sdp: add rollover counters for all sender SSRC

We add different crypto sessions in MIKEY, one for each sender
SSRC. Currently, all of them will have the same security policy, 0.

The rollover counters are obtained from the srtpenc element using the
"stats" property.

https://bugzilla.gnome.org/show_bug.cgi?id=730539

gst/rtsp-server/rtsp-sdp.c

@@ -16,6 +16,9 @@
  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */
+
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 /**
  * SECTION:rtsp-sdp
  * @short_description: Make SDP messages
@@ -73,7 +76,109 @@ update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
   gst_object_unref (src_pad);
 }
 
-static void
+static guint
+get_roc_from_stats (GstStructure * stats, guint ssrc)
+{
+  const GValue *va, *v;
+  guint i, len;
+  /* initialize roc to something different than 0, so if we don't get
+     the proper ROC from the encoder, streaming should fail initially. */
+  guint roc = -1;
+
+  va = gst_structure_get_value (stats, "streams");
+  if (!va || !G_VALUE_HOLDS (va, GST_TYPE_ARRAY)) {
+    GST_WARNING ("stats doesn't have a valid 'streams' field");
+    return 0;
+  }
+
+  len = gst_value_array_get_size (va);
+
+  /* look if there's any SSRC that matches. */
+  for (i = 0; i < len; i++) {
+    GstStructure *stream;
+    v = gst_value_array_get_value (va, i);
+    if (v && (stream = g_value_get_boxed (v))) {
+      guint stream_ssrc;
+      gst_structure_get_uint (stream, "ssrc", &stream_ssrc);
+      if (stream_ssrc == ssrc) {
+        gst_structure_get_uint (stream, "roc", &roc);
+        break;
+      }
+    }
+  }
+
+  return roc;
+}
+
+static gboolean
+mikey_add_crypto_sessions (GstRTSPStream * stream, GstMIKEYMessage * msg)
+{
+  guint i;
+  GObject *session;
+  GstElement *encoder;
+  GValueArray *sources;
+  gboolean roc_found;
+
+  encoder = gst_rtsp_stream_get_srtp_encoder (stream);
+  if (encoder == NULL) {
+    GST_ERROR ("unable to get SRTP encoder from stream %p", stream);
+    return FALSE;
+  }
+
+  session = gst_rtsp_stream_get_rtpsession (stream);
+  if (session == NULL) {
+    GST_ERROR ("unable to get RTP session from stream %p", stream);
+    return FALSE;
+  }
+
+  roc_found = FALSE;
+  g_object_get (session, "sources", &sources, NULL);
+  for (i = 0; sources && (i < sources->n_values); i++) {
+    GValue *val;
+    GObject *source;
+    guint32 ssrc;
+    gboolean is_sender;
+
+    val = g_value_array_get_nth (sources, i);
+    source = (GObject *) g_value_get_object (val);
+
+    g_object_get (source, "ssrc", &ssrc, "is-sender", &is_sender, NULL);
+
+    if (is_sender) {
+      guint32 roc = -1;
+      GstStructure *stats;
+
+      g_object_get (encoder, "stats", &stats, NULL);
+
+      if (stats) {
+        roc = get_roc_from_stats (stats, ssrc);
+        gst_structure_free (stats);
+      }
+
+      roc_found = !!(roc != -1);
+      if (!roc_found) {
+        GST_ERROR ("unable to obtain ROC for stream %p with SSRC %u",
+            stream, ssrc);
+        goto cleanup;
+      }
+
+      GST_INFO ("stream %p with SSRC %u has a ROC of %u", stream, ssrc, roc);
+
+      gst_mikey_message_add_cs_srtp (msg, 0, ssrc, roc);
+    }
+  }
+
+cleanup:
+  {
+    g_value_array_free (sources);
+
+    gst_object_unref (encoder);
+    g_object_unref (session);
+    return roc_found;
+  }
+}
+
+static gboolean
 make_media (GstSDPMessage * sdp, GstSDPInfo * info,
     GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile)
 {
@@ -86,13 +191,12 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info,
   guint ttl;
   GstClockTime rtx_time;
   gchar *base64;
-  guint32 ssrc;
   GstMIKEYMessage *mikey_msg;
 
   gst_sdp_media_new (&smedia);
 
   if (gst_sdp_media_set_media_from_caps (caps, smedia) != GST_SDP_OK) {
-    goto error;
+    goto caps_error;
   }
 
   gst_sdp_media_set_port_info (smedia, 0, 1);
@@ -155,9 +259,9 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info,
   /* check for srtp */
   mikey_msg = gst_mikey_message_new_from_caps (caps);
   if (mikey_msg) {
-    gst_rtsp_stream_get_ssrc (stream, &ssrc);
-    /* add policy '0' for our SSRC */
-    gst_mikey_message_add_cs_srtp (mikey_msg, 0, ssrc, 0);
+    /* add policy '0' for all sending SSRC */
+    if (!mikey_add_crypto_sessions (stream, mikey_msg))
+      goto crypto_sessions_error;
 
     base64 = gst_mikey_message_base64_encode (mikey_msg);
     if (base64) {
@@ -281,7 +385,7 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info,
 
       s = gst_caps_get_structure (caps, 0);
       if (s == NULL)
-        goto error;
+        goto no_caps_info;
 
