diff --git a/trunk/auto/depends.sh b/trunk/auto/depends.sh
index 57596ecbf..4cf1b7e0d 100755
--- a/trunk/auto/depends.sh
+++ b/trunk/auto/depends.sh
@@ -695,15 +695,16 @@ fi
 # ffmpeg-fit, for WebRTC to transcode AAC with Opus.
 #####################################################################################
 if [[ $SRS_FFMPEG_FIT == YES ]]; then
-    FFMPEG_OPTIONS=""
     if [[ $SRS_CROSS_BUILD == YES ]]; then
-      FFMPEG_CONFIGURE=./configure
+      FFMPEG_CONFIGURE="./configure"
     else
       FFMPEG_CONFIGURE="env PKG_CONFIG_PATH=${SRS_DEPENDS_LIBS}/opus/lib/pkgconfig ./configure"
     fi
 
+    # Disable all features, note that there are still some options need to be disabled.
+    FFMPEG_OPTIONS="--disable-everything"
     # Disable all asm for FFmpeg, to compatible with ARM CPU.
-    FFMPEG_OPTIONS="--disable-asm --disable-x86asm --disable-inline-asm"
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-asm --disable-x86asm --disable-inline-asm"
     # Only build static libraries if no shared FFmpeg.
     if [[ $SRS_SHARED_FFMPEG == YES ]]; then
         FFMPEG_OPTIONS="$FFMPEG_OPTIONS --enable-shared"
@@ -725,6 +726,17 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
     else
         FFMPEG_OPTIONS="$FFMPEG_OPTIONS --enable-decoder=libopus --enable-encoder=libopus --enable-libopus"
     fi
+    # Disable features of ffmpeg.
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-avdevice --disable-avformat --disable-swscale --disable-postproc --disable-avfilter --disable-network"
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-dwt --disable-error-resilience --disable-lsp --disable-lzo --disable-faan --disable-pixelutils"
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-hwaccels --disable-devices --disable-audiotoolbox --disable-videotoolbox --disable-cuvid"
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-d3d11va --disable-dxva2 --disable-ffnvcodec --disable-nvdec --disable-nvenc --disable-v4l2-m2m --disable-vaapi"
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-vdpau --disable-appkit --disable-coreimage --disable-avfoundation --disable-securetransport --disable-iconv"
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --disable-lzma --disable-sdl2"
+    # Enable FFmpeg native AAC encoder and decoder.
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --enable-decoder=aac --enable-decoder=aac_fixed --enable-decoder=aac_latm --enable-encoder=aac"
+    # Enable FFmpeg native MP3 decoder, which depends on dct.
+    FFMPEG_OPTIONS="$FFMPEG_OPTIONS --enable-decoder=mp3 --enable-dct"
 
