diff --git a/conf/config.ini b/conf/config.ini
index c7330ed2..35a6cc58 100644
--- a/conf/config.ini
+++ b/conf/config.ini
@@ -211,6 +211,8 @@ timeoutSec=15
 timeoutSec=15
 #本机对rtc客户端的可见ip，作为服务器时一般为公网ip，置空时，会自动获取网卡ip
 externIP=
+#设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质
+rembBitRate=1000000
 
 [rtsp]
 #rtsp专有鉴权方式是采用base64还是md5方式
diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp
index 6317c87b..8a671f66 100644
--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -1241,10 +1241,18 @@ const RtcMedia *RtcSession::getMedia(TrackType type) const{
     return nullptr;
 }
 
+bool RtcSession::supportRtcpFb(const string &name, TrackType type) const {
+    auto media = getMedia(type);
+    if (!media) {
+        return false;
+    }
+    auto &ref = media->plan[0].rtcp_fb;
+    return ref.find(name) != ref.end();
+}
+
 static string const kTWCCRtcpFb = "transport-cc";
 static string const kTWCCExtMap = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
 
-
 void RtcConfigure::RtcTrackConfigure::enableTWCC(bool enable){
     if (!enable) {
         rtcp_fb.erase(kTWCCRtcpFb);
diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h
index 6c9784b0..bcd1783c 100644
--- a/webrtc/Sdp.h
+++ b/webrtc/Sdp.h
@@ -667,6 +667,7 @@ public:
     string toString() const;
     string toRtspSdp() const;
     const  RtcMedia *getMedia(TrackType type) const;
+    bool supportRtcpFb(const string &name, TrackType type = TrackType::TrackVideo) const;
 
 private:
     RtcSessionSdp::Ptr toRtcSessionSdp() const;
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index 80b3d6ae..178daca6 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -11,7 +11,9 @@
 #include "WebRtcTransport.h"
 #include <iostream>
 #include "Rtcp/Rtcp.h"
+#include "Rtcp/RtcpFCI.h"
 #include "Rtsp/RtpReceiver.h"
+
 #define RTX_SSRC_OFFSET 2
 #define RTP_CNAME "zlmediakit-rtp"
 #define RTX_CNAME "zlmediakit-rtx"
@@ -23,10 +25,13 @@ namespace RTC {
 const string kTimeOutSec = RTC_FIELD"timeoutSec";
 //服务器外网ip
 const string kExternIP = RTC_FIELD"externIP";
+//设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质
+const string kRembBitRate = RTC_FIELD"rembBitRate";
 
 static onceToken token([]() {
     mINI::Instance()[kTimeOutSec] = 15;
     mINI::Instance()[kExternIP] = "";
+    mINI::Instance()[kRembBitRate] = 0;
 });
 
 }//namespace RTC
@@ -134,6 +139,22 @@ RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{
     return  _ice_server->GetSelectedTuple();
 }
 
+void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
+    auto remb = FCI_REMB::create({ssrc}, (uint32_t)bit_rate);
+    auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size());
+    fb->ssrc = htonl(0);
+    fb->ssrc_media = htonl(ssrc);
+    sendRtcpPacket((char *) fb.get(), fb->getSize(), true);
+    TraceL << ssrc << " " << bit_rate;
+}
+
+void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
+    auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
+    pli->ssrc = htonl(0);
+    pli->ssrc_media = htonl(ssrc);
+    sendRtcpPacket((char *) pli.get(), pli->getSize(), true);
+}
+
 string getFingerprint(const string &algorithm_str, const std::shared_ptr<RTC::DtlsTransport> &transport){
     auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str);
     for (auto &finger_prints : transport->GetLocalFingerprints()) {
@@ -163,6 +184,12 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){
     }
 }
 
+void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
+    //开启remb后关闭twcc，因为开启twcc后remb无效
+    GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
+    configure.enableTWCC(!remb_bit_rate);
+}
+
 std::string WebRtcTransport::getAnswerSdp(const string &offer){
     try {
         //// 解析offer sdp ////
@@ -389,6 +416,10 @@ void WebRtcTransportImp::onStartWebRTC() {
 
     if (canRecvRtp()) {
         _push_src->setSdp(getSdp(SdpType::answer).toRtspSdp());
+        GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
+        if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb("goog-remb")) {
+            sendRtcpRemb(_recv_video_ssrc, remb_bit_rate);
+        }
     }
     if (canSendRtp()) {
         _reader = _play_src->getRing()->attach(getPoller(), true);
@@ -592,10 +623,7 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr
     if (_pli_ticker.elapsedTime() > 2000) {
         //定期发送pli请求关键帧，方便非rtc等协议
         _pli_ticker.resetTime();
-        auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
-        pli->ssrc = htonl(0);
-        pli->ssrc_media = htonl(_recv_video_ssrc);
-        sendRtcpPacket((char *) pli.get(), pli->getSize(), true);
+        sendRtcpPli(_recv_video_ssrc);
     }
     if (_push_src) {
         _push_src->onWrite(std::move(rtp), false);
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index 50f45604..9cc64664 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -93,7 +93,7 @@ protected:
 
 protected:
     virtual void onStartWebRTC() = 0;
-    virtual void onRtcConfigure(RtcConfigure &configure) const {}
+    virtual void onRtcConfigure(RtcConfigure &configure) const;
     virtual void onCheckSdp(SdpType type, RtcSession &sdp);
     virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0;
 
@@ -104,6 +104,8 @@ protected:
 protected:
     const RtcSession& getSdp(SdpType type) const;
     RTC::TransportTuple* getSelectedTuple() const;
+    void sendRtcpRemb(uint32_t ssrc, size_t bit_rate);
+    void sendRtcpPli(uint32_t ssrc);
 
 private:
     void onSendSockData(const char *buf, size_t len, bool flush = true);