diff --git a/modules/audio_device/ios/voice_processing_audio_unit.mm b/modules/audio_device/ios/voice_processing_audio_unit.mm
index 2bb637c66b..c350969335 100644
--- a/modules/audio_device/ios/voice_processing_audio_unit.mm
+++ b/modules/audio_device/ios/voice_processing_audio_unit.mm
@@ -424,7 +424,7 @@ AudioStreamBasicDescription VoiceProcessingAudioUnit::GetFormat(
   // - avoid resampling in the I/O unit by using the hardware sample rate
   // - linear PCM => noncompressed audio data format with one frame per packet
   // - no need to specify interleaving since only mono is supported
-  AudioStreamBasicDescription format = {0};
+  AudioStreamBasicDescription format;
   RTC_DCHECK_EQ(1, kRTCAudioSessionPreferredNumberOfChannels);
   format.mSampleRate = sample_rate;
   format.mFormatID = kAudioFormatLinearPCM;
diff --git a/p2p/base/port.h b/p2p/base/port.h
index 61fd750691..9e5c22355a 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -432,7 +432,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
   // A virtual cost perceived by the user, usually based on the network type
   // (WiFi. vs. Cellular). It takes precedence over the priority when
   // comparing two connections.
-  uint16_t network_cost_;
+  int16_t network_cost_;
   State state_ = State::INIT;
   int64_t last_time_all_connections_removed_ = 0;
 
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index d4151580ee..23f626fb47 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -42,6 +42,10 @@ if (is_ios || is_mac) {
       "objc/Framework/Classes/VideoToolbox",
       "objc/Framework/Headers",
     ]
+    cflags = [
+      "-Wstrict-overflow",
+      "-Wmissing-field-initializers",
+    ]
   }
 
   rtc_static_library("common_objc") {