webrtc: Initial support for stream addition/removal

Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.

.gitignore

@@ -75,6 +75,7 @@ gst*orc.h
 /tests/examples/webrtc/webrtc
 /tests/examples/webrtc/webrtcbidirectional
 /tests/examples/webrtc/webrtcswap
+/tests/examples/webrtc/webrtcrenego
 /tests/examples/webrtc/webrtctransceiver
 
 Build

ext/webrtc/gstwebrtcbin.h

@@ -131,6 +131,8 @@ struct _GstWebRTCBinPrivate
   /* count of the number of media streams we've offered for uniqueness */
   /* FIXME: overflow? */
   guint media_counter;
+  /* the number of times create_offer has been called for the version field */
+  guint offer_count;
 
   GstStructure *stats;
 };

ext/webrtc/transportstream.c

@@ -75,6 +75,36 @@ transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name)
   return ret;
 }
 
+int *
+transport_stream_get_all_pt (TransportStream * stream,
+    const gchar * encoding_name, gsize * pt_len)
+{
+  guint i;
+  gsize ret_i = 0;
+  gsize ret_size = 8;
+  int *ret = NULL;
+
+  for (i = 0; i < stream->ptmap->len; i++) {
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+    if (!gst_caps_is_empty (item->caps)) {
+      GstStructure *s = gst_caps_get_structure (item->caps, 0);
+      if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
+              encoding_name)) {
+        if (!ret)
+          ret = g_new0 (int, ret_size);
+        if (ret_i >= ret_size) {
+          ret_size *= 2;
+          ret = g_realloc_n (ret, ret_size, sizeof (int));
+        }
+        ret[ret_i++] = item->pt;
+      }
+    }
+  }
+
+  *pt_len = ret_i;
+  return ret;
+}
+
 static void
 transport_stream_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
@@ -152,6 +182,14 @@ transport_stream_dispose (GObject * object)
     gst_object_unref (stream->rtcp_transport);
   stream->rtcp_transport = NULL;
 
+  if (stream->rtxsend)
+    gst_object_unref (stream->rtxsend);
+  stream->rtxsend = NULL;
+
+  if (stream->rtxreceive)
+    gst_object_unref (stream->rtxreceive);
+  stream->rtxreceive = NULL;
+
   GST_OBJECT_PARENT (object) = NULL;
 
   G_OBJECT_CLASS (parent_class)->dispose (object);

ext/webrtc/transportstream.h

@@ -61,6 +61,10 @@ struct _TransportStream
 
   GArray                   *ptmap;                  /* array of PtMapItem's */
   GArray                   *remote_ssrcmap;         /* array of SsrcMapItem's */
+  gboolean                  output_connected;       /* whether receive bin is connected to rtpbin */
+
+  GstElement               *rtxsend;
+  GstElement               *rtxreceive;
 };
 
 struct _TransportStreamClass
@@ -72,6 +76,9 @@ TransportStream *       transport_stream_new        (GstWebRTCBin * webrtc,
                                                      guint session_id);
 int                     transport_stream_get_pt     (TransportStream * stream,
                                                      const gchar * encoding_name);
+int *                   transport_stream_get_all_pt (TransportStream * stream,
+                                                     const gchar * encoding_name,
+                                                     gsize * pt_len);
 GstCaps *               transport_stream_get_caps_for_pt    (TransportStream * stream,
                                                              guint pt);
 

ext/webrtc/utils.c

@@ -21,6 +21,8 @@
 # include "config.h"
 #endif
 
+#include <stdlib.h>
+
 #include "utils.h"
 #include "gstwebrtcbin.h"
 
@@ -53,7 +55,22 @@ _find_pad_template (GstElement * element, GstPadDirection direction,
 }
 
 GstSDPMessage *
-_get_latest_sdp (GstWebRTCBin * webrtc)
+_get_latest_offer (GstWebRTCBin * webrtc)
+{
+  if (webrtc->current_local_description &&
+      webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
+    return webrtc->current_local_description->sdp;
+  }
+  if (webrtc->current_remote_description &&
+      webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
+    return webrtc->current_remote_description->sdp;
+  }
+
+  return NULL;
+}
+
+GstSDPMessage *
+_get_latest_answer (GstWebRTCBin * webrtc)
 {
   if (webrtc->current_local_description &&
       webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
@@ -63,14 +80,19 @@ _get_latest_sdp (GstWebRTCBin * webrtc)
       webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
     return webrtc->current_remote_description->sdp;
   }
-  if (webrtc->current_local_description &&
+
-      webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
+  return NULL;
-    return webrtc->current_local_description->sdp;
+}
-  }
+
-  if (webrtc->current_remote_description &&
+GstSDPMessage *
-      webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
+_get_latest_sdp (GstWebRTCBin * webrtc)
-    return webrtc->current_remote_description->sdp;
+{
-  }
+  GstSDPMessage *ret = NULL;
+
+  if ((ret = _get_latest_answer (webrtc)))
+    return ret;
+  if ((ret = _get_latest_offer (webrtc)))
+    return ret;
 
   return NULL;
 }
@@ -142,3 +164,31 @@ _g_checksum_to_webrtc_string (GChecksumType type)
       return NULL;
   }
 }
+
+GstCaps *
+_rtp_caps_from_media (const GstSDPMedia * media)
+{
+  GstCaps *ret;
+  int i, j;
+
+  ret = gst_caps_new_empty ();
+  for (i = 0; i < gst_sdp_media_formats_len (media); i++) {
+    guint pt = atoi (gst_sdp_media_get_format (media, i));
+    GstCaps *caps;
+
+    caps = gst_sdp_media_get_caps_from_media (media, pt);
+
+    /* gst_sdp_media_get_caps_from_media() produces caps with name
+     * "application/x-unknown" which will fail intersection with
+     * "application/x-rtp" caps so mangle the returns caps to have the
+     * correct name here */
+    for (j = 0; j < gst_caps_get_size (caps); j++) {
+      GstStructure *s = gst_caps_get_structure (caps, j);
+      gst_structure_set_name (s, "application/x-rtp");
+    }
+
+    gst_caps_append (ret, caps);
+  }
+
+  return ret;
+}

ext/webrtc/utils.h

@@ -47,6 +47,8 @@ GstPadTemplate *        _find_pad_template          (GstElement * element,
                                                      const gchar * name);
 
