From d6e11706eca9fc94f5751fe53aaa4521279b48c8 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?=E5=A4=8F=E7=AB=8B=E6=96=B0?= <xialx@yuntongxun.com>
Date: Mon, 17 Feb 2020 18:46:05 +0800
Subject: [PATCH 01/16] Fix disconnect RTSP connection has assertion, resulting
 in program exit

---
 trunk/src/app/srs_app_rtsp.cpp   | 20 ++++++++++++++++++--
 trunk/src/app/srs_app_rtsp.hpp   |  5 ++++-
 trunk/src/app/srs_app_server.cpp |  1 +
 trunk/src/app/srs_app_server.hpp |  3 ++-
 4 files changed, 25 insertions(+), 4 deletions(-)

diff --git a/trunk/src/app/srs_app_rtsp.cpp b/trunk/src/app/srs_app_rtsp.cpp
index 4488b6d98..2577f3128 100644
--- a/trunk/src/app/srs_app_rtsp.cpp
+++ b/trunk/src/app/srs_app_rtsp.cpp
@@ -240,6 +240,11 @@ srs_error_t SrsRtspConn::serve()
     return err;
 }
 
+std::string SrsRtspConn::remote_ip()
+{
+    return "";
+}
+
 srs_error_t SrsRtspConn::do_cycle()
 {
     srs_error_t err = srs_success;
@@ -681,6 +686,7 @@ SrsRtspCaster::SrsRtspCaster(SrsConfDirective* c)
     output = _srs_config->get_stream_caster_output(c);
     local_port_min = _srs_config->get_stream_caster_rtp_port_min(c);
     local_port_max = _srs_config->get_stream_caster_rtp_port_max(c);
+    manager = new SrsCoroutineManager();
 }
 
 SrsRtspCaster::~SrsRtspCaster()
@@ -688,10 +694,20 @@ SrsRtspCaster::~SrsRtspCaster()
     std::vector<SrsRtspConn*>::iterator it;
     for (it = clients.begin(); it != clients.end(); ++it) {
         SrsRtspConn* conn = *it;
-        srs_freep(conn);
+        manager->remove(conn);
     }
     clients.clear();
     used_ports.clear();
+
+    srs_freep(manager);
+}
+
+srs_error_t SrsRtspCaster::initialize()
+{
+    srs_error_t err = srs_success;
+    if ((err = manager->start()) != srs_success) {
+        return srs_error_wrap(err, "start manager");
+    }
 }
 
 srs_error_t SrsRtspCaster::alloc_port(int* pport)
@@ -744,6 +760,6 @@ void SrsRtspCaster::remove(SrsRtspConn* conn)
     }
     srs_info("rtsp: remove connection from caster.");
     
-    srs_freep(conn);
+    manager->remove(conn);
 }
 
diff --git a/trunk/src/app/srs_app_rtsp.hpp b/trunk/src/app/srs_app_rtsp.hpp
index 7bbd52651..0ca748535 100644
--- a/trunk/src/app/srs_app_rtsp.hpp
+++ b/trunk/src/app/srs_app_rtsp.hpp
@@ -100,7 +100,7 @@ public:
 };
 
 // The rtsp connection serve the fd.
-class SrsRtspConn : public ISrsCoroutineHandler
+class SrsRtspConn : public ISrsCoroutineHandler, public ISrsConnection
 {
 private:
     std::string output_template;
@@ -143,6 +143,7 @@ public:
     virtual ~SrsRtspConn();
 public:
     virtual srs_error_t serve();
+    virtual std::string remote_ip();
 private:
     virtual srs_error_t do_cycle();
 // internal methods
@@ -179,6 +180,7 @@ private:
     std::map<int, bool> used_ports;
 private:
     std::vector<SrsRtspConn*> clients;
+    SrsCoroutineManager* manager;
 public:
     SrsRtspCaster(SrsConfDirective* c);
     virtual ~SrsRtspCaster();
@@ -188,6 +190,7 @@ public:
     virtual srs_error_t alloc_port(int* pport);
     // Free the alloced rtp port.
     virtual void free_port(int lpmin, int lpmax);
+    virtual srs_error_t initialize();
 // Interface ISrsTcpHandler
 public:
     virtual srs_error_t on_tcp_client(srs_netfd_t stfd);
diff --git a/trunk/src/app/srs_app_server.cpp b/trunk/src/app/srs_app_server.cpp
index 61eeb2309..2edc525aa 100644
--- a/trunk/src/app/srs_app_server.cpp
+++ b/trunk/src/app/srs_app_server.cpp
@@ -179,6 +179,7 @@ SrsRtspListener::SrsRtspListener(SrsServer* svr, SrsListenerType t, SrsConfDirec
     srs_assert(type == SrsListenerRtsp);
     if (type == SrsListenerRtsp) {
         caster = new SrsRtspCaster(c);
+        caster->initialize();
     }
 }
 
diff --git a/trunk/src/app/srs_app_server.hpp b/trunk/src/app/srs_app_server.hpp
index fd8cbe3ad..f43d7fc04 100644
--- a/trunk/src/app/srs_app_server.hpp
+++ b/trunk/src/app/srs_app_server.hpp
@@ -50,6 +50,7 @@ class ISrsUdpHandler;
 class SrsUdpListener;
 class SrsTcpListener;
 class SrsAppCasterFlv;
+class SrsRtspCaster;
 class SrsCoroutineManager;
 
 // The listener type for server to identify the connection,
@@ -107,7 +108,7 @@ class SrsRtspListener : virtual public SrsListener, virtual public ISrsTcpHandle
 {
 private:
     SrsTcpListener* listener;
-    ISrsTcpHandler* caster;
+    SrsRtspCaster* caster;
 public:
     SrsRtspListener(SrsServer* svr, SrsListenerType t, SrsConfDirective* c);
     virtual ~SrsRtspListener();

From bbdbcf173cbe983817424782d8dcd4ad96f64c70 Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Wed, 26 Feb 2020 09:06:53 +0800
Subject: [PATCH 02/16] For #1615, refactor the version and API for SRT.

