diff --git a/3rdpart/ZLToolKit b/3rdpart/ZLToolKit
index 82f82e72..9300828f 160000
--- a/3rdpart/ZLToolKit
+++ b/3rdpart/ZLToolKit
@@ -1 +1 @@
-Subproject commit 82f82e723eda127777aaf91ad0e4615e8d56e8d5
+Subproject commit 9300828fd9df9f581d1422171a5f13bdc2a4e6ab
diff --git a/player/test_player.cpp b/player/test_player.cpp
index 40d3fcbb..b0bba649 100644
--- a/player/test_player.cpp
+++ b/player/test_player.cpp
@@ -53,7 +53,7 @@ int main(int argc, char *argv[]) {
     Logger::Instance().add(std::make_shared<ConsoleChannel>());
     Logger::Instance().setWriter(std::make_shared<AsyncLogWriter>());
 
-    if (argc != 3) {
+    if (argc < 3) {
         ErrorL << "\r\n测试方法：./test_player rtxp_url rtp_type\r\n"
                << "例如：./test_player rtsp://admin:123456@127.0.0.1/live/0 0\r\n"
                << endl;
@@ -119,6 +119,9 @@ int main(int argc, char *argv[]) {
     (*player)[Client::kRtpType] = atoi(argv[2]);
     //不等待track ready再回调播放成功事件，这样可以加快秒开速度
     (*player)[Client::kWaitTrackReady] = false;
+    if (argc > 3) {
+        (*player)[Client::kPlayTrack] = atoi(argv[3]);
+    }
     player->play(argv[1]);
     SDLDisplayerHelper::Instance().runLoop();
     return 0;
diff --git a/src/Common/config.cpp b/src/Common/config.cpp
index 5dd4a02c..4520f798 100644
--- a/src/Common/config.cpp
+++ b/src/Common/config.cpp
@@ -317,6 +317,7 @@ const string kMediaTimeoutMS = "media_timeout_ms";
 const string kBeatIntervalMS = "beat_interval_ms";
 const string kBenchmarkMode = "benchmark_mode";
 const string kWaitTrackReady = "wait_track_ready";
+const string kPlayTrack = "play_track";
 } // namespace Client
 
 } // namespace mediakit
diff --git a/src/Common/config.h b/src/Common/config.h
index 7e6b68a9..65176b5e 100644
--- a/src/Common/config.h
+++ b/src/Common/config.h
@@ -375,6 +375,9 @@ extern const std::string kBeatIntervalMS;
 extern const std::string kBenchmarkMode;
 // 播放器在触发播放成功事件时，是否等待所有track ready时再回调
 extern const std::string kWaitTrackReady;
+// rtsp播放指定track，可选项有0(不指定，默认)、1(视频)、2(音频)
+// 设置方法:player[Client::kPlayTrack] = 0/1/2;
+extern const std::string kPlayTrack;
 } // namespace Client
 } // namespace mediakit
 
diff --git a/src/Rtsp/RtspPlayer.cpp b/src/Rtsp/RtspPlayer.cpp
index 38b53669..e94e8dcd 100644
--- a/src/Rtsp/RtspPlayer.cpp
+++ b/src/Rtsp/RtspPlayer.cpp
@@ -197,13 +197,24 @@ void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
         _content_base.pop_back();
     }
 
-    SdpParser sdpParser(parser.Content());
     //解析sdp
-    _sdp_track = sdpParser.getAvailableTrack();
+    SdpParser sdpParser(parser.Content());
+
+    string sdp;
+    auto play_track = (TrackType)((int)(*this)[Client::kPlayTrack] - 1);
+    if (play_track != TrackInvalid) {
+        auto track = sdpParser.getTrack(play_track);
+        _sdp_track.emplace_back(track);
+        sdp = track->toString();
+    } else {
+        _sdp_track = sdpParser.getAvailableTrack();
+        sdp = sdpParser.toString();
+    }
+
     if (_sdp_track.empty()) {
         throw std::runtime_error("无有效的Sdp Track");
     }
-    if (!onCheckSDP(sdpParser.toString())) {
+    if (!onCheckSDP(sdp)) {
         throw std::runtime_error("onCheckSDP faied");
     }
     _rtcp_context.clear();
diff --git a/src/Rtsp/RtspSession.cpp b/src/Rtsp/RtspSession.cpp
index 196d222f..e5d2d9db 100644
--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
@@ -801,13 +801,14 @@ void RtspSession::handleReq_Play(const Parser &parser) {
         InfoP(this) << "rtsp seekTo(ms):" << iStartTime;
     }
 
