diff --git a/modules/rtp_rtcp/source/rtp_video_header.cc b/modules/rtp_rtcp/source/rtp_video_header.cc
index a3ee8baa83..bb9413ddd5 100644
--- a/modules/rtp_rtcp/source/rtp_video_header.cc
+++ b/modules/rtp_rtcp/source/rtp_video_header.cc
@@ -12,7 +12,7 @@
 
 namespace webrtc {
 
-RTPVideoHeader::RTPVideoHeader() : playout_delay(), video_timing() {}
+RTPVideoHeader::RTPVideoHeader() : video_timing() {}
 RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default;
 RTPVideoHeader::~RTPVideoHeader() = default;
 
diff --git a/modules/rtp_rtcp/source/rtp_video_header.h b/modules/rtp_rtcp/source/rtp_video_header.h
index 4426c41895..ddf5942e39 100644
--- a/modules/rtp_rtcp/source/rtp_video_header.h
+++ b/modules/rtp_rtcp/source/rtp_video_header.h
@@ -59,7 +59,7 @@ struct RTPVideoHeader {
   uint8_t simulcastIdx = 0;
   VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
 
-  PlayoutDelay playout_delay;
+  PlayoutDelay playout_delay = {-1, -1};
   VideoSendTiming video_timing;
   FrameMarking frame_marking;
   RTPVideoTypeHeader video_type_header;
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 8383c07372..f85ae2d59e 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -530,8 +530,6 @@ void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
   webrtc_rtp_header.video_header().content_type = VideoContentType::UNSPECIFIED;
   webrtc_rtp_header.video_header().video_timing.flags =
       VideoSendTiming::kInvalid;
-  webrtc_rtp_header.video_header().playout_delay.min_ms = -1;
-  webrtc_rtp_header.video_header().playout_delay.max_ms = -1;
   webrtc_rtp_header.video_header().is_last_packet_in_frame =
       webrtc_rtp_header.header.markerBit;
 
diff --git a/video/rtp_video_stream_receiver_unittest.cc b/video/rtp_video_stream_receiver_unittest.cc
index 81342b826a..10c5e7eebc 100644
--- a/video/rtp_video_stream_receiver_unittest.cc
+++ b/video/rtp_video_stream_receiver_unittest.cc
@@ -626,4 +626,13 @@ TEST_F(RtpVideoStreamReceiverTest, RepeatedSecondarySinkDisallowed) {
 }
 #endif
 
+// Initialization of WebRtcRTPHeader is a bit convoluted, with some fields
+// zero-initialized. RtpVideoStreamReceiver depends on proper default values for
+// the playout delay.
+TEST(WebRtcRTPHeader, DefaultPlayoutDelayIsUnspecified) {
+  WebRtcRTPHeader webrtc_rtp_header = {};
+  EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.min_ms, -1);
+  EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.max_ms, -1);
+}
+
 }  // namespace webrtc