diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 5f8d2df..854a18a 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -197,6 +197,7 @@
     FrameEncryptorInterface* frame_encryptor,
     const CryptoOptions& crypto_options,
     rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    bool use_deferred_fec,
     const WebRtcKeyValueConfig& trials) {
   RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
 
@@ -244,7 +245,9 @@
     std::unique_ptr<VideoFecGenerator> fec_generator =
         MaybeCreateFecGenerator(clock, rtp_config, suspended_ssrcs, i, trials);
     configuration.fec_generator = fec_generator.get();
-    video_config.fec_generator = fec_generator.get();
+    if (!use_deferred_fec) {
+      video_config.fec_generator = fec_generator.get();
+    }
 
     configuration.rtx_send_ssrc =
         rtp_config.GetRtxSsrcAssociatedWithMediaSsrc(rtp_config.ssrcs[i]);
@@ -338,6 +341,9 @@
           field_trials_.Lookup("WebRTC-UseEarlyLossDetection"),
           "Disabled")),
       has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
+      use_deferred_fec_(
+          absl::StartsWith(field_trials_.Lookup("WebRTC-DeferredFecGeneration"),
+                           "Enabled")),
       active_(false),
       module_process_thread_(nullptr),
       suspended_ssrcs_(std::move(suspended_ssrcs)),
@@ -356,6 +362,7 @@
                                           frame_encryptor,
                                           crypto_options,
                                           std::move(frame_transformer),
+                                          use_deferred_fec_,
                                           field_trials_)),
       rtp_config_(rtp_config),
       codec_type_(GetVideoCodecType(rtp_config)),
@@ -848,14 +855,26 @@
   *sent_nack_rate_bps = 0;
   *sent_fec_rate_bps = 0;
   for (const RtpStreamSender& stream : rtp_streams_) {
-    if (stream.fec_generator) {
-      stream.fec_generator->SetProtectionParameters(*delta_params, *key_params);
-      *sent_fec_rate_bps += stream.fec_generator->CurrentFecRate().bps();
+    if (use_deferred_fec_) {
+      stream.rtp_rtcp->SetFecProtectionParams(*delta_params, *key_params);
+
+      auto send_bitrate = stream.rtp_rtcp->GetSendRates();
+      *sent_video_rate_bps += send_bitrate[RtpPacketMediaType::kVideo].bps();
+      *sent_fec_rate_bps +=
+          send_bitrate[RtpPacketMediaType::kForwardErrorCorrection].bps();
+      *sent_nack_rate_bps +=
+          send_bitrate[RtpPacketMediaType::kRetransmission].bps();
+    } else {
+      if (stream.fec_generator) {
+        stream.fec_generator->SetProtectionParameters(*delta_params,
+                                                      *key_params);
+        *sent_fec_rate_bps += stream.fec_generator->CurrentFecRate().bps();
+      }
+      *sent_video_rate_bps += stream.sender_video->VideoBitrateSent();
+      *sent_nack_rate_bps +=
+          stream.rtp_rtcp->GetSendRates()[RtpPacketMediaType::kRetransmission]
+              .bps<uint32_t>();
     }
-    *sent_video_rate_bps += stream.sender_video->VideoBitrateSent();
-    *sent_nack_rate_bps +=
-        stream.rtp_rtcp->GetSendRates()[RtpPacketMediaType::kRetransmission]
-            .bps<uint32_t>();
   }
   return 0;
 }
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 0c277d6..e364729 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -176,6 +176,7 @@
   const bool account_for_packetization_overhead_;
   const bool use_early_loss_detection_;
   const bool has_packet_feedback_;
+  const bool use_deferred_fec_;
 
   // TODO(holmer): Remove crit_ once RtpVideoSender runs on the
   // transport task queue.
diff --git a/modules/pacing/pacing_controller.cc b/modules/pacing/pacing_controller.cc
index 07e265b..7e7a01b 100644
--- a/modules/pacing/pacing_controller.cc
+++ b/modules/pacing/pacing_controller.cc
@@ -441,6 +441,9 @@
         keepalive_data_sent +=
             DataSize::Bytes(packet->payload_size() + packet->padding_size());
         packet_sender_->SendPacket(std::move(packet), PacedPacketInfo());
+        for (auto& packet : packet_sender_->FetchFec()) {
+          EnqueuePacket(std::move(packet));
+        }
       }
       OnPaddingSent(keepalive_data_sent);
     }
@@ -559,8 +562,11 @@
       packet_size += DataSize::Bytes(rtp_packet->headers_size()) +
                      transport_overhead_per_packet_;
     }
-    packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
 
+    packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
+    for (auto& packet : packet_sender_->FetchFec()) {
+      EnqueuePacket(std::move(packet));
+    }
     data_sent += packet_size;
 
     // Send done, update send/process time to the target send time.
diff --git a/modules/pacing/pacing_controller.h b/modules/pacing/pacing_controller.h
index 6e361ae..775fdc9 100644
--- a/modules/pacing/pacing_controller.h
+++ b/modules/pacing/pacing_controller.h
@@ -57,6 +57,8 @@
     virtual ~PacketSender() = default;
     virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                             const PacedPacketInfo& cluster_info) = 0;
+    // Should be called after each call to SendPacket().
+    virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() = 0;
     virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
         DataSize size) = 0;
   };
diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc
index bc4d473..9194d07 100644
--- a/modules/pacing/pacing_controller_unittest.cc
+++ b/modules/pacing/pacing_controller_unittest.cc
@@ -97,6 +97,10 @@
                int64_t capture_timestamp,
                bool retransmission,
                bool padding));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFec,
+              (),
+              (override));
   MOCK_METHOD(size_t, SendPadding, (size_t target_size));
 };
 