 GstSDPMessage *         _get_latest_sdp             (GstWebRTCBin * webrtc);
+GstSDPMessage *         _get_latest_offer           (GstWebRTCBin * webrtc);
+GstSDPMessage *         _get_latest_answer          (GstWebRTCBin * webrtc);
 
 GstWebRTCICEStream *    _find_ice_stream_for_session            (GstWebRTCBin * webrtc,
                                                                  guint session_id);
@@ -74,6 +76,8 @@ G_GNUC_INTERNAL
 gchar *                 _enum_value_to_string       (GType type, guint value);
 G_GNUC_INTERNAL
 const gchar *           _g_checksum_to_webrtc_string (GChecksumType type);
+G_GNUC_INTERNAL
+GstCaps *               _rtp_caps_from_media        (const GstSDPMedia * media);
 
 G_END_DECLS
 

ext/webrtc/webrtcsdp.c

@@ -212,7 +212,7 @@ _media_has_mid (const GstSDPMedia * media, guint media_idx, GError ** error)
   return TRUE;
 }
 
-static const gchar *
+const gchar *
 _media_get_ice_ufrag (const GstSDPMessage * msg, guint media_idx)
 {
   const gchar *ice_ufrag;
@@ -227,7 +227,7 @@ _media_get_ice_ufrag (const GstSDPMessage * msg, guint media_idx)
   return ice_ufrag;
 }
 
-static const gchar *
+const gchar *
 _media_get_ice_pwd (const GstSDPMessage * msg, guint media_idx)
 {
   const gchar *ice_pwd;
@@ -437,11 +437,13 @@ _media_replace_direction (GstSDPMedia * media,
 
     if (g_strcmp0 (attr->key, "sendonly") == 0
         || g_strcmp0 (attr->key, "sendrecv") == 0
-        || g_strcmp0 (attr->key, "recvonly") == 0) {
+        || g_strcmp0 (attr->key, "recvonly") == 0
+        || g_strcmp0 (attr->key, "inactive") == 0) {
       GstSDPAttribute new_attr = { 0, };
       GST_TRACE ("replace %s with %s", attr->key, dir_str);
       gst_sdp_attribute_set (&new_attr, dir_str, "");
       gst_sdp_media_replace_attribute (media, i, &new_attr);
+      g_free (dir_str);
       return;
     }
   }
@@ -768,6 +770,21 @@ _message_media_is_datachannel (const GstSDPMessage * msg, guint media_id)
   return TRUE;
 }
 
+guint
+_message_get_datachannel_index (const GstSDPMessage * msg)
+{
+  guint i;
+
+  for (i = 0; i < gst_sdp_message_medias_len (msg); i++) {
+    if (_message_media_is_datachannel (msg, i)) {
+      g_assert (i < G_MAXUINT);
+      return i;
+    }
+  }
+
+  return G_MAXUINT;
+}
+
 void
 _get_ice_credentials_from_sdp_media (const GstSDPMessage * sdp, guint media_idx,
     gchar ** ufrag, gchar ** pwd)
@@ -833,6 +850,8 @@ _parse_bundle (GstSDPMessage * sdp, GStrv * bundled)
     if (!(*bundled)[0]) {
       GST_ERROR ("Invalid format for BUNDLE group, expected at least "
           "one mid (%s)", group);
+      g_strfreev (*bundled);
+      *bundled = NULL;
       goto done;
     }
   } else {

ext/webrtc/webrtcsdp.h

@@ -89,6 +89,8 @@ void                                _get_ice_credentials_from_sdp_media     (con
 G_GNUC_INTERNAL
 gboolean                            _message_media_is_datachannel           (const GstSDPMessage * msg,
                                                                              guint media_id);
+G_GNUC_INTERNAL
+guint                               _message_get_datachannel_index          (const GstSDPMessage * msg);
 
 G_GNUC_INTERNAL
 gboolean                            _get_bundle_index                       (GstSDPMessage * sdp,
@@ -98,4 +100,11 @@ G_GNUC_INTERNAL
 gboolean                            _parse_bundle                           (GstSDPMessage * sdp,
                                                                              GStrv * bundled);
 
+G_GNUC_INTERNAL
+const gchar *                       _media_get_ice_pwd                  (const GstSDPMessage * msg,
+                                                                             guint media_idx);
+G_GNUC_INTERNAL
+const gchar *                       _media_get_ice_ufrag                (const GstSDPMessage * msg,
+                                                                             guint media_idx);
+
 #endif /* __WEBRTC_UTILS_H__ */

ext/webrtc/webrtctransceiver.c

@@ -25,9 +25,14 @@
 #include "utils.h"
 #include "webrtctransceiver.h"
 
+#define GST_CAT_DEFAULT webrtc_transceiver_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
 #define webrtc_transceiver_parent_class parent_class
-G_DEFINE_TYPE (WebRTCTransceiver, webrtc_transceiver,
+G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
-    GST_TYPE_WEBRTC_RTP_TRANSCEIVER);
+    GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
+    GST_DEBUG_CATEGORY_INIT (webrtc_transceiver_debug,
+        "webrtctransceiver", 0, "webrtctransceiver"););
 
 #define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
 #define DEFAULT_DO_NACK FALSE
@@ -172,6 +177,8 @@ webrtc_transceiver_finalize (GObject * object)
     gst_structure_free (trans->local_rtx_ssrc_map);
   trans->local_rtx_ssrc_map = NULL;
 
+  gst_caps_replace (&trans->last_configured_caps, NULL);
+
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 

ext/webrtc/webrtctransceiver.h

@@ -44,6 +44,8 @@ struct _WebRTCTransceiver
   GstWebRTCFECType         fec_type;
   guint                    fec_percentage;
   gboolean                 do_nack;
+
+  GstCaps                  *last_configured_caps;
 };
 
 struct _WebRTCTransceiverClass

gst-libs/gst/webrtc/rtptransceiver.c

@@ -39,7 +39,7 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCRTPTransceiver,
     gst_webrtc_rtp_transceiver, GST_TYPE_OBJECT,
     GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_transceiver_debug,
-        "webrtctransceiver", 0, "webrtctransceiver");
+        "webrtcrtptransceiver", 0, "webrtcrtptransceiver");
     );
 
 enum

tests/check/elements/webrtcbin.c

@@ -41,6 +41,8 @@
 #define TEST_GET_OFFEROR(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc1 : t->webrtc2)
 #define TEST_GET_ANSWERER(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc2 : t->webrtc1)
 
