diff --git a/conf/config.ini b/conf/config.ini
index c1c85b79..d49d586e 100644
--- a/conf/config.ini
+++ b/conf/config.ini
@@ -50,9 +50,9 @@ auto_close=0
 #此参数不应大于播放器超时时间;单位毫秒
 continue_push_ms=15000
 # 是否启用音频转码
-# 转码主要基于包过滤机制，代码实现详见 MultiMediaSourceMuxer 的 onTrackReady 和 onTrackFrame 方法，当前已实现
-# 1. AAC转Opus, 然后传入rtc中
-# 2. 非AAC音频转aac，然后传入rtmp中
+# 主要实现进出RTC音频流的自动转码，代码实现详见 RtcMediaSource.h/cpp，当前实现
+# 1. 将RTC收到的Opus音频流，转成AAC流，并传到MultiMediaSourceMuxer中的其他流中
+# 2. 将从MultiMediaSourceMuxer中收到的AAC音频流，转成OPUS，并送到RTC流中
 # 音频转码底层使用FFMPEG来实现，需要打开FFMPEG, 即编译时必须指定 -DENABLE_FFMPEG=1,  当前已知支持FFMPEG 4.x 5.x 和 6.0，
 # 在ubuntu中可通过以下指令来安装: apt-get install libavcodec-dev libavutil-dev libswscale-dev libresample-dev
 # 此外音频转码正常都是用于webrtc的，一般也会开启WEBRTC, 即-DENABLE_WEBRTC=1, 此前必须自己装好libsrtp库, 安装过程详见wiki
@@ -354,6 +354,10 @@ preferredCodecA=PCMU,PCMA,opus,mpeg4-generic
 #rtc支持的视频codec类型,在前面的优先级更高
 #以下范例为所有支持的视频codec
 preferredCodecV=H264,H265,AV1,VP9,VP8
+# 是否开启RTC协议的G711转码，开启后
+# 能将传给rtc的g711音频转成opus
+# 将由rtc流入g711音频转成aac，并转给其他协议流
+transcodeG711=0
 
 [srt]
 #srt播放推流、播放超时时间,单位秒
diff --git a/src/Common/MultiMediaSourceMuxer.cpp b/src/Common/MultiMediaSourceMuxer.cpp
index 11b51236..448bed88 100644
--- a/src/Common/MultiMediaSourceMuxer.cpp
+++ b/src/Common/MultiMediaSourceMuxer.cpp
@@ -11,9 +11,6 @@
 #include <math.h>
 #include "Common/config.h"
 #include "MultiMediaSourceMuxer.h"
-#include "Extension/AAC.h"
-#include "Extension/Opus.h"
-#include "Extension/G711.h"
 #include "Rtp/RtpSender.h"
 #include "Record/HlsRecorder.h"
 #include "Record/HlsMediaSource.h"
@@ -21,9 +18,8 @@
 #include "Rtmp/RtmpMediaSourceMuxer.h"
 #include "TS/TSMediaSourceMuxer.h"
 #include "FMP4/FMP4MediaSourceMuxer.h"
-
-#ifdef ENABLE_FFMPEG
-#include "Codec/Transcode.h"
+#if defined(ENABLE_WEBRTC)
+#include "webrtc/RtcMediaSource.h"
 #endif
 using namespace std;
 using namespace toolkit;
@@ -122,7 +118,7 @@ MultiMediaSourceMuxer::MultiMediaSourceMuxer(const MediaTuple& tuple, float dur_
     }
     if (option.enable_rtc) {
 #if defined(ENABLE_WEBRTC)
-        _rtc = std::make_shared<RtspMediaSourceMuxer>(_tuple, option, std::make_shared<TitleSdp>(dur_sec), RTC_SCHEMA);
+        _rtc = std::make_shared<RtcMediaSourceMuxer>(_tuple, option, std::make_shared<TitleSdp>(dur_sec));
 #endif
     }
     if (option.enable_hls) {
@@ -137,14 +133,6 @@ MultiMediaSourceMuxer::MultiMediaSourceMuxer(const MediaTuple& tuple, float dur_
     if (option.enable_ts) {
         _ts = dynamic_pointer_cast<TSMediaSourceMuxer>(Recorder::createRecorder(Recorder::type_ts, _tuple, option));
     }
-    if (option.audio_transcode) {
-#if defined(ENABLE_FFMPEG)
-        InfoL << "enable audio_transcode";
-#else
-        InfoL << "without ffmpeg disable audio_transcode";
-        _option.audio_transcode = false;
-#endif
-    }
 