@@ -109,6 +113,11 @@
                const PacedPacketInfo& cluster_info),
               (override));
   MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFec,
+              (),
+              (override));
+
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
               GeneratePadding,
               (DataSize target_size),
               (override));
@@ -125,6 +134,10 @@
     total_bytes_sent_ += packet->payload_size();
   }
 
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() override {
+    return {};
+  }
+
   std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
       DataSize target_size) override {
     size_t num_packets =
@@ -158,6 +171,10 @@
     }
   }
 
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() override {
+    return {};
+  }
+
   std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
       DataSize target_size) override {
     // From RTPSender:
@@ -299,7 +316,7 @@
   }
 
   SimulatedClock clock_;
-  MockPacingControllerCallback callback_;
+  ::testing::NiceMock<MockPacingControllerCallback> callback_;
   std::unique_ptr<PacingController> pacer_;
 };
 
@@ -2029,6 +2046,38 @@
   AdvanceTimeAndProcess();
 }
 
+TEST_P(PacingControllerTest, SendsDeferredFecPackets) {
+  ScopedFieldTrials trial("WebRTC-DeferredFecGeneration/Enabled/");
+  SetUp();
+
+  const uint32_t kSsrc = 12345;
+  const uint32_t kFlexSsrc = 54321;
+  uint16_t sequence_number = 1234;
+  uint16_t flexfec_sequence_number = 4321;
+  const size_t kPacketSize = 123;
+
+  // Set pacing rate to 1000 packet/s, no padding.
+  pacer_->SetPacingRates(
+      DataSize::Bytes(1000 * kPacketSize) / TimeDelta::Seconds(1),
+      DataRate::Zero());
+
+  int64_t now = clock_.TimeInMilliseconds();
+  Send(RtpPacketMediaType::kVideo, kSsrc, sequence_number, now, kPacketSize);
+  EXPECT_CALL(callback_, SendPacket(kSsrc, sequence_number, now, false, false));
+  EXPECT_CALL(callback_, FetchFec).WillOnce([&]() {
+    EXPECT_CALL(callback_, SendPacket(kFlexSsrc, flexfec_sequence_number, now,
+                                      false, false));
+    EXPECT_CALL(callback_, FetchFec);
+    std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets;
+    fec_packets.push_back(
+        BuildPacket(RtpPacketMediaType::kForwardErrorCorrection, kFlexSsrc,
+                    flexfec_sequence_number, now, kPacketSize));
+    return fec_packets;
+  });
+  AdvanceTimeAndProcess();
+  AdvanceTimeAndProcess();
+}
+
 INSTANTIATE_TEST_SUITE_P(
     WithAndWithoutIntervalBudget,
     PacingControllerTest,
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index e75b5a3..833c56d 100644
--- a/modules/pacing/packet_router.cc
+++ b/modules/pacing/packet_router.cc
@@ -42,7 +42,9 @@
       bitrate_bps_(0),
       max_bitrate_bps_(std::numeric_limits<decltype(max_bitrate_bps_)>::max()),
       active_remb_module_(nullptr),
-      transport_seq_(start_transport_seq) {}
+      transport_seq_(start_transport_seq) {
+  send_thread_checker_.Detach();
+}
 
 PacketRouter::~PacketRouter() {
   RTC_DCHECK(send_modules_map_.empty());
@@ -139,6 +141,7 @@
 
 void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                               const PacedPacketInfo& cluster_info) {
+  RTC_DCHECK_RUN_ON(&send_thread_checker_);
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"), "PacketRouter::SendPacket",
                "sequence_number", packet->SequenceNumber(), "rtp_timestamp",
                packet->Timestamp());
@@ -171,6 +174,18 @@
     // properties needed for payload based padding. Cache it for later use.
     last_send_module_ = rtp_module;
   }
+
+  for (auto& packet : rtp_module->FetchFecPackets()) {
+    pending_fec_packets_.push_back(std::move(packet));
+  }
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::FetchFec() {
+  RTC_DCHECK_RUN_ON(&send_thread_checker_);
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      std::move(pending_fec_packets_);
+  pending_fec_packets_.clear();
+  return fec_packets;
 }
 
 std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
diff --git a/modules/pacing/packet_router.h b/modules/pacing/packet_router.h
index 73837f2..b939507 100644
--- a/modules/pacing/packet_router.h
+++ b/modules/pacing/packet_router.h
@@ -29,6 +29,7 @@
 #include "rtc_base/constructor_magic.h"
 #include "rtc_base/critical_section.h"
 #include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/synchronization/sequence_checker.h"
 #include "rtc_base/thread_annotations.h"
 
 namespace webrtc {
@@ -57,6 +58,7 @@
 
   void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                   const PacedPacketInfo& cluster_info) override;
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() override;
   std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
       DataSize size) override;
 
@@ -128,6 +130,10 @@
 
   uint64_t transport_seq_ RTC_GUARDED_BY(modules_mutex_);
 
+  SequenceChecker send_thread_checker_;
+  std::vector<std::unique_ptr<RtpPacketToSend>> pending_fec_packets_
+      RTC_GUARDED_BY(send_thread_checker_);
+
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };
 }  // namespace webrtc
diff --git a/modules/pacing/task_queue_paced_sender_unittest.cc b/modules/pacing/task_queue_paced_sender_unittest.cc
index 876cd96..b02f387 100644
--- a/modules/pacing/task_queue_paced_sender_unittest.cc
+++ b/modules/pacing/task_queue_paced_sender_unittest.cc
@@ -44,6 +44,10 @@
                const PacedPacketInfo& cluster_info),
               (override));
   MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFec,
+              (),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
               GeneratePadding,
               (DataSize target_size),
               (override));
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 869f3d2..46c310e 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -33,6 +33,7 @@
 
 namespace webrtc {
 class RtpPacket;
+class RtpPacketToSend;
 namespace rtcp {
 class TransportFeedback;
 }
@@ -466,5 +467,15 @@
                             int64_t capture_time_ms,
                             uint32_t ssrc) = 0;
 };
+
+// Interface for a class that can assign RTP sequence numbers for a packet
+// to be sent.
+class SequenceNumberAssigner {
+ public:
+  SequenceNumberAssigner() = default;
+  virtual ~SequenceNumberAssigner() = default;
+
+  virtual void AssignSequenceNumber(RtpPacketToSend* packet) = 0;
+};
 }  // namespace webrtc
 #endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 08b38ee..d597b1e 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -99,6 +99,15 @@
               (RtpPacketToSend * packet, const PacedPacketInfo& pacing_info),
               (override));
   MOCK_METHOD(void,
+              SetFecProtectionParams,
+              (const FecProtectionParams& delta_params,
+               const FecProtectionParams& key_params),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFecPackets,
+              (),
+              (override));
+  MOCK_METHOD(void,
               OnPacketsAcknowledged,
               (rtc::ArrayView<const uint16_t>),
               (override));
diff --git a/modules/rtp_rtcp/source/rtp_packet_to_send.h b/modules/rtp_rtcp/source/rtp_packet_to_send.h
index 8997bce..9aaf9a52 100644
--- a/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -108,6 +108,15 @@
   void set_is_key_frame(bool is_key_frame) { is_key_frame_ = is_key_frame; }
   bool is_key_frame() const { return is_key_frame_; }
 
+  // Indicates if packets should be protected by FEC (Forward Error Correction).
+  void set_fec_protect_packet(bool protect) { fec_protect_packet_ = protect; }
+  bool fec_protect_packet() const { return fec_protect_packet_; }
+
+  // Indicates if packet is using RED encapsulation, in accordance with
+  // https://tools.ietf.org/html/rfc2198
+  void set_is_red(bool is_red) { is_red_ = is_red; }
+  bool is_red() const { return is_red_; }
+
  private:
   int64_t capture_time_ms_ = 0;
   absl::optional<RtpPacketMediaType> packet_type_;
@@ -116,6 +125,8 @@
   std::vector<uint8_t> application_data_;
   bool is_first_packet_of_frame_ = false;
   bool is_key_frame_ = false;
+  bool fec_protect_packet_ = false;
+  bool is_red_ = false;
 };
 
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 7b9586a..9f8bac5 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -389,6 +389,17 @@
   return true;
 }
 
+void ModuleRtpRtcpImpl::SetFecProtectionParams(const FecProtectionParams&,
+                                               const FecProtectionParams&) {
+  // Deferred FEC not supported in deprecated RTP module.
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl::FetchFecPackets() {
+  // Deferred FEC not supported in deprecated RTP module.
+  return {};
+}
+
 void ModuleRtpRtcpImpl::OnPacketsAcknowledged(
     rtc::ArrayView<const uint16_t> sequence_numbers) {
   RTC_DCHECK(rtp_sender_);
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 989b8d3..1fa57db 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -139,6 +139,11 @@
   bool TrySendPacket(RtpPacketToSend* packet,
                      const PacedPacketInfo& pacing_info) override;
 
+  void SetFecProtectionParams(const FecProtectionParams& delta_params,
+                              const FecProtectionParams& key_params) override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
+
   void OnPacketsAcknowledged(
       rtc::ArrayView<const uint16_t> sequence_numbers) override;
 
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 70f05d7..67cb70e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -42,11 +42,15 @@
     const RtpRtcpInterface::Configuration& config)
     : packet_history(config.clock, config.enable_rtx_padding_prioritization),
       packet_sender(config, &packet_history),
-      non_paced_sender(&packet_sender),
+      non_paced_sender(&packet_sender, this),
       packet_generator(
           config,
           &packet_history,
           config.paced_sender ? config.paced_sender : &non_paced_sender) {}
+void ModuleRtpRtcpImpl2::RtpSenderContext::AssignSequenceNumber(
+    RtpPacketToSend* packet) {
+  packet_generator.AssignSequenceNumber(packet);
+}
 
 ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
     : worker_queue_(TaskQueueBase::Current()),
@@ -333,6 +337,31 @@
   return true;
 }
 
