sdp: add rollover counters for all sender SSRCsrtp

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

3 files changed, 114 insertions, 4 deletions

diff --git a/gst/rtsp-server/rtsp-sdp.c b/gst/rtsp-server/rtsp-sdp.c
index 348cab3..b5a789b 100644
--- a/gst/rtsp-server/rtsp-sdp.c
+++ b/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
@@ -73,6 +76,87 @@ update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
   gst_object_unref (src_pad);
 }
 
+static guint
+get_roc_from_stats (GstStructure * stats, guint ssrc)
+{
+  const GValue *va, *v;
+  guint i, len, roc = 0;
+
+  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 void
+mikey_add_crypto_sessions (GstRTSPStream * stream, GstMIKEYMessage * msg)
+{
+  guint i;
+  GObject *session;
+  GstElement *encoder;
+  GValueArray *sources;
+
+  session = gst_rtsp_stream_get_rtpsession (stream);
+  if (session == NULL)
+    return;
+
+  encoder = gst_rtsp_stream_get_srtp_encoder (stream);
+  if (encoder == NULL) {
+    g_object_unref (session);
+    return;
+  }
+
+  g_object_get (session, "sources", &sources, NULL);
+  for (i = 0; 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 = 0;
+      GstStructure *stats;
+
+      g_object_get (encoder, "stats", &stats, NULL);
+
+      if (stats) {
+        roc = get_roc_from_stats (stats, ssrc);
+        gst_structure_free (stats);
+      }
+
+      gst_mikey_message_add_cs_srtp (msg, 0, ssrc, roc);
+    }
+  }
+  g_value_array_free (sources);
+
+  gst_object_unref (encoder);
+  g_object_unref (session);
+}
+
 static void
 make_media (GstSDPMessage * sdp, GstSDPInfo * info,
     GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile)
@@ -86,7 +170,6 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info,
   guint ttl;
   GstClockTime rtx_time;
   gchar *base64;
-  guint32 ssrc;
   GstMIKEYMessage *mikey_msg;
 
   gst_sdp_media_new (&smedia);
@@ -155,9 +238,8 @@ 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 */
+    mikey_add_crypto_sessions (stream, mikey_msg);
 
     base64 = gst_mikey_message_base64_encode (mikey_msg);
     if (base64) {
diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index 27ad5ac..5389456 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1672,6 +1672,32 @@ gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
 }
 
 /**
+ * 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
  * @ssrc: (out): result ssrc
diff --git a/gst/rtsp-server/rtsp-stream.h b/gst/rtsp-server/rtsp-stream.h
index a6ab1ff..9ef887a 100644
--- a/gst/rtsp-server/rtsp-stream.h
+++ b/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);