diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index 349fe68039..7edc135037 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -57,7 +57,6 @@ struct RtpSenderObservers {
   BitrateStatisticsObserver* bitrate_observer;
   FrameCountObserver* frame_count_observer;
   RtcpPacketTypeCounterObserver* rtcp_type_observer;
-  SendSideDelayObserver* send_delay_observer;
   SendPacketObserver* send_packet_observer;
 };
 
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index bf0611db0c..438fa1b81f 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -224,7 +224,6 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
       observers.report_block_data_observer;
   configuration.paced_sender = transport->packet_sender();
   configuration.send_bitrate_observer = observers.bitrate_observer;
-  configuration.send_side_delay_observer = observers.send_delay_observer;
   configuration.send_packet_observer = observers.send_packet_observer;
   configuration.event_log = event_log;
   configuration.retransmission_rate_limiter = retransmission_rate_limiter;
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index d69c48b70f..0a7634ce7b 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -66,8 +66,7 @@ RtpSenderObservers CreateObservers(
     StreamDataCountersCallback* rtp_stats,
     BitrateStatisticsObserver* bitrate_observer,
     FrameCountObserver* frame_count_observer,
-    RtcpPacketTypeCounterObserver* rtcp_type_observer,
-    SendSideDelayObserver* send_delay_observer) {
+    RtcpPacketTypeCounterObserver* rtcp_type_observer) {
   RtpSenderObservers observers;
   observers.rtcp_rtt_stats = nullptr;
   observers.intra_frame_callback = intra_frame_callback;
@@ -77,7 +76,6 @@ RtpSenderObservers CreateObservers(
   observers.bitrate_observer = bitrate_observer;
   observers.frame_count_observer = frame_count_observer;
   observers.rtcp_type_observer = rtcp_type_observer;
-  observers.send_delay_observer = send_delay_observer;
   observers.send_packet_observer = nullptr;
   return observers;
 }
@@ -145,8 +143,7 @@ class RtpVideoSenderTestFixture {
         time_controller_.GetClock(), suspended_ssrcs, suspended_payload_states,
         config_.rtp, config_.rtcp_report_interval_ms, &transport_,
         CreateObservers(&encoder_feedback_, &stats_proxy_, &stats_proxy_,
-                        &stats_proxy_, frame_count_observer, &stats_proxy_,
-                        &stats_proxy_),
+                        &stats_proxy_, frame_count_observer, &stats_proxy_),
         &transport_controller_, &event_log_, &retransmission_rate_limiter_,
         std::make_unique<FecControllerDefault>(time_controller_.GetClock()),
         nullptr, CryptoOptions{}, frame_transformer,
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 6804ac3e2e..249cf835ba 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -406,19 +406,8 @@ class BitrateStatisticsObserver {
                       uint32_t ssrc) = 0;
 };
 
-// Callback, used to notify an observer whenever the send-side delay is updated.
-class SendSideDelayObserver {
- public:
-  virtual ~SendSideDelayObserver() {}
-  virtual void SendSideDelayUpdated(int avg_delay_ms,
-                                    int max_delay_ms,
-                                    uint32_t ssrc) = 0;
-};
-
 // Callback, used to notify an observer whenever a packet is sent to the
 // transport.
-// TODO(asapersson): This class will remove the need for SendSideDelayObserver.
-// Remove SendSideDelayObserver once possible.
 class SendPacketObserver {
  public:
   virtual ~SendPacketObserver() = default;
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
index e2531bb1b1..59ff9d70f1 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
@@ -23,7 +23,6 @@
 namespace webrtc {
 namespace {
 constexpr uint32_t kTimestampTicksPerMs = 90;
-constexpr int kSendSideDelayWindowMs = 1000;
 constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1);
 constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
 
@@ -73,15 +72,12 @@ DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
       is_audio_(config.audio),
       need_rtp_packet_infos_(config.need_rtp_packet_infos),
       transport_feedback_observer_(config.transport_feedback_callback),
-      send_side_delay_observer_(config.send_side_delay_observer),
       send_packet_observer_(config.send_packet_observer),
       rtp_stats_callback_(config.rtp_stats_callback),
       bitrate_callback_(config.send_bitrate_observer),
       media_has_been_sent_(false),
       force_part_of_allocation_(false),
       timestamp_offset_(0),
-      max_delay_it_(send_delays_.end()),
-      sum_delays_ms_(0),
       send_rates_(kNumMediaTypes, BitrateTracker(kBitrateStatisticsWindow)),
       rtp_sequence_number_map_(need_rtp_packet_infos_
                                    ? std::make_unique<RtpSequenceNumberMap>(
@@ -180,7 +176,6 @@ void DEPRECATED_RtpSenderEgress::SendPacket(
 
