diff --git a/src/Rtcp/RtcpContext.cpp b/src/Rtcp/RtcpContext.cpp
index 9b4e31cc..384b15b2 100644
--- a/src/Rtcp/RtcpContext.cpp
+++ b/src/Rtcp/RtcpContext.cpp
@@ -18,8 +18,7 @@ void RtcpContext::clear() {
     memset(this, 0, sizeof(RtcpContext));
 }
 
-RtcpContext::RtcpContext(uint32_t sample_rate, bool is_receiver) {
-    _sample_rate = sample_rate;
+RtcpContext::RtcpContext(bool is_receiver) {
     _is_receiver = is_receiver;
 }
 
@@ -35,7 +34,6 @@ void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, size_t bytes) {
                 diff = -diff;
             }
             //抖动单位为采样次数
-            diff *= (_sample_rate / 1000.0);
             _jitter += (diff - _jitter) / 16.0;
         } else {
             _jitter = 0;
@@ -129,7 +127,7 @@ Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
     rtcp->setNtpStamp(tv);
 
     //转换成rtp时间戳
-    rtcp->rtpts = htonl(uint32_t(_last_rtp_stamp * (_sample_rate / 1000.0)));
+    rtcp->rtpts = htonl(_last_rtp_stamp);
     rtcp->packet_count = htonl((uint32_t) _packets);
     rtcp->octet_count = htonl((uint32_t) _bytes);
     return RtcpHeader::toBuffer(std::move(rtcp));
diff --git a/src/Rtcp/RtcpContext.h b/src/Rtcp/RtcpContext.h
index 85be0c29..d9f705ef 100644
--- a/src/Rtcp/RtcpContext.h
+++ b/src/Rtcp/RtcpContext.h
@@ -22,15 +22,14 @@ public:
     using Ptr = std::shared_ptr<RtcpContext>;
     /**
      * 创建rtcp上下文
-     * @param sample_rate 音频采用率，视频一般为90000
      * @param is_receiver 是否为rtp接收者，接收者更消耗性能
      */
-    RtcpContext(uint32_t sample_rate, bool is_receiver);
+    RtcpContext(bool is_receiver);
 
     /**
      * 输出或输入rtp时调用
      * @param seq rtp的seq
-     * @param stamp rtp的时间戳，单位毫秒
+     * @param stamp rtp的时间戳，单位采样数(非毫秒)
      * @param bytes rtp数据长度
      */
     void onRtp(uint16_t seq, uint32_t stamp, size_t bytes);
@@ -87,8 +86,6 @@ private:
     bool _is_receiver;
     //时间戳抖动值
     double _jitter = 0;
-    //视频默认90000,音频为采样率
-    uint32_t _sample_rate;
     //收到或发送的rtp的字节数
     size_t _bytes = 0;
     //收到或发送的rtp的个数
diff --git a/src/Rtp/RtpServer.cpp b/src/Rtp/RtpServer.cpp
index 6462abac..6827138f 100644
--- a/src/Rtp/RtpServer.cpp
+++ b/src/Rtp/RtpServer.cpp
@@ -27,7 +27,7 @@ class RtcpHelper : public RtcpContext, public std::enable_shared_from_this<RtcpH
 public:
     using Ptr = std::shared_ptr<RtcpHelper>;
 
-    RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) : RtcpContext(sample_rate, true){
+    RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) : RtcpContext(true){
         _rtcp_sock = std::move(rtcp_sock);
         _sample_rate = sample_rate;
     }
@@ -35,7 +35,7 @@ public:
     void onRecvRtp(const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len){
         //统计rtp接受情况，用于发送rr包
         auto header = (RtpHeader *) buf->data();
-        onRtp(ntohs(header->seq), ntohl(header->stamp) * uint64_t(1000) / _sample_rate, buf->size());
+        onRtp(ntohs(header->seq), ntohl(header->stamp), buf->size());
         sendRtcp(ntohl(header->ssrc), addr, addr_len);
     }
 
diff --git a/src/Rtsp/RtspPlayer.cpp b/src/Rtsp/RtspPlayer.cpp
index f3b7004f..45d0f445 100644
--- a/src/Rtsp/RtspPlayer.cpp
+++ b/src/Rtsp/RtspPlayer.cpp
@@ -205,7 +205,7 @@ void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
     }
     _rtcp_context.clear();
     for (auto &track : _sdp_track) {
-        _rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
+        _rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
     }
     sendSetup(0);
 }
@@ -591,7 +591,7 @@ void RtspPlayer::sendRtspRequest(const string &cmd, const string &url,const StrC
 
 void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx){
     auto &rtcp_ctx = _rtcp_context[track_idx];
-    rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     auto &ticker = _rtcp_send_ticker[track_idx];
     if (ticker.elapsedTime() < 3 * 1000) {
diff --git a/src/Rtsp/RtspPusher.cpp b/src/Rtsp/RtspPusher.cpp
index c19b0ce7..2d0a8abe 100644
--- a/src/Rtsp/RtspPusher.cpp
+++ b/src/Rtsp/RtspPusher.cpp
@@ -179,7 +179,7 @@ void RtspPusher::sendAnnounce() {
     }
     _rtcp_context.clear();
     for (auto &track : _track_vec) {
-        _rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, false));
+        _rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
     }
     _on_res_func = std::bind(&RtspPusher::handleResAnnounce, this, placeholders::_1);
     sendRtspRequest("ANNOUNCE", _url, {}, src->getSdp());
@@ -360,7 +360,7 @@ void RtspPusher::updateRtcpContext(const RtpPacket::Ptr &rtp){
     int track_index = getTrackIndexByTrackType(rtp->type);
     auto &ticker = _rtcp_send_ticker[track_index];
     auto &rtcp_ctx = _rtcp_context[track_index];
-    rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     //send rtcp every 5 second
     if (ticker.elapsedTime() > 5 * 1000) {
diff --git a/src/Rtsp/RtspSession.cpp b/src/Rtsp/RtspSession.cpp
index 1e12adee..e86eb399 100644
--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
@@ -252,7 +252,7 @@ void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
         }
         _rtcp_context.clear();
         for (auto &track : _sdp_track) {
-            _rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
+            _rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
         }
         _push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
         _push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
@@ -413,7 +413,7 @@ void RtspSession::onAuthSuccess() {
         }
         strongSelf->_rtcp_context.clear();
         for (auto &track : strongSelf->_sdp_track) {
-            strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, false));
+            strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
         }
         strongSelf->_sessionid = makeRandStr(12);
         strongSelf->_play_src = rtsp_src;
@@ -1126,7 +1126,7 @@ void RtspSession::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index){
 void RtspSession::updateRtcpContext(const RtpPacket::Ptr &rtp){
     int track_index = getTrackIndexByTrackType(rtp->type);
     auto &rtcp_ctx = _rtcp_context[track_index];
-    rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     auto &ticker = _rtcp_send_tickers[track_index];
     //send rtcp every 5 second
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index b5891ab2..2a1b14f3 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -408,19 +408,19 @@ void WebRtcTransportImp::onStartWebRTC() {
         info->offer_ssrc_rtx = m_offer->getRtxSSRC();
         info->plan_rtp = &m_answer.plan[0];;
         info->plan_rtx = m_answer.getRelatedRtxPlan(info->plan_rtp->pt);
-        info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
+        info->rtcp_context_send = std::make_shared<RtcpContext>(false);
 
         //send ssrc --> MediaTrack
-        _rtp_info_ssrc[info->answer_ssrc_rtp] = info;
+        _ssrc_to_track[info->answer_ssrc_rtp] = info;
 
         //recv ssrc --> MediaTrack
-        _rtp_info_ssrc[info->offer_ssrc_rtp] = info;
+        _ssrc_to_track[info->offer_ssrc_rtp] = info;
 
