diff --git a/modules/rtp_rtcp/source/rtp_packet_history.cc b/modules/rtp_rtcp/source/rtp_packet_history.cc
index 253e9862be..d63d8032df 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -362,6 +362,11 @@ bool RtpPacketHistory::SetPendingTransmission(uint16_t sequence_number) {
   return true;
 }
 
+void RtpPacketHistory::Clear() {
+  rtc::CritScope cs(&lock_);
+  Reset();
+}
+
 void RtpPacketHistory::Reset() {
   packet_history_.clear();
   padding_priority_.clear();
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.h b/modules/rtp_rtcp/source/rtp_packet_history.h
index e477b45593..54c774e663 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -129,6 +129,10 @@ class RtpPacketHistory {
   // Returns true if status was set, false if packet was not found.
   bool SetPendingTransmission(uint16_t sequence_number);
 
+  // Remove all pending packets from the history, but keep storage mode and
+  // capacity.
+  void Clear();
+
  private:
   struct MoreUseful;
   class StoredPacket;
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 870cc2a869..69066b8342 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1417,16 +1417,21 @@ uint32_t RTPSender::TimestampOffset() const {
 }
 
 void RTPSender::SetSSRC(uint32_t ssrc) {
-  // This is configured via the API.
-  rtc::CritScope lock(&send_critsect_);
+  {
+    rtc::CritScope lock(&send_critsect_);
+    if (ssrc_ == ssrc) {
+      return;  // Since it's the same SSRC, don't reset anything.
+    }
 
-  if (ssrc_ == ssrc) {
-    return;  // Since it's same ssrc, don't reset anything.
-  }
-  ssrc_.emplace(ssrc);
-  if (!sequence_number_forced_) {
-    sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+    ssrc_.emplace(ssrc);
+    if (!sequence_number_forced_) {
+      sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+    }
   }
+
+  // Clear RTP packet history, since any packets there belong to the old SSRC
+  // and they may conflict with packets from the new one.
+  packet_history_.Clear();
 }
 
 uint32_t RTPSender::SSRC() const {
@@ -1459,9 +1464,21 @@ void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
 }
 
 void RTPSender::SetSequenceNumber(uint16_t seq) {
-  rtc::CritScope lock(&send_critsect_);
-  sequence_number_forced_ = true;
-  sequence_number_ = seq;
+  bool updated_sequence_number = false;
+  {
+    rtc::CritScope lock(&send_critsect_);
+    sequence_number_forced_ = true;
+    if (sequence_number_ != seq) {
+      updated_sequence_number = true;
+    }
+    sequence_number_ = seq;
+  }
+
+  if (updated_sequence_number) {
+    // Sequence number series has been reset to a new value, clear RTP packet
+    // history, since any packets there may conflict with new ones.
+    packet_history_.Clear();
+  }
 }
 
 uint16_t RTPSender::SequenceNumber() const {
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index ec12780756..a93a1a6415 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -2708,6 +2708,63 @@ TEST_P(RtpSenderTest, SetsCaptureTimeAndPopulatesTransmissionOffset) {
   EXPECT_EQ(*transmission_time_extension, 2 * kOffsetMs * kTimestampTicksPerMs);
 }
 
+TEST_P(RtpSenderTestWithoutPacer, ClearHistoryOnSsrcChange) {
+  const int64_t kRtt = 10;
+
+  rtp_sender_->SetSendingMediaStatus(true);
+  rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+  rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+  rtp_sender_->SetRtt(kRtt);
+  rtp_sender_->SetSSRC(kSsrc);
+
+  // Send a packet and record its sequence numbers.
+  SendGenericPacket();
+  ASSERT_EQ(1u, transport_.sent_packets_.size());
+  const uint16_t packet_seqence_number =
+      transport_.sent_packets_.back().SequenceNumber();
+
+  // Advance time and make sure it can be retransmitted, even if we try to set
+  // the ssrc the what it already is.
+  rtp_sender_->SetSSRC(kSsrc);
+  fake_clock_.AdvanceTimeMilliseconds(kRtt);
+  EXPECT_GT(rtp_sender_->ReSendPacket(packet_seqence_number), 0);
+
+  // Change the SSRC, then move the time and try to retransmit again. The old
+  // packet should now be gone.
+  rtp_sender_->SetSSRC(kSsrc + 1);
+  fake_clock_.AdvanceTimeMilliseconds(kRtt);
+  EXPECT_EQ(rtp_sender_->ReSendPacket(packet_seqence_number), 0);
+}
+
+TEST_P(RtpSenderTestWithoutPacer, ClearHistoryOnSequenceNumberCange) {
+  const int64_t kRtt = 10;
+
+  rtp_sender_->SetSendingMediaStatus(true);
+  rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+  rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+  rtp_sender_->SetRtt(kRtt);
+
+  // Send a packet and record its sequence numbers.
+  SendGenericPacket();
+  ASSERT_EQ(1u, transport_.sent_packets_.size());
+  const uint16_t packet_seqence_number =
+      transport_.sent_packets_.back().SequenceNumber();
+
+  // Advance time and make sure it can be retransmitted, even if we try to set
+  // the ssrc the what it already is.
+  rtp_sender_->SetSequenceNumber(rtp_sender_->SequenceNumber());
+  fake_clock_.AdvanceTimeMilliseconds(kRtt);
+  EXPECT_GT(rtp_sender_->ReSendPacket(packet_seqence_number), 0);
+
+  // Change the sequence number, then move the time and try to retransmit again.
+  // The old packet should now be gone.
+  rtp_sender_->SetSequenceNumber(rtp_sender_->SequenceNumber() - 1);
+  fake_clock_.AdvanceTimeMilliseconds(kRtt);
+  EXPECT_EQ(rtp_sender_->ReSendPacket(packet_seqence_number), 0);
+}
+
 INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
                          RtpSenderTest,
                          ::testing::Values(TestConfig{false, false},