diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
index af1d5595be..65d6db7cb7 100644
--- a/webrtc/modules/pacing/packet_router.cc
+++ b/webrtc/modules/pacing/packet_router.cc
@@ -30,7 +30,13 @@ void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
   rtc::CritScope cs(&modules_crit_);
   RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
              rtp_modules_.end());
-  rtp_modules_.push_back(rtp_module);
+  // Put modules which can use regular payload packets (over rtx) instead of
+  // padding first as it's less of a waste
+  if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) {
+    rtp_modules_.push_front(rtp_module);
+  } else {
+    rtp_modules_.push_back(rtp_module);
+  }
 }
 
 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
@@ -64,6 +70,7 @@ size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
   RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
   size_t total_bytes_sent = 0;
   rtc::CritScope cs(&modules_crit_);
+  // Rtp modules are ordered by which stream can most benefit from padding.
   for (RtpRtcp* module : rtp_modules_) {
     if (module->SendingMedia()) {
       size_t bytes_sent = module->TimeToSendPadding(
diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
index 456e1def4f..9011f81b36 100644
--- a/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -108,34 +108,37 @@ TEST_F(PacketRouterTest, TimeToSendPadding) {
   const uint16_t kSsrc2 = 4567;
 
   MockRtpRtcp rtp_1;
+  EXPECT_CALL(rtp_1, RtxSendStatus()).WillOnce(Return(kRtxOff));
   EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
   MockRtpRtcp rtp_2;
+  // rtp_2 will be prioritized for padding.
+  EXPECT_CALL(rtp_2, RtxSendStatus()).WillOnce(Return(kRtxRedundantPayloads));
   EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
   packet_router_->AddRtpModule(&rtp_1);
   packet_router_->AddRtpModule(&rtp_2);
 
   // Default configuration, sending padding on all modules sending media,
-  // ordered by SSRC.
+  // ordered by priority (based on rtx mode).
   const size_t requested_padding_bytes = 1000;
   const size_t sent_padding_bytes = 890;
-  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
-  EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes, 111))
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes, 111))
       .Times(1)
       .WillOnce(Return(sent_padding_bytes));
-  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
-  EXPECT_CALL(rtp_2, TimeToSendPadding(
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_1, TimeToSendPadding(
                          requested_padding_bytes - sent_padding_bytes, 111))
       .Times(1)
       .WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
   EXPECT_EQ(requested_padding_bytes,
             packet_router_->TimeToSendPadding(requested_padding_bytes, 111));
 
-  // Let only the second module be sending and verify the padding request is
-  // routed there.
-  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
-  EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes, _)).Times(0);
-  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
-  EXPECT_CALL(rtp_2, TimeToSendPadding(_, _))
+  // Let only the lower priority module be sending and verify the padding
+  // request is routed there.
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes, _)).Times(0);
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_1, TimeToSendPadding(_, _))
       .Times(1)
       .WillOnce(Return(sent_padding_bytes));
   EXPECT_EQ(sent_padding_bytes,
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index aa67532046..9189b98e5a 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -815,9 +815,12 @@ VideoSendStreamImpl::VideoSendStreamImpl(
       config_->periodic_alr_bandwidth_probing);
 
   // RTP/RTCP initialization.
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+
+  // We add the highest spatial layer first to ensure it'll be prioritized
+  // when sending padding, with the hope that the packet rate will be smaller,
+  // and that it's more important to protect than the lower layers.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     packet_router_->AddRtpModule(rtp_rtcp);
-  }
 
   for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
     const std::string& extension = config_->rtp.extensions[i].uri;