diff --git a/pkg/h264/rtp.go b/pkg/h264/rtp.go
index 08e61a78..6044caae 100644
--- a/pkg/h264/rtp.go
+++ b/pkg/h264/rtp.go
@@ -75,11 +75,8 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 				// https://github.com/AlexxIT/WebRTC/issues/391
 				// https://github.com/AlexxIT/WebRTC/issues/392
 				for i := 0; i < len(payload); {
-					switch NALUType(payload[i:]) {
-					case NALUTypeSPS, NALUTypePPS, NALUTypeIFrame, NALUTypePFrame:
-					default:
-						payload = payload[:i]
-						continue
+					if i+4 >= len(payload) {
+						break
 					}
 
 					size := bytes.Index(payload[i+4:], []byte{0, 0, 0, 1})
@@ -89,6 +86,7 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 						}
 						size = len(payload) - (i + 4)
 					}
+
 					binary.BigEndian.PutUint32(payload[i:], uint32(size))
 
 					i += size + 4
diff --git a/www/video-rtc.js b/www/video-rtc.js
index 0bb9e471..e04d5b66 100644
--- a/www/video-rtc.js
+++ b/www/video-rtc.js
@@ -395,11 +395,11 @@ export class VideoRTC extends HTMLElement {
                         bufLen = 0;
                         sb.appendBuffer(data);
                     } else if (sb.buffered && sb.buffered.length) {
-                        const end = sb.buffered.end(sb.buffered.length - 1) - 5;
+                        const end = sb.buffered.end(sb.buffered.length - 1) - 15;
                         const start = sb.buffered.start(0);
                         if (end > start) {
                             sb.remove(start, end);
-                            ms.setLiveSeekableRange(end, end + 5);
+                            ms.setLiveSeekableRange(end, end + 15);
                         }
                         // console.debug("VideoRTC.buffered", start, end);
                     }