     if [[ -f ${SRS_OBJS}/${SRS_PLATFORM}/3rdpatry/ffmpeg/lib/libavcodec.a ]]; then
         rm -rf ${SRS_OBJS}/ffmpeg && cp -rf ${SRS_OBJS}/${SRS_PLATFORM}/3rdpatry/ffmpeg ${SRS_OBJS}/ &&
@@ -736,18 +748,9 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
         cp -rf ${SRS_WORKDIR}/3rdparty/ffmpeg-4-fit ${SRS_OBJS}/${SRS_PLATFORM}/ &&
         (
             cd ${SRS_OBJS}/${SRS_PLATFORM}/ffmpeg-4-fit &&
-            $FFMPEG_CONFIGURE \
-              --prefix=${SRS_DEPENDS_LIBS}/${SRS_PLATFORM}/3rdpatry/ffmpeg --pkg-config=pkg-config \
-              --pkg-config-flags="--static" --extra-libs="-lpthread" --extra-libs="-lm" \
-              --disable-everything ${FFMPEG_OPTIONS} \
-              --disable-programs --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages \
-              --disable-avdevice --disable-avformat --disable-swscale --disable-postproc --disable-avfilter --disable-network \
-              --disable-dct --disable-dwt --disable-error-resilience --disable-lsp --disable-lzo --disable-faan --disable-pixelutils \
-              --disable-hwaccels --disable-devices --disable-audiotoolbox --disable-videotoolbox --disable-cuvid \
-              --disable-d3d11va --disable-dxva2 --disable-ffnvcodec --disable-nvdec --disable-nvenc --disable-v4l2-m2m --disable-vaapi \
-              --disable-vdpau --disable-appkit --disable-coreimage --disable-avfoundation --disable-securetransport --disable-iconv \
-              --disable-lzma --disable-sdl2 --enable-decoder=aac --enable-decoder=aac_fixed --enable-decoder=aac_latm \
-              --enable-encoder=aac
+            $FFMPEG_CONFIGURE --prefix=${SRS_DEPENDS_LIBS}/${SRS_PLATFORM}/3rdpatry/ffmpeg \
+                --pkg-config=pkg-config --pkg-config-flags='--static' --extra-libs='-lpthread' --extra-libs='-lm' \
+                ${FFMPEG_OPTIONS}
         ) &&
         # See https://www.laoyuyu.me/2019/05/23/android/clang_compile_ffmpeg/
         if [[ $SRS_CROSS_BUILD == YES ]]; then
diff --git a/trunk/conf/mp3.rtc.conf b/trunk/conf/mp3.rtc.conf
new file mode 100644
index 000000000..6268e0ae5
--- /dev/null
+++ b/trunk/conf/mp3.rtc.conf
@@ -0,0 +1,32 @@
+listen 1935;
+max_connections 1000;
+daemon off;
+srs_log_tank console;
+http_server {
+    enabled on;
+    listen 8080;
+    dir ./objs/nginx/html;
+}
+rtc_server {
+    enabled on;
+    listen 8000; # UDP port
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+    candidate $CANDIDATE;
+}
+vhost __defaultVhost__ {
+    http_remux {
+        enabled on;
+        mount [vhost]/[app]/[stream].flv;
+    }
+    hls {
+        enabled on;
+        hls_acodec mp3;
+    }
+    rtc {
+        enabled     on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
+        rtc_to_rtmp on;
+    }
+}
diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
index d63452553..ad9dd1970 100644
--- a/trunk/doc/CHANGELOG.md
+++ b/trunk/doc/CHANGELOG.md
@@ -21,6 +21,7 @@ The changelog for SRS.
 
 ## SRS 5.0 Changelog
 
+* v5.0, 2022-12-26, For [#296](https://github.com/ossrs/srs/issues/296): MP3: Convert RTMP(MP3) to WebRTC(OPUS). v5.0.118
 * v5.0, 2022-12-25, For [#296](https://github.com/ossrs/srs/issues/296): MP3: Support dump stream information. v5.0.117
 * v5.0, 2022-12-25, For [#296](https://github.com/ossrs/srs/issues/296): MP3: Support mp3 for RTMP/HLS/HTTP-FLV/HTTP-TS/HLS etc. v5.0.116
 * v5.0, 2022-12-24, Fix [#3328](https://github.com/ossrs/srs/issues/3328): Docker: Avoiding duplicated copy files. v5.0.115
diff --git a/trunk/src/app/srs_app_rtc_codec.cpp b/trunk/src/app/srs_app_rtc_codec.cpp
index 9a8b2575c..50e983321 100644
--- a/trunk/src/app/srs_app_rtc_codec.cpp
+++ b/trunk/src/app/srs_app_rtc_codec.cpp
@@ -14,6 +14,8 @@ static const AVCodec* srs_find_decoder_by_id(SrsAudioCodecId id)
 {
     if (id == SrsAudioCodecIdAAC) {
         return avcodec_find_decoder_by_name("aac");
+    } else if (id == SrsAudioCodecIdMP3) {
+        return avcodec_find_decoder_by_name("mp3");
     } else if (id == SrsAudioCodecIdOpus) {
         const AVCodec* codec = avcodec_find_decoder_by_name("libopus");
         if (!codec) {
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index 0c760ccd2..4f57e5944 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -717,6 +717,7 @@ SrsRtcFromRtmpBridge::SrsRtcFromRtmpBridge(SrsRtcSource* source)
     source_ = source;
     format = new SrsRtmpFormat();
     codec_ = new SrsAudioTranscoder();
+    latest_codec_ = SrsAudioCodecIdForbidden;
     rtmp_to_rtc = false;
     keep_bframe = false;
     merge_nalus = false;
@@ -766,12 +767,6 @@ srs_error_t SrsRtcFromRtmpBridge::initialize(SrsRequest* r)
 
         // Setup the SPS/PPS parsing strategy.
         format->try_annexb_first = _srs_config->try_annexb_first(r->vhost);
-
-        int bitrate = 48000; // The output bitrate in bps.
-        if ((err = codec_->initialize(SrsAudioCodecIdAAC, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate,
-                                      bitrate)) != srs_success) {
-            return srs_error_wrap(err, "init codec");
-        }
     }
 
     keep_bframe = _srs_config->get_rtc_keep_bframe(req->vhost);
@@ -831,6 +826,11 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
         return srs_error_wrap(err, "format consume audio");
     }
 
+    // Try to init codec when startup or codec changed.
+    if (format->acodec && (err = init_codec(format->acodec->id)) != srs_success) {
+        return srs_error_wrap(err, "init codec");
+    }
+
     // Ignore if no format->acodec, it means the codec is not parsed, or unknown codec.
     // @issue https://github.com/ossrs/srs/issues/1506#issuecomment-562079474
     if (!format->acodec) {
@@ -843,14 +843,18 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
         return err;
     }
 
+    // ignore sequence header
+    srs_assert(format->audio);
+
+    if (format->acodec->id == SrsAudioCodecIdMP3) {
+        return transcode(format->audio);
+    }
+
     // When drop aac audio packet, never transcode.
     if (acodec != SrsAudioCodecIdAAC) {
         return err;
     }
 
-    // ignore sequence header
-    srs_assert(format->audio);
-
     char* adts_audio = NULL;
     int nn_adts_audio = 0;
     // TODO: FIXME: Reserve 7 bytes header when create shared message.
@@ -875,6 +879,35 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
     return err;
 }
 
+srs_error_t SrsRtcFromRtmpBridge::init_codec(SrsAudioCodecId codec)
+{
+    srs_error_t err = srs_success;
+
+    // Ignore if not changed.
+    if (latest_codec_ == codec) return err;
+
+    // Create a new codec.
+    srs_freep(codec_);
+    codec_ = new SrsAudioTranscoder();
+
+    // Initialize the codec according to the codec in stream.
+    int bitrate = 48000; // The output bitrate in bps.
+    if ((err = codec_->initialize(codec, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate, bitrate)) != srs_success) {
+        return srs_error_wrap(err, "init codec=%d", codec);
+    }
+
+    // Update the latest codec in stream.
+    if (latest_codec_ == SrsAudioCodecIdForbidden) {
+        srs_trace("RTMP2RTC: Init audio codec to %d(%s)", codec, srs_audio_codec_id2str(codec).c_str());
+    } else {
+        srs_trace("RTMP2RTC: Switch audio codec %d(%s) to %d(%s)", latest_codec_, srs_audio_codec_id2str(latest_codec_).c_str(),
+            codec, srs_audio_codec_id2str(codec).c_str());
+    }
+    latest_codec_ = codec;
+
+    return err;
+}
+
 srs_error_t SrsRtcFromRtmpBridge::transcode(SrsAudioFrame* audio)
 {
     srs_error_t err = srs_success;
diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
index d7ec8dc07..d38e6d2ee 100644
--- a/trunk/src/app/srs_app_rtc_source.hpp
+++ b/trunk/src/app/srs_app_rtc_source.hpp
@@ -254,6 +254,7 @@ private:
     SrsMetaCache* meta;
 private:
     bool rtmp_to_rtc;
+    SrsAudioCodecId latest_codec_;
     SrsAudioTranscoder* codec_;
     bool keep_bframe;
     bool merge_nalus;
@@ -272,6 +273,7 @@ public:
     virtual void on_unpublish();
     virtual srs_error_t on_audio(SrsSharedPtrMessage* msg);
 private:
+    srs_error_t init_codec(SrsAudioCodecId codec);
     srs_error_t transcode(SrsAudioFrame* audio);
     srs_error_t package_opus(SrsAudioFrame* audio, SrsRtpPacket* pkt);
 public:
diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp
index eff14fa38..25a9a934d 100644
--- a/trunk/src/core/srs_core_version5.hpp
+++ b/trunk/src/core/srs_core_version5.hpp
@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       5
 #define VERSION_MINOR       0
-#define VERSION_REVISION    117
+#define VERSION_REVISION    118
 
 #endif