       /* get payload type and clock rate */
       gst_structure_get_int (s, "payload", &caps_pt);
@@ -306,21 +410,36 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info,
   gst_sdp_message_add_media (sdp, smedia);
   gst_sdp_media_free (smedia);
 
-  return;
+  return TRUE;
 
   /* ERRORS */
+caps_error:
+  {
+    gst_sdp_media_free (smedia);
+    GST_ERROR ("unable to set media from caps for stream %d",
+        gst_rtsp_stream_get_index (stream));
+    return FALSE;
+  }
 no_multicast:
   {
     gst_sdp_media_free (smedia);
-    g_warning ("ignoring stream %d without multicast address",
+    GST_ERROR ("stream %d has no multicast address",
         gst_rtsp_stream_get_index (stream));
-    return;
+    return FALSE;
   }
-error:
+no_caps_info:
   {
     gst_sdp_media_free (smedia);
-    g_warning ("ignoring stream %d", gst_rtsp_stream_get_index (stream));
-    return;
+    GST_ERROR ("caps for stream %d have no structure",
+        gst_rtsp_stream_get_index (stream));
+    return FALSE;
+  }
+crypto_sessions_error:
+  {
+    gst_sdp_media_free (smedia);
+    GST_ERROR ("unable to add MIKEY crypto sessions for stream %d",
+        gst_rtsp_stream_get_index (stream));
+    return FALSE;
   }
 }
 
@@ -341,6 +460,7 @@ gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
 {
   guint i, n_streams;
   gchar *rangestr;
+  gboolean res;
 
   n_streams = gst_rtsp_media_n_streams (media);
 
@@ -351,11 +471,16 @@ gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
   gst_sdp_message_add_attribute (sdp, "range", rangestr);
   g_free (rangestr);
 
-  for (i = 0; i < n_streams; i++) {
+  res = TRUE;
+  for (i = 0; res && (i < n_streams); i++) {
     GstRTSPStream *stream;
 
     stream = gst_rtsp_media_get_stream (media, i);
-    gst_rtsp_sdp_from_stream (sdp, info, stream);
+    res = gst_rtsp_sdp_from_stream (sdp, info, stream);
+    if (!res) {
+      GST_ERROR ("could not get SDP from stream %p", stream);
+      goto sdp_error;
+    }
   }
 
   {
@@ -382,7 +507,7 @@ gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
     }
   }
 
-  return TRUE;
+  return res;
 
   /* ERRORS */
 not_prepared:
@@ -390,6 +515,11 @@ not_prepared:
     GST_ERROR ("media %p is not prepared", media);
     return FALSE;
   }
+sdp_error:
+  {
+    GST_ERROR ("could not get SDP from media %p", media);
+    return FALSE;
+  }
 }
 
 /**
@@ -400,32 +530,37 @@ not_prepared:
  *
  * Add info from @stream to @sdp.
  *
+ * Returns: TRUE on success.
  */
-void
+gboolean
 gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info,
     GstRTSPStream * stream)
 {
   GstCaps *caps;
   GstRTSPProfile profiles;
   guint mask;
+  gboolean res;
 
   caps = gst_rtsp_stream_get_caps (stream);
 
   if (caps == NULL) {
-    g_warning ("ignoring stream without caps");
-    return;
+    GST_ERROR ("stream %p has no caps", stream);
+    return FALSE;
   }
 
   /* make a new media for each profile */
   profiles = gst_rtsp_stream_get_profiles (stream);
   mask = 1;
-  while (profiles >= mask) {
+  res = TRUE;
+  while (res && (profiles >= mask)) {
     GstRTSPProfile prof = profiles & mask;
 
     if (prof)
-      make_media (sdp, info, stream, caps, prof);
+      res = make_media (sdp, info, stream, caps, prof);
 
     mask <<= 1;
   }
   gst_caps_unref (caps);
+
+  return res;
 }

gst/rtsp-server/rtsp-sdp.h

@@ -34,7 +34,7 @@ typedef struct {
 
 /* creating SDP */
 gboolean            gst_rtsp_sdp_from_media  (GstSDPMessage *sdp, GstSDPInfo *info, GstRTSPMedia * media);
-void                gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPStream *stream);
+gboolean            gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPStream *stream);
 
 G_END_DECLS
 

gst/rtsp-server/rtsp-stream.c

@@ -1671,6 +1671,32 @@ gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
   return session;
 }
 
+/**
+ * gst_rtsp_stream_get_encoder:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the SRTP encoder for this stream.
+ *
+ * Returns: (transfer full): The SRTP encoder for this stream. Unref after usage.
+ */
+GstElement *
+gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  GstElement *encoder;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((encoder = priv->srtpenc))
+    g_object_ref (encoder);
+  g_mutex_unlock (&priv->lock);
+
+  return encoder;
+}
+
 /**
  * gst_rtsp_stream_get_ssrc:
  * @stream: a #GstRTSPStream

gst/rtsp-server/rtsp-stream.h

@@ -129,6 +129,8 @@ GstRTSPAddress *  gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
 
 GObject *         gst_rtsp_stream_get_rtpsession   (GstRTSPStream *stream);
 
+GstElement *      gst_rtsp_stream_get_srtp_encoder (GstRTSPStream *stream);
+
 void              gst_rtsp_stream_get_ssrc         (GstRTSPStream *stream,
                                                     guint *ssrc);