diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index b19ce3880a..3977ee4f20 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -55,6 +55,7 @@ using rtcp::ReportBlock;
 const int kRrTimeoutIntervals = 3;
 
 const int64_t kMaxWarningLogIntervalMs = 10000;
+const int64_t kRtcpMinFrameLengthMs = 17;
 
 }  // namespace
 
@@ -80,9 +81,6 @@ struct RTCPReceiver::ReceiveInformation {
 
   int64_t last_time_received_ms = 0;
 
-  int64_t last_fir_request_ms = 0;
-  int32_t last_fir_sequence_number = -1;
-
   bool ready_for_delete = false;
 
   std::vector<rtcp::TmmbItem> tmmbn;
@@ -99,6 +97,13 @@ struct RTCPReceiver::ReportBlockWithRtt {
   size_t num_rtts = 0;
 };
 
+struct RTCPReceiver::LastFirStatus {
+  LastFirStatus(int64_t now_ms, uint8_t sequence_number)
+      : request_ms(now_ms), sequence_number(sequence_number) {}
+  int64_t request_ms;
+  uint8_t sequence_number;
+};
+
 RTCPReceiver::RTCPReceiver(
     Clock* clock,
     bool receiver_only,
@@ -683,6 +688,7 @@ void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
   if (receive_info)
     receive_info->ready_for_delete = true;
 
+  last_fir_.erase(bye.sender_ssrc());
   received_cnames_.erase(bye.sender_ssrc());
   xr_rr_rtt_ms_ = 0;
 }
@@ -880,8 +886,6 @@ void RTCPReceiver::HandleFir(const CommonHeader& rtcp_block,
     return;
   }
 
-  ReceiveInformation* receive_info = GetReceiveInformation(fir.sender_ssrc());
-
   for (const rtcp::Fir::Request& fir_request : fir.requests()) {
     // Is it our sender that is requested to generate a new keyframe.
     if (main_ssrc_ != fir_request.ssrc)
@@ -889,18 +893,22 @@ void RTCPReceiver::HandleFir(const CommonHeader& rtcp_block,
 
     ++packet_type_counter_.fir_packets;
 
-    if (receive_info) {
+    int64_t now_ms = clock_->TimeInMilliseconds();
+    auto inserted = last_fir_.insert(std::make_pair(
+        fir.sender_ssrc(), LastFirStatus(now_ms, fir_request.seq_nr)));
+    if (!inserted.second) {  // There was already an entry.
+      LastFirStatus* last_fir = &inserted.first->second;
+
       // Check if we have reported this FIRSequenceNumber before.
-      if (fir_request.seq_nr == receive_info->last_fir_sequence_number)
+      if (fir_request.seq_nr == last_fir->sequence_number)
         continue;
 
-      int64_t now_ms = clock_->TimeInMilliseconds();
       // Sanity: don't go crazy with the callbacks.
-      if (now_ms - receive_info->last_fir_request_ms < RTCP_MIN_FRAME_LENGTH_MS)
+      if (now_ms - last_fir->request_ms < kRtcpMinFrameLengthMs)
         continue;
 
-      receive_info->last_fir_request_ms = now_ms;
-      receive_info->last_fir_sequence_number = fir_request.seq_nr;
+      last_fir->request_ms = now_ms;
+      last_fir->sequence_number = fir_request.seq_nr;
     }
     // Received signal that we need to send a new key frame.
     packet_information->packet_type_flags |= kRtcpFir;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index b86b03060c..26defae680 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -119,6 +119,7 @@ class RTCPReceiver {
   struct PacketInformation;
   struct ReceiveInformation;
   struct ReportBlockWithRtt;
+  struct LastFirStatus;
   // Mapped by remote ssrc.
   using ReceivedInfoMap = std::map<uint32_t, ReceiveInformation>;
   // RTCP report blocks mapped by remote SSRC.
@@ -245,6 +246,7 @@ class RTCPReceiver {
   ReportBlockMap received_report_blocks_ GUARDED_BY(rtcp_receiver_lock_);
   ReceivedInfoMap received_infos_ GUARDED_BY(rtcp_receiver_lock_);
   int64_t oldest_received_info_ms_ GUARDED_BY(rtcp_receiver_lock_);
+  std::map<uint32_t, LastFirStatus> last_fir_ GUARDED_BY(rtcp_receiver_lock_);
   std::map<uint32_t, std::string> received_cnames_
       GUARDED_BY(rtcp_receiver_lock_);
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
index f09bf55617..5d0505816e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -23,7 +23,6 @@ enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
 enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
 enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
 enum { RTCP_MAX_REPORT_BLOCKS = 31 };  // RFC 3550 page 37
-enum { RTCP_MIN_FRAME_LENGTH_MS = 17 };
 enum {
   kRtcpAppCode_DATA_SIZE = 32 * 4
 };  // multiple of 4, this is not a limitation of the size