---
 trunk/src/srt/srt_server.cpp | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/trunk/src/srt/srt_server.cpp b/trunk/src/srt/srt_server.cpp
index eb41866e0..7576d3bf5 100644
--- a/trunk/src/srt/srt_server.cpp
+++ b/trunk/src/srt/srt_server.cpp
@@ -257,8 +257,9 @@ void srt_server::on_work()
         }
     }
 
-    // @see 2020-01-28 https://github.com/Haivision/srt/commit/b8c70ec801a56bea151ecce9c09c4ebb720c2f68#diff-fb66028e8746fea578788532533a296bR786
-#if SRT_VERSION_MAJOR > 1 || SRT_VERSION_MINOR > 4 || SRT_VERSION_PATCH > 1
+    // New API at 2020-01-28, >1.4.1
+    // @see https://github.com/Haivision/srt/commit/b8c70ec801a56bea151ecce9c09c4ebb720c2f68#diff-fb66028e8746fea578788532533a296bR786
+#if (SRT_VERSION_MAJOR<<24 | SRT_VERSION_MINOR<<16 | SRT_VERSION_PATCH<<8) > 0x01040100
     srt_epoll_clear_usocks(_pollid);
 #endif
 }

From 4bd37d43b464a39a3a5fc5fb62a174b4870a4745 Mon Sep 17 00:00:00 2001
From: runner365 <shi.weibd@hotmail.com>
Date: Wed, 26 Feb 2020 11:14:05 +0800
Subject: [PATCH 03/16] solve dts==0 bugs; solve large aac 2930bytes timestamp
 bugs

---
 output.txt                    |  0
 trunk/conf/srt.conf           |  4 +++
 trunk/src/srt/srt_to_rtmp.cpp | 66 ++++++++++++++++++++++++++++-------
 trunk/src/srt/srt_to_rtmp.hpp |  2 ++
 trunk/src/srt/ts_demux.cpp    |  1 -
 5 files changed, 59 insertions(+), 14 deletions(-)
 create mode 100644 output.txt

diff --git a/output.txt b/output.txt
new file mode 100644
index 000000000..e69de29bb
diff --git a/trunk/conf/srt.conf b/trunk/conf/srt.conf
index b9fcc9d32..2aa29aa36 100644
--- a/trunk/conf/srt.conf
+++ b/trunk/conf/srt.conf
@@ -27,6 +27,10 @@ srt_server {
 
 # @doc https://github.com/ossrs/srs/issues/1147#issuecomment-577607026
 vhost __defaultVhost__ {
+    play {
+        #atc                     on;
+        mix_correct             on;
+    }
 }
 vhost srs.srt.com.cn {
 }
diff --git a/trunk/src/srt/srt_to_rtmp.cpp b/trunk/src/srt/srt_to_rtmp.cpp
index 13e83056d..71a903b94 100644
--- a/trunk/src/srt/srt_to_rtmp.cpp
+++ b/trunk/src/srt/srt_to_rtmp.cpp
@@ -381,7 +381,7 @@ srs_error_t rtmp_client::write_audio_raw_frame(char* frame, int frame_size, SrsR
     if ((err = _aac_ptr->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != srs_success) {
         return srs_error_wrap(err, "mux aac to flv");
     }
-    
+
     return rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
 }
 
@@ -410,7 +410,12 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
     if ((err = connect()) != srs_success) {
         return srs_error_wrap(err, "connect");
     }
-    
+    dts = dts / 90;
+    pts = pts / 90;
+
+    if (dts == 0) {
+        dts = pts;
+    }
     // send each frame.
     while (!avs_ptr->empty()) {
         char* frame = NULL;
@@ -480,14 +485,46 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
     return err;
 }
 
+int rtmp_client::get_sample_rate(char sound_rate) {
+    int sample_rate = 44100;
+
+    switch (sound_rate)
+    {
+    case SrsAudioSampleRate44100:
+        sample_rate = 44100;
+        break;
+    case SrsAudioSampleRate22050:
+        sample_rate = 22050;
+        break;
+    case SrsAudioSampleRate11025:
+        sample_rate = 11025;
+        break;
+    case SrsAudioSampleRate5512:
+        sample_rate = 5512;
+        break;
+    default:
+        break;
+    }
+    return sample_rate;
+}
+
 srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
     srs_error_t err = srs_success;
+    uint64_t last_dts;
+    uint64_t real_dts;
+    int index = 0;
+    int sample_size = 1024;
 
     // ensure rtmp connected.
     if ((err = connect()) != srs_success) {
         return srs_error_wrap(err, "connect");
     }
     
+    last_dts = dts/90;
+    if (last_dts == 0) {
+        last_dts = pts/90;
+    }
+
     // send each frame.
     while (!avs_ptr->empty()) {
         char* frame = NULL;
@@ -496,16 +533,21 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
         if ((err = _aac_ptr->adts_demux(avs_ptr.get(), &frame, &frame_size, codec)) != srs_success) {
             return srs_error_wrap(err, "demux adts");
         }
-        //srs_trace("audio annexb demux sampling_frequency_index:%d, aac_packet_type:%d, sound_rate:%d, sound_size:%d", 
-        //    codec.sampling_frequency_index, codec.aac_packet_type, codec.sound_rate,
-        //    codec.sound_size);
-        //srs_trace_data(frame, frame_size, "audio annexb demux:");
-        // ignore invalid frame,
-        //  * atleast 1bytes for aac to decode the data.
+
         if (frame_size <= 0) {
             continue;
         }
-        
+        int sample_rate = get_sample_rate(codec.sound_rate);
+
+        if (codec.aac_packet_type > SrsAudioOpusFrameTraitRaw) {
+            sample_size = 2048;
+        } else {
+            sample_size = 1024;
+        }
+
+        real_dts = last_dts + index * 1000.0 * sample_size / sample_rate;
+        index++;
+
         // generate sh.
         if (_aac_specific_config.empty()) {
             std::string sh;
@@ -516,14 +558,14 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
             
             codec.aac_packet_type = 0;
             
-            if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, dts)) != srs_success) {
+            if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, real_dts)) != srs_success) {
                 return srs_error_wrap(err, "write raw audio frame");
             }
         }
         
