diff --git a/webrtc/WebRtcPusher.cpp b/webrtc/WebRtcPusher.cpp
index 8fe6549c..5731c870 100644
--- a/webrtc/WebRtcPusher.cpp
+++ b/webrtc/WebRtcPusher.cpp
@@ -130,14 +130,21 @@ void WebRtcPusher::onDestory() {
         }
     }
 
-    if (_push_src && _continue_push_ms) {
+    WebRtcTransportImp::onDestory();
+}
+
+void WebRtcPusher::onShutdown(const SockException &ex) {
+
+    if (_push_src && _continue_push_ms && ex.getErrCode() != Err_shutdown) {
         //取消所有权
         _push_src_ownership = nullptr;
-        //延时10秒注销流
+        //延时注销流
         auto push_src = std::move(_push_src);
         getPoller()->doDelayTask(_continue_push_ms, [push_src]() { return 0; });
     }
-    WebRtcTransportImp::onDestory();
+
+    WebRtcTransportImp::onShutdown(ex);
+
 }
 
 void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
@@ -150,14 +157,5 @@ float WebRtcPusher::getLossRate(MediaSource &sender,TrackType type){
     return WebRtcTransportImp::getLossRate(type);
 }
 
-void WebRtcPusher::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
-   //主动关闭推流，那么不等待重推
-    _push_src = nullptr;
-    WebRtcTransportImp::OnDtlsTransportClosed(dtlsTransport);
-}
-
-void WebRtcPusher::onRtcpBye(){
-     WebRtcTransportImp::onRtcpBye();
-}
 
 }// namespace mediakit
\ No newline at end of file
diff --git a/webrtc/WebRtcPusher.h b/webrtc/WebRtcPusher.h
index f00e0483..677da8b2 100644
--- a/webrtc/WebRtcPusher.h
+++ b/webrtc/WebRtcPusher.h
@@ -27,11 +27,9 @@ protected:
     ///////WebRtcTransportImp override///////
     void onStartWebRTC() override;
     void onDestory() override;
+    void onShutdown(const SockException &ex) override;
     void onRtcConfigure(RtcConfigure &configure) const override;
     void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override;
-    void onRtcpBye() override;
-    ////  dtls相关的回调 ////
-   void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
 
 protected:
     ///////MediaSourceEvent override///////
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index 99f9f55d..cb579118 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -802,7 +802,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                 }
                 _ssrc_to_track.erase(it);
             }
-            onRtcpBye();
             // bye 会在 sender audio track mute 时出现, 因此不能作为 shutdown 的依据
             break;
         }
@@ -1075,7 +1074,6 @@ uint64_t WebRtcTransportImp::getDuration() const {
     return _alive_ticker.createdTime() / 1000;
 }
 
-void WebRtcTransportImp::onRtcpBye(){}
 
 /////////////////////////////////////////////////////////////////////////////////////////////
 
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index dfae8012..30fa5a8b 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -156,7 +156,6 @@ protected:
     virtual void onShutdown(const SockException &ex) = 0;
     virtual void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) = 0;
     virtual void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) = 0;
-    virtual void onRtcpBye() = 0;
 
 protected:
     void sendRtcpRemb(uint32_t ssrc, size_t bit_rate);
@@ -266,7 +265,6 @@ protected:
     virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
     void updateTicker();
     float getLossRate(TrackType type);
-    void onRtcpBye() override;
 
 private:
     void onSortedRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp);