diff --git a/api/source/mk_player.cpp b/api/source/mk_player.cpp
index 9fc0abc1..4d900c82 100755
--- a/api/source/mk_player.cpp
+++ b/api/source/mk_player.cpp
@@ -188,7 +188,7 @@ API_EXPORT void API_CALL mk_player_seektoByPos(mk_player ctx, int seekPos)
 }
 
 static void mk_player_set_on_event(mk_player ctx, on_mk_play_event cb, void *user_data, int type) {
-    assert(ctx && cb);
+    assert(ctx);
     MediaPlayerForC &obj = **((MediaPlayerForC::Ptr *)ctx);
     obj.setOnEvent(cb,user_data, type);
 }
@@ -202,7 +202,7 @@ API_EXPORT void API_CALL mk_player_set_on_shutdown(mk_player ctx, on_mk_play_eve
 }
 
 API_EXPORT void API_CALL mk_player_set_on_data(mk_player ctx, on_mk_play_data cb, void *user_data) {
-    assert(ctx && cb);
+    assert(ctx);
     MediaPlayerForC &obj = **((MediaPlayerForC::Ptr *)ctx);
     obj.setOnData(cb,user_data);
 }
diff --git a/conf/config.ini b/conf/config.ini
index accde647..02fc096c 100644
--- a/conf/config.ini
+++ b/conf/config.ini
@@ -29,8 +29,6 @@ maxStreamWaitMS=15000
 #某个流无人观看时，触发hook.on_stream_none_reader事件的最大等待时间，单位毫秒
 #在配合hook.on_stream_none_reader事件时，可以做到无人观看自动停止拉流或停止接收推流
 streamNoneReaderDelayMS=20000
-#是否开启低延时模式，该模式下禁用MSG_MORE,启用TCP_NODEALY，延时将降低，但数据发送性能将降低
-ultraLowDelay=1
 #拉流代理是否添加静音音频(直接拉流模式本协议无效)
 addMuteAudio=1
 #拉流代理时如果断流再重连成功是否删除前一次的媒体流数据，如果删除将重新开始，
@@ -43,8 +41,8 @@ publishToHls=1
 #是否默认推流时mp4录像，hook接口(on_publish)中可以覆盖该设置
 publishToMP4=0
 #合并写缓存大小(单位毫秒)，合并写指服务器缓存一定的数据后才会一次性写入socket，这样能提高性能，但是会提高延时
-#在开启低延时模式后，该参数不起作用
-mergeWriteMS=300
+#开启后会同时关闭TCP_NODELAY并开启MSG_MORE
+mergeWriteMS=0
 
 [hls]
 #hls写文件的buf大小，调整参数可以提高文件io性能
@@ -150,6 +148,8 @@ keepAliveSecond=15
 modifyStamp=0
 #rtmp服务器监听端口
 port=1935
+#rtmps服务器监听地址
+sslport=19350
 
 [rtp]
 #音频mtu大小，该参数限制rtp最大字节数，推荐不要超过1400
diff --git a/server/main.cpp b/server/main.cpp
index 55ea061e..cd1ee8a0 100644
--- a/server/main.cpp
+++ b/server/main.cpp
@@ -75,8 +75,10 @@ onceToken token1([](){
 namespace Rtmp {
 #define RTMP_FIELD "rtmp."
 const string kPort = RTMP_FIELD"port";
+const string kSSLPort = RTMP_FIELD"sslport";
 onceToken token1([](){
     mINI::Instance()[kPort] = 1935;
+    mINI::Instance()[kSSLPort] = 19350;
 },nullptr);
 } //namespace RTMP
 
@@ -255,9 +257,10 @@ int start_main(int argc,char *argv[]) {
         uint16_t rtspPort = mINI::Instance()[Rtsp::kPort];
         uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort];
         uint16_t rtmpPort = mINI::Instance()[Rtmp::kPort];
+        uint16_t rtmpsPort = mINI::Instance()[Rtmp::kSSLPort];
         uint16_t httpPort = mINI::Instance()[Http::kPort];
         uint16_t httpsPort = mINI::Instance()[Http::kSSLPort];
-        uint16_t rtp_proxy = mINI::Instance()[RtpProxy::kPort];
+        uint16_t rtpPort = mINI::Instance()[RtpProxy::kPort];
 
         //设置poller线程数,该函数必须在使用ZLToolKit网络相关对象之前调用才能生效
         EventPollerPool::setPoolSize(threads);
@@ -265,38 +268,51 @@ int start_main(int argc,char *argv[]) {
         //简单的telnet服务器，可用于服务器调试，但是不能使用23端口，否则telnet上了莫名其妙的现象
         //测试方法:telnet 127.0.0.1 9000
         TcpServer::Ptr shellSrv(new TcpServer());
+
+        //rtsp[s]服务器, 可用于诸如亚马逊echo show这样的设备访问
         TcpServer::Ptr rtspSrv(new TcpServer());
-        TcpServer::Ptr rtmpSrv(new TcpServer());
-        TcpServer::Ptr httpSrv(new TcpServer());
-        //如果支持ssl，还可以开启https服务器
-        TcpServer::Ptr httpsSrv(new TcpServer());
-        //支持ssl加密的rtsp服务器，可用于诸如亚马逊echo show这样的设备访问
         TcpServer::Ptr rtspSSLSrv(new TcpServer());
 
+        //rtmp[s]服务器
+        TcpServer::Ptr rtmpSrv(new TcpServer());
+        TcpServer::Ptr rtmpsSrv(new TcpServer());
+
+        //http[s]服务器
+        TcpServer::Ptr httpSrv(new TcpServer());
+        TcpServer::Ptr httpsSrv(new TcpServer());
+
 #if defined(ENABLE_RTPPROXY)
+        //GB28181 rtp推流端口，支持UDP/TCP
         UdpRecver recver;
         TcpServer::Ptr tcpRtpServer(new TcpServer());
 #endif//defined(ENABLE_RTPPROXY)
 
         try {
             //rtsp服务器，端口默认554
-            rtspSrv->start<RtspSession>(rtspPort);//默认554
+            if(rtspPort) { rtspSrv->start<RtspSession>(rtspPort); }
             //rtsps服务器，端口默认322
-            rtspSSLSrv->start<RtspSessionWithSSL>(rtspsPort);
+            if(rtspsPort) { rtspSSLSrv->start<RtspSessionWithSSL>(rtspsPort); }
+
             //rtmp服务器，端口默认1935
-            rtmpSrv->start<RtmpSession>(rtmpPort);
+            if(rtmpPort) { rtmpSrv->start<RtmpSession>(rtmpPort); }
+            //rtmps服务器，端口默认19350
+            if(rtmpsPort) { rtmpsSrv->start<RtmpSessionWithSSL>(rtmpsPort); }
+
             //http服务器，端口默认80
-            httpSrv->start<HttpSession>(httpPort);
+            if(httpPort) { httpSrv->start<HttpSession>(httpPort); }
             //https服务器，端口默认443
-            httpsSrv->start<HttpsSession>(httpsPort);
+            if(httpsPort) { httpsSrv->start<HttpsSession>(httpsPort); }
+
             //telnet远程调试服务器
-            shellSrv->start<ShellSession>(shellPort);
+            if(shellPort) { shellSrv->start<ShellSession>(shellPort); }
 
 #if defined(ENABLE_RTPPROXY)
-            //创建rtp udp服务器
-            recver.initSock(rtp_proxy);
-            //创建rtp tcp服务器
-            tcpRtpServer->start<RtpSession>(rtp_proxy);
+            if(rtpPort){
+                //创建rtp udp服务器
+                recver.initSock(rtpPort);
+                //创建rtp tcp服务器
+                tcpRtpServer->start<RtpSession>(rtpPort);
+            }
 #endif//defined(ENABLE_RTPPROXY)
 
         }catch (std::exception &ex){
diff --git a/src/Common/MediaSource.cpp b/src/Common/MediaSource.cpp
index 3c41c0ec..fafb13dd 100644
--- a/src/Common/MediaSource.cpp
+++ b/src/Common/MediaSource.cpp
@@ -516,9 +516,8 @@ static bool isFlushAble_merge(bool is_audio, uint32_t last_stamp, uint32_t new_s
 
 bool FlushPolicy::isFlushAble(uint32_t new_stamp, int cache_size) {
     bool ret = false;
-    GET_CONFIG(bool, ultraLowDelay, General::kUltraLowDelay);
     GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
-    if (ultraLowDelay || mergeWriteMS <= 0) {
+    if (mergeWriteMS <= 0) {
         //关闭了合并写或者合并写阈值小于等于0
         ret = isFlushAble_default(_is_audio, _last_stamp, new_stamp, cache_size);
     } else {
diff --git a/src/Common/MultiMediaSourceMuxer.cpp b/src/Common/MultiMediaSourceMuxer.cpp
index 0f56ddbe..58217d2b 100644
--- a/src/Common/MultiMediaSourceMuxer.cpp
+++ b/src/Common/MultiMediaSourceMuxer.cpp
@@ -22,17 +22,21 @@ MultiMuxerPrivate::MultiMuxerPrivate(const string &vhost,
                                      bool enable_mp4) {
     if (enable_rtmp) {
         _rtmp = std::make_shared<RtmpMediaSourceMuxer>(vhost, app, stream, std::make_shared<TitleMeta>(dur_sec));
+        _enable_rtxp = true;
     }
     if (enable_rtsp) {
         _rtsp = std::make_shared<RtspMediaSourceMuxer>(vhost, app, stream, std::make_shared<TitleSdp>(dur_sec));
+        _enable_rtxp = true;
     }
 
     if (enable_hls) {
         _hls = Recorder::createRecorder(Recorder::type_hls, vhost, app, stream);
+        _enable_record = true;
     }
 
     if (enable_mp4) {
         _mp4 = Recorder::createRecorder(Recorder::type_mp4, vhost, app, stream);
+        _enable_record = true;
     }
 }
 
@@ -77,8 +81,8 @@ int MultiMuxerPrivate::totalReaderCount() const {
     return (_rtsp ? _rtsp->readerCount() : 0) + (_rtmp ? _rtmp->readerCount() : 0) + (hls_src ? hls_src->readerCount() : 0);
 }
 
-static std::shared_ptr<MediaSinkInterface> makeRecorder(const vector<Track::Ptr> &tracks, Recorder::type type, MediaSource &sender){
-    auto recorder = Recorder::createRecorder(type, sender.getVhost(), sender.getApp(), sender.getId());
+static std::shared_ptr<MediaSinkInterface> makeRecorder(const vector<Track::Ptr> &tracks, Recorder::type type, const string &custom_path, MediaSource &sender){
+    auto recorder = Recorder::createRecorder(type, sender.getVhost(), sender.getApp(), sender.getId(), custom_path);
     for (auto &track : tracks) {
         recorder->addTrack(track);
     }
@@ -91,7 +95,7 @@ bool MultiMuxerPrivate::setupRecord(MediaSource &sender, Recorder::type type, bo
         case Recorder::type_hls : {
             if (start && !_hls) {
                 //开始录制
-                _hls = makeRecorder(getTracks(true), type, sender);
+                _hls = makeRecorder(getTracks(true), type, custom_path, sender);
                 auto hls_src = getHlsMediaSource();
                 if (hls_src) {
                     //设置HlsMediaSource的事件监听器
@@ -102,20 +106,21 @@ bool MultiMuxerPrivate::setupRecord(MediaSource &sender, Recorder::type type, bo
                 //停止录制
                 _hls = nullptr;
             }
+            _enable_record = _hls || _mp4;
             return true;
         }
         case Recorder::type_mp4 : {
             if (start && !_mp4) {
                 //开始录制
-                _mp4 = makeRecorder(getTracks(true), type, sender);;
+                _mp4 = makeRecorder(getTracks(true), type, custom_path, sender);
             } else if (!start && _mp4) {
                 //停止录制
                 _mp4 = nullptr;
             }
+            _enable_record = _hls || _mp4;
             return true;
         }
-        default:
-            return false;
+        default : return false;
     }
 }
 
@@ -163,6 +168,10 @@ void MultiMuxerPrivate::onTrackReady(const Track::Ptr &track) {
     }
 }
 
+bool MultiMuxerPrivate::isEnabled(){
+    return _enable_rtxp || _enable_record;
+}
+
 void MultiMuxerPrivate::onTrackFrame(const Frame::Ptr &frame) {
     if (_rtmp) {
         _rtmp->inputFrame(frame);
@@ -293,4 +302,9 @@ void MultiMediaSourceMuxer::inputFrame(const Frame::Ptr &frame) {
     _muxer->inputFrame(frame);
 }
 
