diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 2b7da83..cc2fa46 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -673,6 +673,9 @@
     // Added to be able to control rollout of this feature.
     bool enable_implicit_rollback = false;
 
+    // Whether network condition based codec switching is allowed.
+    absl::optional<bool> allow_codec_switching;
+
     //
     // Don't forget to update operator== if adding something.
     //
diff --git a/media/base/media_channel.cc b/media/base/media_channel.cc
index 1bd20dc..579cbc6 100644
--- a/media/base/media_channel.cc
+++ b/media/base/media_channel.cc
@@ -47,6 +47,8 @@
   // Placeholder should be pure virtual once internal supports it.
 }
 
+void MediaChannel::SetVideoCodecSwitchingEnabled(bool enabled) {}
+
 MediaSenderInfo::MediaSenderInfo() = default;
 MediaSenderInfo::~MediaSenderInfo() = default;
 
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 582d29c..c49f2ec 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -243,6 +243,9 @@
       uint32_t ssrc,
       rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor);
 
+  // Enable network condition based codec switching.
+  virtual void SetVideoCodecSwitchingEnabled(bool enabled);
+
   // Base method to send packet using NetworkInterface.
   bool SendPacket(rtc::CopyOnWriteBuffer* packet,
                   const rtc::PacketOptions& options) {
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 9ea80cc..cab4e12 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -739,6 +739,12 @@
   invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [this, conf] {
     RTC_DCHECK_RUN_ON(&thread_checker_);
 
+    if (!allow_codec_switching_) {
+      RTC_LOG(LS_INFO) << "Encoder switch requested but codec switching has"
+                       << " not been enabled.";
+      return;
+    }
+
     for (VideoCodecSettings codec_setting : negotiated_codecs_) {
       if (codec_setting.codec.name == conf.codec_name) {
         if (conf.param) {
@@ -1678,6 +1684,13 @@
   }
 }
 
+void WebRtcVideoChannel::SetVideoCodecSwitchingEnabled(bool enabled) {
+  invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [this, enabled] {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    allow_codec_switching_ = enabled;
+  });
+}
+
 bool WebRtcVideoChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
                                                       int delay_ms) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 5e5ab6e..2493edb 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -172,6 +172,8 @@
                          rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
                              frame_encryptor) override;
 
+  void SetVideoCodecSwitchingEnabled(bool enabled) override;
+
   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
 
   absl::optional<int> GetBaseMinimumPlayoutDelayMs(
@@ -569,6 +571,8 @@
   std::unique_ptr<UnhandledPacketsBuffer> unknown_ssrc_packet_buffer_
       RTC_GUARDED_BY(thread_checker_);
 
+  bool allow_codec_switching_ = false;
+
   // In order for the |invoker_| to protect other members from being destructed
   // as they are used in asynchronous tasks it has to be destructed first.
   rtc::AsyncInvoker invoker_;
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 62bbf24..b60ab95 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -2151,6 +2151,7 @@
   parameters.codecs.push_back(vp8);
 
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
+  channel_->SetVideoCodecSwitchingEnabled(true);
 
   VideoCodec codec;
   ASSERT_TRUE(channel_->GetSendCodec(&codec));
@@ -2188,6 +2189,7 @@
   parameters.codecs.push_back(vp8);
 
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
+  channel_->SetVideoCodecSwitchingEnabled(true);
 
   VideoCodec codec;
   ASSERT_TRUE(channel_->GetSendCodec(&codec));
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 494a649..08fbe41 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -784,6 +784,7 @@
     bool offer_extmap_allow_mixed;
     std::string turn_logging_id;
     bool enable_implicit_rollback;
+    absl::optional<bool> allow_codec_switching;
   };
   static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
                 "Did you add something to RTCConfiguration and forget to "
@@ -851,7 +852,8 @@
          crypto_options == o.crypto_options &&
          offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
          turn_logging_id == o.turn_logging_id &&
-         enable_implicit_rollback == o.enable_implicit_rollback;
+         enable_implicit_rollback == o.enable_implicit_rollback &&
+         allow_codec_switching == o.allow_codec_switching;
 }
 
 bool PeerConnectionInterface::RTCConfiguration::operator!=(
@@ -3704,6 +3706,7 @@
   modified_config.use_datagram_transport_for_data_channels_receive_only =
       configuration.use_datagram_transport_for_data_channels_receive_only;
   modified_config.turn_logging_id = configuration.turn_logging_id;
+  modified_config.allow_codec_switching = configuration.allow_codec_switching;
   if (configuration != modified_config) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
                          "Modifying the configuration in an unsupported way.");
@@ -3791,6 +3794,11 @@
         modified_config.active_reset_srtp_params);
   }
 
+  if (modified_config.allow_codec_switching.has_value()) {
+    video_media_channel()->SetVideoCodecSwitchingEnabled(
+        *modified_config.allow_codec_switching);
+  }
+
   configuration_ = modified_config;
   use_media_transport_ = configuration.use_media_transport;
   return RTCError::OK();
diff --git a/sdk/android/api/org/webrtc/PeerConnection.java b/sdk/android/api/org/webrtc/PeerConnection.java
index b981520..e675d13 100644
--- a/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/sdk/android/api/org/webrtc/PeerConnection.java
@@ -548,6 +548,11 @@
     // every offer/answer negotiation.This is only intended to be a workaround for crbug.com/835958
     public boolean activeResetSrtpParams;
 
+    // Whether this client is allowed to switch encoding codec mid-stream. This is a workaround for
+    // a WebRTC bug where the receiver could get confussed if a codec switch happened mid-call.
+    // Null indicates no change to currently configured value.
+    @Nullable public Boolean allowCodecSwitching;
+
     /*
      * Experimental flag that enables a use of media transport. If this is true, the media transport
      * factory MUST be provided to the PeerConnectionFactory.
@@ -619,6 +624,7 @@
       useMediaTransportForDataChannels = false;
       cryptoOptions = null;
       turnLoggingId = null;
+      allowCodecSwitching = null;
     }
 
     @CalledByNative("RTCConfiguration")
@@ -828,6 +834,12 @@
       return activeResetSrtpParams;
     }
 
+    @Nullable
+    @CalledByNative("RTCConfiguration")
+    Boolean getAllowCodecSwitching() {
+      return allowCodecSwitching;
+    }
+
     @CalledByNative("RTCConfiguration")
     boolean getUseMediaTransport() {
       return useMediaTransport;
diff --git a/sdk/android/src/jni/pc/peer_connection.cc b/sdk/android/src/jni/pc/peer_connection.cc
index f40a7bf..9b1cce6 100644
--- a/sdk/android/src/jni/pc/peer_connection.cc
+++ b/sdk/android/src/jni/pc/peer_connection.cc
@@ -279,6 +279,9 @@
   rtc_config->crypto_options =
       JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
 
+  rtc_config->allow_codec_switching = JavaToNativeOptionalBool(
+      jni, Java_RTCConfiguration_getAllowCodecSwitching(jni, j_rtc_config));
+
   ScopedJavaLocalRef<jstring> j_turn_logging_id =
       Java_RTCConfiguration_getTurnLoggingId(jni, j_rtc_config);
   if (!IsNull(jni, j_turn_logging_id)) {