+void ModuleRtpRtcpImpl2::SetFecProtectionParams(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_sender.SetFecProtectionParameters(delta_params,
+                                                        key_params);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl2::FetchFecPackets() {
+  RTC_DCHECK(rtp_sender_);
+  auto fec_packets = rtp_sender_->packet_sender.FetchFecPackets();
+  if (!fec_packets.empty()) {
+    // Don't assign sequence numbers for FlexFEC packets.
+    const bool generate_sequence_numbers =
+        !rtp_sender_->packet_sender.FlexFecSsrc().has_value();
+    if (generate_sequence_numbers) {
+      for (auto& fec_packet : fec_packets) {
+        rtp_sender_->packet_generator.AssignSequenceNumber(fec_packet.get());
+      }
+    }
+  }
+  return fec_packets;
+}
+
 void ModuleRtpRtcpImpl2::OnPacketsAcknowledged(
     rtc::ArrayView<const uint16_t> sequence_numbers) {
   RTC_DCHECK(rtp_sender_);
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index c04edfc..932c436 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -148,6 +148,11 @@
   bool TrySendPacket(RtpPacketToSend* packet,
                      const PacedPacketInfo& pacing_info) override;
 
+  void SetFecProtectionParams(const FecProtectionParams& delta_params,
+                              const FecProtectionParams& key_params) override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
+
   void OnPacketsAcknowledged(
       rtc::ArrayView<const uint16_t> sequence_numbers) override;
 
@@ -268,8 +273,9 @@
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImpl2Test, Rtt);
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImpl2Test, RttForReceiverOnly);
 
-  struct RtpSenderContext {
+  struct RtpSenderContext : public SequenceNumberAssigner {
     explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
+    void AssignSequenceNumber(RtpPacketToSend* packet) override;
     // Storage of packets, for retransmissions and padding, if applicable.
     RtpPacketHistory packet_history;
     // Handles final time timestamping/stats/etc and handover to Transport.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 440837f..f763da2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -293,6 +293,17 @@
   virtual bool TrySendPacket(RtpPacketToSend* packet,
                              const PacedPacketInfo& pacing_info) = 0;
 
+  // Update the FEC protection parameters to use for delta- and key-frames.
+  // Only used when deferred FEC is active.
+  virtual void SetFecProtectionParams(
+      const FecProtectionParams& delta_params,
+      const FecProtectionParams& key_params) = 0;
+
+  // If deferred FEC generation is enabled, this method should be called after
+  // calling TrySendPacket(). Any generated FEC packets will be removed and
+  // returned from the FEC generator.
+  virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0;
+
   virtual void OnPacketsAcknowledged(
       rtc::ArrayView<const uint16_t> sequence_numbers) = 0;
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 7196dcd..c8f10ee 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -10,6 +10,7 @@
 
 #include "modules/rtp_rtcp/source/rtp_sender_egress.h"
 
+#include <algorithm>
 #include <limits>
 #include <memory>
 #include <utility>
@@ -38,21 +39,45 @@
 }  // namespace
 
 RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
-    RtpSenderEgress* sender)
-    : transport_sequence_number_(0), sender_(sender) {}
+    RtpSenderEgress* sender,
+    SequenceNumberAssigner* sequence_number_assigner)
+    : transport_sequence_number_(0),
+      sender_(sender),
+      sequence_number_assigner_(sequence_number_assigner) {
+  RTC_DCHECK(sequence_number_assigner_);
+}
 RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() = default;
 
 void RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
     std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
   for (auto& packet : packets) {
-    if (!packet->SetExtension<TransportSequenceNumber>(
-            ++transport_sequence_number_)) {
-      --transport_sequence_number_;
-    }
-    packet->ReserveExtension<TransmissionOffset>();
-    packet->ReserveExtension<AbsoluteSendTime>();
+    PrepareForSend(packet.get());
     sender_->SendPacket(packet.get(), PacedPacketInfo());
   }
+  auto fec_packets = sender_->FetchFecPackets();
+  if (!fec_packets.empty()) {
+    // Don't generate sequence numbers for flexfec, they are already running on
+    // an internally maintained sequence.
+    const bool generate_sequence_numbers = !sender_->FlexFecSsrc().has_value();
+
+    for (auto& packet : fec_packets) {
+      if (generate_sequence_numbers) {
+        sequence_number_assigner_->AssignSequenceNumber(packet.get());
+      }
+      PrepareForSend(packet.get());
+    }
+    EnqueuePackets(std::move(fec_packets));
+  }
+}
+
+void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
+    RtpPacketToSend* packet) {
+  if (!packet->SetExtension<TransportSequenceNumber>(
+          ++transport_sequence_number_)) {
+    --transport_sequence_number_;
+  }
+  packet->ReserveExtension<TransmissionOffset>();
+  packet->ReserveExtension<AbsoluteSendTime>();
 }
 
 RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
@@ -73,6 +98,10 @@
       is_audio_(config.audio),
 #endif
       need_rtp_packet_infos_(config.need_rtp_packet_infos),
+      fec_generator_(
+          IsEnabled("WebRTC-DeferredFecGeneration", config.field_trials)
+              ? config.fec_generator
+              : nullptr),
       transport_feedback_observer_(config.transport_feedback_callback),
       send_side_delay_observer_(config.send_side_delay_observer),
       send_packet_observer_(config.send_packet_observer),
@@ -142,6 +171,41 @@
         }));
   }
 
