diff --git a/src/plugins/janus_audiobridge.c b/src/plugins/janus_audiobridge.c
index aa59b7f055..5af9139b43 100644
--- a/src/plugins/janus_audiobridge.c
+++ b/src/plugins/janus_audiobridge.c
@@ -1729,9 +1729,9 @@ typedef struct janus_audiobridge_participant {
 	OpusDecoder *decoder;		/* Opus decoder instance */
 	gboolean fec;				/* Opus FEC status */
 	int expected_loss;			/* Percentage of expected loss, to configure libopus FEC behaviour (default=0, no FEC even if negotiated) */
-	uint16_t expected_seq;		/* Expected sequence number */
 	uint16_t probation; 		/* Used to determine new ssrc validity */
 	uint32_t last_timestamp;	/* Last in seq timestamp */
+	uint16_t last_seq; 		/* Last sequence number */
 	gboolean reset;				/* Whether or not the Opus context must be reset, without re-joining the room */
 	GThread *thread;			/* Encoding thread for this participant */
 	gboolean mjr_active;		/* Whether this participant has to be recorded to an mjr file or not */
@@ -2189,6 +2189,7 @@ static int janus_audiobridge_resample(int16_t *input, int input_num, int input_r
 /* Jitter Buffer and queue-in settings */
 #define JITTER_BUFFER_MIN_PACKETS 2
 #define JITTER_BUFFER_MAX_PACKETS 50
+#define JITTER_BUFFER_MAX_GAP_SIZE 20
 #define JITTER_BUFFER_CHECK_USECS 1*G_USEC_PER_SEC
 #define QUEUE_IN_MAX_PACKETS 4
 
@@ -6603,9 +6604,9 @@ static void *janus_audiobridge_handler(void *data) {
 				participant->decoder = NULL;
 				participant->reset = FALSE;
 				participant->fec = FALSE;
-				participant->expected_seq = 0;
 				participant->probation = 0;
 				participant->last_timestamp = 0;
+				participant->last_seq = 0;
 				janus_mutex_init(&participant->qmutex);
 				participant->arc = NULL;
 				janus_audiobridge_plainrtp_media_cleanup(&participant->plainrtp_media);
@@ -8747,8 +8748,9 @@ static void *janus_audiobridge_participant_thread(void *data) {
 	int jitter_ticks = 0;
 	janus_rtp_header *rtp = NULL;
 	gint64 now = janus_get_monotonic_time(), before = now;
-	gboolean first = TRUE, use_fec = FALSE;
+	gboolean first = TRUE;
 	int ret = 0;
+	int lost_packets_gap = 0;
 
 	/* Start working: check both the incoming queue (to decode and queue) and the outgoing one (to encode and send) */
 	while(!g_atomic_int_get(&stopping) && g_atomic_int_get(&session->destroyed) == 0) {
@@ -8779,8 +8781,54 @@ static void *janus_audiobridge_participant_thread(void *data) {
 				jitter_buffer_tick(participant->jitter);
 				janus_mutex_unlock(&participant->qmutex);
 				if(ret != JITTER_BUFFER_OK) {
-					/* No packet in the jitter buffer? Move on the talking detection, if needed */
-					janus_audiobridge_participant_istalking(session, participant, NULL, NULL);
+					if(!first && participant->codec == JANUS_AUDIOCODEC_OPUS && lost_packets_gap <= JITTER_BUFFER_MAX_GAP_SIZE && !participant->muted) {
+						lost_packets_gap += 1;
+
+						if(!g_atomic_int_compare_and_exchange(&participant->decoding, 0, 1)) {
+							/* This means we're cleaning up, so don't try to decode */
+							janus_audiobridge_buffer_packet_destroy(bpkt);
+							break;
+						}
+
+						int32_t output_samples;
+						opus_decoder_ctl(participant->decoder, OPUS_GET_LAST_PACKET_DURATION(&output_samples));
+
+						pkt = g_malloc(sizeof(janus_audiobridge_rtp_relay_packet));
+						pkt->data = g_malloc0(BUFFER_SAMPLES * sizeof(opus_int16));
+						pkt->ssrc = 0;
+						pkt->timestamp = participant->last_timestamp + OPUS_SAMPLES;
+						pkt->seq_number = participant->last_seq + 1;
+						/* This is a redundant packet, so we can't parse any extension info */
+						pkt->silence = FALSE;
+						janus_audiobridge_participant_istalking(session, participant, NULL, NULL);
+						pkt->length = opus_decode(participant->decoder, NULL, 0, (opus_int16 *)pkt->data, output_samples, 0);
+#ifdef HAVE_RNNOISE
+						/* Check if we need to denoise this packet */
+						if(participant->denoise)
+							janus_audiobridge_participant_denoise(participant, (char *)pkt->data, pkt->length);