+#define TEST_SDP_IS_LOCAL(t, e, d) ((TEST_IS_OFFER_ELEMENT (t, e) ^ ((d)->type == GST_WEBRTC_SDP_TYPE_OFFER)) == 0)
+
 typedef enum
 {
   STATE_NEW,
@@ -626,12 +628,12 @@ _pad_added_fakesink (struct test_webrtc *t, GstElement * element,
 }
 
 static void
-_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
+on_negotiation_needed_hit (struct test_webrtc *t, GstElement * element,
-    GstWebRTCSessionDescription * desc, gpointer user_data)
+    gpointer user_data)
 {
-  guint expected = GPOINTER_TO_UINT (user_data);
+  guint *flag = (guint *) user_data;
 
-  fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), expected);
+  *flag = 1;
 }
 
 typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
@@ -645,6 +647,9 @@ struct validate_sdp
   struct validate_sdp *next;
 };
 
+#define VAL_SDP_INIT(name,func,data,next) \
+    struct validate_sdp name = { func, data, next }
+
 static GstWebRTCSessionDescription *
 _check_validate_sdp (struct test_webrtc *t, GstElement * element,
     GstPromise * promise, gpointer user_data)
@@ -698,13 +703,20 @@ test_validate_sdp (struct test_webrtc *t, struct validate_sdp *offer,
   fail_unless (t->state == STATE_ANSWER_CREATED);
 }
 
+static void
+_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  guint expected = GPOINTER_TO_UINT (user_data);
+
+  fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), expected);
+}
+
 GST_START_TEST (test_sdp_no_media)
 {
   struct test_webrtc *t = test_webrtc_new ();
-  struct validate_sdp offer =
+  VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (0), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (0), NULL };
+  VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (0), NULL);
-  struct validate_sdp answer =
-      { _count_num_sdp_media, GUINT_TO_POINTER (0), NULL };
 
   /* check that a no stream connection creates 0 media sections */
 
@@ -756,10 +768,8 @@ create_audio_test (void)
 GST_START_TEST (test_audio)
 {
   struct test_webrtc *t = create_audio_test ();
-  struct validate_sdp offer =
+  VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (1), NULL };
+  VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
-  struct validate_sdp answer =
-      { _count_num_sdp_media, GUINT_TO_POINTER (1), NULL };
 
   /* check that a single stream connection creates the associated number
    * of media sections */
@@ -786,10 +796,8 @@ create_audio_video_test (void)
 GST_START_TEST (test_audio_video)
 {
   struct test_webrtc *t = create_audio_video_test ();
-  struct validate_sdp offer =
+  VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
+  VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
-  struct validate_sdp answer =
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
 
   /* check that a dual stream connection creates the associated number
    * of media sections */
@@ -850,15 +858,14 @@ GST_START_TEST (test_media_direction)
   struct test_webrtc *t = create_audio_video_test ();
   const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
   const gchar *expected_answer[] = { "sendrecv", "recvonly" };
-  struct validate_sdp offer_direction =
-      { on_sdp_media_direction, expected_offer, NULL };
-  struct validate_sdp offer =
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), &offer_direction };
-  struct validate_sdp answer_direction =
-      { on_sdp_media_direction, expected_answer, NULL };
-  struct validate_sdp answer =
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), &answer_direction };
   GstHarness *h;
+  VAL_SDP_INIT (offer_direction, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2),
+      &offer_direction);
+  VAL_SDP_INIT (answer_direction, on_sdp_media_direction, expected_answer,
+      NULL);
+  VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (2),
+      &answer_direction);
 
   /* check the default media directions for transceivers */
 
@@ -905,9 +912,8 @@ on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
 GST_START_TEST (test_payload_types)
 {
   struct test_webrtc *t = create_audio_video_test ();
-  struct validate_sdp payloads = { on_sdp_media_payload_types, NULL, NULL };
+  VAL_SDP_INIT (payloads, on_sdp_media_payload_types, NULL, NULL);
-  struct validate_sdp offer =
+  VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads };
   GstWebRTCRTPTransceiver *trans;
   GArray *transceivers;
 
@@ -956,8 +962,8 @@ GST_START_TEST (test_media_setup)
   struct test_webrtc *t = create_audio_test ();
   const gchar *expected_offer[] = { "actpass" };
   const gchar *expected_answer[] = { "active" };
-  struct validate_sdp offer = { on_sdp_media_setup, expected_offer, NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_setup, expected_offer, NULL);
-  struct validate_sdp answer = { on_sdp_media_setup, expected_answer, NULL };
+  VAL_SDP_INIT (answer, on_sdp_media_setup, expected_answer, NULL);
 
   /* check the default dtls setup negotiation values */
   test_validate_sdp (t, &offer, &answer);
@@ -1333,9 +1339,8 @@ GST_START_TEST (test_add_recvonly_transceiver)
   GstWebRTCRTPTransceiver *trans;
   const gchar *expected_offer[] = { "recvonly" };
   const gchar *expected_answer[] = { "sendonly" };
-  struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
-  struct validate_sdp answer =
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
-      { on_sdp_media_direction, expected_answer, NULL };
   GstCaps *caps;
   GstHarness *h;
 
@@ -1372,9 +1377,8 @@ GST_START_TEST (test_recvonly_sendonly)
   GstWebRTCRTPTransceiver *trans;
   const gchar *expected_offer[] = { "recvonly", "sendonly" };
   const gchar *expected_answer[] = { "sendonly", "recvonly" };
-  struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
-  struct validate_sdp answer =
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
-      { on_sdp_media_direction, expected_answer, NULL };
   GstCaps *caps;
   GstHarness *h;
   GArray *transceivers;
@@ -1446,8 +1450,8 @@ GST_START_TEST (test_data_channel_create)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
   gchar *label;
 
   t->on_negotiation_needed = NULL;
@@ -1500,8 +1504,8 @@ GST_START_TEST (test_data_channel_remote_notify)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
 
   t->on_negotiation_needed = NULL;
   t->offer_data = &offer;
@@ -1575,8 +1579,8 @@ GST_START_TEST (test_data_channel_transfer_string)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
 
   t->on_negotiation_needed = NULL;
   t->offer_data = &offer;
@@ -1657,8 +1661,8 @@ GST_START_TEST (test_data_channel_transfer_data)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
 
   t->on_negotiation_needed = NULL;
   t->offer_data = &offer;
@@ -1715,8 +1719,8 @@ GST_START_TEST (test_data_channel_create_after_negotiate)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
 
   t->on_negotiation_needed = NULL;
   t->offer_data = &offer;
@@ -1776,8 +1780,8 @@ GST_START_TEST (test_data_channel_low_threshold)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
 
   t->on_negotiation_needed = NULL;
   t->offer_data = &offer;
@@ -1849,8 +1853,8 @@ GST_START_TEST (test_data_channel_max_message_size)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
 
   t->on_negotiation_needed = NULL;
   t->offer_data = &offer;
@@ -1904,8 +1908,8 @@ GST_START_TEST (test_data_channel_pre_negotiated)
 {
   struct test_webrtc *t = test_webrtc_new ();
   GObject *channel1 = NULL, *channel2 = NULL;
-  struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
-  struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+  VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
   GstStructure *s;
   gint n_ready = 0;
 