     if (option.enable_fmp4) {
         _fmp4 = dynamic_pointer_cast<FMP4MediaSourceMuxer>(Recorder::createRecorder(Recorder::type_fmp4, _tuple, option));
@@ -407,79 +395,9 @@ std::shared_ptr<MultiMediaSourceMuxer> MultiMediaSourceMuxer::getMuxer(MediaSour
 
 bool MultiMediaSourceMuxer::onTrackReady(const Track::Ptr &track) {
     bool ret = false;
-    auto rtmp = _rtmp;
-    auto rtc = _rtc;
-#if defined(ENABLE_FFMPEG)
-    if (_option.audio_transcode) {
-        if (track->getCodecId() == CodecAAC) {
-            if (rtmp) {
-                rtmp->addTrack(track);
-                rtmp = nullptr;
-            }
-            _audio_dec = nullptr;
-            _audio_enc = nullptr;
-            _opus_mute_maker = nullptr;
-            if (rtc) {
-                Track::Ptr newTrack(new OpusTrack());
-                GET_CONFIG(int, bitrate, General::kOpusBitrate);
-                newTrack->setBitRate(bitrate);
-                rtc->addTrack(newTrack);
-                rtc = nullptr;
-                if (!hasMuteAudio()) {
-                    // aac to opus
-                    _audio_dec.reset(new FFmpegDecoder(track));
-                    _audio_enc.reset(new FFmpegEncoder(newTrack));
-                    _audio_dec->setOnDecode([this](const FFmpegFrame::Ptr & frame) {
-                        _audio_enc->inputFrame(frame, false);
-                    });
-                    _audio_enc->setOnEncode([this](const Frame::Ptr& frame) {
-                        // fill data to _rtc
-                        if (_rtc && _rtc->isEnabled())
-                            _rtc->inputFrame(frame);
-                     });
-                }
-                else {
-                    _opus_mute_maker = std::make_shared<MuteAudioMaker>(CodecOpus);
-                    _opus_mute_maker->addDelegate([this](const Frame::Ptr &frame) {
-                        if (_rtc && _rtc->isEnabled())
-                            _rtc->inputFrame(frame);
-                        return true;
-                    });
-                }
-            }
-        }
-        else if (track->getTrackType() == TrackAudio) {
-            if (rtc) {
-                rtc->addTrack(track);
-                rtc = nullptr;
-            }
-            _audio_dec = nullptr;
-            _audio_enc = nullptr;
-            _opus_mute_maker = nullptr;
-            if (rtmp) {
-                Track::Ptr newTrack(new AACTrack(44100, std::dynamic_pointer_cast<AudioTrack>(track)->getAudioChannel()));
-                GET_CONFIG(int, bitrate, General::kAacBitrate);
-                newTrack->setBitRate(bitrate);
-                rtmp->addTrack(newTrack);
-                rtmp = nullptr;
-
-                _audio_dec.reset(new FFmpegDecoder(track));
-                _audio_enc.reset(new FFmpegEncoder(newTrack));
-                _audio_dec->setOnDecode([this](const FFmpegFrame::Ptr & frame) {
-                    _audio_enc->inputFrame(frame, false);
-                });
-                _audio_enc->setOnEncode([this](const Frame::Ptr& frame) {
-                    // fill aac frame to rtmp
-                    if (_rtmp && _rtmp->isEnabled())
-                        _rtmp->inputFrame(frame);
-                });
-            }
-        }
-    }
-#endif
-    if (rtc && rtc->addTrack(track))
+    if (_rtc && _rtc->addTrack(track))
         ret = true;
-    if (rtmp && rtmp->addTrack(track))
+    if (_rtmp && _rtmp->addTrack(track))
         ret = true;
     if (_rtsp && _rtsp->addTrack(track))
         ret = true;
@@ -579,11 +497,6 @@ void MultiMediaSourceMuxer::resetTracks() {
     if (_rtc) {
         _rtc->resetTracks();
     }
-#if defined(ENABLE_FFMPEG)
-    _audio_dec = nullptr;
-    _audio_dec = nullptr;
-    _opus_mute_maker = nullptr;
-#endif
     if (_fmp4) {
         _fmp4->resetTracks();
     }
@@ -606,36 +519,10 @@ bool MultiMediaSourceMuxer::onTrackFrame(const Frame::Ptr &frame_in) {
     }
 
     bool ret = false;
-    RtspMediaSourceMuxer::Ptr rtc;
-    RtmpMediaSourceMuxer::Ptr rtmp;
-    if (_rtc && _rtc->isEnabled())
-        rtc = _rtc;
-    if (_rtmp && _rtmp->isEnabled())
-        rtmp = _rtmp;
-#if defined(ENABLE_FFMPEG)
-    if (_option.audio_transcode) {
-        if (frame->getCodecId() == CodecAAC) {
-            if (rtc) {
-                if (_audio_dec && rtc->readerCount())
-                    _audio_dec->inputFrame(frame, true, false, false);
-                rtc = nullptr;
-            }
-        }
-        else if (frame->getTrackType() == TrackAudio) {
-            if (rtmp) {
-                if (_audio_dec && (rtmp->readerCount() || !rtmp->isRegisted()))
-                    _audio_dec->inputFrame(frame, true, false, false);
-                rtmp = nullptr;
-            }
-        } else if (_opus_mute_maker && rtc) {
-            _opus_mute_maker->inputFrame(frame);
-        }
-    }
-#endif
-    if (rtc && rtc->inputFrame(frame))
+    if (_rtc && _rtc->inputFrame(frame))
       ret = true;
 
-    if (rtmp && rtmp->inputFrame(frame))
+    if (_rtmp && _rtmp->inputFrame(frame))
       ret = true;
 
     if (_rtsp && _rtsp->inputFrame(frame))
diff --git a/src/Common/MultiMediaSourceMuxer.h b/src/Common/MultiMediaSourceMuxer.h
index 5042ef65..755bc33a 100644
--- a/src/Common/MultiMediaSourceMuxer.h
+++ b/src/Common/MultiMediaSourceMuxer.h
@@ -24,10 +24,6 @@ class RtmpMediaSourceMuxer;
 class TSMediaSourceMuxer;
 class FMP4MediaSourceMuxer;
 class RtpSender;
-#ifdef ENABLE_FFMPEG
-class FFmpegDecoder;
-class FFmpegEncoder;
-#endif
 
 class MultiMediaSourceMuxer : public MediaSourceEventInterceptor, public MediaSink, public std::enable_shared_from_this<MultiMediaSourceMuxer>{
 public:
@@ -178,11 +174,6 @@ private:
     std::shared_ptr<RtspMediaSourceMuxer> _rtsp;
     std::shared_ptr<TSMediaSourceMuxer> _ts;
     std::shared_ptr<RtspMediaSourceMuxer> _rtc;
-#if defined(ENABLE_FFMPEG)
-    MuteAudioMaker::Ptr _opus_mute_maker;
-    std::shared_ptr<FFmpegDecoder> _audio_dec;
-    std::shared_ptr<FFmpegEncoder> _audio_enc;
-#endif
     MediaSinkInterface::Ptr _mp4;
     std::shared_ptr<HlsRecorder> _hls;
     std::shared_ptr<HlsFMP4Recorder> _hls_fmp4;
diff --git a/webrtc/RtcMediaSource.cpp b/webrtc/RtcMediaSource.cpp
new file mode 100644
index 00000000..6b050daa
--- /dev/null
+++ b/webrtc/RtcMediaSource.cpp
@@ -0,0 +1,199 @@
+#include "RtcMediaSource.h"
+#include "Common/config.h"
+#include "Codec/Transcode.h"
+#include "Extension/AAC.h"
+#include "Extension/Opus.h"
+#include "Extension/G711.h"
+// for RTC configure
+#include "WebRtcTransport.h"
+namespace mediakit {
+
+bool needTransToOpus(CodecId codec) {
+  GET_CONFIG(int, transG711, Rtc::kTranscodeG711);
+  switch (codec)
+  {
+  case CodecG711U:
+  case CodecG711A:
+      return transG711;
+  case CodecAAC:
+      return true;
+  default:
+      return false;
+  }
+}
+
+bool needTransToAac(CodecId codec) {
+    GET_CONFIG(int, transG711, Rtc::kTranscodeG711);
+    switch (codec)
+    {
+    case CodecG711U:
+    case CodecG711A:
+        return transG711;
+    case CodecOpus:
+        return true;
+    default:
+        return false;
+    }
+}
+
+RtcMediaSourceMuxer::RtcMediaSourceMuxer(const MediaTuple& tuple, const ProtocolOption &option, const TitleSdp::Ptr &title) 
+  : RtspMediaSourceMuxer(tuple, option, title, RTC_SCHEMA)
+{
+  if (_option.audio_transcode) {
+#ifndef ENABLE_FFMPEG
+      WarnL << "without ffmpeg, skip transcode setting";
+      _option.audio_transcode = false;
+#endif
+  }
+}
+
+RtspMediaSource::Ptr RtcMediaSourceImp::clone(const std::string &stream) {
+    auto tuple = _tuple;
+    tuple.stream = stream;
+    auto src_imp = std::make_shared<RtcMediaSourceImp>(tuple);
+    src_imp->setSdp(getSdp());
+    src_imp->setProtocolOption(getProtocolOption());
+    return src_imp;
+}
+
+bool RtcMediaSourceMuxer::inputFrame(const Frame::Ptr &frame)
+{
+  if (_clear_cache && _on_demand) {
+    _clear_cache = false;
+    _media_src->clearCache();
+  }
+  if (_enabled || !_on_demand) {
+#if defined(ENABLE_FFMPEG)
+    if (_option.audio_transcode && needTransToOpus(frame->getCodecId())) {
+      if (!_audio_dec) { // addTrack可能没调, 这边根据情况再调一次
+        Track::Ptr track;
+        switch (frame->getCodecId())
+        {
+        case CodecAAC:
+          if (frame->prefixSize()) {
+            std::string cfg = makeAacConfig((uint8_t *)(frame->data()), frame->prefixSize());
+            track = std::make_shared<AACTrack>(cfg);
+          }
+          else {
+            track = std::make_shared<AACTrack>(44100, 2);
+          }
+          break;
+        case CodecG711A:
+        case CodecG711U:
+          track.reset(new G711Track(frame->getCodecId()));
+          break;
+        default:
+          break;
+        }
+        if (track)
+          addTrack(track);
+        if (!_audio_dec) return false;
+      }
+      if (readerCount()) {
+        _audio_dec->inputFrame(frame, true, false);
+        if (!_count)
+          InfoL << "start transcode " << frame->getCodecName() << "," << frame->pts() << "->Opus";
+        _count++;
+      }
+      else if (_count) {
+        InfoL << "stop transcode with " << _count << " items";
+        _count = 0;
+      }
+      return true;
+    }
+#endif
+    return RtspMuxer::inputFrame(frame);
+  }
+  return false;
+}
+
+#if defined(ENABLE_FFMPEG)
+
+bool RtcMediaSourceMuxer::addTrack(const Track::Ptr & track)
+{
+  Track::Ptr newTrack = track;
+  if (_option.audio_transcode && needTransToOpus(track->getCodecId())) {
+    newTrack = std::make_shared<OpusTrack>();
+    GET_CONFIG(int, bitrate, General::kOpusBitrate);
+    newTrack->setBitRate(bitrate);
+    _audio_dec.reset(new FFmpegDecoder(track));
+    _audio_enc.reset(new FFmpegEncoder(newTrack));
+    // aac to opus
+    _audio_dec->setOnDecode([this](const FFmpegFrame::Ptr & frame) {
+      _audio_enc->inputFrame(frame, false);
+    });
+    _audio_enc->setOnEncode([this](const Frame::Ptr& frame) {
+        RtspMuxer::inputFrame(frame);
+    });
+  }
+  return RtspMuxer::addTrack(newTrack);
+}
+
+
+void RtcMediaSourceMuxer::resetTracks()
+{
+  RtspMuxer::resetTracks();
+  _audio_dec = nullptr;
+  _audio_enc = nullptr;
+  if (_count) {
+    InfoL << "stop transcode with " << _count << " items";
+    _count = 0;
+  }
+}
+
+bool RtcMediaSourceImp::addTrack(const Track::Ptr &track)
+{
+  if (_muxer) {
+    Track::Ptr newTrack = track;
+    if (_option.audio_transcode && needTransToAac(track->getCodecId())) {
+      newTrack.reset(new AACTrack(44100, 2));
+      GET_CONFIG(int, bitrate, General::kAacBitrate);
+      newTrack->setBitRate(bitrate);
+      _audio_dec.reset(new FFmpegDecoder(track));
+      _audio_enc.reset(new FFmpegEncoder(newTrack));
+      // hook data to newTack
+      track->addDelegate([this](const Frame::Ptr &frame) -> bool {
+        if (_all_track_ready && 0 == _muxer->totalReaderCount()) {
+          if (_count) {
+            InfoL << "stop transcode with " << _count << " items";
+            _count = 0;
+          }
+          return true;
+        }
+        if (_audio_dec) {
+          if (!_count)
+            InfoL << "start transcode " << frame->getCodecName() << "," << frame->pts() << "->AAC";
+          _count++;
+          _audio_dec->inputFrame(frame, true, false);
+        }
+        return true;
+      });
+      _audio_dec->setOnDecode([this](const FFmpegFrame::Ptr & frame) {
+        _audio_enc->inputFrame(frame, false);
+      });
+      _audio_enc->setOnEncode([newTrack](const Frame::Ptr& frame) {
+        newTrack->inputFrame(frame);
+      });
+    }
+
+    if (_muxer->addTrack(newTrack)) {
+      newTrack->addDelegate(_muxer);
+      return true;
+    }
+  }
+  return false;
+}
+
+void RtcMediaSourceImp::resetTracks()
+{
+  RtspMediaSourceImp::resetTracks();
+  _audio_dec = nullptr;
+  _audio_enc = nullptr;
+  if (_count) {
+    InfoL << "stop transcode with " << _count << " items";
+    _count = 0;
+  }
+}
+
+#endif
+}
diff --git a/webrtc/RtcMediaSource.h b/webrtc/RtcMediaSource.h
new file mode 100644
index 00000000..73109d10
--- /dev/null
+++ b/webrtc/RtcMediaSource.h
@@ -0,0 +1,60 @@
+#ifndef ZLMEDIAKIT_RTCMEDIASOURCE_H
+#define ZLMEDIAKIT_RTCMEDIASOURCE_H
+
+#include "Rtsp/RtspMediaSourceMuxer.h"
+#include "Rtsp/RtspMediaSourceImp.h"
+namespace mediakit {
+class FFmpegDecoder;
+class FFmpegEncoder;
+bool needTransToOpus(CodecId codec);
+bool needTransToAac(CodecId codec);
+
+class RtcMediaSourceImp : public RtspMediaSourceImp {
+public:
+    typedef std::shared_ptr<RtcMediaSourceImp> Ptr;
+
+    RtcMediaSourceImp(const MediaTuple& tuple, int ringSize = RTP_GOP_SIZE)
+        : RtspMediaSourceImp(tuple, RTC_SCHEMA, ringSize) {
+    }
+
+    RtspMediaSource::Ptr clone(const std::string &stream);
+#if defined(ENABLE_FFMPEG)
+    ~RtcMediaSourceImp() override { resetTracks(); }
+    /**
+     * _demuxer触发的添加Track事件
+     */
+    bool addTrack(const Track::Ptr &track) override;
+    void resetTracks() override;
+private:
+    int _count = 0;
+    std::shared_ptr<FFmpegDecoder> _audio_dec;
+    std::shared_ptr<FFmpegEncoder> _audio_enc;
+#endif
+};
+
+class RtcMediaSourceMuxer : public RtspMediaSourceMuxer {
+public:
+    typedef std::shared_ptr<RtcMediaSourceMuxer> Ptr;
+
+    RtcMediaSourceMuxer( const MediaTuple& tuple,
+                         const ProtocolOption &option,
+                         const TitleSdp::Ptr &title = nullptr);
+
+
+    bool inputFrame(const Frame::Ptr &frame) override;
+
+#if defined(ENABLE_FFMPEG)
+    ~RtcMediaSourceMuxer() override{resetTracks();}
+
+    bool addTrack(const Track::Ptr & track) override;
+    void resetTracks() override;
+
+private:
+    int _count = 0;
+    std::shared_ptr<FFmpegDecoder> _audio_dec;
+    std::shared_ptr<FFmpegEncoder> _audio_enc;
+#endif
+};
+
+}
+#endif
\ No newline at end of file
diff --git a/webrtc/WebRtcPusher.cpp b/webrtc/WebRtcPusher.cpp
index 0a62c0f3..3253bb26 100644
--- a/webrtc/WebRtcPusher.cpp
+++ b/webrtc/WebRtcPusher.cpp
@@ -10,7 +10,7 @@
 
 #include "WebRtcPusher.h"
 #include "Common/config.h"
-#include "Rtsp/RtspMediaSourceImp.h"
+#include "RtcMediaSource.h"
 
 using namespace std;
 
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index 2f8f627a..4d540ede 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -18,7 +18,7 @@
 #include "Rtsp/Rtsp.h"
 #include "Rtsp/RtpReceiver.h"
 #include "WebRtcTransport.h"
-
+#include "RtcMediaSource.h"
 #include "WebRtcEchoTest.h"
 #include "WebRtcPlayer.h"
 #include "WebRtcPusher.h"
@@ -45,6 +45,9 @@ const string kTimeOutSec = RTC_FIELD "timeoutSec";
 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";
+// 是否转码G711音频，做到: 出rtc将g711转成aac，入rtc将g711转成opus
+const string kTranscodeG711 = RTC_FIELD "transcodeG711";
+
 // webrtc单端口udp服务器
 const string kPort = RTC_FIELD "port";
 
@@ -56,6 +59,7 @@ static onceToken token([]() {
     mINI::Instance()[kRembBitRate] = 0;
     mINI::Instance()[kPort] = 0;
     mINI::Instance()[kTcpPort] = 0;
+    mINI::Instance()[kTranscodeG711] = 0;
 });
 
 } // namespace RTC
@@ -1206,7 +1210,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
         }
 
         if (!push_src) {
-            push_src = std::make_shared<RtspMediaSourceImp>(info, schema);
+            push_src = std::make_shared<RtcMediaSourceImp>(info);
             push_src_ownership = push_src->getOwnership();
             push_src->setProtocolOption(option);
         }
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index 2534ca3b..96891dfe 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -33,6 +33,7 @@ namespace Rtc {
 extern const std::string kPort;
 extern const std::string kTcpPort;
 extern const std::string kTimeOutSec;
+extern const std::string kTranscodeG711;
 }//namespace RTC
 
 class WebRtcInterface {