+bool MultiMediaSourceMuxer::isEnabled(){
+    return _muxer->isEnabled();
+}
+
+
 }//namespace mediakit
\ No newline at end of file
diff --git a/src/Common/MultiMediaSourceMuxer.h b/src/Common/MultiMediaSourceMuxer.h
index 5ba6b8fe..04109745 100644
--- a/src/Common/MultiMediaSourceMuxer.h
+++ b/src/Common/MultiMediaSourceMuxer.h
@@ -45,6 +45,7 @@ private:
     void setTrackListener(Listener *listener);
     bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const string &custom_path);
     bool isRecording(MediaSource &sender, Recorder::type type);
+    bool isEnabled();
 private:
     void onTrackReady(const Track::Ptr & track) override;
     void onTrackFrame(const Frame::Ptr &frame) override;
@@ -57,6 +58,8 @@ private:
     MediaSinkInterface::Ptr _mp4;
     Listener *_listener = nullptr;
     std::weak_ptr<MediaSourceEvent> _meida_listener;
+    bool _enable_rtxp = false;
+    bool _enable_record = false;
 };
 
 class MultiMediaSourceMuxer : public MediaSourceEvent, public MediaSinkInterface, public TrackSource, public std::enable_shared_from_this<MultiMediaSourceMuxer>{
@@ -167,6 +170,11 @@ public:
      * @param frame 帧
      */
     void inputFrame(const Frame::Ptr &frame) override;
+
+    /**
+     * 判断是否生效(是否正在转其他协议)
+     */
+    bool isEnabled();
 private:
     MultiMuxerPrivate::Ptr _muxer;
     std::weak_ptr<MediaSourceEvent> _listener;
diff --git a/src/Common/Stamp.cpp b/src/Common/Stamp.cpp
index 7ed1474f..0bbaf958 100644
--- a/src/Common/Stamp.cpp
+++ b/src/Common/Stamp.cpp
@@ -96,58 +96,84 @@ int64_t Stamp::getRelativeStamp() const {
     return _relativeStamp;
 }
 
-
 bool DtsGenerator::getDts(uint32_t pts, uint32_t &dts){
     bool ret = false;
-    if(pts == _last_pts){
-        //pts未变，返回上次结果
-        if(_last_dts){
+    if (pts == _last_pts) {
+        //pts未变，说明dts也不会变，返回上次dts
+        if (_last_dts) {
             dts = _last_dts;
             ret = true;
         }
-        return ret;
+    } else {
+        //pts变了，尝试计算dts
+        ret = getDts_l(pts, dts);
+        if (ret) {
+            //获取到了dts，保存本次结果
+            _last_dts = dts;
+        }
     }
 
-    ret = getDts_l(pts,dts);
-    if(ret){
-        //保存本次结果
-        _last_dts = dts;
+    if (!ret) {
+        //pts排序列队长度还不知道，也就是不知道有没有B帧，
+        //那么先强制dts == pts，这样可能导致有B帧的情况下，起始画面有几帧回退
+        dts = pts;
     }
+
     //记录上次pts
     _last_pts = pts;
     return ret;
 }
 
+//该算法核心思想是对pts进行排序，排序好的pts就是dts。
+//排序有一定的滞后性，那么需要加上排序导致的时间戳偏移量
 bool DtsGenerator::getDts_l(uint32_t pts, uint32_t &dts){
     if(_sorter_max_size == 1){
-        //没有B帧
+        //没有B帧，dts就等于pts
         dts = pts;
         return true;
     }
 
     if(!_sorter_max_size){
+        //尚未计算出pts排序列队长度(也就是P帧间B帧个数)
         if(pts > _last_max_pts){
+            //pts时间戳增加了，那么说明这帧画面不是B帧(说明是P帧或关键帧)
             if(_frames_since_last_max_pts && _count_sorter_max_size++ > 0){
+                //已经出现多次非B帧的情况，那么我们就能知道P帧间B帧的个数
                 _sorter_max_size = _frames_since_last_max_pts;
+                //我们记录P帧间时间间隔(也就是多个B帧时间戳增量累计)
                 _dts_pts_offset = (pts - _last_max_pts) / 2;
             }
+            //遇到P帧或关键帧，连续B帧计数清零
             _frames_since_last_max_pts = 0;
+            //记录上次非B帧的pts时间戳(同时也是dts)，用于统计连续B帧时间戳增量
             _last_max_pts = pts;
         }
+        //如果pts时间戳小于上一个P帧，那么断定这个是B帧,我们记录B帧连续个数
         ++_frames_since_last_max_pts;
     }
 
