diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 6d19cea88..64980051c 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -51,9 +51,39 @@ struct VideoPayload {
   H264::Profile h264_profile;
 };
 
-union PayloadUnion {
-    AudioPayload Audio;
-    VideoPayload Video;
+class PayloadUnion {
+ public:
+  explicit PayloadUnion(const AudioPayload& payload)
+      : Audio(payload), is_audio_(true) {}
+  explicit PayloadUnion(const VideoPayload& payload)
+      : Video(payload), is_audio_(false) {}
+
+  bool is_audio() const { return is_audio_; }
+  bool is_video() const { return !is_audio_; }
+  const AudioPayload& audio_payload() const {
+    RTC_DCHECK(is_audio_);
+    return Audio;
+  }
+  const VideoPayload& video_payload() const {
+    RTC_DCHECK(!is_audio_);
+    return Video;
+  }
+  AudioPayload& audio_payload() {
+    RTC_DCHECK(is_audio_);
+    return Audio;
+  }
+  VideoPayload& video_payload() {
+    RTC_DCHECK(!is_audio_);
+    return Video;
+  }
+
+ public:
+  // These two are public for backwards compatibilty. They'll go private soon.
+  AudioPayload Audio;
+  VideoPayload Video;
+
+ private:
+  bool is_audio_;
 };
 
 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
diff --git a/modules/rtp_rtcp/source/rtp_payload_registry.cc b/modules/rtp_rtcp/source/rtp_payload_registry.cc
index fcd0276c0..9effcc291 100644
--- a/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -46,15 +46,11 @@ bool PayloadIsCompatible(const RtpUtility::Payload& payload,
 }
 
 RtpUtility::Payload CreatePayloadType(const CodecInst& audio_codec) {
-  RtpUtility::Payload payload;
-  payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
-  strncpy(payload.name, audio_codec.plname, RTP_PAYLOAD_NAME_SIZE - 1);
   RTC_DCHECK_GE(audio_codec.plfreq, 1000);
-  payload.typeSpecific.Audio.frequency = audio_codec.plfreq;
-  payload.typeSpecific.Audio.channels = audio_codec.channels;
-  payload.typeSpecific.Audio.rate = 0;
-  payload.audio = true;
-  return payload;
+  return {audio_codec.plname,
+          PayloadUnion(
+              AudioPayload{rtc::dchecked_cast<uint32_t>(audio_codec.plfreq),
+                           audio_codec.channels, 0})};
 }
 
 RtpVideoCodecTypes ConvertToRtpVideoCodecType(VideoCodecType type) {
@@ -74,15 +70,11 @@ RtpVideoCodecTypes ConvertToRtpVideoCodecType(VideoCodecType type) {
 }
 
 RtpUtility::Payload CreatePayloadType(const VideoCodec& video_codec) {
-  RtpUtility::Payload payload;
-  payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
-  strncpy(payload.name, video_codec.plName, RTP_PAYLOAD_NAME_SIZE - 1);
-  payload.typeSpecific.Video.videoCodecType =
-      ConvertToRtpVideoCodecType(video_codec.codecType);
+  VideoPayload p;
+  p.videoCodecType = ConvertToRtpVideoCodecType(video_codec.codecType);
   if (video_codec.codecType == kVideoCodecH264)
-    payload.typeSpecific.Video.h264_profile = video_codec.H264().profile;
-  payload.audio = false;
-  return payload;
+    p.h264_profile = video_codec.H264().profile;
+  return {video_codec.plName, PayloadUnion(p)};
 }
 
 bool IsPayloadTypeValid(int8_t payload_type) {
@@ -172,7 +164,9 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
   // Audio codecs must be unique.
   DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(audio_codec);
 
-  payload_type_map_[audio_codec.pltype] = CreatePayloadType(audio_codec);
+  const auto insert_status = payload_type_map_.emplace(
+      audio_codec.pltype, CreatePayloadType(audio_codec));
+  RTC_DCHECK(insert_status.second);  // Insertion succeeded.
   *created_new_payload = true;
 
   // Successful set of payload type, clear the value of last received payload
@@ -205,7 +199,9 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
     return -1;
   }
 
-  payload_type_map_[video_codec.plType] = CreatePayloadType(video_codec);
+  const auto insert_status = payload_type_map_.emplace(
+      video_codec.plType, CreatePayloadType(video_codec));
+  RTC_DCHECK(insert_status.second);  // Insertion succeeded.
 