         // audio raw data.
         codec.aac_packet_type = 1;
-        if ((err = write_audio_raw_frame(frame, frame_size, &codec, dts)) != srs_success) {
+        if ((err = write_audio_raw_frame(frame, frame_size, &codec, real_dts)) != srs_success) {
             return srs_error_wrap(err, "write audio raw frame");
         }
         _last_live_ts = now_ms();
@@ -541,8 +583,6 @@ void rtmp_client::on_data_callback(SRT_DATA_MSG_PTR data_ptr, unsigned int media
     }
 
     auto avs_ptr = std::make_shared<SrsBuffer>((char*)data_ptr->get_data(), data_ptr->data_len());
-    dts = dts / 90;
-    pts = pts / 90;
 
     if (media_type == STREAM_TYPE_VIDEO_H264) {
         on_ts_video(avs_ptr, dts, pts);
diff --git a/trunk/src/srt/srt_to_rtmp.hpp b/trunk/src/srt/srt_to_rtmp.hpp
index 08735c09d..3a3dea3d9 100644
--- a/trunk/src/srt/srt_to_rtmp.hpp
+++ b/trunk/src/srt/srt_to_rtmp.hpp
@@ -46,6 +46,8 @@ private:
     virtual srs_error_t write_h264_ipb_frame(char* frame, int frame_size, uint32_t dts, uint32_t pts);
     virtual srs_error_t write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts);
 
+    int get_sample_rate(char sound_rate);
+
 private:
     virtual srs_error_t rtmp_write_packet(char type, uint32_t timestamp, char* data, int size);
 
diff --git a/trunk/src/srt/ts_demux.cpp b/trunk/src/srt/ts_demux.cpp
index 13cee93ec..3eed4f74f 100644
--- a/trunk/src/srt/ts_demux.cpp
+++ b/trunk/src/srt/ts_demux.cpp
@@ -291,7 +291,6 @@ int ts_demux::decode(SRT_DATA_MSG_PTR data_ptr, TS_DATA_CALLBACK_PTR callback)
         return -1;
     }
 
-
     unsigned int count = data_ptr->data_len()/188;
     path = data_ptr->get_path();
     for (unsigned int index = 0; index < count; index++)

From 9e90870779f3488e2370fbdf7d8c2bcbc3962624 Mon Sep 17 00:00:00 2001
From: runner365 <shi.weibd@hotmail.com>
Date: Wed, 26 Feb 2020 11:14:34 +0800
Subject: [PATCH 04/16] solve dts==0 bugs; solve large aac 2930bytes timestamp
 bugs

---
 output.txt | 0
 1 file changed, 0 insertions(+), 0 deletions(-)
 delete mode 100644 output.txt

diff --git a/output.txt b/output.txt
deleted file mode 100644
index e69de29bb..000000000

From 7f4cc6221c232d2b790a015edb75d8602cd3d5eb Mon Sep 17 00:00:00 2001
From: runner365 <shi.weibd@hotmail.com>
Date: Sat, 29 Feb 2020 16:47:20 +0800
Subject: [PATCH 05/16] 1,solve audio timestamp bugs;2,solve video dts increase
 bugs

---
 trunk/conf/srt.conf              |   4 -
 trunk/src/app/srs_app_config.cpp |  16 +++-
 trunk/src/app/srs_app_config.hpp |   2 +
 trunk/src/srt/srt_handle.cpp     |   1 +
 trunk/src/srt/srt_to_rtmp.cpp    | 141 +++++++++++++++++++++++++++----
 trunk/src/srt/srt_to_rtmp.hpp    |  37 ++++++++
 trunk/src/srt/ts_demux.cpp       |   8 +-
 7 files changed, 186 insertions(+), 23 deletions(-)

diff --git a/trunk/conf/srt.conf b/trunk/conf/srt.conf
index 2aa29aa36..b9fcc9d32 100644
--- a/trunk/conf/srt.conf
+++ b/trunk/conf/srt.conf
@@ -27,10 +27,6 @@ srt_server {
 
 # @doc https://github.com/ossrs/srs/issues/1147#issuecomment-577607026
 vhost __defaultVhost__ {
-    play {
-        #atc                     on;
-        mix_correct             on;
-    }
 }
 vhost srs.srt.com.cn {
 }
diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index 0eaabb4c9..627a01580 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -3530,7 +3530,7 @@ srs_error_t SrsConfig::check_normal_config()
                 && n != "mss" && n != "latency" && n != "recvlatency"
                 && n != "peerlatency" && n != "tlpkdrop" && n != "connect_timeout"
                 && n != "sendbuf" && n != "recvbuf" && n != "payloadsize"
-                && n != "default_app") {
+                && n != "default_app" && n != "mix_correct") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal srt_stream.%s", n.c_str());
             }
         }
@@ -6754,6 +6754,20 @@ unsigned short SrsConfig::get_srt_listen_port()
     return (unsigned short)atoi(conf->arg0().c_str());
 }
 
+bool SrsConfig::get_srt_mix_correct() {
+    static bool DEFAULT = true;
+    SrsConfDirective* conf = root->get("srt_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("mix_correct");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
 int SrsConfig::get_srto_maxbw() {
     static int64_t DEFAULT = -1;
     SrsConfDirective* conf = root->get("srt_server");
diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
index dde708f8e..4360c41da 100644
--- a/trunk/src/app/srs_app_config.hpp
+++ b/trunk/src/app/srs_app_config.hpp
@@ -625,6 +625,8 @@ public:
     virtual int get_srto_payloadsize();
     // Get the default app.
     virtual std::string get_default_app_name();
+    // Get the mix_correct
+    virtual bool get_srt_mix_correct();
 
 // http_hooks section
 private:
diff --git a/trunk/src/srt/srt_handle.cpp b/trunk/src/srt/srt_handle.cpp
index 4592785e3..634376275 100644
--- a/trunk/src/srt/srt_handle.cpp
+++ b/trunk/src/srt/srt_handle.cpp
@@ -145,6 +145,7 @@ void srt_handle::add_newconn(SRT_CONN_PTR conn_ptr, int events) {
     srs_trace("srto SRTO_RCVBUF=%d", val_i);
     srt_getsockopt(conn_ptr->get_conn(), 0, SRTO_MAXBW, &val_i, &opt_len);
     srs_trace("srto SRTO_MAXBW=%d", val_i);
+    srs_trace("srt mix_correct is %s", _srs_config->get_srt_mix_correct() ? "enable" : "disable");
 
     if (conn_ptr->get_mode() == PULL_SRT_MODE) {
         add_new_puller(conn_ptr, conn_ptr->get_subpath());
diff --git a/trunk/src/srt/srt_to_rtmp.cpp b/trunk/src/srt/srt_to_rtmp.cpp
index 71a903b94..ec2d7d828 100644
--- a/trunk/src/srt/srt_to_rtmp.cpp
+++ b/trunk/src/srt/srt_to_rtmp.cpp
@@ -322,12 +322,13 @@ srs_error_t rtmp_client::write_h264_sps_pps(uint32_t dts, uint32_t pts) {
         return srs_error_wrap(err, "avc to flv");
     }
     
-    // the timestamp in rtmp message header is dts.
-    uint32_t timestamp = dts;
-    if ((err = rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv)) != srs_success) {
-        return srs_error_wrap(err, "write packet");
+    if (_srs_config->get_srt_mix_correct()) {
+        _rtmp_queue.insert_rtmp_data((unsigned char*)flv, nb_flv, (int64_t)dts, SrsFrameTypeVideo);
+        rtmp_write_work();
+    } else {
+        rtmp_write_packet(SrsFrameTypeVideo, dts, flv, nb_flv);
     }
-    
+
     // reset sps and pps.
     _h264_sps_changed = false;
     _h264_pps_changed = false;
@@ -367,10 +368,14 @@ srs_error_t rtmp_client::write_h264_ipb_frame(char* frame, int frame_size, uint3
     if ((err = _avc_ptr->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
         return srs_error_wrap(err, "mux avc to flv");
     }
-    
-    // the timestamp in rtmp message header is dts.
-    uint32_t timestamp = dts;
-    return rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv);
+    if (_srs_config->get_srt_mix_correct()) {
+        _rtmp_queue.insert_rtmp_data((unsigned char*)flv, nb_flv, (int64_t)dts, SrsFrameTypeVideo);
+        rtmp_write_work();
+    } else {
+        rtmp_write_packet(SrsFrameTypeVideo, dts, flv, nb_flv);
+    }
+
+    return err;
 }
 
 srs_error_t rtmp_client::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts) {
@@ -381,14 +386,20 @@ srs_error_t rtmp_client::write_audio_raw_frame(char* frame, int frame_size, SrsR
     if ((err = _aac_ptr->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != srs_success) {
         return srs_error_wrap(err, "mux aac to flv");
     }
+    if (_srs_config->get_srt_mix_correct()) {
+        _rtmp_queue.insert_rtmp_data((unsigned char*)data, size, (int64_t)dts, SrsFrameTypeAudio);
+        rtmp_write_work();
+    } else {
+        rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
+    }
 
-    return rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
+    return err;
 }
 
 srs_error_t rtmp_client::rtmp_write_packet(char type, uint32_t timestamp, char* data, int size) {
     srs_error_t err = srs_success;
     SrsSharedPtrMessage* msg = NULL;
-    
+
     if ((err = srs_rtmp_create_msg(type, timestamp, data, size, _rtmp_conn_ptr->sid(), &msg)) != srs_success) {
         return srs_error_wrap(err, "create message");
     }
@@ -403,6 +414,19 @@ srs_error_t rtmp_client::rtmp_write_packet(char type, uint32_t timestamp, char*
     return err;
 }
 
+void rtmp_client::rtmp_write_work() {
+    rtmp_packet_info_s packet_info;
+    bool ret = false;
+    
+    do {
+        ret = _rtmp_queue.get_rtmp_data(packet_info);
+        if (ret) {
+            rtmp_write_packet(packet_info._type, packet_info._dts, (char*)packet_info._data, packet_info._len);
+        }
+    } while(ret);
+    return;
+}
+
 srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
     srs_error_t err = srs_success;
 
@@ -416,6 +440,7 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
     if (dts == 0) {
         dts = pts;
     }
+
     // send each frame.
     while (!avs_ptr->empty()) {
         char* frame = NULL;
@@ -510,8 +535,9 @@ int rtmp_client::get_sample_rate(char sound_rate) {
 
 srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
     srs_error_t err = srs_success;
-    uint64_t last_dts;
+    uint64_t base_dts;
     uint64_t real_dts;
+    uint64_t first_dts;
     int index = 0;
     int sample_size = 1024;
 
@@ -520,11 +546,11 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
         return srs_error_wrap(err, "connect");
     }
     
-    last_dts = dts/90;
-    if (last_dts == 0) {
-        last_dts = pts/90;
+    base_dts = dts/90;
+    if (base_dts == 0) {
+        base_dts = pts/90;
     }
-
+    
     // send each frame.
     while (!avs_ptr->empty()) {
         char* frame = NULL;
@@ -545,7 +571,10 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
             sample_size = 1024;
         }
 
-        real_dts = last_dts + index * 1000.0 * sample_size / sample_rate;
+        real_dts = base_dts + index * 1000.0 * sample_size / sample_rate;
+        if (index == 0) {
+            first_dts = real_dts;
+        }
         index++;
 
         // generate sh.
@@ -571,6 +600,12 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
         _last_live_ts = now_ms();
     }
     
+    uint64_t diff_t = real_dts - first_dts;
+    diff_t += 100;
+    if ((diff_t > 200) && (diff_t < 600)) {
+        srs_info("set_queue_timeout timeout:%lu", diff_t);
+        _rtmp_queue.set_queue_timeout(diff_t);
+    }
     return err;
 }
 