+    //pts放入排序缓存列队，缓存列队最大等于连续B帧个数
     _pts_sorter.emplace(pts);
+
     if(_sorter_max_size && _pts_sorter.size() >  _sorter_max_size){
+        //如果启用了pts排序(意味着存在B帧)，并且pts排序缓存列队长度大于连续B帧个数，
+        //意味着后续的pts都会比最早的pts大，那么说明可以取出最早的pts了，这个pts将当做该帧的dts基准
         auto it = _pts_sorter.begin();
+
+        //由于该pts是前面偏移了个_sorter_max_size帧的pts(也就是那帧画面的dts),
+        //那么我们加上时间戳偏移量，基本等于该帧的dts
         dts = *it + _dts_pts_offset;
         if(dts > pts){
             //dts不能大于pts(基本不可能到达这个逻辑)
             dts = pts;
         }
+
+        //pts排序缓存出列
         _pts_sorter.erase(it);
         return true;
     }
+
+    //排序缓存尚未满
     return false;
 }
 
diff --git a/src/Common/config.cpp b/src/Common/config.cpp
index 819901fb..8a22bea1 100644
--- a/src/Common/config.cpp
+++ b/src/Common/config.cpp
@@ -61,7 +61,6 @@ const string kFlowThreshold = GENERAL_FIELD"flowThreshold";
 const string kStreamNoneReaderDelayMS = GENERAL_FIELD"streamNoneReaderDelayMS";
 const string kMaxStreamWaitTimeMS = GENERAL_FIELD"maxStreamWaitMS";
 const string kEnableVhost = GENERAL_FIELD"enableVhost";
-const string kUltraLowDelay = GENERAL_FIELD"ultraLowDelay";
 const string kAddMuteAudio = GENERAL_FIELD"addMuteAudio";
 const string kResetWhenRePlay = GENERAL_FIELD"resetWhenRePlay";
 const string kPublishToRtxp = GENERAL_FIELD"publishToRtxp";
@@ -74,13 +73,12 @@ onceToken token([](){
     mINI::Instance()[kStreamNoneReaderDelayMS] = 20 * 1000;
     mINI::Instance()[kMaxStreamWaitTimeMS] = 15 * 1000;
     mINI::Instance()[kEnableVhost] = 0;
-    mINI::Instance()[kUltraLowDelay] = 1;
     mINI::Instance()[kAddMuteAudio] = 1;
     mINI::Instance()[kResetWhenRePlay] = 1;
     mINI::Instance()[kPublishToRtxp] = 1;
     mINI::Instance()[kPublishToHls] = 1;
     mINI::Instance()[kPublishToMP4] = 0;
-    mINI::Instance()[kMergeWriteMS] = 300;
+    mINI::Instance()[kMergeWriteMS] = 0;
 },nullptr);
 
 }//namespace General
diff --git a/src/Common/config.h b/src/Common/config.h
index aa42acaa..3948879f 100644
--- a/src/Common/config.h
+++ b/src/Common/config.h
@@ -160,8 +160,6 @@ extern const string kStreamNoneReaderDelayMS;
 extern const string kMaxStreamWaitTimeMS;
 //是否启动虚拟主机
 extern const string kEnableVhost;
-//超低延时模式，默认打开，打开后会降低延时但是转发性能会稍差
-extern const string kUltraLowDelay;
 //拉流代理时是否添加静音音频
 extern const string kAddMuteAudio;
 //拉流代理时如果断流再重连成功是否删除前一次的媒体流数据，如果删除将重新开始，
@@ -174,7 +172,7 @@ extern const string kPublishToHls ;
 //是否默认推流时mp4录像，hook接口(on_publish)中可以覆盖该设置
 extern const string kPublishToMP4 ;
 //合并写缓存大小(单位毫秒)，合并写指服务器缓存一定的数据后才会一次性写入socket，这样能提高性能，但是会提高延时
-//在开启低延时模式后，该参数不起作用
+//开启后会同时关闭TCP_NODELAY并开启MSG_MORE
 extern const string kMergeWriteMS ;
 }//namespace General
 
diff --git a/src/Extension/H264.cpp b/src/Extension/H264.cpp
index d1f66f9a..7cfb086a 100644
--- a/src/Extension/H264.cpp
+++ b/src/Extension/H264.cpp
@@ -50,6 +50,9 @@ void splitH264(const char *ptr, int len, const std::function<void(const char *,
     while(true) {
         auto next_nal = memfind(nal + 3,end - nal - 3,"\x0\x0\x1",3);
         if(next_nal){
+            if(*(next_nal - 1) == 0x00){
+                next_nal -= 1;
+            }
             cb(nal,next_nal - nal);
             nal = next_nal;
             continue;
@@ -59,6 +62,18 @@ void splitH264(const char *ptr, int len, const std::function<void(const char *,
     }
 }
 
+#if 0
+//splitH264函数测试程序
+static onceToken s_token([](){
+    char buf[] = "\x00\x00\x00\x01\x12\x23\x34\x45\x56"
+                 "\x00\x00\x00\x01\x12\x23\x34\x45\x56"
+                 "\x00\x00\x00\x01\x12\x23\x34\x45\x56"
+                 "\x00\x00\x01\x12\x23\x34\x45\x56";
+    splitH264(buf, sizeof(buf) - 1, [](const char *ptr, int len){
+        cout << hexdump(ptr, len) << endl;
+    });
+});
+#endif //0
 
 Sdp::Ptr H264Track::getSdp() {
     if(!ready()){
diff --git a/src/Extension/H264Rtp.cpp b/src/Extension/H264Rtp.cpp
index 1a3f08a8..d9b4b9f6 100644
--- a/src/Extension/H264Rtp.cpp
+++ b/src/Extension/H264Rtp.cpp
@@ -191,19 +191,10 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
 }
 
 void H264RtpDecoder::onGetH264(const H264Frame::Ptr &frame) {
-    auto flag = _dts_generator.getDts(frame->_pts,frame->_dts);
-    if(!flag){
-        if(frame->configFrame() || frame->keyFrame()){
-            flag = true;
-            frame->_dts = frame->_pts;
-        }
-    }
-
-    //根据pts计算dts
-    if(flag){
-        //写入环形缓存
-        RtpCodec::inputFrame(frame);
-    }
+    //rtsp没有dts，那么根据pts排序算法生成dts
+    _dts_generator.getDts(frame->_pts,frame->_dts);
+    //写入环形缓存
+    RtpCodec::inputFrame(frame);
     _h264frame = obtainFrame();
 }
 
diff --git a/src/Extension/H265.h b/src/Extension/H265.h
index 71069d11..fca162c3 100644
--- a/src/Extension/H265.h
+++ b/src/Extension/H265.h
@@ -46,6 +46,9 @@ public:
         NAL_IDR_W_RADL = 19,
         NAL_IDR_N_LP = 20,
         NAL_CRA_NUT = 21,
+        NAL_RSV_IRAP_VCL22 = 22,
+        NAL_RSV_IRAP_VCL23 = 23,
+
         NAL_VPS = 32,
         NAL_SPS = 33,
         NAL_PPS = 34,
@@ -101,17 +104,7 @@ public:
     }
 
     static bool isKeyFrame(int type) {
-        switch (type) {
-            case NAL_BLA_N_LP:
-            case NAL_BLA_W_LP:
-            case NAL_BLA_W_RADL:
-            case NAL_CRA_NUT:
-            case NAL_IDR_N_LP:
-            case NAL_IDR_W_RADL:
-                return true;
-            default:
-                return false;
-        }
+        return type >= NAL_BLA_W_LP && type <= NAL_RSV_IRAP_VCL23;
     }
 
 public:
diff --git a/src/Extension/H265Rtp.cpp b/src/Extension/H265Rtp.cpp
index b41b4cfa..daa190a0 100644
--- a/src/Extension/H265Rtp.cpp
+++ b/src/Extension/H265Rtp.cpp
@@ -127,18 +127,10 @@ bool H265RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
 }
 