+  if (fec_generator_ && packet->fec_protect_packet()) {
+    // Deferred fec generation is used, add packet to generator.
+    RTC_DCHECK(fec_generator_);
+    RTC_DCHECK(packet->packet_type() == RtpPacketMediaType::kVideo);
+    absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
+        new_fec_params;
+    {
+      rtc::CritScope lock(&lock_);
+      new_fec_params.swap(pending_fec_params_);
+    }
+    if (new_fec_params) {
+      fec_generator_->SetProtectionParameters(new_fec_params->first,
+                                              new_fec_params->second);
+    }
+    if (packet->is_red()) {
+      RtpPacketToSend unpacked_packet(*packet);
+
+      const rtc::CopyOnWriteBuffer buffer = packet->Buffer();
+      // Grab media payload type from RED header.
+      const size_t headers_size = packet->headers_size();
+      unpacked_packet.SetPayloadType(buffer[headers_size]);
+
+      // Copy the media payload into the unpacked buffer.
+      uint8_t* payload_buffer =
+          unpacked_packet.SetPayloadSize(packet->payload_size() - 1);
+      std::copy(&packet->payload()[0] + 1,
+                &packet->payload()[0] + packet->payload_size(), payload_buffer);
+
+      fec_generator_->AddPacketAndGenerateFec(unpacked_packet);
+    } else {
+      // If not RED encapsulated - we can just insert packet directly.
+      fec_generator_->AddPacketAndGenerateFec(*packet);
+    }
+  }
+
   // Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
   // the pacer, these modifications of the header below are happening after the
   // FEC protection packets are calculated. This will corrupt recovered packets
@@ -208,9 +272,14 @@
   }
 
   if (send_success) {
-    // TODO(tommi): Is this assuming is_media is true?
+    // |media_has_been_sent_| is used by RTPSender to figure out if it can send
+    // padding in the absence of transport-cc or abs-send-time.
+    // In those cases media must be sent first to set a reference timestamp.
     media_has_been_sent_ = true;
 
+    // TODO(sprang): Add support for FEC protecting all header extensions, add
+    // media packet to generator here instead.
+
     RTC_DCHECK(packet->packet_type().has_value());
     RtpPacketMediaType packet_type = *packet->packet_type();
     RtpPacketCounter counter(*packet);
@@ -295,6 +364,24 @@
   return results;
 }
 
+void RtpSenderEgress::SetFecProtectionParameters(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  // TODO(sprang): Post task to pacer queue instead, one pacer is fully
+  // migrated to a task queue.
+  rtc::CritScope lock(&lock_);
+  pending_fec_params_.emplace(delta_params, key_params);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+RtpSenderEgress::FetchFecPackets() {
+  RTC_DCHECK_RUN_ON(&pacer_checker_);
+  if (fec_generator_) {
+    return fec_generator_->GetFecPackets();
+  }
+  return {};
+}
+
 bool RtpSenderEgress::HasCorrectSsrc(const RtpPacketToSend& packet) const {
   switch (*packet.packet_type()) {
     case RtpPacketMediaType::kAudio:
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index 5b50ddf..e2cc00f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -13,6 +13,7 @@
 
 #include <map>
 #include <memory>
+#include <utility>
 #include <vector>
 
 #include "absl/types/optional.h"
@@ -41,15 +42,18 @@
   // without passing through an actual paced sender.
   class NonPacedPacketSender : public RtpPacketSender {
    public:
-    explicit NonPacedPacketSender(RtpSenderEgress* sender);
+    NonPacedPacketSender(RtpSenderEgress* sender,
+                         SequenceNumberAssigner* sequence_number_assigner);
     virtual ~NonPacedPacketSender();
 
     void EnqueuePackets(
         std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
 
    private:
+    void PrepareForSend(RtpPacketToSend* packet);
     uint16_t transport_sequence_number_;
     RtpSenderEgress* const sender_;
+    SequenceNumberAssigner* sequence_number_assigner_;
   };
 
   RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
@@ -82,6 +86,10 @@
       rtc::ArrayView<const uint16_t> sequence_numbers) const
       RTC_LOCKS_EXCLUDED(lock_);
 
+  void SetFecProtectionParameters(const FecProtectionParams& delta_params,
+                                  const FecProtectionParams& key_params);
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets();
+
  private:
   // Maps capture time in milliseconds to send-side delay in milliseconds.
   // Send-side delay is the difference between transmission time and capture
@@ -133,6 +141,7 @@
   const bool is_audio_;
 #endif
   const bool need_rtp_packet_infos_;
+  VideoFecGenerator* const fec_generator_ RTC_GUARDED_BY(pacer_checker_);
 
   TransportFeedbackObserver* const transport_feedback_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
@@ -154,6 +163,8 @@
   StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
   // One element per value in RtpPacketMediaType, with index matching value.
   std::vector<RateStatistics> send_rates_ RTC_GUARDED_BY(lock_);
+  absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
+      pending_fec_params_ RTC_GUARDED_BY(lock_);
 
   // Maps sent packets' sequence numbers to a tuple consisting of:
   // 1. The timestamp, without the randomizing offset mandated by the RFC.
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index dbd474c..9e1aefa 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -32,6 +32,7 @@
 #include "modules/rtp_rtcp/source/rtp_sender_egress.h"
 #include "modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/strings/string_builder.h"
@@ -141,8 +142,10 @@
 }
 
 struct TestConfig {
-  explicit TestConfig(bool with_overhead) : with_overhead(with_overhead) {}
+  TestConfig(bool with_overhead, bool deferred_fec)
+      : with_overhead(with_overhead), deferred_fec(deferred_fec) {}
   bool with_overhead = false;
+  bool deferred_fec = false;
 };
 
 class MockRtpPacketPacer : public RtpPacketSender {
@@ -212,15 +215,18 @@
 // Mimics ModuleRtpRtcp::RtpSenderContext.
 // TODO(sprang): Split up unit tests and test these components individually
 // wherever possible.
-struct RtpSenderContext {
+struct RtpSenderContext : public SequenceNumberAssigner {
   explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config)
       : packet_history_(config.clock, config.enable_rtx_padding_prioritization),
         packet_sender_(config, &packet_history_),
-        non_paced_sender_(&packet_sender_),
+        non_paced_sender_(&packet_sender_, this),
         packet_generator_(
             config,
             &packet_history_,
             config.paced_sender ? config.paced_sender : &non_paced_sender_) {}
+  void AssignSequenceNumber(RtpPacketToSend* packet) override {
+    packet_generator_.AssignSequenceNumber(packet);
+  }
   RtpPacketHistory packet_history_;
   RtpSenderEgress packet_sender_;
   RtpSenderEgress::NonPacedPacketSender non_paced_sender_;
@@ -229,10 +235,14 @@
 
 class FieldTrialConfig : public WebRtcKeyValueConfig {
  public:
-  FieldTrialConfig() : overhead_enabled_(false), max_padding_factor_(1200) {}
+  FieldTrialConfig()
+      : overhead_enabled_(false),
+        deferred_fec_(false),
+        max_padding_factor_(1200) {}
   ~FieldTrialConfig() override {}
 
   void SetOverHeadEnabled(bool enabled) { overhead_enabled_ = enabled; }
+  void UseDeferredFec(bool enabled) { deferred_fec_ = enabled; }
   void SetMaxPaddingFactor(double factor) { max_padding_factor_ = factor; }
 
   std::string Lookup(absl::string_view key) const override {
@@ -243,12 +253,15 @@
       return ssb.str();
     } else if (key == "WebRTC-SendSideBwe-WithOverhead") {
       return overhead_enabled_ ? "Enabled" : "Disabled";
+    } else if (key == "WebRTC-DeferredFecGeneration") {
+      return deferred_fec_ ? "Enabled" : "Disabled";
     }
     return "";
   }
 
  private:
   bool overhead_enabled_;
+  bool deferred_fec_;
   double max_padding_factor_;
 };
 
@@ -269,6 +282,7 @@
                         &fake_clock_),
         kMarkerBit(true) {
     field_trials_.SetOverHeadEnabled(GetParam().with_overhead);
+    field_trials_.UseDeferredFec(GetParam().deferred_fec);
   }
 
   void SetUp() override { SetUpRtpSender(true, false, false); }
@@ -286,12 +300,20 @@
   void SetUpRtpSender(bool pacer,
                       bool populate_network2,
                       bool always_send_mid_and_rid) {
+    SetUpRtpSender(pacer, populate_network2, always_send_mid_and_rid,
+                   &flexfec_sender_);
+  }
+
+  void SetUpRtpSender(bool pacer,
+                      bool populate_network2,
+                      bool always_send_mid_and_rid,
+                      VideoFecGenerator* fec_generator) {
     RtpRtcpInterface::Configuration config;
     config.clock = &fake_clock_;
     config.outgoing_transport = &transport_;
     config.local_media_ssrc = kSsrc;
     config.rtx_send_ssrc = kRtxSsrc;
-    config.fec_generator = &flexfec_sender_;
+    config.fec_generator = fec_generator;
     config.event_log = &mock_rtc_event_log_;
     config.send_packet_observer = &send_packet_observer_;
     config.retransmission_rate_limiter = &retransmission_rate_limiter_;
@@ -1250,6 +1272,7 @@
   config.event_log = &mock_rtc_event_log_;
   config.send_packet_observer = &send_packet_observer_;
   config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+  config.field_trials = &field_trials_;
   rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
 
   rtp_sender()->SetSequenceNumber(kSeqNum);
@@ -1260,7 +1283,11 @@
   RTPSenderVideo::Config video_config;
   video_config.clock = &fake_clock_;
   video_config.rtp_sender = rtp_sender();
-  video_config.fec_generator = &flexfec_sender;
+  if (!GetParam().deferred_fec) {
+    video_config.fec_generator = &flexfec_sender;
+  }
+  video_config.fec_type = flexfec_sender.GetFecType();
+  video_config.fec_overhead_bytes = flexfec_sender.MaxPacketOverhead();
   video_config.fec_type = flexfec_sender.GetFecType();
   video_config.fec_overhead_bytes = flexfec_sender.MaxPacketOverhead();
   video_config.field_trials = &field_trials;
@@ -1286,11 +1313,21 @@
             EXPECT_EQ(packet->Ssrc(), kSsrc);
             EXPECT_EQ(packet->SequenceNumber(), kSeqNum);
             media_packet = std::move(packet);
+            if (GetParam().deferred_fec) {
+              // Simulate RtpSenderEgress adding packet to fec generator.
+              flexfec_sender.AddPacketAndGenerateFec(*media_packet);
+              auto fec_packets = flexfec_sender.GetFecPackets();
+              EXPECT_EQ(fec_packets.size(), 1u);
+              fec_packet = std::move(fec_packets[0]);
+              EXPECT_EQ(fec_packet->packet_type(),
+                        RtpPacketMediaType::kForwardErrorCorrection);
+              EXPECT_EQ(fec_packet->Ssrc(), kFlexFecSsrc);
+            }
           } else {
             EXPECT_EQ(packet->packet_type(),
                       RtpPacketMediaType::kForwardErrorCorrection);
-            EXPECT_EQ(packet->Ssrc(), kFlexFecSsrc);
             fec_packet = std::move(packet);
+            EXPECT_EQ(fec_packet->Ssrc(), kFlexFecSsrc);
           }
         }
       });
@@ -1338,6 +1375,7 @@
   config.event_log = &mock_rtc_event_log_;
   config.send_packet_observer = &send_packet_observer_;
   config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+  config.field_trials = &field_trials_;
   rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
 
   rtp_sender()->SetSequenceNumber(kSeqNum);
@@ -1346,7 +1384,9 @@
   RTPSenderVideo::Config video_config;
   video_config.clock = &fake_clock_;
   video_config.rtp_sender = rtp_sender();
-  video_config.fec_generator = &flexfec_sender;
+  if (!GetParam().deferred_fec) {
+    video_config.fec_generator = &flexfec_sender;
+  }
   video_config.fec_type = flexfec_sender.GetFecType();
   video_config.fec_overhead_bytes = flexfec_sender_.MaxPacketOverhead();
   video_config.field_trials = &field_trials;
@@ -1357,7 +1397,11 @@
   params.fec_rate = 15;
   params.max_fec_frames = 1;
   params.fec_mask_type = kFecMaskRandom;
-  flexfec_sender.SetProtectionParameters(params, params);
+  if (GetParam().deferred_fec) {
+    rtp_egress()->SetFecProtectionParameters(params, params);
+  } else {
+    flexfec_sender.SetProtectionParameters(params, params);
+  }
 
   EXPECT_CALL(mock_rtc_event_log_,
               LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
@@ -1662,25 +1706,16 @@
                                kNoRtpExtensions, kNoRtpExtensionSizes,
                                nullptr /* rtp_state */, &fake_clock_);
 
-  // Reset |rtp_sender_| to use FlexFEC.
-  RtpRtcpInterface::Configuration config;
-  config.clock = &fake_clock_;
-  config.outgoing_transport = &transport_;
-  config.paced_sender = &mock_paced_sender_;
-  config.local_media_ssrc = kSsrc;
-  config.fec_generator = &flexfec_sender;
-  config.event_log = &mock_rtc_event_log_;
-  config.send_packet_observer = &send_packet_observer_;
-  config.retransmission_rate_limiter = &retransmission_rate_limiter_;
-  rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
-
-  rtp_sender()->SetSequenceNumber(kSeqNum);
+  // Reset |rtp_sender_| to use this FlexFEC instance.
+  SetUpRtpSender(false, false, false, &flexfec_sender);
 
   FieldTrialBasedConfig field_trials;
   RTPSenderVideo::Config video_config;
   video_config.clock = &fake_clock_;
   video_config.rtp_sender = rtp_sender();
-  video_config.fec_generator = &flexfec_sender;
+  if (!GetParam().deferred_fec) {
+    video_config.fec_generator = &flexfec_sender;
+  }
   video_config.fec_type = flexfec_sender.GetFecType();
   video_config.fec_overhead_bytes = flexfec_sender.MaxPacketOverhead();
   video_config.field_trials = &field_trials;
@@ -1690,12 +1725,15 @@
   params.fec_rate = 15;
   params.max_fec_frames = 1;
   params.fec_mask_type = kFecMaskRandom;
-  flexfec_sender.SetProtectionParameters(params, params);
+  if (GetParam().deferred_fec) {
+    rtp_egress()->SetFecProtectionParameters(params, params);
+  } else {
+    flexfec_sender.SetProtectionParameters(params, params);
+  }
 
   constexpr size_t kNumMediaPackets = 10;
   constexpr size_t kNumFecPackets = kNumMediaPackets;
   constexpr int64_t kTimeBetweenPacketsMs = 10;
-  EXPECT_CALL(mock_paced_sender_, EnqueuePackets).Times(kNumMediaPackets);
   for (size_t i = 0; i < kNumMediaPackets; ++i) {
     RTPVideoHeader video_header;
 
@@ -1713,9 +1751,21 @@
   constexpr size_t kPayloadLength = sizeof(kPayloadData);
   constexpr size_t kPacketLength = kRtpHeaderLength + kFlexfecHeaderLength +
                                    kGenericCodecHeaderLength + kPayloadLength;
-  EXPECT_NEAR(kNumFecPackets * kPacketLength * 8 /
-                  (kNumFecPackets * kTimeBetweenPacketsMs / 1000.0f),
-              flexfec_sender.CurrentFecRate().bps<double>(), 500);
+
+  loop_.Flush();
+  if (GetParam().deferred_fec) {
+    EXPECT_NEAR(
+        kNumFecPackets * kPacketLength * 8 /
+            (kNumFecPackets * kTimeBetweenPacketsMs / 1000.0f),
+        rtp_egress()
+            ->GetSendRates()[RtpPacketMediaType::kForwardErrorCorrection]
+            .bps<double>(),
+        500);
+  } else {
+    EXPECT_NEAR(kNumFecPackets * kPacketLength * 8 /
+                    (kNumFecPackets * kTimeBetweenPacketsMs / 1000.0f),
+                flexfec_sender.CurrentFecRate().bps<double>(), 500);
+  }
 }
 
 TEST_P(RtpSenderTest, BitrateCallbacks) {
@@ -1862,15 +1912,18 @@
   const uint8_t kUlpfecPayloadType = 97;
   const uint8_t kPayloadType = 127;
   const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
-  FieldTrialBasedConfig field_trials;
+
   UlpfecGenerator ulpfec_generator(kRedPayloadType, kUlpfecPayloadType,
                                    &fake_clock_);
+  SetUpRtpSender(false, false, false, &ulpfec_generator);
   RTPSenderVideo::Config video_config;
   video_config.clock = &fake_clock_;
   video_config.rtp_sender = rtp_sender();
-  video_config.field_trials = &field_trials;
+  video_config.field_trials = &field_trials_;
   video_config.red_payload_type = kRedPayloadType;
-  video_config.fec_generator = &ulpfec_generator;
+  if (!GetParam().deferred_fec) {
+    video_config.fec_generator = &ulpfec_generator;
+  }
   video_config.fec_type = ulpfec_generator.GetFecType();
   video_config.fec_overhead_bytes = ulpfec_generator.MaxPacketOverhead();
   RTPSenderVideo rtp_sender_video(video_config);
@@ -1887,7 +1940,11 @@
   fec_params.fec_mask_type = kFecMaskRandom;
   fec_params.fec_rate = 1;
   fec_params.max_fec_frames = 1;
-  ulpfec_generator.SetProtectionParameters(fec_params, fec_params);
+  if (GetParam().deferred_fec) {
+    rtp_egress()->SetFecProtectionParameters(fec_params, fec_params);
+  } else {
+    ulpfec_generator.SetProtectionParameters(fec_params, fec_params);
+  }
   video_header.frame_type = VideoFrameType::kVideoFrameDelta;
   ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321,
                                          payload, nullptr, video_header,
@@ -2713,12 +2770,16 @@
 
 INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
                          RtpSenderTest,
-                         ::testing::Values(TestConfig{false},
-                                           TestConfig{true}));
+                         ::testing::Values(TestConfig{false, false},
+                                           TestConfig{false, true},
+                                           TestConfig{true, false},
+                                           TestConfig{false, false}));
 
 INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
                          RtpSenderTestWithoutPacer,
-                         ::testing::Values(TestConfig{false},
-                                           TestConfig{true}));
+                         ::testing::Values(TestConfig{false, false},
+                                           TestConfig{false, true},
+                                           TestConfig{true, false},
+                                           TestConfig{false, false}));
 
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 9ebfa77..7d870d2 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -598,8 +598,8 @@
       if (fec_generator_) {
         fec_generator_->AddPacketAndGenerateFec(*packet);
       } else {
-        // TODO(sprang): When deferred FEC generation is enabled, just mark the
-        // packet as protected here.
+        // Deferred FEC generation, just mark packet.
+        packet->set_fec_protect_packet(true);
       }
     }
 
@@ -607,6 +607,7 @@
       std::unique_ptr<RtpPacketToSend> red_packet(new RtpPacketToSend(*packet));
       BuildRedPayload(*packet, red_packet.get());
       red_packet->SetPayloadType(*red_payload_type_);
+      red_packet->set_is_red(true);
 
       // Send |red_packet| instead of |packet| for allocated sequence number.
       red_packet->set_packet_type(RtpPacketMediaType::kVideo);
diff --git a/modules/rtp_rtcp/source/ulpfec_generator.cc b/modules/rtp_rtcp/source/ulpfec_generator.cc
index 265fa4d..04cb59d 100644
--- a/modules/rtp_rtcp/source/ulpfec_generator.cc
+++ b/modules/rtp_rtcp/source/ulpfec_generator.cc
@@ -230,6 +230,8 @@
     total_fec_size_bytes += red_packet->size();
     red_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
     red_packet->set_allow_retransmission(false);
+    red_packet->set_is_red(true);
+    red_packet->set_fec_protect_packet(false);
     fec_packets.push_back(std::move(red_packet));
   }
 
diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc
index 37afe1b..873ef63 100644
--- a/test/scenario/video_stream_unittest.cc
+++ b/test/scenario/video_stream_unittest.cc
@@ -9,6 +9,7 @@
  */
 #include <atomic>
 
+#include "test/field_trial.h"
 #include "test/gtest.h"
 #include "test/scenario/scenario.h"
 
@@ -170,6 +171,25 @@
   EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
 }
 
+TEST(VideoStreamTest, SendsFecWithDeferredFlexFec) {
+  ScopedFieldTrials trial("WebRTC-DeferredFecGeneration/Enabled/");
+  Scenario s;
+  auto route =
+      s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+                     {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+                       c->loss_rate = 0.1;
+                       c->delay = TimeDelta::Millis(100);
+                     })},
+                     s.CreateClient("callee", CallClientConfig()),
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+    c->stream.use_flexfec = true;
+  });
+  s.RunFor(TimeDelta::Seconds(5));
+  VideoSendStream::Stats video_stats = video->send()->GetStats();
+  EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
+}
+
 TEST(VideoStreamTest, ResolutionAdaptsToAvailableBandwidth) {
   // Declared before scenario to avoid use after free.
   std::atomic<size_t> num_qvga_frames_(0);