diff --git a/call/call.cc b/call/call.cc
index 0d58ed8ed9..b30b92f861 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -466,6 +466,10 @@ class Call final : public webrtc::Call,
 
   bool is_started_ RTC_GUARDED_BY(worker_thread_) = false;
 
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
+  absl::optional<rtc::SentPacket> last_sent_packet_
+      RTC_GUARDED_BY(sent_packet_sequence_checker_);
+
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
@@ -814,6 +818,7 @@ Call::Call(Clock* clock,
   RTC_DCHECK(worker_thread_->IsCurrent());
 
   send_transport_sequence_checker_.Detach();
+  sent_packet_sequence_checker_.Detach();
 
   // Do not remove this call; it is here to convince the compiler that the
   // WebRTC source timestamp string needs to be in the final binary.
@@ -1382,6 +1387,20 @@ void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream,
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  RTC_DCHECK_RUN_ON(&sent_packet_sequence_checker_);
+  // When bundling is in effect, multiple senders may be sharing the same
+  // transport. It means every |sent_packet| will be multiply notified from
+  // different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel. Record
+  // |last_sent_packet_| to deduplicate redundant notifications to downstream.
+  // (https://crbug.com/webrtc/13437): Pass all packets without a |packet_id| to
+  // downstream.
+  if (last_sent_packet_.has_value() && last_sent_packet_->packet_id != -1 &&
+      last_sent_packet_->packet_id == sent_packet.packet_id &&
+      last_sent_packet_->send_time_ms == sent_packet.send_time_ms) {
+    return;
+  }
+  last_sent_packet_ = sent_packet;
+
   // In production and with most tests, this method will be called on the
   // network thread. However some test classes such as DirectTransport don't
   // incorporate a network thread. This means that tests for RtpSenderEgress