         //rtp pt --> MediaTrack
-        _rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info));
+        _pt_to_track.emplace(info->plan_rtp->pt, std::make_pair(false, info));
         if (info->plan_rtx) {
             //rtx pt --> MediaTrack
-            _rtp_info_pt.emplace(info->plan_rtx->pt, std::make_pair(true, info));
+            _pt_to_track.emplace(info->plan_rtx->pt, std::make_pair(true, info));
         }
         if (m_offer->type != TrackApplication) {
             //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id
@@ -464,10 +464,10 @@ void WebRtcTransportImp::onStartWebRTC() {
             }
             auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
             if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
-                auto it = _rtp_info_pt.find(m.plan[0].pt);
-                CHECK(it != _rtp_info_pt.end());
+                auto it = _pt_to_track.find(m.plan[0].pt);
+                CHECK(it != _pt_to_track.end());
                 //记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
-                _send_rtp_info[m.type] = it->second.second;
+                _type_to_track[m.type] = it->second.second;
             }
         }
     }
@@ -558,8 +558,7 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
 
 class RtpChannel : public RtpReceiver {
 public:
-    uint32_t ssrc;
-    RtcpContext::Ptr rtcp_context;
+    uint32_t rtp_ssrc;
 
 public:
     RtpChannel(function<void(RtpPacket::Ptr rtp)> on_rtp, function<void(const FCI_NACK &nack)> on_nack) {
@@ -576,11 +575,16 @@ public:
             //统计rtp接受情况，便于生成nack rtcp包
             nack_ctx.received(seq);
             //统计rtp收到的情况，好做rr汇报
-            rtcp_context->onRtp(seq, ntohl(rtp->stamp) * uint64_t(1000) / sample_rate, len);
+            rtcp_context.onRtp(seq, ntohl(rtp->stamp), len);
         }
         return handleOneRtp((int) type, type, sample_rate, ptr, len);
     }
 
+    Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
+        rtcp_context.onRtcp(sr);
+        return rtcp_context.createRtcpRR(ssrc, rtp_ssrc);
+    }
+
 protected:
     void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override {
         _on_sort(std::move(rtp));
@@ -588,6 +592,7 @@ protected:
 
 private:
     NackContext nack_ctx;
+    RtcpContext rtcp_context{true};
     function<void(RtpPacket::Ptr rtp)> _on_sort;
 };
 