+#endif
+						/* Update the details */
+						participant->last_seq = pkt->seq_number;
+						participant->last_timestamp = pkt->timestamp;
+						g_atomic_int_set(&participant->decoding, 0);
+						if(pkt->length < 0) {
+							JANUS_LOG(LOG_ERR, "[Opus] Ops! got an error decoding the Opus frame: %d (%s)\n", pkt->length, opus_strerror(pkt->length));
+							g_free(pkt->data);
+							g_free(pkt);
+							break;
+						}
+						/* Queue the decoded packet for the mixer */
+						/* Do not let queue-in grow too much */
+						guint count = g_list_length(participant->inbuf);
+						if((int) count > QUEUE_IN_MAX_PACKETS) {
+							JANUS_LOG(LOG_WARN, "Participant queue-in contains too many packets, clearing now (count=%u)\n", count);
+							janus_audiobridge_participant_clear_inbuf(participant);
+						}
+						participant->inbuf = g_list_append(participant->inbuf, pkt);
+					} else {
+						/* No packet in the jitter buffer? Move on the talking detection, if needed */
+						janus_audiobridge_participant_istalking(session, participant, NULL, NULL);
+					}
 				} else {
 					/* Decode the audio packet */
 					bpkt = (janus_audiobridge_buffer_packet *)jbp.data;
@@ -8802,42 +8850,8 @@ static void *janus_audiobridge_participant_thread(void *data) {
 						break;
 					}
 					rtp = (janus_rtp_header *)buffer;
-					/* If this is Opus, check if there's a packet gap we should fix with FEC */
-					use_fec = FALSE;
-					/* FIXME Disable inbound FEC due to potential deadlocks on qmutex */
-					/* TODO replacing FEC with PLC will eliminate the issue */
-					//if(!first && participant->codec == JANUS_AUDIOCODEC_OPUS && participant->fec) {
-					//	if(ntohs(rtp->seq_number) == (participant->expected_seq + 1)) {
-					//		/* Lost a packet here? Use FEC to recover */
-					//		use_fec = TRUE;
-					//	}
-					//}
 					first = FALSE;
-					if(use_fec) {
-						/* There was a gap, try to get decode from redundant info first */
-						pkt = g_malloc(sizeof(janus_audiobridge_rtp_relay_packet));
-						pkt->data = g_malloc0(BUFFER_SAMPLES*sizeof(opus_int16));
-						pkt->ssrc = 0;
-						pkt->timestamp = participant->last_timestamp + 960;	/* FIXME */
-						pkt->seq_number = participant->expected_seq;		/* FIXME */
-						/* This is a redundant packet, so we can't parse any extension info */
-						pkt->silence = FALSE;
-						/* Decode the lost packet using fec=1 */
-						int32_t output_samples;
-						opus_decoder_ctl(participant->decoder, OPUS_GET_LAST_PACKET_DURATION(&output_samples));
-						pkt->length = opus_decode(participant->decoder, payload, plen, (opus_int16 *)pkt->data, output_samples, 1);
-#ifdef HAVE_RNNOISE
-						/* Check if we need to denoise this packet */
-						if(participant->denoise)
-							janus_audiobridge_participant_denoise(participant, (char *)pkt->data, pkt->length);
-#endif
-						/* Queue the decoded redundant packet for the mixer */
-						/* FIXME This mutex lock can race with hangup_media_internal */
-						janus_mutex_lock(&participant->qmutex);
-						participant->inbuf = g_list_append(participant->inbuf, pkt);
-						janus_mutex_unlock(&participant->qmutex);
-						/* Now we can process the next packet */
-					}
+					lost_packets_gap = 0;
 					/* Decode the packet */
 					pkt = g_malloc(sizeof(janus_audiobridge_rtp_relay_packet));
 					pkt->data = g_malloc0(BUFFER_SAMPLES*sizeof(opus_int16));
@@ -8882,8 +8896,8 @@ static void *janus_audiobridge_participant_thread(void *data) {
 					/* Get rid of the buffered packet */
 					janus_audiobridge_buffer_packet_destroy(bpkt);
 					/* Update the details */
+					participant->last_seq = pkt->seq_number;
 					participant->last_timestamp = pkt->timestamp;
-					participant->expected_seq = pkt->seq_number + 1;
 					g_atomic_int_set(&participant->decoding, 0);
 					if(pkt->length < 0) {
 						if(participant->codec == JANUS_AUDIOCODEC_OPUS) {