diff --git a/webrtc/WebRtcPlayer.cpp b/webrtc/WebRtcPlayer.cpp
index 7531332d..1bd8b513 100644
--- a/webrtc/WebRtcPlayer.cpp
+++ b/webrtc/WebRtcPlayer.cpp
@@ -33,12 +33,6 @@ void WebRtcPlayer::onStartWebRTC() {
     CHECK(_play_src);
     WebRtcTransportImp::onStartWebRTC();
     if (canSendRtp()) {
-        //确保该rtp codec类型对方支持
-        memset(_can_send_rtp, 0, sizeof(_can_send_rtp));
-        for (auto &m : _answer_sdp->media) {
-            _can_send_rtp[m.type] = m.direction == RtpDirection::sendonly || m.direction == RtpDirection::sendrecv;
-        }
-
         _play_src->pause(false);
         _reader = _play_src->getRing()->attach(getPoller(), true);
         weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
@@ -49,7 +43,7 @@ void WebRtcPlayer::onStartWebRTC() {
             }
             size_t i = 0;
             pkt->for_each([&](const RtpPacket::Ptr &rtp) {
-                strongSelf->beforeSendRtp(rtp, ++i == pkt->size());
+                strongSelf->onSendRtp(rtp, ++i == pkt->size());
             });
         });
         _reader->setDetachCB([weak_self]() {
@@ -91,10 +85,3 @@ void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
     configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
     configure.setPlayRtspInfo(_play_src->getSdp());
 }
-
-void WebRtcPlayer::beforeSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx) {
-    if (!_can_send_rtp[rtp->type]) {
-        return;
-    }
-    onSendRtp(rtp, flush, rtx);
-}
diff --git a/webrtc/WebRtcPlayer.h b/webrtc/WebRtcPlayer.h
index ade67388..beb25ef2 100644
--- a/webrtc/WebRtcPlayer.h
+++ b/webrtc/WebRtcPlayer.h
@@ -28,10 +28,8 @@ protected:
 
 private:
     WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
-    void beforeSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
 
 private:
-    bool _can_send_rtp[TrackMax];
     //媒体相关元数据
     MediaInfo _media_info;
     //播放的rtsp源
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index c7a8106f..28ce411c 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -372,7 +372,10 @@ void WebRtcTransportImp::onStartWebRTC() {
         track->rtcp_context_send = std::make_shared<RtcpContextForSend>();
 
         //rtp track type --> MediaTrack
-        _type_to_track[m_answer.type] = track;
+        if (m_answer.direction == RtpDirection::sendonly || m_answer.direction == RtpDirection::sendrecv) {
+            //该类型的track 才支持发送
+            _type_to_track[m_answer.type] = track;
+        }
         //send ssrc --> MediaTrack
         _ssrc_to_track[track->answer_ssrc_rtp] = track;
         _ssrc_to_track[track->answer_ssrc_rtx] = track;