diff --git a/CMakeLists.txt b/CMakeLists.txt
index dea60289..76bc9b34 100755
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -84,10 +84,6 @@ if (NOT EXISTS ${WEBRTCOBJS}/${CMAKE_STATIC_LIBRARY_PREFIX}webrtc${CMAKE_STATIC_
   file(READ ${WEBRTCROOT}/src/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc filecontent)
   string(REPLACE "config.emplace()" "config.emplace(Config())" filecontent "${filecontent}")
   file(WRITE ${WEBRTCROOT}/src/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc "${filecontent}")
-  file(READ ${WEBRTCROOT}/src/build_overrides/build.gni filecontent)
-  string(REPLACE "enable_safe_libcxx = true\n" "enable_safe_libcxx = true\nenable_safe_libstdcxx=true\n" filecontent "${filecontent}")
-  file(WRITE ${WEBRTCROOT}/src/build_overrides/build.gni "${filecontent}")
-
   
   
 
diff --git a/inc/liveaudiosource.h b/inc/liveaudiosource.h
index eb1d0752..fc75ca07 100644
--- a/inc/liveaudiosource.h
+++ b/inc/liveaudiosource.h
@@ -50,47 +50,24 @@ class LiveAudioSource : public webrtc::Notifier<webrtc::AudioSourceInterface>, p
     }
 
     // overide RTSPConnection::Callback
-    virtual bool onNewSession(const char *id, const char *media, const char *codec, const char *sdp) override
+    virtual bool onNewSession(const char *id, const char *media, const char *codec, const char *sdp, unsigned int rtpfrequency, unsigned int channels) override
     {
         bool success = false;
         if (strcmp(media, "audio") == 0)
         {
             RTC_LOG(LS_INFO) << "LiveAudioSource::onNewSession " << media << "/" << codec << " " << sdp;
 
-            // parse sdp to extract freq and channel
-            std::string fmt(sdp);
-            std::transform(fmt.begin(), fmt.end(), fmt.begin(), [](unsigned char c) { return std::tolower(c); });
-            std::string codecstr(codec);
-            std::transform(codecstr.begin(), codecstr.end(), codecstr.begin(), [](unsigned char c) { return std::tolower(c); });
-            size_t pos = fmt.find(codecstr);
-            if (pos != std::string::npos)
-            {
-                fmt.erase(0, pos + strlen(codec));
-                fmt.erase(fmt.find_first_of(" \r\n"));
-                std::istringstream is(fmt);
-                std::string dummy;
-                std::getline(is, dummy, '/');
-                std::string freq;
-                std::getline(is, freq, '/');
-                if (!freq.empty())
-                {
-                    m_freq = std::stoi(freq);
-                }
-                std::string channel;
-                std::getline(is, channel, '/');
-                if (!channel.empty())
-                {
-                    m_channel = std::stoi(channel);
-                }
-            }
-            RTC_LOG(LS_INFO) << "LiveAudioSource::onNewSession codec:" << codecstr << " freq:" << m_freq << " channel:" << m_channel;
+            m_freq = rtpfrequency;
+            m_channel = channels;
+
+            RTC_LOG(LS_INFO) << "LiveAudioSource::onNewSession codec:" << " freq:" << m_freq << " channel:" << m_channel;
             std::map<std::string, std::string> params;
             if (m_channel == 2)
             {
                 params["stereo"] = "1";
             }
 
-            webrtc::SdpAudioFormat format = webrtc::SdpAudioFormat(codecstr, m_freq, m_channel, std::move(params));
+            webrtc::SdpAudioFormat format = webrtc::SdpAudioFormat(codec, m_freq, m_channel, std::move(params));
             if (m_factory->IsSupportedDecoder(format))
             {
                 m_decoder = m_factory->MakeAudioDecoder(format, absl::optional<webrtc::AudioCodecPairId>());
diff --git a/live555helper b/live555helper
index af76e869..834115dc 160000
--- a/live555helper
+++ b/live555helper
@@ -1 +1 @@
-Subproject commit af76e8693cd13ee3f6d8a030616daba6ae873503
+Subproject commit 834115dc53fea5c9ce0121374874ab05bc472d60