 void H265RtpDecoder::onGetH265(const H265Frame::Ptr &frame) {
-    //计算dts
-    auto flag = _dts_generator.getDts(frame->_pts,frame->_dts);
-    if(!flag){
-        if(frame->configFrame() || frame->keyFrame()){
-            flag = true;
-            frame->_dts = frame->_pts;
-        }
-    }
-    if(flag){
-        //写入环形缓存
-        RtpCodec::inputFrame(frame);
-    }
+    //rtsp没有dts，那么根据pts排序算法生成dts
+    _dts_generator.getDts(frame->_pts,frame->_dts);
+    //写入环形缓存
+    RtpCodec::inputFrame(frame);
     _h265frame = obtainFrame();
 }
 
diff --git a/src/Http/HttpSession.cpp b/src/Http/HttpSession.cpp
index 26056ed0..65ed9b3a 100644
--- a/src/Http/HttpSession.cpp
+++ b/src/Http/HttpSession.cpp
@@ -606,8 +606,8 @@ void HttpSession::sendNotFound(bool bClose) {
 }
 
 void HttpSession::setSocketFlags(){
-    GET_CONFIG(bool,ultraLowDelay,General::kUltraLowDelay);
-    if(!ultraLowDelay) {
+    GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
+    if(mergeWriteMS > 0) {
         //推流模式下，关闭TCP_NODELAY会增加推流端的延时，但是服务器性能将提高
         SockUtil::setNoDelay(_sock->rawFD(), false);
         //播放模式下，开启MSG_MORE会增加延时，但是能提高发送性能
diff --git a/src/Record/Recorder.cpp b/src/Record/Recorder.cpp
index 38d6b98a..2ab341b7 100644
--- a/src/Record/Recorder.cpp
+++ b/src/Record/Recorder.cpp
@@ -56,7 +56,7 @@ string Recorder::getRecordPath(Recorder::type type, const string &vhost, const s
 }
 
 std::shared_ptr<MediaSinkInterface> Recorder::createRecorder(type type, const string &vhost, const string &app, const string &stream_id, const string &customized_path){
-    auto path = Recorder::getRecordPath(type, vhost, app, stream_id);
+    auto path = Recorder::getRecordPath(type, vhost, app, stream_id, customized_path);
     switch (type) {
         case Recorder::type_hls: {
 #if defined(ENABLE_HLS)
diff --git a/src/Rtmp/RtmpMediaSourceImp.h b/src/Rtmp/RtmpMediaSourceImp.h
index 954180b9..e78f1286 100644
--- a/src/Rtmp/RtmpMediaSourceImp.h
+++ b/src/Rtmp/RtmpMediaSourceImp.h
@@ -57,7 +57,14 @@ public:
      * 输入rtmp并解析
      */
     void onWrite(const RtmpPacket::Ptr &pkt,bool key_pos = true) override {
-        key_pos = _demuxer->inputRtmp(pkt);
+        if(_all_track_ready && !_muxer->isEnabled()){
+            //获取到所有Track后，并且未开启转协议，那么不需要解复用rtmp
+            key_pos = pkt->isVideoKeyFrame();
+        }else{
+            //需要解复用rtmp
+            key_pos = _demuxer->inputRtmp(pkt);
+        }
+
         RtmpMediaSource::onWrite(pkt,key_pos);
     }
 
@@ -138,10 +145,12 @@ public:
      */
     void onAllTrackReady() override{
         setTrackSource(_muxer);
+        _all_track_ready = true;
     }
 private:
     RtmpDemuxer::Ptr _demuxer;
     MultiMediaSourceMuxer::Ptr _muxer;
+    bool _all_track_ready = false;
 };
 } /* namespace mediakit */
 
diff --git a/src/Rtmp/RtmpPusher.cpp b/src/Rtmp/RtmpPusher.cpp
index 3722e23d..725868b5 100644
--- a/src/Rtmp/RtmpPusher.cpp
+++ b/src/Rtmp/RtmpPusher.cpp
@@ -228,8 +228,8 @@ inline void RtmpPusher::send_metaData(){
 }
 
 void RtmpPusher::setSocketFlags(){
-    GET_CONFIG(bool,ultraLowDelay,General::kUltraLowDelay);
-    if(!ultraLowDelay) {
+    GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
+    if(mergeWriteMS > 0) {
         //提高发送性能
         setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
         SockUtil::setNoDelay(_sock->rawFD(), false);
diff --git a/src/Rtmp/RtmpSession.cpp b/src/Rtmp/RtmpSession.cpp
index d606f118..a062c5d6 100644
--- a/src/Rtmp/RtmpSession.cpp
+++ b/src/Rtmp/RtmpSession.cpp
@@ -529,8 +529,8 @@ int RtmpSession::totalReaderCount(MediaSource &sender) {
 }
 
