diff --git a/android/hal-audio.c b/android/hal-audio.c
index a261871..3ff3b9d 100644
--- a/android/hal-audio.c
+++ b/android/hal-audio.c
@@ -45,6 +45,8 @@
 
 #define MAX_FRAMES_IN_PAYLOAD 15
 
+#define MAX_DELAY	100000 /* 100ms */
+
 static const uint8_t a2dp_src_uuid[] = {
 		0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
 		0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
@@ -236,6 +238,8 @@ struct audio_endpoint {
 	uint16_t seq;
 	uint32_t samples;
 	struct timespec start;
+
+	bool resync;
 };
 
 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
@@ -914,6 +918,7 @@ static bool resume_endpoint(struct audio_endpoint *ep)
 		return false;
 
 	ep->samples = 0;
+	ep->resync = false;
 
 	return true;
 }
@@ -1031,10 +1036,15 @@ static bool write_data(struct a2dp_stream_out *out, const void *buffer,
 		audio_sent = ep->samples * 1000000ll / out->cfg.rate;
 		audio_passed = timespec_diff_us(&current, &ep->start);
 
-		/* if we're ahead of stream then wait for next write point */
+		/* if we're ahead of stream then wait for next write point
+		 * if we're lagging more than 100ms then stop writing and just
+		 * skip data until we're back in sync
+		 */
 		if (audio_sent > audio_passed) {
 			struct timespec anchor;
 
+			ep->resync = false;
+
 			timespec_add(&ep->start, audio_sent, &anchor);
 
 			while (true) {
@@ -1051,18 +1061,28 @@ static bool write_data(struct a2dp_stream_out *out, const void *buffer,
 					return false;
 				}
 			}
-		}
+		} else if (!ep->resync) {
+			uint64_t diff = audio_passed - audio_sent;
 
-		/* wait some time for socket to be ready for write,
-		 * but we'll just skip writing data if timeout occurs
-		 */
-		if (!wait_for_endpoint(ep, &do_write))
-			return false;
+			if (diff > MAX_DELAY) {
+				warn("lag is %jums, resyncing", diff / 1000);
+				ep->resync = true;
+			}
+		}
 
-		if (do_write)
-			if (!write_to_endpoint(ep, written))
+		/* in resync mode we'll just drop mediapackets */
+		if (!ep->resync) {
+			/* wait some time for socket to be ready for write,
+			 * but we'll just skip writing data if timeout occurs
+			 */
+			if (!wait_for_endpoint(ep, &do_write))
 				return false;
 
+			if (do_write)
+				if (!write_to_endpoint(ep, written))
+					return false;
+		}
+
 		/* AudioFlinger provides 16bit PCM, so sample size is 2 bytes
 		 * multiplied by number of channels. Number of channels is
 		 * simply number of bits set in channels mask.