diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md
index e845aa33..c91f6307 100644
--- a/RELEASENOTES-DEVELOPMENT.md
+++ b/RELEASENOTES-DEVELOPMENT.md
@@ -1,3 +1,7 @@
+Version-4.1-dev-115-g8656149d
+====
+* Simplify and improve timing when a play session from a new device is started.
+
 Version-4.1-dev-106-ge98111fb
 ====
 #### Enhancement
diff --git a/player.c b/player.c
index 1feee08f..2e9819c9 100644
--- a/player.c
+++ b/player.c
@@ -1062,17 +1062,23 @@ 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;
-              
-              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);
 
+              if (lt < 150000000) {
+                debug(1, "Connection %d: Short lead time for first frame %" PRId64 ": %f seconds. Flushing 0.5 seconds",
+                    conn->connection_number, conn->first_packet_timestamp, lt * 0.000000001);
+                do_flush(conn->first_packet_timestamp + 5 * 4410, conn);
+              } else {
+                debug(1, "Connection %d: 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;
               if (lateness > 0) {
                 debug(1, "First packet is %" PRId64 " nanoseconds late! Flushing 0.5 seconds",
                       lateness);
-                do_flush(conn->first_packet_timestamp + 5 * 4410, conn);
+
               }
+*/
             }
 
             if (conn->first_packet_time_to_play != 0) {
diff --git a/rtp.c b/rtp.c
index 4bd07a60..348ac869 100644
--- a/rtp.c
+++ b/rtp.c
@@ -1280,7 +1280,7 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
   if (response == clock_ok) {
     uint64_t time_now = get_absolute_time_in_ns();
     int64_t time_since_sample = time_now - actual_time_of_sample;
-    if (time_since_sample > 10000000000) {
+    if (time_since_sample > 300000000000) {
       if (long_time_notifcation_done == 0) {
         debug(1, "The last PTP timing sample is pretty old: %f seconds.",
               0.000000001 * time_since_sample);
@@ -1305,7 +1305,7 @@ 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 < 2000000000) {
+        if (duration_of_mastership < 1500000000) {
           debug(3, "master not old enough yet: %f ms", 0.000001 * duration_of_mastership);
           response = clock_not_ready;
         } else if ((duration_of_mastership > 5000000000) ||
diff --git a/rtsp.c b/rtsp.c
index 6b46310a..cd76c537 100644
--- a/rtsp.c
+++ b/rtsp.c
@@ -2538,7 +2538,7 @@ void handle_teardown_2(rtsp_conn_info *conn, __attribute__((unused)) rtsp_messag
     resp->respcode = 451; // don't know what to do here
   }
   // debug(1,"Bogus exit for valgrind -- remember to comment it out!.");
-  // exit(EXIT_SUCCESS); // 
+  // exit(EXIT_SUCCESS); //
 }
 #endif
 
@@ -2679,6 +2679,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
         // if it's a full service PTP stream, we get groupUUID, groupContainsGroupLeader and
         // timingPeerList
         if (conn->airplay_stream_category == ptp_stream) {
+          ptp_send_control_message_string("T"); // remove all previous history
           debug_log_rtsp_message(2, "SETUP \"PTP\" message", req);
           plist_t groupUUID = plist_dict_get_item(messagePlist, "groupUUID");
           if (groupUUID) {
@@ -2836,8 +2837,8 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
             debug(2, "Connection %d: SETUP mdns_update on %s.", conn->connection_number,
                   get_category_string(conn->airplay_stream_category));
             mdns_update(NULL, secondary_txt_records);
-            
-            
+
+
 
             resp->respcode = 200;
           } else {