diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp
index 91d822bf..42e85241 100644
--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -1496,8 +1496,18 @@ void RtcConfigure::setPlayRtspInfo(const string &sdp){
     session.loadFrom(sdp, false);
     for (auto &m : session.media) {
         switch (m.type) {
-            case TrackVideo : _rtsp_video_plan = std::make_shared<RtcCodecPlan>(m.plan[0]); break;
-            case TrackAudio : _rtsp_audio_plan = std::make_shared<RtcCodecPlan>(m.plan[0]); break;
+            case TrackVideo : {
+                _rtsp_video_plan = std::make_shared<RtcCodecPlan>(m.plan[0]);
+                video.preferred_codec.clear();
+                video.preferred_codec.emplace_back(getCodecId(_rtsp_video_plan->codec));
+                break;
+            }
+            case TrackAudio : {
+                _rtsp_audio_plan = std::make_shared<RtcCodecPlan>(m.plan[0]);
+                audio.preferred_codec.clear();
+                audio.preferred_codec.emplace_back(getCodecId(_rtsp_audio_plan->codec));
+                break;
+            }
             default: break;
         }
     }
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index 7aa6417b..2dbbef3b 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -298,11 +298,14 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
         return;
     }
 
-    //添加answer sdp的ssrc信息，并且记录发送rtp的pt
+    RtcSession rtsp_send_sdp;
+    rtsp_send_sdp.loadFrom(_src->getSdp(), false);
+
     for (auto &m : sdp.media) {
         if (m.type == TrackApplication) {
             continue;
         }
+        //添加answer sdp的ssrc信息
         m.rtp_ssrc.ssrc = _src->getSsrc(m.type);
         m.rtp_ssrc.cname = RTP_CNAME;
         //todo 先屏蔽rtx，因为chrome报错
@@ -310,8 +313,9 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
             m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc;
             m.rtx_ssrc.cname = RTX_CNAME;
         }
-        auto rtsp_media = _rtsp_send_sdp.getMedia(m.type);
+        auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
         if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
+            //记录发送rtp的pt
             _send_rtp_pt[m.type] = m.plan[0].pt;
         }
     }
@@ -325,24 +329,6 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
         configure.video.direction = RtpDirection::sendonly;
         configure.audio.direction = RtpDirection::sendonly;
         configure.setPlayRtspInfo(_src->getSdp());
-        _rtsp_send_sdp.loadFrom(_src->getSdp(), false);
-        //根据rtsp流的相关信息，设置rtc最佳编码
-        for (auto &m : _rtsp_send_sdp.media) {
-            switch (m.type) {
-                case TrackVideo: {
-                    configure.video.preferred_codec.clear();
-                    configure.video.preferred_codec.emplace_back(getCodecId(m.plan[0].codec));
-                    break;
-                }
-                case TrackAudio: {
-                    configure.audio.preferred_codec.clear();
-                    configure.audio.preferred_codec.emplace_back(getCodecId(m.plan[0].codec));
-                    break;
-                }
-                default:
-                    break;
-            }
-        }
     } else {
         //这是推流
         configure.video.direction = RtpDirection::recvonly;
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index 000c4cc7..28086a01 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -159,7 +159,6 @@ private:
     Socket::Ptr _socket;
     RtcSession _answer_sdp;
     RtspMediaSource::Ptr _src;
-    mutable RtcSession _rtsp_send_sdp;
     RtspMediaSource::RingType::RingReader::Ptr _reader;
     unordered_map<uint8_t, RtpPayloadInfo> _rtp_info_pt;
     unordered_map<uint32_t, RtpPayloadInfo*> _rtp_info_ssrc;