+    vector<TrackType> inited_tracks;
     _StrPrinter rtp_info;
     for (auto &track : _sdp_track) {
         if (track->_inited == false) {
-            //还有track没有setup
-            shutdown(SockException(Err_shutdown, "track not setuped"));
-            return;
+            //为支持播放器播放单一track, 不校验没有发setup的track
+            continue;
         }
+        inited_tracks.emplace_back(track->_type);
         track->_ssrc = play_src->getSsrc(track->_type);
         track->_seq = play_src->getSeqence(track->_type);
         track->_time_stamp = play_src->getTimeStamp(track->_type);
@@ -824,6 +825,12 @@ void RtspSession::handleReq_Play(const Parser &parser) {
     res_header.emplace("Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << play_src->getTimeStamp(TrackInvalid) / 1000.0);
     sendRtspResponse("200 OK", res_header);
 
+    //设置播放track
+    if (inited_tracks.size() == 1) {
+        _target_play_track = inited_tracks[0];
+        InfoP(this) << "指定播放track:" << _target_play_track;
+    }
+
     //在回复rtsp信令后再恢复播放
     play_src->pause(false);
 
@@ -956,13 +963,17 @@ void RtspSession::onRcvPeerUdpData(int interleaved, const Buffer::Ptr &buf, cons
         } else if (!_udp_connected_flags.count(interleaved)) {
             //这是rtsp播放器的rtp打洞包
             _udp_connected_flags.emplace(interleaved);
-            _rtp_socks[interleaved / 2]->bindPeerAddr((struct sockaddr *)&addr);
+            if (_rtp_socks[interleaved / 2]) {
+                _rtp_socks[interleaved / 2]->bindPeerAddr((struct sockaddr *)&addr);
+            }
         }
     } else {
         //rtcp包
         if (!_udp_connected_flags.count(interleaved)) {
             _udp_connected_flags.emplace(interleaved);
-            _rtcp_socks[(interleaved - 1) / 2]->bindPeerAddr((struct sockaddr *)&addr);
+            if (_rtcp_socks[(interleaved - 1) / 2]) {
+                _rtcp_socks[(interleaved - 1) / 2]->bindPeerAddr((struct sockaddr *)&addr);
+            }
         }
         onRtcpPacket((interleaved - 1) / 2, _sdp_track[(interleaved - 1) / 2], buf->data(), buf->size());
     }
@@ -1190,32 +1201,39 @@ void RtspSession::updateRtcpContext(const RtpPacket::Ptr &rtp){
 void RtspSession::sendRtpPacket(const RtspMediaSource::RingDataType &pkt) {
     switch (_rtp_type) {
         case Rtsp::RTP_TCP: {
-            size_t i = 0;
-            auto size = pkt->size();
             setSendFlushFlag(false);
             pkt->for_each([&](const RtpPacket::Ptr &rtp) {
-                updateRtcpContext(rtp);
-                if (++i == size) {
-                    setSendFlushFlag(true);
+                if (_target_play_track == TrackInvalid || _target_play_track == rtp->type) {
+                    updateRtcpContext(rtp);
+                    send(rtp);
                 }
-                send(rtp);
             });
+            flushAll();
+            setSendFlushFlag(true);
         }
             break;
         case Rtsp::RTP_UDP: {
-            size_t i = 0;
-            auto size = pkt->size();
+            //下标0表示视频，1表示音频
+            Socket::Ptr rtp_socks[2];
+            rtp_socks[TrackVideo] = _rtp_socks[getTrackIndexByTrackType(TrackVideo)];
+            rtp_socks[TrackAudio] = _rtp_socks[getTrackIndexByTrackType(TrackAudio)];
             pkt->for_each([&](const RtpPacket::Ptr &rtp) {
-                updateRtcpContext(rtp);
-                int track_index = getTrackIndexByTrackType(rtp->type);
-                auto &pSock = _rtp_socks[track_index];
-                if (!pSock) {
-                    shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
-                    return;
+                if (_target_play_track == TrackInvalid || _target_play_track == rtp->type) {
+                    updateRtcpContext(rtp);
+                    auto &sock = rtp_socks[rtp->type];
+                    if (!sock) {
+                        shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
+                        return;
+                    }
+                    _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
+                    sock->send(std::make_shared<BufferRtp>(rtp, RtpPacket::kRtpTcpHeaderSize), nullptr, 0, false);
                 }
-                _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
-                pSock->send(std::make_shared<BufferRtp>(rtp, RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
             });
+            for (auto &sock : rtp_socks) {
+                if (sock) {
+                    sock->flushAll();
+                }
+            }
         }
             break;
         default:
diff --git a/src/Rtsp/RtspSession.h b/src/Rtsp/RtspSession.h
index b07428aa..a4efabe5 100644
--- a/src/Rtsp/RtspSession.h
+++ b/src/Rtsp/RtspSession.h
@@ -195,6 +195,8 @@ private:
     RtspMediaSource::RingType::RingReader::Ptr _play_reader;
     //sdp里面有效的track,包含音频或视频
     std::vector<SdpTrack::Ptr> _sdp_track;
+    //播放器setup指定的播放track,默认为TrackInvalid表示不指定即音视频都推
+    TrackType _target_play_track = TrackInvalid;
 
     ////////RTP over udp////////
     //RTP端口,trackid idx 为数组下标