diff --git a/player.c b/player.c
index 6e1ee80d..6131ad4c 100644
--- a/player.c
+++ b/player.c
@@ -1062,8 +1062,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
               conn->first_packet_time_to_play = should_be_time;
 
               int64_t lt = conn->first_packet_time_to_play - local_time_now;
-
-              debug(1, "Connection %d: Lead time for first frame %" PRId64 ": %f seconds.",
+              
+              if (lt < 100000000)
+                debug(1, "Connection %d: Short lead time for first frame %" PRId64 ": %f seconds.",
                     conn->connection_number, conn->first_packet_timestamp, lt * 0.000000001);
 
               int64_t lateness = local_time_now - conn->first_packet_time_to_play;
diff --git a/rtp.c b/rtp.c
index f80e32ae..401f4dba 100644
--- a/rtp.c
+++ b/rtp.c
@@ -1310,7 +1310,7 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
         // if the master clock and the anchor clock are the same
         // wait at least this time before using the new master clock
         // note that mastershipo may be backdated
-        if (duration_of_mastership < 100000000) {
+        if (duration_of_mastership < 275000000) {
           debug(3, "master not old enough yet: %f ms", 0.000001 * duration_of_mastership);
           response = clock_not_ready;
         } else if ((duration_of_mastership > 5000000000) ||
@@ -2254,7 +2254,7 @@ void *rtp_buffered_audio_processor(void *arg) {
   int streaming_has_started = 0;
   int play_enabled = 0;
   uint32_t flush_from_timestamp;
-  double requested_lead_time = 0.0; // normal lead time minimum -- maybe  it should be about 0.1
+  double requested_lead_time = 0.1; // normal lead time minimum -- maybe  it should be about 0.1
 
   // wait until our timing information is valid
 
@@ -2642,7 +2642,7 @@ void *rtp_buffered_audio_processor(void *arg) {
                   if (pkt->size) {
                     // if (0) {
                     if (pkt->size <= 7) { // no idea about this...
-                      debug(1, "malformed AAC packet skipped.");
+                      debug(2, "malformed AAC packet skipped.");
                     } else {
                       ret = avcodec_send_packet(codec_context, pkt);
 
diff --git a/rtsp.c b/rtsp.c
index 177a4b81..4b0827bf 100644
--- a/rtsp.c
+++ b/rtsp.c
@@ -697,7 +697,7 @@ void cancel_all_RTSP_threads(airplay_stream_c stream_category, int except_this_o
         ((stream_category == unspecified_stream_category) ||
          (stream_category == conns[i]->airplay_stream_category))) {
       pthread_cancel(conns[i]->thread);
-      debug(1, "Connection %d: cancelled.", conns[i]->connection_number);
+      debug(2, "Connection %d: cancelled.", conns[i]->connection_number);
     }
   }
   for (i = 0; i < nconns; i++) {
@@ -706,7 +706,7 @@ void cancel_all_RTSP_threads(airplay_stream_c stream_category, int except_this_o
         ((stream_category == unspecified_stream_category) ||
          (stream_category == conns[i]->airplay_stream_category))) {
       pthread_join(conns[i]->thread, NULL);
-      debug(1, "Connection %d: joined.", conns[i]->connection_number);
+      debug(2, "Connection %d: joined.", conns[i]->connection_number);
       free(conns[i]);
       conns[i] = NULL;
     }
@@ -2625,7 +2625,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
             uint8_t isRemoteControlOnlyBoolean = 0;
             plist_get_bool_val(isRemoteControlOnly, &isRemoteControlOnlyBoolean);
             if (isRemoteControlOnlyBoolean != 0) {
-              debug(1, "Connection %d: SETUP: Remote Control setup detected.",
+              debug(2, "Connection %d: SETUP: Remote Control setup detected.",
                     conn->connection_number);
               conn->airplay_stream_category = remote_control_stream;
             } else {
@@ -2825,7 +2825,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
                 conn->connection_number, err);
           }
 
-          debug(1, "Connection %d: TCP Remote Control event port opened: %u.",
+          debug(2, "Connection %d: TCP Remote Control event port opened: %u.",
                 conn->connection_number, conn->local_event_port);
           if (conn->rtp_event_thread != NULL)
             debug(1, "previous rtp_event_thread allocation not freed, it seems.");
@@ -3056,7 +3056,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
       resp->respcode = 200;
     } else if (conn->airplay_stream_category == remote_control_stream) {
       debug(2, "Connection %d: SETUP: Remote Control Stream received.", conn->connection_number);
-      debug_log_rtsp_message(1, "Remote Control Stream stream (second) message", req);
+      debug_log_rtsp_message(2, "Remote Control Stream stream (second) message", req);
 
       resp->respcode = 200;
     } else {