@@ -594,3 +629,75 @@ void rtmp_client::on_data_callback(SRT_DATA_MSG_PTR data_ptr, unsigned int media
     }
     return;
 }
+
+rtmp_packet_queue::rtmp_packet_queue():_queue_timeout(QUEUE_DEF_TIMEOUT)
+    ,_queue_maxlen(QUEUE_LEN_MAX)
+    ,_first_packet_t(-1)
+    ,_first_local_t(-1) {
+
+}
+
+rtmp_packet_queue::~rtmp_packet_queue() {
+    for (auto item : _send_map) {
+        rtmp_packet_info_s info = item.second;
+        if (info._data) {
+            delete info._data;
+        }
+    }
+    _send_map.clear();
+}
+
+void rtmp_packet_queue::set_queue_timeout(int64_t queue_timeout) {
+    _queue_timeout = queue_timeout;
+}
+
+void rtmp_packet_queue::insert_rtmp_data(unsigned char* data, int len, int64_t dts, char media_type) {
+    rtmp_packet_info_s packet_info;
+
+    packet_info._data = data;
+    packet_info._len  = len;
+    packet_info._dts  = dts;
+    packet_info._type = media_type;
+
+    if (_first_packet_t == -1) {
+        _first_packet_t = dts;
+        _first_local_t = (int64_t)now_ms();
+    }
+
+    _send_map.insert(std::make_pair(dts, packet_info));
+    return;
+}
+
+bool rtmp_packet_queue::is_ready() {
+    if (!_srs_config->get_srt_mix_correct() && !_send_map.empty()) {
+        return true;
+    }
+    if (_send_map.size() < 2) {
+        return false;
+    }
+
+    if (_send_map.size() >= (size_t)_queue_maxlen) {
+        return true;
+    }
+
+    auto first_item = _send_map.begin();
+    int64_t now_t = (int64_t)now_ms();
+
+    int64_t diff_t = (now_t - _first_local_t) - (first_item->first - _first_packet_t);
+
+    if (diff_t >= _queue_timeout) {
+        return true;
+    }
+    return false;
+}
+
+bool rtmp_packet_queue::get_rtmp_data(rtmp_packet_info_s& packet_info) {
+    if (!is_ready()) {
+        return false;
+    }
+    auto iter = _send_map.begin();
+    packet_info = iter->second;
+    _send_map.erase(iter);
+
+    return true;
+}
\ No newline at end of file
diff --git a/trunk/src/srt/srt_to_rtmp.hpp b/trunk/src/srt/srt_to_rtmp.hpp
index 3a3dea3d9..5fcaf6acf 100644
--- a/trunk/src/srt/srt_to_rtmp.hpp
+++ b/trunk/src/srt/srt_to_rtmp.hpp
@@ -4,6 +4,7 @@
 #include <string>
 #include <thread>
 #include <queue>
+#include <map>
 #include <mutex>
 #include <condition_variable>
 #include <srs_kernel_ts.hpp>
@@ -24,6 +25,37 @@ typedef std::shared_ptr<SrsRawAacStream> AAC_PTR;
 
 #define DEFAULT_VHOST "__default_host__"
 
+#define QUEUE_DEF_TIMEOUT 500
+#define QUEUE_LEN_MAX     100
+
+typedef struct {
+    unsigned char* _data;
+    int _len;
+    int64_t _dts;
+    char _type;
+    char reserve[3];
+} rtmp_packet_info_s;
+
+class rtmp_packet_queue {
+public:
+    rtmp_packet_queue();
+    ~rtmp_packet_queue();
+
+    void set_queue_timeout(int64_t queue_timeout);
+    void insert_rtmp_data(unsigned char* data, int len, int64_t dts, char media_type);
+    bool get_rtmp_data(rtmp_packet_info_s& packet_info);
+
+private:
+    bool is_ready();
+
+private:
+    int64_t _queue_timeout;
+    int64_t _queue_maxlen;
+    int64_t _first_packet_t;
+    int64_t _first_local_t;
+    std::multimap<int64_t, rtmp_packet_info_s> _send_map;//key:dts, value:rtmp_packet_info
+};
+
 class rtmp_client : public ts_media_data_callback_I, public std::enable_shared_from_this<rtmp_client> {
 public:
     rtmp_client(std::string key_path);
@@ -48,6 +80,8 @@ private:
 
     int get_sample_rate(char sound_rate);
 
+    void rtmp_write_work();
+
 private:
     virtual srs_error_t rtmp_write_packet(char type, uint32_t timestamp, char* data, int size);
 
@@ -73,6 +107,9 @@ private:
     RTMP_CONN_PTR _rtmp_conn_ptr;
     bool _connect_flag;
     int64_t _last_live_ts;
+
+private:
+    rtmp_packet_queue _rtmp_queue;
 };
 
 typedef std::shared_ptr<rtmp_client> RTMP_CLIENT_PTR;
diff --git a/trunk/src/srt/ts_demux.cpp b/trunk/src/srt/ts_demux.cpp
index 3eed4f74f..ead598bbe 100644
--- a/trunk/src/srt/ts_demux.cpp
+++ b/trunk/src/srt/ts_demux.cpp
@@ -259,11 +259,17 @@ int ts_demux::decode_unit(unsigned char* data_p, std::string key_path, TS_DATA_C
                     if(ts_header_info._payload_unit_start_indicator){
                         unsigned char* ret_data_p = nullptr;
                         size_t ret_size = 0;
+                        uint64_t dts = 0;
+                        uint64_t pts = 0;
                         
                         //callback last media data in data buffer
                         on_callback(callback, _last_pid, key_path, _last_dts, _last_pts);
 
-                        pes_parse(data_p+npos, npos, &ret_data_p, ret_size, _last_dts, _last_pts);
+                        pes_parse(data_p+npos, npos, &ret_data_p, ret_size, dts, pts);
+
+                        _last_pts = pts;
+                        _last_dts = (dts == 0) ? pts : dts;
+
                         if ((ret_data_p != nullptr) && (ret_size > 0)) {
                             insert_into_databuf(ret_data_p, ret_size, key_path, ts_header_info._PID);
                         }

From f917cc1e0cee76c22a6e44f2b24f0856ab9742ae Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sun, 1 Mar 2020 10:09:44 +0800
Subject: [PATCH 06/16] For SRT, perfer mix_correct to true as default.

