * stream, gint * channel)
   return -1;
 }
 
-static gint
-find_stream_by_pt (GstRTSPStream * stream, gint * pt)
-{
-  if (stream->pt == *pt)
-    return 0;
-
-  return -1;
-}
-
 static gint
 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
 {
@@ -1299,28 +1296,72 @@ gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
   }
 }
 
+
+/*   m=<media> <UDP port> RTP/AVP <payload>
+ */
 static void
 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
     const GstSDPMedia * media, GstRTSPStream * stream)
 {
-  const gchar *payload;
+  guint i, len;
+  const gchar *proto;
+
+  /* get proto */
+  proto = gst_sdp_media_get_proto (media);
+  if (proto == NULL)
+    goto no_proto;
+
+  if (g_str_equal (proto, "RTP/AVP"))
+    stream->profile = GST_RTSP_PROFILE_AVP;
+  else if (g_str_equal (proto, "RTP/SAVP"))
+    stream->profile = GST_RTSP_PROFILE_SAVP;
+  else if (g_str_equal (proto, "RTP/AVPF"))
+    stream->profile = GST_RTSP_PROFILE_AVPF;
+  else if (g_str_equal (proto, "RTP/SAVPF"))
+    stream->profile = GST_RTSP_PROFILE_SAVPF;
+  else
+    goto unknown_proto;
+
+  len = gst_sdp_media_formats_len (media);
+  for (i = 0; i < len; i++) {
+    gint pt;
+    GstCaps *caps;
+    GstStructure *s;
+    const gchar *enc;
+    PtMapItem item;
+
+    pt = atoi (gst_sdp_media_get_format (media, i));
+
+    GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
 
-  /* we must have a payload. No payload means we cannot create caps */
-  /* FIXME, handle multiple formats. The problem here is that we just want to
-   * take the first available format that we can handle but in order to do that
-   * we need to scan for depayloader plugins. Scanning for payloader plugins is
-   * also suboptimal because the user maybe just wants to save the raw stream
-   * and then we don't care. */
-  if ((payload = gst_sdp_media_get_format (media, 0))) {
-    stream->pt = atoi (payload);
     /* convert caps */
-    stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
+    caps = gst_rtspsrc_media_to_caps (pt, media);
+    if (caps == NULL) {
+      GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
+      continue;
+    }
 
+    /* do some tweaks */
+    s = gst_caps_get_structure (caps, 0);
+    if ((enc = gst_structure_get_string (s, "encoding-name"))) {
+      stream->is_real = (strstr (enc, "-REAL") != NULL);
+      if (strcmp (enc, "X-ASF-PF") == 0)
+        stream->container = TRUE;
+    }
     GST_DEBUG ("mapping sdp session level attributes to caps");
-    gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
+    gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
     GST_DEBUG ("mapping sdp media level attributes to caps");
-    gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
+    gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
 
+    /* the first pt will be the default */
+    if (stream->ptmap->len == 0)
+      stream->default_pt = pt;
+
+    item.pt = pt;
+    item.caps = caps;
+    g_array_append_val (stream->ptmap, item);
+
+#if 0
     if (stream->pt >= 96) {
       /* If we have a dynamic payload type, see if we have a stream with the
        * same payload number. If there is one, they are part of the same
@@ -1331,6 +1372,19 @@ gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
             stream->pt);
       }
     }
+#endif
+  }
+  return;
+
+no_proto:
+  {
+    GST_ERROR_OBJECT (src, "can't find proto in media");
+    return;
+  }
+unknown_proto:
+  {
+    GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
+    return;
   }
 }
 
@@ -1375,6 +1429,8 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
   stream->discont = TRUE;
   stream->seqbase = -1;
   stream->timebase = -1;
+  stream->profile = GST_RTSP_PROFILE_AVP;
+  stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
 
   /* collect bandwidth information for this steam. FIXME, configure in the RTP
    * session manager to scale RTCP. */
@@ -1383,6 +1439,7 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
   /* collect connection info */
   gst_rtspsrc_collect_connections (src, sdp, media, stream);
 
+  /* make the payload type map */
   gst_rtspsrc_collect_payloads (src, sdp, media, stream);
 
   /* collect port number */
@@ -1396,10 +1453,8 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
     control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
 
   GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
-  GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
   GST_DEBUG_OBJECT (src, " port: %d", stream->port);
   GST_DEBUG_OBJECT (src, " container: %d", stream->container);
-  GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
   GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
 
   if (control_url != NULL) {
@@ -1447,8 +1502,7 @@ gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
 
   GST_DEBUG_OBJECT (src, "free stream %p", stream);
 
-  if (stream->caps)
-    gst_caps_unref (stream->caps);
+  g_array_free (stream->ptmap, TRUE);
 
   g_free (stream->destination);
   g_free (stream->control_url);
@@ -1678,10 +1732,31 @@ gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
   }
 }
 
+static const gchar *
+rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
+    gint pt)
+{
+  guint i;
+
+  for (i = 0;; i++) {
+    const gchar *attr;
+    gint val;
+
+    if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
+      break;
+
+    if (sscanf (attr, "%d ", &val) != 1)
+      continue;
+
+    if (val == pt)
+      return attr;
+  }
+  return NULL;
+}
+
 /*
  *  Mapping of caps to and from SDP fields:
  *
- *   m=<media> <UDP port> RTP/AVP <payload>
  *   a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
  *   a=fmtp:<payload> <param>[=<value>];...
  */
@@ -1700,29 +1775,17 @@ gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
   gboolean ret;
 
   /* get and parse rtpmap */
-  if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
-    ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
-    if (ret) {
-      if (payload != pt) {
-        /* we ignore the rtpmap if the payload type is different. */
-        g_warning ("rtpmap of wrong payload type, ignoring");
-        name = NULL;
-        rate = -1;
-        params = NULL;
-      }
-    } else {
-      /* if we failed to parse the rtpmap for a dynamic payload type, we have an
-       * error */
-      if (pt >= 96)
-        goto no_rtpmap;
-      /* else we can ignore */
-      g_warning ("error parsing rtpmap, ignoring");
-    }
-  } else {
-    /* dynamic payloads need rtpmap or we fail */
-    if (pt >= 96)
-      goto no_rtpmap;
+  rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
+
+  /* dynamic payloads need rtpmap or we fail */
+  if (rtpmap == NULL && pt >= 96)
+    goto no_rtpmap;
+
+  ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
+  if (!ret) {
+    g_warning ("error parsing rtpmap, ignoring");
   }
+
   /* check if we have a rate, if not, we need to look up the rate from the
    * default rates based on the payload types. */
   if (rate == -1) {
@@ -1770,7 +1833,7 @@ gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
   }
 
   /* parse optional fmtp: field */
-  if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
+  if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
     gchar *p;
     gint payload = 0;
 
@@ -2599,6 +2662,20 @@ unknown_stream:
   }
 }
 
+static GstCaps *
+stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
+{
+  guint i, len;
+
+  len = stream->ptmap->len;
+  for (i = 0; i < len; i++) {
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+    if (item->pt == pt)
+      return item->caps;
+  }
+  return NULL;
+}
+
 static GstCaps *
 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
 {
@@ -2612,8 +2689,7 @@ request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
   if (!stream)
     goto unknown_stream;
 
-  caps = stream->caps;
-  if (caps)
+  if ((caps = stream_get_caps_for_pt (stream, pt)))
     gst_caps_ref (caps);
   GST_RTSP_STATE_UNLOCK (src);
 
@@ -2767,9 +2843,6 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
     /* configure the manager */
     if (src->manager == NULL) {
       GObjectClass *klass;
-      GstStructure *s;
-      const gchar *encoding;
-      gboolean need_slave;
 
       if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
         /* fallback */
@@ -2818,15 +2891,11 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
        * with such input times, e.g. container ones, most notably ASF */
       /* TODO alternatives are having an event that indicates these shifts,
        * or having rtsp extensions provide suggestion on buffer mode */
-      need_slave = stream->container;
-      if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
-          (encoding = gst_structure_get_string (s, "encoding-name")))
-        need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
       /* valid duration implies not likely live pipeline,
        * so slaving in jitterbuffer does not make much sense
        * (and might mess things up due to bursts) */
       if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
-          src->segment.duration && !need_slave) {
+          src->segment.duration && !stream->container) {
         src->use_buffering = TRUE;
       } else {
         src->use_buffering = FALSE;
@@ -3423,15 +3492,13 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
   GstPad *outpad = NULL;
   GstPadTemplate *template;
   gchar *name;
-  GstStructure *s;
   const gchar *media_type;
+  guint i, len;
 
   src = stream->parent;
 
   GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
 
-  s = gst_caps_get_structure (stream->caps, 0);
-
   /* get the proper media type for this stream now */
   if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
     goto unknown_transport;
@@ -3440,7 +3507,18 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
 
   /* configure the final media type */
   GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
-  gst_structure_set_name (s, media_type);
+
+  len = stream->ptmap->len;
+  for (i = 0; i < len; i++) {
+    GstStructure *s;
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+
+    if (item->caps == NULL)
+      continue;
+
+    s = gst_caps_get_structure (item->caps, 0);
+    gst_structure_set_name (s, media_type);
+  }
 
   /* try to get and configure a manager, channelpad[0-1] will be configured with
    * the pads for the manager, or NULL when no manager is needed. */
@@ -3561,8 +3639,11 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
       /* if we don't have a session manager, set the caps now. If we have a
        * session, we will get a notification of the pad and the caps. */
       if (!src->manager) {
+        GstCaps *caps;
+
+        caps = stream_get_caps_for_pt (stream, stream->default_pt);
         GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
-        gst_pad_set_caps (stream->srcpad, stream->caps);
+        gst_pad_set_caps (stream->srcpad, caps);
       }
       /* add the pad */
       if (!stream->added) {
@@ -3604,10 +3685,17 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
 
   for (walk = src->streams; walk; walk = g_list_next (walk)) {
     GstRTSPStream *stream = (GstRTSPStream *) walk->data;
-    GstCaps *caps;
+    guint j, len;
 
-    if ((caps = stream->caps)) {
-      caps = gst_caps_make_writable (caps);
+    len = stream->ptmap->len;
+    for (j = 0; j < len; j++) {
+      GstCaps *caps;
+      PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
+
+      if (item->caps == NULL)
+        continue;
+
+      caps = gst_caps_make_writable (item->caps);
       /* update caps */
       if (stream->timebase != -1)
         gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
@@ -3621,9 +3709,10 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
       gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
       gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
 
-      stream->caps = caps;
+      item->caps = caps;
+      GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
+          item->pt, caps);
     }
-    GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
   }
   if (reset_manager && src->manager) {
     GST_DEBUG_OBJECT (src, "clear session");
@@ -5330,7 +5419,7 @@ static guint protocol_masks[] = {
 
 static GstRTSPResult
 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
-    GstRTSPLowerTrans protocols, gchar ** transports)
+    GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
 {
   GstRTSPResult res;
   GString *result;
@@ -5354,23 +5443,35 @@ gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
   add_udp_str = FALSE;
 
   /* the default RTSP transports */
-  result = g_string_new ("");
+  result = g_string_new ("RTP");
+
+  switch (profile) {
+    case GST_RTSP_PROFILE_AVP:
+      g_string_append (result, "/AVP");
+      break;
+    case GST_RTSP_PROFILE_SAVP:
+      g_string_append (result, "/SAVP");
+      break;
+    case GST_RTSP_PROFILE_AVPF:
+      g_string_append (result, "/AVPF");
+      break;
+    case GST_RTSP_PROFILE_SAVPF:
+      g_string_append (result, "/SAVPF");
+      break;
+    default:
+      break;
+  }
+
   if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
     GST_DEBUG_OBJECT (src, "adding UDP unicast");
-
-    g_string_append (result, "RTP/AVP");
     if (add_udp_str)
       g_string_append (result, "/UDP");
     g_string_append (result, ";unicast;client_port=%%u1-%%u2");
   } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
     GST_DEBUG_OBJECT (src, "adding UDP multicast");
-
     /* we don't have to allocate any UDP ports yet, if the selected transport
      * turns out to be multicast we can create them and join the multicast
      * group indicated in the transport reply */
-    if (result->len > 0)
-      g_string_append (result, ",");
-    g_string_append (result, "RTP/AVP");
     if (add_udp_str)
       g_string_append (result, "/UDP");
     g_string_append (result, ";multicast");
@@ -5385,9 +5486,7 @@ gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
   } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
     GST_DEBUG_OBJECT (src, "adding TCP");
 