@@ -611,15 +616,14 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
             case RtcpType::RTCP_SR : {
                 //对方汇报rtp发送情况
                 RtcpSR *sr = (RtcpSR *) rtcp;
-                auto it = _rtp_info_ssrc.find(sr->ssrc);
-                if (it != _rtp_info_ssrc.end()) {
+                auto it = _ssrc_to_track.find(sr->ssrc);
+                if (it != _ssrc_to_track.end()) {
                     auto &info = it->second;
                     auto rtp_chn = info->getRtpChannel(sr->ssrc);
                     if(!rtp_chn){
                         WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
                     } else {
-                        rtp_chn->rtcp_context->onRtcp(sr);
-                        auto rr = rtp_chn->rtcp_context->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc);
+                        auto rr = rtp_chn->createRtcpRR(sr, info->answer_ssrc_rtp);
                         sendRtcpPacket(rr->data(), rr->size(), true);
                     }
                 } else {
@@ -632,8 +636,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                 //对方汇报rtp接收情况
                 RtcpRR *rr = (RtcpRR *) rtcp;
                 for (auto item : rr->getItemList()) {
-                    auto it = _rtp_info_ssrc.find(item->ssrc);
-                    if (it != _rtp_info_ssrc.end()) {
+                    auto it = _ssrc_to_track.find(item->ssrc);
+                    if (it != _ssrc_to_track.end()) {
                         auto &info = it->second;
                         auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp);
                         sendRtcpPacket(sr->data(), sr->size(), true);
@@ -647,12 +651,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                 //对方汇报停止发送rtp
                 RtcpBye *bye = (RtcpBye *) rtcp;
                 for (auto ssrc : bye->getSSRC()) {
-                    auto it = _rtp_info_ssrc.find(*ssrc);
-                    if (it == _rtp_info_ssrc.end()) {
+                    auto it = _ssrc_to_track.find(*ssrc);
+                    if (it == _ssrc_to_track.end()) {
                         WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
                         continue;
                     }
-                    _rtp_info_ssrc.erase(it);
+                    _ssrc_to_track.erase(it);
                 }
                 onShutdown(SockException(Err_eof, "rtcp bye message received"));
                 break;
@@ -666,8 +670,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                 switch ((RTPFBType) rtcp->report_count) {
                     case RTPFBType::RTCP_RTPFB_NACK : {
                         RtcpFB *fb = (RtcpFB *) rtcp;
-                        auto it = _rtp_info_ssrc.find(fb->ssrc_media);
-                        if (it == _rtp_info_ssrc.end()) {
+                        auto it = _ssrc_to_track.find(fb->ssrc_media);
+                        if (it == _ssrc_to_track.end()) {
                             WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
                             return;
                         }
@@ -752,11 +756,9 @@ void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, cons
         onSendNack(*info, nack, ssrc);
     });
     //rid --> rtp ssrc
-    ref->ssrc = ssrc;
-    //rtp ssrc --> RtcpContext
-    ref->rtcp_context = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
+    ref->rtp_ssrc = ssrc;
     //rtp ssrc --> MediaTrack
-    _rtp_info_ssrc[ssrc] = info;
+    _ssrc_to_track[ssrc] = info;
     InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << info->plan_rtp->codec;
 }
 
@@ -766,8 +768,8 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
 
     RtpHeader *rtp = (RtpHeader *) buf;
     //根据接收到的rtp的pt信息，找到该流的信息
-    auto it = _rtp_info_pt.find(rtp->pt);
-    if (it == _rtp_info_pt.end()) {
+    auto it = _pt_to_track.find(rtp->pt);
+    if (it == _pt_to_track.end()) {
         WarnL << "unknown rtp pt:" << (int)rtp->pt;
         return;
     }
@@ -822,7 +824,7 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
     //rtx 转换为 rtp
     rtp->pt = info->plan_rtp->pt;
     rtp->seq = htons(origin_seq);
-    rtp->ssrc = htonl(ref->ssrc);
+    rtp->ssrc = htonl(ref->rtp_ssrc);
 
     memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
     buf += 2;
@@ -878,14 +880,14 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &info, const string &rid, RtpPac
 ///////////////////////////////////////////////////////////////////
 
 void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
-    auto &info = _send_rtp_info[rtp->type];
+    auto &info = _type_to_track[rtp->type];
     if (!info) {
         //忽略，对方不支持该编码类型
         return;
     }
     if (!rtx) {
         //统计rtp发送情况，好做sr汇报
-        info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+        info->rtcp_context_send->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
         info->nack_list.push_back(rtp);
 #if 0
         //此处模拟发送丢包
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index 4b24f488..87450ff3 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -384,11 +384,11 @@ private:
     //播放rtsp源的reader对象
     RtspMediaSource::RingType::RingReader::Ptr _reader;
     //根据发送rtp的track类型获取相关信息
-    MediaTrack::Ptr _send_rtp_info[2];
+    MediaTrack::Ptr _type_to_track[2];
     //根据接收rtp的pt获取相关信息
-    unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,MediaTrack::Ptr> > _rtp_info_pt;
+    unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,MediaTrack::Ptr> > _pt_to_track;
     //根据rtcp的ssrc获取相关信息，只记录rtp的ssrc，rtx的ssrc不记录
-    unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _rtp_info_ssrc;
+    unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _ssrc_to_track;
     //发送rtp时需要修改rtp ext id
     map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
     //接收rtp时需要修改rtp ext id