diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 5902886..ea33eb5 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -40,6 +40,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/system/fallthrough.h"
+#include "rtc_base/time_utils.h"
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 #include "video/receive_statistics_proxy.h"
@@ -52,6 +53,10 @@
 constexpr int kPacketBufferStartSize = 512;
 constexpr int kPacketBufferMaxSize = 2048;
 
+// Maximum time between frames before resetting reference_finder to avoid RTP
+// fields wraparounds to affect FrameBuffer.
+constexpr TimeDelta kInactiveStreamThreshold = TimeDelta::Seconds<5>();
+
 int PacketBufferMaxSize() {
   // The group here must be a positive power of 2, in which case that is used as
   // size. All other values shall result in the default value being used.
@@ -214,7 +219,9 @@
       rtcp_feedback_buffer_(this, nack_sender, this),
       packet_buffer_(clock_, kPacketBufferStartSize, PacketBufferMaxSize()),
       has_received_frame_(false),
-      frames_decryptable_(false) {
+      frames_decryptable_(false),
+      last_completed_picture_id_(0),
+      last_assembled_frame_time_(Timestamp::MinusInfinity()) {
   constexpr bool remb_candidate = true;
   if (packet_router_)
     packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
@@ -597,6 +604,15 @@
   RTC_DCHECK_RUN_ON(&network_tc_);
   RTC_DCHECK(frame);
 
+  bool recreate_reference_finder_requested = false;
+
+  Timestamp now = clock_->CurrentTime();
+  if (last_assembled_frame_time_.IsFinite() &&
+      now - last_assembled_frame_time_ > kInactiveStreamThreshold) {
+    recreate_reference_finder_requested = true;
+  }
+  last_assembled_frame_time_ = now;
+
   absl::optional<RtpGenericFrameDescriptor> descriptor =
       frame->GetGenericFrameDescriptor();
 
@@ -629,14 +645,7 @@
 
     if (frame->codec_type() != current_codec_) {
       if (frame_is_newer) {
-        // When we reset the |reference_finder_| we don't want new picture ids
-        // to overlap with old picture ids. To ensure that doesn't happen we
-        // start from the |last_completed_picture_id_| and add an offset in case
-        // of reordering.
-        reference_finder_ =
-            std::make_unique<video_coding::RtpFrameReferenceFinder>(
-                this, last_completed_picture_id_ +
-                          std::numeric_limits<uint16_t>::max());
+        recreate_reference_finder_requested = true;
         current_codec_ = frame->codec_type();
       } else {
         // Old frame from before the codec switch, discard it.
@@ -652,6 +661,16 @@
     last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
   }
 
+  if (recreate_reference_finder_requested) {
+    // When we reset the |reference_finder_| we don't want new picture ids
+    // to overlap with old picture ids. To ensure that doesn't happen we
+    // start from the |last_completed_picture_id_| and add an offset in case
+    // of reordering.
+    reference_finder_ = std::make_unique<video_coding::RtpFrameReferenceFinder>(
+        this,
+        last_completed_picture_id_ + std::numeric_limits<uint16_t>::max());
+  }
+
   if (buffered_frame_decryptor_ == nullptr) {
     reference_finder_->ManageFrame(std::move(frame));
   } else {
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 7021c3c..b7f38f8 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -319,7 +319,8 @@
   std::atomic<bool> frames_decryptable_;
   absl::optional<ColorSpace> last_color_space_;
 
-  int64_t last_completed_picture_id_ = 0;
+  int64_t last_completed_picture_id_;
+  Timestamp last_assembled_frame_time_ RTC_GUARDED_BY(network_tc_);
 };
 
 }  // namespace webrtc
diff --git a/video/video_receive_stream.cc b/video/video_receive_stream.cc
index a60bb07..aef1c51 100644
--- a/video/video_receive_stream.cc
+++ b/video/video_receive_stream.cc
@@ -159,10 +159,9 @@
   int64_t RenderTime() const override { return 0; }
 };
 
-// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
 // Maximum time between frames before resetting the FrameBuffer to avoid RTP
-// timestamps wraparound to affect FrameBuffer.
-constexpr int kInactiveStreamThresholdMs = 600000;  //  10 minutes.
+// timestamps and picture IDs wraparounds to affect FrameBuffer.
+constexpr int kInactiveStreamThresholdMs = 5000;  //  5 seconds.
 
 }  // namespace
 
@@ -546,7 +545,9 @@
 void VideoReceiveStream::OnCompleteFrame(
     std::unique_ptr<video_coding::EncodedFrame> frame) {
   RTC_DCHECK_RUN_ON(&network_sequence_checker_);
-  // TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
+
+  // Resetting of stream state if there was long enough pause in the stream.
+  // This is done to avoid undetected wraparounds in RTP fields.
   int64_t time_now_ms = rtc::TimeMillis();
   if (last_complete_frame_time_ms_ > 0 &&
       time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) {