@@ -2028,17 +2032,16 @@ GST_START_TEST (test_bundle_audio_video_max_bundle_max_bundle)
   const gchar *offer_bundle_only[] = { "video1", NULL };
   const gchar *answer_bundle_only[] = { NULL };
 
-  struct validate_sdp count =
+  VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
+  VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
-  struct validate_sdp bundle_tag = { _check_bundle_tag, bundle, &count };
+  VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
-  struct validate_sdp offer_non_reject =
+      GUINT_TO_POINTER (1), &bundle_tag);
-      { _count_non_rejected_media, GUINT_TO_POINTER (1), &bundle_tag };
+  VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
-  struct validate_sdp answer_non_reject =
+      GUINT_TO_POINTER (2), &bundle_tag);
-      { _count_non_rejected_media, GUINT_TO_POINTER (2), &bundle_tag };
+  VAL_SDP_INIT (offer, _check_bundle_only_media, &offer_bundle_only,
-  struct validate_sdp offer =
+      &offer_non_reject);
-      { _check_bundle_only_media, &offer_bundle_only, &offer_non_reject };
+  VAL_SDP_INIT (answer, _check_bundle_only_media, &answer_bundle_only,
-  struct validate_sdp answer =
+      &answer_non_reject);
-      { _check_bundle_only_media, &answer_bundle_only, &answer_non_reject };
 
   /* We set a max-bundle policy on the offering webrtcbin,
    * this means that all the offered medias should be part
@@ -2068,13 +2071,12 @@ GST_START_TEST (test_bundle_audio_video_max_compat_max_bundle)
   const gchar *bundle[] = { "audio0", "video1", NULL };
   const gchar *bundle_only[] = { NULL };
 
-  struct validate_sdp count =
+  VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
+  VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
-  struct validate_sdp bundle_tag = { _check_bundle_tag, bundle, &count };
+  VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
-  struct validate_sdp count_non_reject =
+      GUINT_TO_POINTER (2), &bundle_tag);
-      { _count_non_rejected_media, GUINT_TO_POINTER (2), &bundle_tag };
+  VAL_SDP_INIT (bundle_sdp, _check_bundle_only_media, &bundle_only,
-  struct validate_sdp bundle_sdp =
+      &count_non_reject);
-      { _check_bundle_only_media, &bundle_only, &count_non_reject };
 
   /* We set a max-compat policy on the offering webrtcbin,
    * this means that all the offered medias should be part
@@ -2106,18 +2108,17 @@ GST_START_TEST (test_bundle_audio_video_max_bundle_none)
   const gchar *answer_bundle[] = { NULL };
   const gchar *answer_bundle_only[] = { NULL };
 
-  struct validate_sdp count =
+  VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
+  VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
-  struct validate_sdp count_non_reject =
+      GUINT_TO_POINTER (1), &count);
-      { _count_non_rejected_media, GUINT_TO_POINTER (1), &count };
+  VAL_SDP_INIT (offer_bundle_tag, _check_bundle_tag, offer_bundle,
-  struct validate_sdp offer_bundle_tag =
+      &count_non_reject);
-      { _check_bundle_tag, offer_bundle, &count_non_reject };
+  VAL_SDP_INIT (answer_bundle_tag, _check_bundle_tag, answer_bundle,
-  struct validate_sdp answer_bundle_tag =
+      &count_non_reject);
-      { _check_bundle_tag, answer_bundle, &count_non_reject };
+  VAL_SDP_INIT (offer, _check_bundle_only_media, &offer_bundle_only,
-  struct validate_sdp offer =
+      &offer_bundle_tag);
-      { _check_bundle_only_media, &offer_bundle_only, &offer_bundle_tag };
+  VAL_SDP_INIT (answer, _check_bundle_only_media, &answer_bundle_only,
-  struct validate_sdp answer =
+      &answer_bundle_tag);
-      { _check_bundle_only_media, &answer_bundle_only, &answer_bundle_tag };
 
   /* We set a max-bundle policy on the offering webrtcbin,
    * this means that all the offered medias should be part
@@ -2148,17 +2149,16 @@ GST_START_TEST (test_bundle_audio_video_data)
   const gchar *answer_bundle_only[] = { NULL };
   GObject *channel = NULL;
 
-  struct validate_sdp count =
+  VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (3), NULL);
-      { _count_num_sdp_media, GUINT_TO_POINTER (3), NULL };
+  VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
-  struct validate_sdp bundle_tag = { _check_bundle_tag, bundle, &count };
+  VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
-  struct validate_sdp offer_non_reject =
+      GUINT_TO_POINTER (1), &bundle_tag);
-      { _count_non_rejected_media, GUINT_TO_POINTER (1), &bundle_tag };
+  VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
-  struct validate_sdp answer_non_reject =
+      GUINT_TO_POINTER (3), &bundle_tag);
-      { _count_non_rejected_media, GUINT_TO_POINTER (3), &bundle_tag };
+  VAL_SDP_INIT (offer, _check_bundle_only_media, &offer_bundle_only,
-  struct validate_sdp offer =
+      &offer_non_reject);
-      { _check_bundle_only_media, &offer_bundle_only, &offer_non_reject };
+  VAL_SDP_INIT (answer, _check_bundle_only_media, &answer_bundle_only,
-  struct validate_sdp answer =
+      &answer_non_reject);
-      { _check_bundle_only_media, &answer_bundle_only, &answer_non_reject };
 
   /* We set a max-bundle policy on the offering webrtcbin,
    * this means that all the offered medias should be part
@@ -2196,19 +2196,23 @@ GST_START_TEST (test_duplicate_nego)
   struct test_webrtc *t = create_audio_video_test ();
   const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
   const gchar *expected_answer[] = { "sendrecv", "recvonly" };
-  struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
-  struct validate_sdp answer =
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
-      { on_sdp_media_direction, expected_answer, NULL };
   GstHarness *h;
+  guint negotiation_flag = 0;
 
   /* check that negotiating twice succeeds */
 
+  t->on_negotiation_needed = on_negotiation_needed_hit;
+  t->negotiation_data = &negotiation_flag;
+
   h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
   add_fake_audio_src_harness (h, 96);
   t->harnesses = g_list_prepend (t->harnesses, h);
 
-  t->on_negotiation_needed = NULL;
   test_validate_sdp (t, &offer, &answer);
+  fail_unless_equals_int (negotiation_flag, 1);
+
   test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
   test_validate_sdp (t, &offer, &answer);
 
@@ -2222,9 +2226,8 @@ GST_START_TEST (test_dual_audio)
   struct test_webrtc *t = create_audio_test ();
   const gchar *expected_offer[] = { "sendrecv", "sendrecv", };
   const gchar *expected_answer[] = { "sendrecv", "recvonly" };
-  struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
-  struct validate_sdp answer =
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
-      { on_sdp_media_direction, expected_answer, NULL };
   GstHarness *h;
   GstWebRTCRTPTransceiver *trans;
   GArray *transceivers;
@@ -2258,6 +2261,431 @@ GST_START_TEST (test_dual_audio)
   test_webrtc_free (t);
 }
 