 void RtmpSession::setSocketFlags(){
-    GET_CONFIG(bool,ultraLowDelay,General::kUltraLowDelay);
-    if(!ultraLowDelay) {
+    GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
+    if(mergeWriteMS > 0) {
         //推流模式下，关闭TCP_NODELAY会增加推流端的延时，但是服务器性能将提高
         SockUtil::setNoDelay(_sock->rawFD(), false);
         //播放模式下，开启MSG_MORE会增加延时，但是能提高发送性能
diff --git a/src/Rtmp/RtmpSession.h b/src/Rtmp/RtmpSession.h
index da76c627..f88d2eb6 100644
--- a/src/Rtmp/RtmpSession.h
+++ b/src/Rtmp/RtmpSession.h
@@ -91,6 +91,12 @@ private:
 
 };
 
+
+/**
+ * 支持ssl加密的rtmp服务器
+ */
+typedef TcpSessionWithSSL<RtmpSession> RtmpSessionWithSSL;
+
 } /* namespace mediakit */
 
 #endif /* SRC_RTMP_RTMPSESSION_H_ */
diff --git a/src/Rtsp/RtspMediaSourceImp.h b/src/Rtsp/RtspMediaSourceImp.h
index 495158f3..91a97992 100644
--- a/src/Rtsp/RtspMediaSourceImp.h
+++ b/src/Rtsp/RtspMediaSourceImp.h
@@ -48,7 +48,14 @@ public:
      * 输入rtp并解析
      */
     void onWrite(const RtpPacket::Ptr &rtp, bool key_pos) override {
-        key_pos = _demuxer->inputRtp(rtp);
+        if(_all_track_ready && !_muxer->isEnabled()){
+            //获取到所有Track后，并且未开启转协议，那么不需要解复用rtp
+            //在关闭rtp解复用后，无法知道是否为关键帧，这样会导致无法秒开，或者开播花屏
+            key_pos = rtp->type == TrackVideo;
+        }else{
+            //需要解复用rtp
+            key_pos = _demuxer->inputRtp(rtp);
+        }
         RtspMediaSource::onWrite(rtp, key_pos);
     }
 
@@ -129,10 +136,12 @@ public:
      */
     void onAllTrackReady() override{
         setTrackSource(_muxer);
+        _all_track_ready = true;
     }
 private:
     RtspDemuxer::Ptr _demuxer;
     MultiMediaSourceMuxer::Ptr _muxer;
+    bool _all_track_ready = false;
 };
 } /* namespace mediakit */
 
diff --git a/src/Rtsp/RtspPusher.cpp b/src/Rtsp/RtspPusher.cpp
index 5243be4c..f259a0d6 100644
--- a/src/Rtsp/RtspPusher.cpp
+++ b/src/Rtsp/RtspPusher.cpp
@@ -392,8 +392,8 @@ void RtspPusher::sendRecord() {
 }
 
 void RtspPusher::setSocketFlags(){
-    GET_CONFIG(bool,ultraLowDelay,General::kUltraLowDelay);
-    if(!ultraLowDelay) {
+    GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
+    if(mergeWriteMS > 0) {
         //提高发送性能
         setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
         SockUtil::setNoDelay(_sock->rawFD(), false);
diff --git a/src/Rtsp/RtspSession.cpp b/src/Rtsp/RtspSession.cpp
index f8591e6c..a9cbcffb 100644
--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
@@ -1236,8 +1236,8 @@ void RtspSession::sendSenderReport(bool overTcp,int iTrackIndex) {
 }
 
 void RtspSession::setSocketFlags(){
-    GET_CONFIG(bool,ultraLowDelay,General::kUltraLowDelay);
-    if(!ultraLowDelay) {
+    GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
+    if(mergeWriteMS > 0) {
         //推流模式下，关闭TCP_NODELAY会增加推流端的延时，但是服务器性能将提高
         SockUtil::setNoDelay(_sock->rawFD(), false);
         //播放模式下，开启MSG_MORE会增加延时，但是能提高发送性能