diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/rtpsession.c b/subprojects/gst-plugins-good/gst/rtpmanager/rtpsession.c
index f9202a7e8a..70f69db881 100644
--- a/subprojects/gst-plugins-good/gst/rtpmanager/rtpsession.c
+++ b/subprojects/gst-plugins-good/gst/rtpmanager/rtpsession.c
@@ -3819,7 +3819,7 @@ typedef struct
   gboolean timeout_inactive_sources;
 } ReportData;
 
-static void
+static gboolean
 session_start_rtcp (RTPSession * sess, ReportData * data)
 {
   GstRTCPPacket *packet = &data->packet;
@@ -3844,8 +3844,11 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
     gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
 
     /* get latest stats */
-    rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
-        &ntptime, &rtptime, &packet_count, &octet_count);
+    if (!rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
+            &ntptime, &rtptime, &packet_count, &octet_count)) {
+      gst_rtcp_buffer_unmap (&data->rtcpbuf);
+      return FALSE;
+    }
     /* store stats */
     rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
         packet_count, octet_count);
@@ -3861,6 +3864,8 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
     gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
     gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
   }
+
+  return TRUE;
 }
 
 /* construct a Sender or Receiver Report */
@@ -4541,7 +4546,10 @@ generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
   data->source = source;
 
   /* open packet */
-  session_start_rtcp (sess, data);
+  if (!session_start_rtcp (sess, data)) {
+    GST_WARNING ("source %08x can not generate RTCP", source->ssrc);
+    return;
+  }
 
   if (source->marked_bye) {
     /* send BYE */
diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/rtpsource.c b/subprojects/gst-plugins-good/gst/rtpmanager/rtpsource.c
index 673f1ad682..2c4fe559bd 100644
--- a/subprojects/gst-plugins-good/gst/rtpmanager/rtpsource.c
+++ b/subprojects/gst-plugins-good/gst/rtpmanager/rtpsource.c
@@ -1627,6 +1627,13 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
   }
 
   if (src->clock_rate != -1) {
+    /* if no running time has been set yet we wait until we get one */
+    if (src->last_rtime == -1) {
+      GST_WARNING ("running time not set, can not create SR for SSRC %u",
+          src->ssrc);
+      return FALSE;
+    }
+
     /* get the diff between the clock running_time and the buffer running_time.
      * This is the elapsed time, as measured against the pipeline clock, between
      * when the rtp timestamp was observed and the current running_time.