+GST_END_TEST;
+
+static void
+sdp_increasing_session_version (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  GstWebRTCSessionDescription *previous;
+  const GstSDPOrigin *our_origin, *previous_origin;
+  const gchar *prop;
+  guint64 our_v, previous_v;
+
+  prop =
+      TEST_SDP_IS_LOCAL (t, element,
+      desc) ? "current-local-description" : "current-remote-description";
+  g_object_get (element, prop, &previous, NULL);
+
+  our_origin = gst_sdp_message_get_origin (desc->sdp);
+  previous_origin = gst_sdp_message_get_origin (previous->sdp);
+
+  our_v = g_ascii_strtoull (our_origin->sess_version, NULL, 10);
+  previous_v = g_ascii_strtoull (previous_origin->sess_version, NULL, 10);
+
+  ck_assert_int_lt (previous_v, our_v);
+
+  gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_equal_session_id (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  GstWebRTCSessionDescription *previous;
+  const GstSDPOrigin *our_origin, *previous_origin;
+  const gchar *prop;
+
+  prop =
+      TEST_SDP_IS_LOCAL (t, element,
+      desc) ? "current-local-description" : "current-remote-description";
+  g_object_get (element, prop, &previous, NULL);
+
+  our_origin = gst_sdp_message_get_origin (desc->sdp);
+  previous_origin = gst_sdp_message_get_origin (previous->sdp);
+
+  fail_unless_equals_string (previous_origin->sess_id, our_origin->sess_id);
+  gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_media_equal_attribute (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, GstWebRTCSessionDescription * previous,
+    const gchar * attr)
+{
+  guint i, n;
+
+  n = MIN (gst_sdp_message_medias_len (previous->sdp),
+      gst_sdp_message_medias_len (desc->sdp));
+
+  for (i = 0; i < n; i++) {
+    const GstSDPMedia *our_media, *other_media;
+    const gchar *our_mid, *other_mid;
+
+    our_media = gst_sdp_message_get_media (desc->sdp, i);
+    other_media = gst_sdp_message_get_media (previous->sdp, i);
+
+    our_mid = gst_sdp_media_get_attribute_val (our_media, attr);
+    other_mid = gst_sdp_media_get_attribute_val (other_media, attr);
+
+    fail_unless_equals_string (our_mid, other_mid);
+  }
+}
+
+static void
+sdp_media_equal_mid (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  GstWebRTCSessionDescription *previous;
+  const gchar *prop;
+
+  prop =
+      TEST_SDP_IS_LOCAL (t, element,
+      desc) ? "current-local-description" : "current-remote-description";
+  g_object_get (element, prop, &previous, NULL);
+
+  sdp_media_equal_attribute (t, element, desc, previous, "mid");
+
+  gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_media_equal_ice_params (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  GstWebRTCSessionDescription *previous;
+  const gchar *prop;
+
+  prop =
+      TEST_SDP_IS_LOCAL (t, element,
+      desc) ? "current-local-description" : "current-remote-description";
+  g_object_get (element, prop, &previous, NULL);
+
+  sdp_media_equal_attribute (t, element, desc, previous, "ice-ufrag");
+  sdp_media_equal_attribute (t, element, desc, previous, "ice-pwd");
+
+  gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_media_equal_fingerprint (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  GstWebRTCSessionDescription *previous;
+  const gchar *prop;
+
+  prop =
+      TEST_SDP_IS_LOCAL (t, element,
+      desc) ? "current-local-description" : "current-remote-description";
+  g_object_get (element, prop, &previous, NULL);
+
+  sdp_media_equal_attribute (t, element, desc, previous, "fingerprint");
+
+  gst_webrtc_session_description_free (previous);
+}
+
+GST_START_TEST (test_renego_add_stream)
+{
+  struct test_webrtc *t = create_audio_video_test ();
+  const gchar *expected_offer[] = { "sendrecv", "sendrecv", "sendrecv" };
+  const gchar *expected_answer[] = { "sendrecv", "recvonly", "recvonly" };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+  VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+  VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+      &renego_mid);
+  VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+  VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+      &renego_sess_id);
+  VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+      &renego_sess_ver);
+  GstHarness *h;
+
+  /* negotiate an AV stream and then renegotiate an extra stream */
+  h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+  add_fake_audio_src_harness (h, 96);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+  add_fake_audio_src_harness (h, 98);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  offer.next = &renego_fingerprint;
+  answer.next = &renego_fingerprint;
+
+  /* renegotiate! */
+  test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+  test_validate_sdp (t, &offer, &answer);
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_renego_stream_add_data_channel)
+{
+  struct test_webrtc *t = create_audio_video_test ();
+  const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
+  const gchar *expected_answer[] = { "sendrecv", "recvonly" };
+
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+  VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+  VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+      &renego_mid);
+  VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+  VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+      &renego_sess_id);
+  VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+      &renego_sess_ver);
+  GObject *channel;
+  GstHarness *h;
+
+  /* negotiate an AV stream and then renegotiate a data channel */
+  h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+  add_fake_audio_src_harness (h, 96);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
+      &channel);
+
+  offer.next = &renego_fingerprint;
+  answer.next = &renego_fingerprint;
+
+  /* renegotiate! */
+  test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+  test_validate_sdp (t, &offer, &answer);
+
+  g_object_unref (channel);
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_renego_data_channel_add_stream)
+{
+  struct test_webrtc *t = test_webrtc_new ();
+  const gchar *expected_offer[] = { NULL, "sendrecv" };
+  const gchar *expected_answer[] = { NULL, "recvonly" };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+  VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+  VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+      &renego_mid);
+  VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+  VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+      &renego_sess_id);
+  VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+      &renego_sess_ver);
+  GObject *channel;
+  GstHarness *h;
+
+  /* negotiate an AV stream and then renegotiate a data channel */
+  t->on_negotiation_needed = NULL;
+  t->on_ice_candidate = NULL;
+  t->on_pad_added = _pad_added_fakesink;
+
+  fail_if (gst_element_set_state (t->webrtc1,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+  fail_if (gst_element_set_state (t->webrtc2,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+
+  g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
+      &channel);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
+  add_fake_audio_src_harness (h, 97);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  offer.next = &renego_fingerprint;
+  answer.next = &renego_fingerprint;
+
+  /* renegotiate! */
+  test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+  test_validate_sdp (t, &offer, &answer);
+
+  g_object_unref (channel);
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bundle_renego_add_stream)
+{
+  struct test_webrtc *t = create_audio_video_test ();
+  const gchar *expected_offer[] = { "sendrecv", "sendrecv", "sendrecv" };
+  const gchar *expected_answer[] = { "sendrecv", "recvonly", "recvonly" };
+  const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
+  const gchar *offer_bundle_only[] = { "video1", "audio2", NULL };
+  const gchar *answer_bundle_only[] = { NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+
+  VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+  VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+      &renego_mid);
+  VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+  VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+      &renego_sess_id);
+  VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+      &renego_sess_ver);
+  VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &renego_fingerprint);
+  VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
+      GUINT_TO_POINTER (1), &bundle_tag);
+  VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
+      GUINT_TO_POINTER (3), &bundle_tag);
+  VAL_SDP_INIT (offer_bundle_only_sdp, _check_bundle_only_media,
+      &offer_bundle_only, &offer_non_reject);
+  VAL_SDP_INIT (answer_bundle_only_sdp, _check_bundle_only_media,
+      &answer_bundle_only, &answer_non_reject);
+  GstHarness *h;
+
+  /* We set a max-bundle policy on the offering webrtcbin,
+   * this means that all the offered medias should be part
+   * of the group:BUNDLE attribute, and they should be marked
+   * as bundle-only
+   */
+  gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+      "max-bundle");
+  /* We also set a max-bundle policy on the answering webrtcbin,
+   * this means that all the offered medias should be part
+   * of the group:BUNDLE attribute, but need not be marked
+   * as bundle-only.
+   */
+  gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
+      "max-bundle");
+
+  /* negotiate an AV stream and then renegotiate an extra stream */
+  h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+  add_fake_audio_src_harness (h, 96);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+  add_fake_audio_src_harness (h, 98);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  offer.next = &offer_bundle_only_sdp;
+  answer.next = &answer_bundle_only_sdp;
+
+  /* renegotiate! */
+  test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+  test_validate_sdp (t, &offer, &answer);
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bundle_max_compat_max_bundle_renego_add_stream)
+{
+  struct test_webrtc *t = create_audio_video_test ();
+  const gchar *expected_offer[] = { "sendrecv", "sendrecv", "sendrecv" };
+  const gchar *expected_answer[] = { "sendrecv", "recvonly", "recvonly" };
+  const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
+  const gchar *bundle_only[] = { NULL };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+
+  VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+  VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+      &renego_mid);
+  VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+  VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+      &renego_sess_id);
+  VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+      &renego_sess_ver);
+  VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &renego_fingerprint);
+  VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
+      GUINT_TO_POINTER (3), &bundle_tag);
+  VAL_SDP_INIT (bundle_sdp, _check_bundle_only_media, &bundle_only,
+      &count_non_reject);
+  GstHarness *h;
+
+  /* We set a max-compat policy on the offering webrtcbin,
+   * this means that all the offered medias should be part
+   * of the group:BUNDLE attribute, and they should *not* be marked
+   * as bundle-only
+   */
+  gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+      "max-compat");
+  /* We set a max-bundle policy on the answering webrtcbin,
+   * this means that all the offered medias should be part
+   * of the group:BUNDLE attribute, but need not be marked
+   * as bundle-only.
+   */
+  gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
+      "max-bundle");
+
+  /* negotiate an AV stream and then renegotiate an extra stream */
+  h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+  add_fake_audio_src_harness (h, 96);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+  add_fake_audio_src_harness (h, 98);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  offer.next = &bundle_sdp;
+  answer.next = &bundle_sdp;
+
+  /* renegotiate! */
+  test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+  test_validate_sdp (t, &offer, &answer);
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_renego_transceiver_set_direction)
+{
+  struct test_webrtc *t = create_audio_test ();
+  const gchar *expected_offer[] = { "sendrecv" };
+  const gchar *expected_answer[] = { "sendrecv" };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+  GstWebRTCRTPTransceiver *transceiver;
+  GstHarness *h;
+  GstPad *pad;
+
+  /* negotiate an AV stream and then change the transceiver direction */
+  h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+  add_fake_audio_src_harness (h, 96);
+  t->harnesses = g_list_prepend (t->harnesses, h);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  /* renegotiate an inactive transceiver! */
+  pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
+  g_object_get (pad, "transceiver", &transceiver, NULL);
+  fail_unless (transceiver != NULL);
+  gst_webrtc_rtp_transceiver_set_direction (transceiver,
+      GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE);
+  expected_offer[0] = "inactive";
+  expected_answer[0] = "inactive";
+
+  /* TODO: also validate EOS events from the inactive change */
+
+  test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+  test_validate_sdp (t, &offer, &answer);
+
+  gst_object_unref (pad);
+  gst_object_unref (transceiver);
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
 static Suite *
 webrtcbin_suite (void)
 {
@@ -2294,6 +2722,10 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_bundle_audio_video_max_compat_max_bundle);
     tcase_add_test (tc, test_dual_audio);
     tcase_add_test (tc, test_duplicate_nego);
+    tcase_add_test (tc, test_renego_add_stream);
+    tcase_add_test (tc, test_bundle_renego_add_stream);
+    tcase_add_test (tc, test_bundle_max_compat_max_bundle_renego_add_stream);
+    tcase_add_test (tc, test_renego_transceiver_set_direction);
     if (sctpenc && sctpdec) {
       tcase_add_test (tc, test_data_channel_create);
       tcase_add_test (tc, test_data_channel_remote_notify);
@@ -2304,6 +2736,8 @@ webrtcbin_suite (void)
       tcase_add_test (tc, test_data_channel_max_message_size);
       tcase_add_test (tc, test_data_channel_pre_negotiated);
       tcase_add_test (tc, test_bundle_audio_video_data);
+      tcase_add_test (tc, test_renego_stream_add_data_channel);
+      tcase_add_test (tc, test_renego_data_channel_add_stream);
     } else {
       GST_WARNING ("Some required elements were not found. "
           "All datachannel tests are disabled. sctpenc %p, sctpdec %p", sctpenc,