   // Successful set of payload type, clear the value of last received payload
   // type since it might mean something else.
diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index d399ad53a..9ccc3ec44 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -35,7 +35,7 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
       cng_fb_payload_type_(-1),
       num_energy_(0),
       current_remote_energy_() {
-  last_payload_.Audio.channels = 1;
+  last_payload_.emplace(AudioPayload{0, 1, 0});
   memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
index de58b615e..6db24c9a0 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
@@ -15,20 +15,20 @@
 namespace webrtc {
 
 RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
-    : data_callback_(data_callback) {
-  memset(&last_payload_, 0, sizeof(last_payload_));
-}
+    : data_callback_(data_callback) {}
 
 void RTPReceiverStrategy::GetLastMediaSpecificPayload(
     PayloadUnion* payload) const {
   rtc::CritScope cs(&crit_sect_);
-  memcpy(payload, &last_payload_, sizeof(*payload));
+  if (last_payload_) {
+    *payload = *last_payload_;
+  }
 }
 
 void RTPReceiverStrategy::SetLastMediaSpecificPayload(
     const PayloadUnion& payload) {
   rtc::CritScope cs(&crit_sect_);
-  memcpy(&last_payload_, &payload, sizeof(last_payload_));
+  last_payload_.emplace(payload);
 }
 
 void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type,
diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index af1868e9f..cc1d1e6e1 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -89,7 +89,7 @@ class RTPReceiverStrategy {
   explicit RTPReceiverStrategy(RtpData* data_callback);
 
   rtc::CriticalSection crit_sect_;
-  PayloadUnion last_payload_;
+  rtc::Optional<PayloadUnion> last_payload_;
   RtpData* data_callback_;
 };
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
index c1cde0f59..f1d5233e2 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
@@ -90,7 +90,7 @@ TEST_F(RtpReceiverTest, GetSources) {
   header.numCSRCs = 2;
   header.arrOfCSRCs[0] = kCsrc1;
   header.arrOfCSRCs[1] = kCsrc2;
-  PayloadUnion payload_specific = {AudioPayload()};
+  const PayloadUnion payload_specific{AudioPayload()};
 
   EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
       header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
@@ -140,7 +140,7 @@ TEST_F(RtpReceiverTest, GetSourcesChangeSSRC) {
   header.payloadType = kPcmuPayloadType;
   header.ssrc = kSsrc1;
   header.timestamp = rtp_timestamp(now_ms);
-  PayloadUnion payload_specific = {AudioPayload()};
+  const PayloadUnion payload_specific{AudioPayload()};
 
   EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
       header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
@@ -191,7 +191,7 @@ TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
   RTPHeader header;
   header.payloadType = kPcmuPayloadType;
   header.timestamp = rtp_timestamp(now_ms);
-  PayloadUnion payload_specific = {AudioPayload()};
+  const PayloadUnion payload_specific{AudioPayload()};
   header.numCSRCs = 1;
   size_t kSourceListSize = 20;
 
@@ -265,7 +265,7 @@ TEST_F(RtpReceiverTest, GetSourcesContainsAudioLevelExtension) {
   header.timestamp = rtp_timestamp(time1_ms);
   header.extension.hasAudioLevel = true;
   header.extension.audioLevel = 10;
-  PayloadUnion payload_specific = {AudioPayload()};
+  const PayloadUnion payload_specific{AudioPayload()};
 
   EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
       header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
@@ -317,7 +317,7 @@ TEST_F(RtpReceiverTest,
   header.timestamp = rtp_timestamp(time1_ms);
   header.extension.hasAudioLevel = true;
   header.extension.audioLevel = 10;
-  PayloadUnion payload_specific = {AudioPayload()};
+  const PayloadUnion payload_specific{AudioPayload()};
 
   EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
       header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index f75452fde..dadf30b53 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -65,13 +65,8 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     dtmf_payload_freq_ = frequency;
     return 0;
   }
-  *payload = new RtpUtility::Payload;
-  (*payload)->typeSpecific.Audio.frequency = frequency;
-  (*payload)->typeSpecific.Audio.channels = channels;
-  (*payload)->typeSpecific.Audio.rate = rate;
-  (*payload)->audio = true;
-  (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
-  strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+  *payload = new RtpUtility::Payload(
+      payloadName, PayloadUnion(AudioPayload{frequency, channels, rate}));
   return 0;
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 888dc7277..60636d1e2 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -92,12 +92,9 @@ RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload(
   } else {
     video_type = kRtpVideoGeneric;
   }
-  RtpUtility::Payload* payload = new RtpUtility::Payload();
-  payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
-  strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1);
-  payload->typeSpecific.Video.videoCodecType = video_type;
-  payload->audio = false;
-  return payload;
+  VideoPayload vp;
+  vp.videoCodecType = video_type;
+  return new RtpUtility::Payload(payload_name, PayloadUnion(vp));
 }
 
 void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
diff --git a/modules/rtp_rtcp/source/rtp_utility.h b/modules/rtp_rtcp/source/rtp_utility.h
index cd4968eff..04eb4383d 100644
--- a/modules/rtp_rtcp/source/rtp_utility.h
+++ b/modules/rtp_rtcp/source/rtp_utility.h
@@ -11,6 +11,7 @@
 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 #define MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 
+#include <cstring>
 #include <map>
 
 #include "modules/rtp_rtcp/include/receive_statistics.h"
@@ -29,6 +30,11 @@ RtpFeedback* NullObjectRtpFeedback();
 namespace RtpUtility {
 
 struct Payload {
+  Payload(const char* name, const PayloadUnion& pu)
+      : audio(pu.is_audio()), typeSpecific(pu) {
+    std::strncpy(this->name, name, sizeof(this->name) - 1);
+    this->name[sizeof(this->name) - 1] = '\0';
+  }
   char name[RTP_PAYLOAD_NAME_SIZE];
   bool audio;
   PayloadUnion typeSpecific;