   if (packet->packet_type() != RtpPacketMediaType::kPadding &&
       packet->packet_type() != RtpPacketMediaType::kRetransmission) {
-    UpdateDelayStatistics(packet->capture_time().ms(), now_ms, packet_ssrc);
     UpdateOnSendPacket(options.packet_id, packet->capture_time().ms(),
                        packet_ssrc);
   }
@@ -317,88 +312,6 @@ void DEPRECATED_RtpSenderEgress::AddPacketToTransportFeedback(
   }
 }
 
-void DEPRECATED_RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
-                                                       int64_t now_ms,
-                                                       uint32_t ssrc) {
-  if (!send_side_delay_observer_ || capture_time_ms <= 0)
-    return;
-
-  int avg_delay_ms = 0;
-  int max_delay_ms = 0;
-  {
-    MutexLock lock(&lock_);
-    // Compute the max and average of the recent capture-to-send delays.
-    // The time complexity of the current approach depends on the distribution
-    // of the delay values. This could be done more efficiently.
-
-    // Remove elements older than kSendSideDelayWindowMs.
-    auto lower_bound =
-        send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
-    for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
-      if (max_delay_it_ == it) {
-        max_delay_it_ = send_delays_.end();
-      }
-      sum_delays_ms_ -= it->second;
-    }
-    send_delays_.erase(send_delays_.begin(), lower_bound);
-    if (max_delay_it_ == send_delays_.end()) {
-      // Removed the previous max. Need to recompute.
-      RecomputeMaxSendDelay();
-    }
-
-    // Add the new element.
-    RTC_DCHECK_GE(now_ms, 0);
-    RTC_DCHECK_LE(now_ms, std::numeric_limits<int64_t>::max() / 2);
-    RTC_DCHECK_GE(capture_time_ms, 0);
-    RTC_DCHECK_LE(capture_time_ms, std::numeric_limits<int64_t>::max() / 2);
-    int64_t diff_ms = now_ms - capture_time_ms;
-    RTC_DCHECK_GE(diff_ms, static_cast<int64_t>(0));
-    RTC_DCHECK_LE(diff_ms, std::numeric_limits<int>::max());
-    int new_send_delay = rtc::dchecked_cast<int>(now_ms - capture_time_ms);
-    SendDelayMap::iterator it;
-    bool inserted;
-    std::tie(it, inserted) =
-        send_delays_.insert(std::make_pair(now_ms, new_send_delay));
-    if (!inserted) {
-      // TODO(terelius): If we have multiple delay measurements during the same
-      // millisecond then we keep the most recent one. It is not clear that this
-      // is the right decision, but it preserves an earlier behavior.
-      int previous_send_delay = it->second;
-      sum_delays_ms_ -= previous_send_delay;
-      it->second = new_send_delay;
-      if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
-        RecomputeMaxSendDelay();
-      }
-    }
-    if (max_delay_it_ == send_delays_.end() ||
-        it->second >= max_delay_it_->second) {
-      max_delay_it_ = it;
-    }
-    sum_delays_ms_ += new_send_delay;
-
-    size_t num_delays = send_delays_.size();
-    RTC_DCHECK(max_delay_it_ != send_delays_.end());
-    max_delay_ms = rtc::dchecked_cast<int>(max_delay_it_->second);
-    int64_t avg_ms = (sum_delays_ms_ + num_delays / 2) / num_delays;
-    RTC_DCHECK_GE(avg_ms, static_cast<int64_t>(0));
-    RTC_DCHECK_LE(avg_ms,
-                  static_cast<int64_t>(std::numeric_limits<int>::max()));
-    avg_delay_ms =
-        rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
-  }
-  send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
-                                                  ssrc);
-}
-
-void DEPRECATED_RtpSenderEgress::RecomputeMaxSendDelay() {
-  max_delay_it_ = send_delays_.begin();
-  for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
-    if (it->second >= max_delay_it_->second) {
-      max_delay_it_ = it;
-    }
-  }
-}
-
 void DEPRECATED_RtpSenderEgress::UpdateOnSendPacket(int packet_id,
                                                     int64_t capture_time_ms,
                                                     uint32_t ssrc) {
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
index e786d90c2f..9d343c2d08 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
@@ -83,20 +83,11 @@ class DEPRECATED_RtpSenderEgress {
       RTC_LOCKS_EXCLUDED(lock_);
 