---
 trunk/src/app/srs_app_config.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index 627a01580..5781d2f1b 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -6765,7 +6765,7 @@ bool SrsConfig::get_srt_mix_correct() {
     if (!conf || conf->arg0().empty()) {
         return DEFAULT;
     }
-    return SRS_CONF_PERFER_FALSE(conf->arg0());
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
 int SrsConfig::get_srto_maxbw() {

From 1b9a705e4b03db4acefbb51f51b6a730144b7858 Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sun, 1 Mar 2020 10:17:03 +0800
Subject: [PATCH 07/16] For #1621, support mix_correct for aggregate aac
 packets. 4.0.10

---
 README.md                            | 2 ++
 trunk/src/core/srs_core_version4.hpp | 2 +-
 2 files changed, 3 insertions(+), 1 deletion(-)

diff --git a/README.md b/README.md
index bec6c8de6..9e54d43a2 100755
--- a/README.md
+++ b/README.md
@@ -153,6 +153,7 @@ For previous versions, please read:
 
 ## V4 changes
 
+* v4.0, 2020-03-01, For [#1621][bug #1621], support mix_correct for aggregate aac packets. 4.0.10
 * v4.0, 2020-02-25, For [#1615][bug #1615], support default app(live) for vmix SRT. 4.0.9
 * v4.0, 2020-02-21, For [#1598][bug #1598], support SLB health checking by TCP. 4.0.8
 * v4.0, 2020-02-19, For [#1579][bug #1579], support rolling update of k8s. 4.0.7
@@ -1687,6 +1688,7 @@ Winlin
 [bug #1579]: https://github.com/ossrs/srs/issues/1579
 [bug #1598]: https://github.com/ossrs/srs/issues/1598
 [bug #1615]: https://github.com/ossrs/srs/issues/1615
+[bug #1621]: https://github.com/ossrs/srs/issues/1621
 [bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
 
 [exo #828]: https://github.com/google/ExoPlayer/pull/828
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index da3e2b804..548632cf5 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -24,6 +24,6 @@
 #ifndef SRS_CORE_VERSION4_HPP
 #define SRS_CORE_VERSION4_HPP
 
-#define SRS_VERSION4_REVISION 9
+#define SRS_VERSION4_REVISION 10
 
 #endif

From 72155fc64523cd7d6b2c8617b0085d31ba993b48 Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sun, 1 Mar 2020 10:34:33 +0800
Subject: [PATCH 08/16] For #1621, support mix_correct for aggregate aac for
 SRT. 4.0.10

---
 README.md | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/README.md b/README.md
index 9e54d43a2..87f6be55f 100755
--- a/README.md
+++ b/README.md
@@ -153,7 +153,7 @@ For previous versions, please read:
 
 ## V4 changes
 
-* v4.0, 2020-03-01, For [#1621][bug #1621], support mix_correct for aggregate aac packets. 4.0.10
+* v4.0, 2020-03-01, For [#1621][bug #1621], support mix_correct for aggregate aac for SRT. 4.0.10
 * v4.0, 2020-02-25, For [#1615][bug #1615], support default app(live) for vmix SRT. 4.0.9
 * v4.0, 2020-02-21, For [#1598][bug #1598], support SLB health checking by TCP. 4.0.8
 * v4.0, 2020-02-19, For [#1579][bug #1579], support rolling update of k8s. 4.0.7

From 40a09958c4d01f30a2231b77bea7096f82fc3155 Mon Sep 17 00:00:00 2001
From: runner365 <shi.weibd@hotmail.com>
Date: Sun, 1 Mar 2020 19:42:32 +0800
Subject: [PATCH 09/16] add checking for pes_parse

---
 trunk/src/srt/ts_demux.cpp | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

diff --git a/trunk/src/srt/ts_demux.cpp b/trunk/src/srt/ts_demux.cpp
index ead598bbe..db486d5ad 100644
--- a/trunk/src/srt/ts_demux.cpp
+++ b/trunk/src/srt/ts_demux.cpp
@@ -265,7 +265,11 @@ int ts_demux::decode_unit(unsigned char* data_p, std::string key_path, TS_DATA_C
                         //callback last media data in data buffer
                         on_callback(callback, _last_pid, key_path, _last_dts, _last_pts);
 
-                        pes_parse(data_p+npos, npos, &ret_data_p, ret_size, dts, pts);
+                        int ret = pes_parse(data_p+npos, npos, &ret_data_p, ret_size, dts, pts);
+                        assert(ret <= 188);
+                        if (ret > 188) {
+                            return -1;
+                        }
 
                         _last_pts = pts;
                         _last_dts = (dts == 0) ? pts : dts;

From b525fc65aa3e79215da3b64510af38048feeb19a Mon Sep 17 00:00:00 2001
From: Xiaofeng Wang <wasphin@gmail.com>
Date: Tue, 3 Mar 2020 21:05:49 +0800
Subject: [PATCH 10/16] Remove unused ip variable

* "inet addr" is not compatible with rhel/centos 7+;
* ifconfig has been replaced by ip-utils in newer distro;
---
 trunk/configure | 1 -
 1 file changed, 1 deletion(-)

diff --git a/trunk/configure b/trunk/configure
index d233f66db..8ba25a8ba 100755
--- a/trunk/configure
+++ b/trunk/configure
@@ -730,7 +730,6 @@ fi
 # next step
 #####################################################################################
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
-    ip=`ifconfig|grep "inet addr"| grep -v "127.0.0.1"|awk '{print $2}'|awk -F ':' 'NR==1 {print $2}'`
     echo ""
     echo "You can run 3rdparty applications:"
     if [ $SRS_HTTP_CALLBACK = YES ]; then

From 4242985b36a50f6ccb46534d2afa1893021217c0 Mon Sep 17 00:00:00 2001
From: runner365 <shi.weibd@hotmail.com>
Date: Sat, 7 Mar 2020 15:15:14 +0800
Subject: [PATCH 11/16] update only support h264+aac

---
 trunk/src/srt/srt_to_rtmp.cpp | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

diff --git a/trunk/src/srt/srt_to_rtmp.cpp b/trunk/src/srt/srt_to_rtmp.cpp
index ec2d7d828..0d13c6733 100644
--- a/trunk/src/srt/srt_to_rtmp.cpp
+++ b/trunk/src/srt/srt_to_rtmp.cpp
@@ -624,8 +624,7 @@ void rtmp_client::on_data_callback(SRT_DATA_MSG_PTR data_ptr, unsigned int media
     } else if (media_type == STREAM_TYPE_AUDIO_AAC) {
         on_ts_audio(avs_ptr, dts, pts);
     } else {
-        srs_error("mpegts demux unkown stream type:0x%02x", media_type);
-        assert(0);
+        srs_error("mpegts demux unkown stream type:0x%02x, only support h264+aac", media_type);
     }
     return;
 }

From a2839aacdf8582a2eda3f951d8ef1dffe6f9d0be Mon Sep 17 00:00:00 2001
From: runner365 <shi.weibd@hotmail.com>
Date: Sat, 7 Mar 2020 16:15:36 +0800
Subject: [PATCH 12/16] add h264 sei filter

---
 trunk/src/app/srs_app_config.cpp | 16 +++++++++++++++-
 trunk/src/app/srs_app_config.hpp |  2 ++
 trunk/src/srt/srt_handle.cpp     |  3 ++-
 trunk/src/srt/srt_to_rtmp.cpp    |  6 ++++++
 4 files changed, 25 insertions(+), 2 deletions(-)

diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index 5781d2f1b..f66ca5cec 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -3530,7 +3530,7 @@ srs_error_t SrsConfig::check_normal_config()
                 && n != "mss" && n != "latency" && n != "recvlatency"
                 && n != "peerlatency" && n != "tlpkdrop" && n != "connect_timeout"
                 && n != "sendbuf" && n != "recvbuf" && n != "payloadsize"
-                && n != "default_app" && n != "mix_correct") {
+                && n != "default_app" && n != "mix_correct" && n != "sei_filter") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal srt_stream.%s", n.c_str());
             }
         }
@@ -6838,6 +6838,20 @@ int SrsConfig::get_srto_peer_latency() {
     return atoi(conf->arg0().c_str());
 }
 
+bool SrsConfig::get_srt_sei_filter() {
+    static bool DEFAULT = true;
+    SrsConfDirective* conf = root->get("srt_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("sei_filter");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
+}
+
 bool SrsConfig::get_srto_tlpkdrop() {
     static bool DEFAULT = true;
     SrsConfDirective* conf = root->get("srt_server");
diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
index 4360c41da..c276b34db 100644
--- a/trunk/src/app/srs_app_config.hpp
+++ b/trunk/src/app/srs_app_config.hpp
@@ -613,6 +613,8 @@ public:
     virtual int get_srto_recv_latency();
     // Get the srt SRTO_PEERLATENCY, peer latency, default is 0..
     virtual int get_srto_peer_latency();
+    // Get the srt h264 sei filter, default is on, it will drop h264 sei packet.
+    virtual bool get_srt_sei_filter();
     // Get the srt SRTO_TLPKDROP, Too-late Packet Drop, default is true.
     virtual bool get_srto_tlpkdrop();
     // Get the srt SRTO_CONNTIMEO, connection timeout, default is 3000ms.
diff --git a/trunk/src/srt/srt_handle.cpp b/trunk/src/srt/srt_handle.cpp
index 634376275..a6351548e 100644
--- a/trunk/src/srt/srt_handle.cpp
+++ b/trunk/src/srt/srt_handle.cpp
@@ -145,7 +145,8 @@ void srt_handle::add_newconn(SRT_CONN_PTR conn_ptr, int events) {
     srs_trace("srto SRTO_RCVBUF=%d", val_i);
     srt_getsockopt(conn_ptr->get_conn(), 0, SRTO_MAXBW, &val_i, &opt_len);
     srs_trace("srto SRTO_MAXBW=%d", val_i);
-    srs_trace("srt mix_correct is %s", _srs_config->get_srt_mix_correct() ? "enable" : "disable");
+    srs_trace("srt mix_correct is %s.", _srs_config->get_srt_mix_correct() ? "enable" : "disable");
+    srs_trace("srt h264 sei filter is %s.", _srs_config->get_srt_sei_filter() ? "enable" : "disable");
 
     if (conn_ptr->get_mode() == PULL_SRT_MODE) {
         add_new_puller(conn_ptr, conn_ptr->get_subpath());
diff --git a/trunk/src/srt/srt_to_rtmp.cpp b/trunk/src/srt/srt_to_rtmp.cpp
index 0d13c6733..0842a7e7f 100644
--- a/trunk/src/srt/srt_to_rtmp.cpp
+++ b/trunk/src/srt/srt_to_rtmp.cpp
@@ -459,6 +459,12 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
         if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
             continue;
         }
+
+        if (_srs_config->get_srt_sei_filter()) {
+            if (nal_unit_type == SrsAvcNaluTypeSEI) {
+                continue;
+            }
+        }
         
         // for sps
         if (_avc_ptr->is_sps(frame, frame_size)) {

From 41acf9ca8e37e6dfd0d4be6e8abfbcb2d14b152d Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sat, 7 Mar 2020 22:25:19 +0800
Subject: [PATCH 13/16] For #1631, support sei_filter for SRT. 4.0.11

---
 README.md                            | 4 ++++
 trunk/src/core/srs_core_version4.hpp | 2 +-
 trunk/src/srt/srt_to_rtmp.cpp        | 1 +
 3 files changed, 6 insertions(+), 1 deletion(-)

diff --git a/README.md b/README.md
index 87f6be55f..fffccb183 100755
--- a/README.md
+++ b/README.md
@@ -153,6 +153,7 @@ For previous versions, please read:
 
 ## V4 changes
 
+* v4.0, 2020-03-07, For [#1631][bug #1631], support sei_filter for SRT. 4.0.11
 * v4.0, 2020-03-01, For [#1621][bug #1621], support mix_correct for aggregate aac for SRT. 4.0.10
 * v4.0, 2020-02-25, For [#1615][bug #1615], support default app(live) for vmix SRT. 4.0.9
 * v4.0, 2020-02-21, For [#1598][bug #1598], support SLB health checking by TCP. 4.0.8
@@ -1691,6 +1692,9 @@ Winlin
 [bug #1621]: https://github.com/ossrs/srs/issues/1621
 [bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
 
