diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 8977137f65..2d0abd8156 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -666,6 +666,10 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
     // Added to be able to control rollout of this feature.
     bool enable_implicit_rollback = false;
 
+    // Whether network condition based codec switching is allowed.
+    // TODO(bugs.webrtc.org/11341): Remove this unsupported config value.
+    absl::optional<bool> allow_codec_switching;
+
     // The delay before doing a usage histogram report for long-lived
     // PeerConnections. Used for testing only.
     absl::optional<int> report_usage_pattern_delay_ms;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 46c28bba37..8afc44eb14 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -329,6 +329,7 @@ RTCErrorOr<PeerConnectionInterface::RTCConfiguration> ApplyConfiguration(
   modified_config.active_reset_srtp_params =
       configuration.active_reset_srtp_params;
   modified_config.turn_logging_id = configuration.turn_logging_id;
+  modified_config.allow_codec_switching = configuration.allow_codec_switching;
   modified_config.stable_writable_connection_ping_interval_ms =
       configuration.stable_writable_connection_ping_interval_ms;
   if (configuration != modified_config) {
@@ -457,6 +458,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
     bool offer_extmap_allow_mixed;
     std::string turn_logging_id;
     bool enable_implicit_rollback;
+    absl::optional<bool> allow_codec_switching;
     absl::optional<int> report_usage_pattern_delay_ms;
     absl::optional<int> stable_writable_connection_ping_interval_ms;
     VpnPreference vpn_preference;
@@ -520,6 +522,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
          offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
          turn_logging_id == o.turn_logging_id &&
          enable_implicit_rollback == o.enable_implicit_rollback &&
+         allow_codec_switching == o.allow_codec_switching &&
          report_usage_pattern_delay_ms == o.report_usage_pattern_delay_ms &&
          stable_writable_connection_ping_interval_ms ==
              o.stable_writable_connection_ping_interval_ms &&
diff --git a/sdk/android/api/org/webrtc/PeerConnection.java b/sdk/android/api/org/webrtc/PeerConnection.java
index d530bc2c0c..5c87fe3ea3 100644
--- a/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/sdk/android/api/org/webrtc/PeerConnection.java
@@ -540,6 +540,11 @@ public class PeerConnection {
     // 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;
+
     /**
      * Defines advanced optional cryptographic settings related to SRTP and
      * frame encryption for native WebRTC. Setting this will overwrite any
@@ -606,6 +611,7 @@ public class PeerConnection {
       activeResetSrtpParams = false;
       cryptoOptions = null;
       turnLoggingId = null;
+      allowCodecSwitching = null;
       enableImplicitRollback = false;
       offerExtmapAllowMixed = true;
     }
@@ -795,6 +801,12 @@ public class PeerConnection {
       return activeResetSrtpParams;
     }
 
+    @Nullable
+    @CalledByNative("RTCConfiguration")
+    Boolean getAllowCodecSwitching() {
+      return allowCodecSwitching;
+    }
+
     @Nullable
     @CalledByNative("RTCConfiguration")
     CryptoOptions getCryptoOptions() {
diff --git a/sdk/android/src/jni/pc/peer_connection.cc b/sdk/android/src/jni/pc/peer_connection.cc
index e5fb11e3b4..a063804ac8 100644
--- a/sdk/android/src/jni/pc/peer_connection.cc
+++ b/sdk/android/src/jni/pc/peer_connection.cc
@@ -267,6 +267,10 @@ void JavaToNativeRTCConfiguration(
       Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
   rtc_config->crypto_options =
       JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
+
+  rtc_config->allow_codec_switching = JavaToNativeOptionalBool(
+      jni, Java_RTCConfiguration_getAllowCodecSwitching(jni, j_rtc_config));
+
   rtc_config->offer_extmap_allow_mixed =
       Java_RTCConfiguration_getOfferExtmapAllowMixed(jni, j_rtc_config);
   rtc_config->enable_implicit_rollback =
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.h b/sdk/objc/api/peerconnection/RTCConfiguration.h
index 011eaa613d..1b0d14baf1 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -184,6 +184,12 @@ RTC_OBJC_EXPORT
  */
 @property(nonatomic, assign) BOOL activeResetSrtpParams;
 
+/** If the remote side support mid-stream codec switches then allow encoder
+ *  switching to be performed.
+ */
+
+@property(nonatomic, assign) BOOL allowCodecSwitching;
+
 /**
  * Defines advanced optional cryptographic settings related to SRTP and
  * frame encryption for native WebRTC. Setting this will overwrite any
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 86ecbabf8d..8e42cb2a82 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -51,6 +51,7 @@
 @synthesize sdpSemantics = _sdpSemantics;
 @synthesize turnCustomizer = _turnCustomizer;
 @synthesize activeResetSrtpParams = _activeResetSrtpParams;
+@synthesize allowCodecSwitching = _allowCodecSwitching;
 @synthesize cryptoOptions = _cryptoOptions;
 @synthesize turnLoggingId = _turnLoggingId;
 @synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@@ -138,6 +139,7 @@
     _turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
     _rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
     _rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
+    _allowCodecSwitching = config.allow_codec_switching.value_or(false);
     _enableImplicitRollback = config.enable_implicit_rollback;
     _offerExtmapAllowMixed = config.offer_extmap_allow_mixed;
     _iceCheckIntervalStrongConnectivity =
@@ -284,6 +286,7 @@
   nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
   nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
   nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
+  nativeConfig->allow_codec_switching = _allowCodecSwitching;
   nativeConfig->enable_implicit_rollback = _enableImplicitRollback;
   nativeConfig->offer_extmap_allow_mixed = _offerExtmapAllowMixed;
   if (_iceCheckIntervalStrongConnectivity != nil) {