-    if (result->len > 0)
-      g_string_append (result, ",");
-    g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
+    g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
   }
   *transports = g_string_free (result, FALSE);
 
@@ -5484,23 +5583,6 @@ failed:
   }
 }
 
-static gboolean
-gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
-{
-  gboolean res = FALSE;
-
-  if (stream->caps) {
-    GstStructure *s;
-    const gchar *enc = NULL;
-
-    s = gst_caps_get_structure (stream->caps, 0);
-    if ((enc = gst_structure_get_string (s, "encoding-name"))) {
-      res = (strstr (enc, "-REAL") != NULL);
-    }
-  }
-  return res;
-}
-
 /* Perform the SETUP request for all the streams.
  *
  * We ask the server for a specific transport, which initially includes all the
@@ -5558,11 +5640,17 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
     gint retry = 0;
     guint mask = 0;
     gboolean selected;
+    GstCaps *caps;
 
     stream = (GstRTSPStream *) walk->data;
+    caps = stream_get_caps_for_pt (stream, stream->default_pt);
+    if (caps == NULL) {
+      GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
+      continue;
+    }
 
     /* see if we need to configure this stream */
-    if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
+    if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
       GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
           stream);
       stream->disabled = TRUE;
@@ -5570,7 +5658,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
     }
 
     g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
-        stream->id, stream->caps, &selected);
+        stream->id, caps, &selected);
     if (!selected) {
       GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
       stream->disabled = TRUE;
@@ -5579,11 +5667,11 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
     stream->disabled = FALSE;
 
     /* merge/overwrite global caps */
-    if (stream->caps) {
+    if (caps) {
       guint j, num;
       GstStructure *s;
 
-      s = gst_caps_get_structure (stream->caps, 0);
+      s = gst_caps_get_structure (caps, 0);
 
       num = gst_structure_n_fields (src->props);
       for (j = 0; j < num; j++) {
@@ -5633,7 +5721,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
     /* create a string with first transport in line */
     transports = NULL;
     res = gst_rtspsrc_create_transports_string (src,
-        protocols & protocol_masks[mask], &transports);
+        protocols & protocol_masks[mask], stream->profile, &transports);
     if (res < 0 || transports == NULL)
       goto setup_transport_failed;
 
@@ -5705,7 +5793,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
          * but not without checking for lost cause/extension so we can
          * post a nicer/more useful error message later */
         if (!unsupported_real)
-          unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
+          unsupported_real = stream->is_real;
         /* select next available protocol, give up on this stream if none */
         mask++;
         while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
@@ -6619,13 +6707,21 @@ gen_range_header (GstRTSPSrc * src, GstSegment * segment)
 static void
 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
 {
+  guint i, len;
+
   stream->timebase = -1;
   stream->seqbase = -1;
-  if (stream->caps) {
+
+  len = stream->ptmap->len;
+  for (i = 0; i < len; i++) {
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
     GstStructure *s;
 
-    stream->caps = gst_caps_make_writable (stream->caps);
-    s = gst_caps_get_structure (stream->caps, 0);
+    if (item->caps == NULL)
+      continue;
+
+    item->caps = gst_caps_make_writable (item->caps);
+    s = gst_caps_get_structure (item->caps, 0);
     gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
   }
 }
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 004d4c2296..1751b30d8c 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -108,7 +108,6 @@ struct _GstRTSPStream {
 
   /* for interleaved mode */
   guint8        channel[2];
-  GstCaps      *caps;
   GstPad       *channelpad[2];
 
   /* our udp sources */
@@ -125,9 +124,12 @@ struct _GstRTSPStream {
   GstElement   *fakesrc;
 
   /* state */
-  gint          pt;
   guint         port;
   gboolean      container;
+  gboolean      is_real;
+  guint8        default_pt;
+  GstRTSPProfile profile;
+  GArray       *ptmap;
   /* original control url */
   gchar        *control_url;
   guint32       ssrc;
@@ -149,6 +151,7 @@ struct _GstRTSPStream {
   gchar        *destination;
   gboolean      is_multicast;
   guint         ttl;
+
 };
 
 /**