tests/examples/webrtc/Makefile.am

@@ -1,5 +1,5 @@
 
-noinst_PROGRAMS = webrtc webrtcbidirectional webrtcswap webrtctransceiver
+noinst_PROGRAMS = webrtc webrtcbidirectional webrtcswap webrtctransceiver webrtcrenego
 
 webrtc_SOURCES = webrtc.c
 webrtc_CFLAGS=\
@@ -52,3 +52,16 @@ webrtctransceiver_LDADD=\
 	$(GST_LIBS) \
 	$(GST_SDP_LIBS) \
 	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la
+
+webrtcrenego_SOURCES = webrtcrenego.c
+webrtcrenego_CFLAGS=\
+	-I$(top_srcdir)/gst-libs \
+	-I$(top_builddir)/gst-libs \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_CFLAGS) \
+	$(GST_SDP_CFLAGS)
+webrtcrenego_LDADD=\
+	$(GST_PLUGINS_BASE_LIBS) \
+	$(GST_LIBS) \
+	$(GST_SDP_LIBS) \
+	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la

tests/examples/webrtc/meson.build

@@ -1,4 +1,4 @@
-examples = ['webrtc', 'webrtcbidirectional', 'webrtcswap', 'webrtctransceiver']
+examples = ['webrtc', 'webrtcbidirectional', 'webrtcswap', 'webrtctransceiver', 'webrtcrenego']
 
