diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index cfd4259c6eb..de5b8c2dd56 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -1780,8 +1780,8 @@ vhost hls.srs.com {
         #   EXT-X-TARGETDURATION = hls_td_ratio * hls_fragment // init
         #   EXT-X-TARGETDURATION = max(ts_duration, EXT-X-TARGETDURATION) // for each ts
         # Overwrite by env SRS_VHOST_HLS_HLS_TD_RATIO for all vhosts.
-        # default: 1.5
-        hls_td_ratio 1.5;
+        # default: 1.0
+        hls_td_ratio 1.0;
         # the audio overflow ratio.
         # for pure audio, the duration to reap the segment.
         # for example, the hls_fragment is 10s, hls_aof_ratio is 2.0,
diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index 176f7f8fd53..1507b028384 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -6973,7 +6973,7 @@ double SrsConfig::get_hls_td_ratio(string vhost)
 {
     SRS_OVERWRITE_BY_ENV_FLOAT("srs.vhost.hls.hls_td_ratio"); // SRS_VHOST_HLS_HLS_TD_RATIO
 
-    static double DEFAULT = 1.5;
+    static double DEFAULT = 1.0;
     
     SrsConfDirective* conf = get_hls(vhost);
     if (!conf) {
diff --git a/trunk/src/app/srs_app_hls.cpp b/trunk/src/app/srs_app_hls.cpp
index 15ec9cd457e..547ebcb567c 100644
--- a/trunk/src/app/srs_app_hls.cpp
+++ b/trunk/src/app/srs_app_hls.cpp
@@ -542,9 +542,12 @@ bool SrsHlsMuxer::is_segment_overflow()
         return false;
     }
     
-    // use N% deviation, to smoother.
+    // Use N% deviation, to smoother.
     srs_utime_t deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0;
-    return current->duration() >= hls_fragment + deviation;
+
+    // Keep in mind that we use max_td for the base duration, not the hls_fragment. To calculate
+    // max_td, multiply hls_fragment by hls_td_ratio.
+    return current->duration() >= max_td + deviation;
 }
 
 bool SrsHlsMuxer::wait_keyframe()
@@ -627,6 +630,11 @@ srs_error_t SrsHlsMuxer::flush_video(SrsTsMessageCache* cache)
     return err;
 }
 
+void SrsHlsMuxer::update_duration(uint64_t dts)
+{
+    current->append(dts / 90);
+}
+
 srs_error_t SrsHlsMuxer::segment_close()
 {
     srs_error_t err = do_segment_close();
@@ -920,8 +928,9 @@ srs_error_t SrsHlsController::on_publish(SrsRequest* req)
     std::string vhost = req->vhost;
     std::string stream = req->stream;
     std::string app = req->app;
-    
+
     srs_utime_t hls_fragment = _srs_config->get_hls_fragment(vhost);
+    double hls_td_ratio = _srs_config->get_hls_td_ratio(vhost);
     srs_utime_t hls_window = _srs_config->get_hls_window(vhost);
     
     // get the hls m3u8 ts list entry prefix config
@@ -965,9 +974,9 @@ srs_error_t SrsHlsController::on_publish(SrsRequest* req)
     // This config item is used in SrsHls, we just log its value here.
     bool hls_dts_directly = _srs_config->get_vhost_hls_dts_directly(req->vhost);
 
-    srs_trace("hls: win=%dms, frag=%dms, prefix=%s, path=%s, m3u8=%s, ts=%s, aof=%.2f, floor=%d, clean=%d, waitk=%d, dispose=%dms, dts_directly=%d",
+    srs_trace("hls: win=%dms, frag=%dms, prefix=%s, path=%s, m3u8=%s, ts=%s, tdr=%.2f, aof=%.2f, floor=%d, clean=%d, waitk=%d, dispose=%dms, dts_directly=%d",
         srsu2msi(hls_window), srsu2msi(hls_fragment), entry_prefix.c_str(), path.c_str(), m3u8_file.c_str(), ts_file.c_str(),
-        hls_aof_ratio, ts_floor, cleanup, wait_keyframe, srsu2msi(hls_dispose), hls_dts_directly);
+        hls_td_ratio, hls_aof_ratio, ts_floor, cleanup, wait_keyframe, srsu2msi(hls_dispose), hls_dts_directly);
     
     return err;
 }
@@ -1017,6 +1026,10 @@ srs_error_t SrsHlsController::write_audio(SrsAudioFrame* frame, int64_t pts)
     if ((err = tsmc->cache_audio(frame, pts)) != srs_success) {
         return srs_error_wrap(err, "hls: cache audio");
     }
+
+    // First, update the duration of the segment, as we might collect the segment. The duration should
+    // cover from the first frame to the last frame.
+    muxer->update_duration(tsmc->audio->dts);
     
     // reap when current source is pure audio.
     // it maybe changed when stream info changed,
@@ -1064,6 +1077,10 @@ srs_error_t SrsHlsController::write_video(SrsVideoFrame* frame, int64_t dts)
     if ((err = tsmc->cache_video(frame, dts)) != srs_success) {
         return srs_error_wrap(err, "hls: cache video");
     }
+
+    // First, update the duration of the segment, as we might collect the segment. The duration should
+    // cover from the first frame to the last frame.
+    muxer->update_duration(tsmc->video->dts);
     
     // when segment overflow, reap if possible.
     if (muxer->is_segment_overflow()) {
diff --git a/trunk/src/app/srs_app_hls.hpp b/trunk/src/app/srs_app_hls.hpp
index e111b68b695..8b74f48db14 100644
--- a/trunk/src/app/srs_app_hls.hpp
+++ b/trunk/src/app/srs_app_hls.hpp
@@ -204,6 +204,10 @@ class SrsHlsMuxer
     virtual bool pure_audio();
     virtual srs_error_t flush_audio(SrsTsMessageCache* cache);
     virtual srs_error_t flush_video(SrsTsMessageCache* cache);
+    // When flushing video or audio, we update the duration. But, we should also update the
+    // duration before closing the segment. Keep in mind that it's fine to update the duration
+    // several times using the same dts timestamp.
+    void update_duration(uint64_t dts);
     // Close segment(ts).
     virtual srs_error_t segment_close();
 private: