diff --git a/player.c b/player.c
index 153ae227..704a23dc 100644
--- a/player.c
+++ b/player.c
@@ -1694,7 +1694,7 @@ static void *player_thread_func(void *arg) {
         //          debug(3, "Play frame %d.", play_number);
         conn->play_number_after_flush++;
         if (inframe->timestamp == 0) {
-          debug(1, "Player has supplied a silent frame, (possibly frame %u) for play number %d.",
+          debug(3, "Player has supplied a silent frame, (possibly frame %u) for play number %d.",
                 SUCCESSOR(conn->last_seqno_read),play_number);
           conn->last_seqno_read = (SUCCESSOR(conn->last_seqno_read) &
                                    0xffff); // manage the packet out of sequence minder
@@ -1848,7 +1848,7 @@ static void *player_thread_func(void *arg) {
                 SUCCESSOR(conn->last_seqno_read); // int32_t from seq_t, i.e. uint16_t, so okay.
             if (inframe->sequence_number !=
                 conn->last_seqno_read) { // seq_t, ei.e. uint16_t and int32_t, so okay
-              debug(1, "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u "
+              debug(2, "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u "
                        "and ab_write: %u.",
                     conn->last_seqno_read, inframe->sequence_number, conn->ab_read, conn->ab_write);
               conn->last_seqno_read = inframe->sequence_number; // reset warning...
diff --git a/rtp.c b/rtp.c
index 0f8d3dcf..446ba53f 100644
--- a/rtp.c
+++ b/rtp.c
@@ -147,7 +147,7 @@ void *rtp_audio_receiver(void *arg) {
             last_seqno = seqno; // reset warning...
           }
         } else {
-          debug(2, "Audio Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
+          debug(3, "Audio Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
         }
 
         uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
@@ -165,7 +165,7 @@ void *rtp_audio_receiver(void *arg) {
               (drand48() > config.diagnostic_drop_packet_fraction))
             player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
           else
-            debug(2, "Dropping audio packet %u to simulate a bad connection.", seqno);
+            debug(3, "Dropping audio packet %u to simulate a bad connection.", seqno);
           continue;
         }
         if (type == 0x56 && seqno == 0) {
@@ -319,7 +319,7 @@ void *rtp_control_receiver(void *arg) {
 
                 if (la != conn->latency) {
                   conn->latency = la;
-                  debug(2, "New latency detected: %" PRId64 ", sync latency: %" PRId64
+                  debug(3, "New latency detected: %" PRId64 ", sync latency: %" PRId64
                            ", minimum latency: %" PRId64 ", maximum "
                            "latency: %" PRId64 ", fixed offset: %" PRId64 ".",
                         la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency,
@@ -366,13 +366,13 @@ void *rtp_control_receiver(void *arg) {
             // debug(1,"Sync Time is %lld us late (local
             // times).",((local_time_now-reference_timestamp_time)*1000000)>>32);
           } else {
-            debug(1, "Sync packet received before we got a timing packet back.");
+            debug(2, "Sync packet received before we got a timing packet back.");
           }
         } else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
           pktp = packet + 4;
           plen -= 4;
           seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
-          debug(2, "Control Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
+          debug(3, "Control Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
 
           uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
           int64_t timestamp = monotonic_timestamp(actual_timestamp, conn);
@@ -391,7 +391,7 @@ void *rtp_control_receiver(void *arg) {
           debug(1, "Control Receiver -- Unknown RTP packet of type 0x%02X length %d, ignored.",
                 packet[1], nread);
       } else {
-        debug(2, "Control Receiver -- dropping a packet to simulate a bad network.");
+        debug(3, "Control Receiver -- dropping a packet to simulate a bad network.");
       }
     } else {
       debug(1, "Control Receiver -- error receiving a packet.");
@@ -469,7 +469,7 @@ void *rtp_timing_sender(void *arg) {
       }
 
     } else {
-      debug(2, "Timing Sender Thread -- dropping outgoing packet to simulate bad network.");
+      debug(3, "Timing Sender Thread -- dropping outgoing packet to simulate bad network.");
     }
 
     request_number++;
@@ -717,14 +717,14 @@ void *rtp_timing_receiver(void *arg) {
             // buffer_occupancy,
             //(return_time*1000000)>>32);
           } else {
-            debug(1, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
+            debug(2, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
                   rtus);
           }
         } else {
           debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
         }
       } else {
-        debug(2, "Timing Receiver Thread -- dropping incoming packet to simulate a bad network.");
+        debug(3, "Timing Receiver Thread -- dropping incoming packet to simulate a bad network.");
       }
     } else {
       debug(1, "Timing receiver -- error receiving a packet.");
@@ -817,7 +817,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
          "SETUP call?");
   else {
 
-    debug(2, "rtp_setup: cport=%d tport=%d.", cport, tport);
+    debug(3, "rtp_setup: cport=%d tport=%d.", cport, tport);
 
     // print out what we know about the client
     void *client_addr = NULL, *self_addr = NULL;
@@ -853,7 +853,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
     inet_ntop(conn->connection_ip_family, self_addr, conn->self_ip_string,
               sizeof(conn->self_ip_string));
 
-    debug(2, "Set up play connection from %s to self at %s on RTSP conversation thread %d.",
+    debug(2, "SETUP connection from %s to self at %s on RTSP conversation thread %d.",
           conn->client_ip_string, conn->self_ip_string, conn->connection_number);
 
     // set up a the record of the remote's control socket
@@ -911,7 +911,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
     conn->local_audio_port = bind_port(conn->connection_ip_family, conn->self_ip_string,
                                        conn->self_scope_id, &conn->audio_socket);
 
-    debug(2, "listening for audio, control and timing on ports %d, %d, %d.", conn->local_audio_port,
+    debug(3, "listening for audio, control and timing on ports %d, %d, %d.", conn->local_audio_port,
           conn->local_control_port, conn->local_timing_port);
 
     conn->reference_timestamp = 0;
@@ -959,7 +959,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
     char req[8]; // *not* a standard RTCP NACK
     req[0] = 0x80;
     req[1] = (char)0x55 | (char)0x80;            // Apple 'resend'
-    *(unsigned short *)(req + 2) = htons(1);     // our seqnum
+    *(unsigned short *)(req + 2) = htons(1);     // our sequence number
     *(unsigned short *)(req + 4) = htons(first); // missed seqnum
     *(unsigned short *)(req + 6) = htons(count); // count
     socklen_t msgsize = sizeof(struct sockaddr_in);
@@ -975,7 +975,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
          resend_error_backoff_time)) {
       if ((config.diagnostic_drop_packet_fraction == 0.0) ||
           (drand48() > config.diagnostic_drop_packet_fraction)) {
-        if (sendto(conn->audio_socket, req, sizeof(req), 0,
+        if (sendto(conn->control_socket, req, sizeof(req), 0,
                    (struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) {
           char em[1024];
           strerror_r(errno, em, sizeof(em));
@@ -985,7 +985,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
           conn->rtp_time_of_last_resend_request_error_fp = 0;
         }
       } else {
-        debug(2, "Dropping resend request packet to simulate a bad network.");
+        debug(3, "Dropping resend request packet to simulate a bad network.");
       }
     }
   } else {