 foreach example : examples
   exe_name = example

tests/examples/webrtc/webrtcrenego.c

@@ -0,0 +1,289 @@
+#include <gst/gst.h>
+#include <gst/sdp/sdp.h>
+#include <gst/webrtc/webrtc.h>
+
+#include <string.h>
+
+static GMainLoop *loop;
+static GstElement *pipe1, *webrtc1, *webrtc2, *extra_src;
+static GstBus *bus1;
+
+#define SEND_SRC(pattern) "videotestsrc is-live=true pattern=" pattern " ! timeoverlay ! queue ! vp8enc ! rtpvp8pay ! queue ! " \
+    "capsfilter caps=application/x-rtp,media=video,payload=96,encoding-name=VP8"
+
+static void
+_element_message (GstElement * parent, GstMessage * msg)
+{
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_EOS:{
+      GstElement *receive, *webrtc;
+      GstPad *pad, *peer;
+
+      g_print ("Got element EOS message from %s parent %s\n",
+          GST_OBJECT_NAME (msg->src), GST_OBJECT_NAME (parent));
+
+      receive = GST_ELEMENT (msg->src);
+
+      pad = gst_element_get_static_pad (receive, "sink");
+      peer = gst_pad_get_peer (pad);
+
+      webrtc = GST_ELEMENT (gst_pad_get_parent (peer));
+      gst_bin_remove (GST_BIN (pipe1), receive);
+
+      gst_pad_unlink (peer, pad);
+      gst_element_release_request_pad (webrtc, peer);
+
+      gst_object_unref (pad);
+      gst_object_unref (peer);
+
+      gst_element_set_state (receive, GST_STATE_NULL);
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+static gboolean
+_bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe)
+{
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_STATE_CHANGED:
+      if (GST_ELEMENT (msg->src) == pipe) {
+        GstState old, new, pending;
+
+        gst_message_parse_state_changed (msg, &old, &new, &pending);
+
+        {
+          gchar *dump_name = g_strconcat ("state_changed-",
+              gst_element_state_get_name (old), "_",
+              gst_element_state_get_name (new), NULL);
+          GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (msg->src),
+              GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
+          g_free (dump_name);
+        }
+      }
+      break;
+    case GST_MESSAGE_ERROR:{
+      GError *err = NULL;
+      gchar *dbg_info = NULL;
+
+      GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
+          GST_DEBUG_GRAPH_SHOW_ALL, "error");
+
+      gst_message_parse_error (msg, &err, &dbg_info);
+      g_printerr ("ERROR from element %s: %s\n",
+          GST_OBJECT_NAME (msg->src), err->message);
+      g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
+      g_error_free (err);
+      g_free (dbg_info);
+      g_main_loop_quit (loop);
+      break;
+    }
+    case GST_MESSAGE_EOS:{
+      GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
+          GST_DEBUG_GRAPH_SHOW_ALL, "eos");
+      g_print ("EOS received\n");
+      g_main_loop_quit (loop);
+      break;
+    }
+    case GST_MESSAGE_ELEMENT:{
+      const GstStructure *s = gst_message_get_structure (msg);
+      if (g_strcmp0 (gst_structure_get_name (s), "GstBinForwarded") == 0) {
+        GstMessage *sub_msg;
+
+        gst_structure_get (s, "message", GST_TYPE_MESSAGE, &sub_msg, NULL);
+        _element_message (GST_ELEMENT (msg->src), sub_msg);
+        gst_message_unref (sub_msg);
+      }
+      break;
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+static void
+_webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe)
+{
+  GstElement *out;
+  GstPad *sink;
+
+  if (GST_PAD_DIRECTION (new_pad) != GST_PAD_SRC)
+    return;
+
+  out = gst_parse_bin_from_description ("queue ! rtpvp8depay ! vp8dec ! "
+      "videoconvert ! queue ! xvimagesink", TRUE, NULL);
+  gst_bin_add (GST_BIN (pipe), out);
+  gst_element_sync_state_with_parent (out);
+
+  sink = out->sinkpads->data;
+
+  gst_pad_link (new_pad, sink);
+}
+
+static void
+_on_answer_received (GstPromise * promise, gpointer user_data)
+{
+  GstWebRTCSessionDescription *answer = NULL;
+  const GstStructure *reply;
+  gchar *desc;
+
+  g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
+  reply = gst_promise_get_reply (promise);
+  gst_structure_get (reply, "answer",
+      GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
+  gst_promise_unref (promise);
+  desc = gst_sdp_message_as_text (answer->sdp);
+  g_print ("Created answer:\n%s\n", desc);
+  g_free (desc);
+
+  /* this is one way to tell webrtcbin that we don't want to be notified when
+   * this task is complete: set a NULL promise */
+  g_signal_emit_by_name (webrtc1, "set-remote-description", answer, NULL);
+  /* this is another way to tell webrtcbin that we don't want to be notified
+   * when this task is complete: interrupt the promise */
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (webrtc2, "set-local-description", answer, promise);
+  gst_promise_interrupt (promise);
+  gst_promise_unref (promise);
+
+  gst_webrtc_session_description_free (answer);
+}
+
+static void
+_on_offer_received (GstPromise * promise, gpointer user_data)
+{
+  GstWebRTCSessionDescription *offer = NULL;
+  const GstStructure *reply;
+  gchar *desc;
+
+  g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
+  reply = gst_promise_get_reply (promise);
+  gst_structure_get (reply, "offer",
+      GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
+  gst_promise_unref (promise);
+  desc = gst_sdp_message_as_text (offer->sdp);
+  g_print ("Created offer:\n%s\n", desc);
+  g_free (desc);
+
+  g_signal_emit_by_name (webrtc1, "set-local-description", offer, NULL);
+  g_signal_emit_by_name (webrtc2, "set-remote-description", offer, NULL);
+
+  promise = gst_promise_new_with_change_func (_on_answer_received, user_data,
+      NULL);
+  g_signal_emit_by_name (webrtc2, "create-answer", NULL, promise);
+
+  gst_webrtc_session_description_free (offer);
+}
+
+static void
+_on_negotiation_needed (GstElement * element, gpointer user_data)
+{
+  GstPromise *promise;
+
+  promise = gst_promise_new_with_change_func (_on_offer_received, user_data,
+      NULL);
+  g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
+}
+
+static void
+_on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate,
+    GstElement * other)
+{
+  g_signal_emit_by_name (other, "add-ice-candidate", mlineindex, candidate);
+}
+
+static gboolean
+stream_change (gpointer data)
+{
+  if (!extra_src) {
+    g_print ("Adding extra stream\n");
+    extra_src =
+        gst_parse_bin_from_description (SEND_SRC ("circular"), TRUE, NULL);
+
+    gst_element_set_locked_state (extra_src, TRUE);
+    gst_bin_add (GST_BIN (pipe1), extra_src);
+    gst_element_link (extra_src, webrtc1);
+    gst_element_set_locked_state (extra_src, FALSE);
+    gst_element_sync_state_with_parent (extra_src);
+    GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe1),
+        GST_DEBUG_GRAPH_SHOW_ALL, "add");
+  } else {
+    GstPad *pad, *peer;
+    GstWebRTCRTPTransceiver *transceiver;
+
+    g_print ("Removing extra stream\n");
+    pad = gst_element_get_static_pad (extra_src, "src");
+    peer = gst_pad_get_peer (pad);
+    gst_element_send_event (extra_src, gst_event_new_eos ());
+
+    g_object_get (peer, "transceiver", &transceiver, NULL);
+    gst_webrtc_rtp_transceiver_set_direction (transceiver,
+        GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE);
+
+    gst_element_set_locked_state (extra_src, TRUE);
+    gst_element_set_state (extra_src, GST_STATE_NULL);
+    gst_pad_unlink (pad, peer);
+    gst_element_release_request_pad (webrtc1, peer);
+
+    gst_object_unref (peer);
+    gst_object_unref (pad);
+
+    gst_bin_remove (GST_BIN (pipe1), extra_src);
+    extra_src = NULL;
+    GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe1),
+        GST_DEBUG_GRAPH_SHOW_ALL, "remove");
+  }
+
+  return G_SOURCE_CONTINUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+  pipe1 = gst_parse_launch (SEND_SRC ("smpte")
+      " ! webrtcbin name=smpte bundle-policy=max-bundle " SEND_SRC ("ball")
+      " ! webrtcbin name=ball bundle-policy=max-bundle", NULL);
+  g_object_set (pipe1, "message-forward", TRUE, NULL);
+  bus1 = gst_pipeline_get_bus (GST_PIPELINE (pipe1));
+  gst_bus_add_watch (bus1, (GstBusFunc) _bus_watch, pipe1);
+
+  webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "smpte");
+  g_signal_connect (webrtc1, "on-negotiation-needed",
+      G_CALLBACK (_on_negotiation_needed), NULL);
+  g_signal_connect (webrtc1, "pad-added", G_CALLBACK (_webrtc_pad_added),
+      pipe1);
+  webrtc2 = gst_bin_get_by_name (GST_BIN (pipe1), "ball");
+  g_signal_connect (webrtc2, "pad-added", G_CALLBACK (_webrtc_pad_added),
+      pipe1);
+  g_signal_connect (webrtc1, "on-ice-candidate",
+      G_CALLBACK (_on_ice_candidate), webrtc2);
+  g_signal_connect (webrtc2, "on-ice-candidate",
+      G_CALLBACK (_on_ice_candidate), webrtc1);
+
+  g_print ("Starting pipeline\n");
+  gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
+
+  g_timeout_add_seconds (5, stream_change, NULL);
+
+  g_main_loop_run (loop);
+
+  gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
+  g_print ("Pipeline stopped\n");
+
+  gst_object_unref (webrtc1);
+  gst_object_unref (webrtc2);
+  gst_bus_remove_watch (bus1);
+  gst_object_unref (bus1);
+  gst_object_unref (pipe1);
+
+  gst_deinit ();
+
+  return 0;
+}