+[bug #1631]: https://github.com/ossrs/srs/issues/1631
+[bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
+
 [exo #828]: https://github.com/google/ExoPlayer/pull/828
 
 [r3.0b1]: https://github.com/ossrs/srs/releases/tag/v3.0-b1
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index 548632cf5..45b47d69a 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -24,6 +24,6 @@
 #ifndef SRS_CORE_VERSION4_HPP
 #define SRS_CORE_VERSION4_HPP
 
-#define SRS_VERSION4_REVISION 10
+#define SRS_VERSION4_REVISION 11
 
 #endif
diff --git a/trunk/src/srt/srt_to_rtmp.cpp b/trunk/src/srt/srt_to_rtmp.cpp
index 0842a7e7f..f36a353ee 100644
--- a/trunk/src/srt/srt_to_rtmp.cpp
+++ b/trunk/src/srt/srt_to_rtmp.cpp
@@ -460,6 +460,7 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
             continue;
         }
 
+        // TODO: FIXME: Should cache this config, it's better not to get it for each video frame.
         if (_srs_config->get_srt_sei_filter()) {
             if (nal_unit_type == SrsAvcNaluTypeSEI) {
                 continue;

From dcb0553cc8c3e495d3d3b4ecad676409954f3945 Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sat, 7 Mar 2020 22:35:11 +0800
Subject: [PATCH 14/16] For #1612, fix crash bug for RTSP. 4.0.12

---
 README.md                            | 2 ++
 trunk/src/app/srs_app_rtsp.cpp       | 1 +
 trunk/src/app/srs_app_server.cpp     | 2 ++
 trunk/src/core/srs_core_version4.hpp | 2 +-
 4 files changed, 6 insertions(+), 1 deletion(-)

diff --git a/README.md b/README.md
index fffccb183..dc6abcfb5 100755
--- a/README.md
+++ b/README.md
@@ -153,6 +153,7 @@ For previous versions, please read:
 
 ## V4 changes
 
+* v4.0, 2020-03-07, For [#1612][bug #1612], fix crash bug for RTSP. 4.0.12
 * v4.0, 2020-03-07, For [#1631][bug #1631], support sei_filter for SRT. 4.0.11
 * v4.0, 2020-03-01, For [#1621][bug #1621], support mix_correct for aggregate aac for SRT. 4.0.10
 * v4.0, 2020-02-25, For [#1615][bug #1615], support default app(live) for vmix SRT. 4.0.9
@@ -1693,6 +1694,7 @@ Winlin
 [bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
 
 [bug #1631]: https://github.com/ossrs/srs/issues/1631
+[bug #1612]: https://github.com/ossrs/srs/issues/1612
 [bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
 
 [exo #828]: https://github.com/google/ExoPlayer/pull/828
diff --git a/trunk/src/app/srs_app_rtsp.cpp b/trunk/src/app/srs_app_rtsp.cpp
index 3dcf20213..c81c71c78 100644
--- a/trunk/src/app/srs_app_rtsp.cpp
+++ b/trunk/src/app/srs_app_rtsp.cpp
@@ -242,6 +242,7 @@ srs_error_t SrsRtspConn::serve()
 
 std::string SrsRtspConn::remote_ip()
 {
+    // TODO: FIXME: Implement it.
     return "";
 }
 
diff --git a/trunk/src/app/srs_app_server.cpp b/trunk/src/app/srs_app_server.cpp
index 0a904d514..c35c27a42 100644
--- a/trunk/src/app/srs_app_server.cpp
+++ b/trunk/src/app/srs_app_server.cpp
@@ -179,6 +179,8 @@ SrsRtspListener::SrsRtspListener(SrsServer* svr, SrsListenerType t, SrsConfDirec
     srs_assert(type == SrsListenerRtsp);
     if (type == SrsListenerRtsp) {
         caster = new SrsRtspCaster(c);
+
+        // TODO: FIXME: Must check error.
         caster->initialize();
     }
 }
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index 45b47d69a..98f2899e1 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -24,6 +24,6 @@
 #ifndef SRS_CORE_VERSION4_HPP
 #define SRS_CORE_VERSION4_HPP
 
-#define SRS_VERSION4_REVISION 11
+#define SRS_VERSION4_REVISION 12
 
 #endif

From 48be5b6245651120ae02394ba0edd9977f35652a Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sat, 7 Mar 2020 22:37:12 +0800
Subject: [PATCH 15/16] Update authors

---
 AUTHORS.txt | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/AUTHORS.txt b/AUTHORS.txt
index 6755f273b..352376b94 100644
--- a/AUTHORS.txt
+++ b/AUTHORS.txt
@@ -53,4 +53,6 @@ CONTRIBUTORS ordered by first contribution.
 * alphonsetai<tyh_123@163.com>
 * Michael.Ma<wnpllr@gmail.com>
 * lam2003<linmin3@yy.com>
-* runner365<shi.weibd@hotmail.com>
\ No newline at end of file
+* runner365<shi.weibd@hotmail.com>
+* XiaofengWang<wasphin@gmail.com>
+* XiaLixin<xialx@yuntongxun.com>
\ No newline at end of file

From 5586c2a128fa469263b8837ae2fc07f390c8ea97 Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Sat, 7 Mar 2020 22:42:23 +0800
Subject: [PATCH 16/16] For #1612, refactor return value for RTSP initialize.

---
 trunk/src/app/srs_app_rtsp.cpp | 1 +
 1 file changed, 1 insertion(+)

diff --git a/trunk/src/app/srs_app_rtsp.cpp b/trunk/src/app/srs_app_rtsp.cpp
index c81c71c78..8c6967115 100644
--- a/trunk/src/app/srs_app_rtsp.cpp
+++ b/trunk/src/app/srs_app_rtsp.cpp
@@ -712,6 +712,7 @@ srs_error_t SrsRtspCaster::initialize()
     if ((err = manager->start()) != srs_success) {
         return srs_error_wrap(err, "start manager");
     }
+    return err;
 }
 
 srs_error_t SrsRtspCaster::alloc_port(int* pport)