  private:
-  // Maps capture time in milliseconds to send-side delay in milliseconds.
-  // Send-side delay is the difference between transmission time and capture
-  // time.
-  typedef std::map<int64_t, int> SendDelayMap;
-
   RtpSendRates GetSendRatesLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
   bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
   void AddPacketToTransportFeedback(uint16_t packet_id,
                                     const RtpPacketToSend& packet,
                                     const PacedPacketInfo& pacing_info);
-  void UpdateDelayStatistics(int64_t capture_time_ms,
-                             int64_t now_ms,
-                             uint32_t ssrc);
-  void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
   void UpdateOnSendPacket(int packet_id,
                           int64_t capture_time_ms,
                           uint32_t ssrc);
@@ -119,7 +110,6 @@ class DEPRECATED_RtpSenderEgress {
   const bool need_rtp_packet_infos_;
 
   TransportFeedbackObserver* const transport_feedback_observer_;
-  SendSideDelayObserver* const send_side_delay_observer_;
   SendPacketObserver* const send_packet_observer_;
   StreamDataCountersCallback* const rtp_stats_callback_;
   BitrateStatisticsObserver* const bitrate_callback_;
@@ -129,10 +119,6 @@ class DEPRECATED_RtpSenderEgress {
   bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
   uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_);
 
-  SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
-  SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
-  // The sum of delays over a kSendSideDelayWindowMs sliding window.
-  int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_);
   StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
   StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
   // One element per value in RtpPacketMediaType, with index matching value.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 198d64f313..924c764ccd 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -95,7 +95,6 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
     VideoFecGenerator* fec_generator = nullptr;
 
     BitrateStatisticsObserver* send_bitrate_observer = nullptr;
-    SendSideDelayObserver* send_side_delay_observer = nullptr;
     RtcEventLog* event_log = nullptr;
     SendPacketObserver* send_packet_observer = nullptr;
     RateLimiter* retransmission_rate_limiter = nullptr;
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 64feec9b8e..7fcea096ec 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -23,7 +23,6 @@
 namespace webrtc {
 namespace {
 constexpr uint32_t kTimestampTicksPerMs = 90;
-constexpr TimeDelta kSendSideDelayWindow = TimeDelta::Seconds(1);
 constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1);
 constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
 constexpr TimeDelta kUpdateInterval = kBitrateStatisticsWindow;
@@ -96,15 +95,12 @@ RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
       need_rtp_packet_infos_(config.need_rtp_packet_infos),
       fec_generator_(config.fec_generator),
       transport_feedback_observer_(config.transport_feedback_callback),
-      send_side_delay_observer_(config.send_side_delay_observer),
       send_packet_observer_(config.send_packet_observer),
       rtp_stats_callback_(config.rtp_stats_callback),
       bitrate_callback_(config.send_bitrate_observer),
       media_has_been_sent_(false),
       force_part_of_allocation_(false),
       timestamp_offset_(0),
-      max_delay_it_(send_delays_.end()),
-      sum_delays_(TimeDelta::Zero()),
       send_rates_(kNumMediaTypes, BitrateTracker(kBitrateStatisticsWindow)),
       rtp_sequence_number_map_(need_rtp_packet_infos_
                                    ? std::make_unique<RtpSequenceNumberMap>(
@@ -263,11 +259,11 @@ void RtpSenderEgress::CompleteSendPacket(const Packet& compound_packet,
 
   options.additional_data = packet->additional_data();
 
-  const uint32_t packet_ssrc = packet->Ssrc();
   if (packet->packet_type() != RtpPacketMediaType::kPadding &&
-      packet->packet_type() != RtpPacketMediaType::kRetransmission) {
-    UpdateDelayStatistics(packet->capture_time(), now, packet_ssrc);
-    UpdateOnSendPacket(packet_id, packet->capture_time(), packet_ssrc);
+      packet->packet_type() != RtpPacketMediaType::kRetransmission &&
+      send_packet_observer_ != nullptr && packet->capture_time().IsFinite()) {
+    send_packet_observer_->OnSendPacket(packet_id, packet->capture_time(),
+                                        packet->Ssrc());
   }
   options.batchable = enable_send_packet_batching_ && !is_audio_;
   options.last_packet_in_batch = last_in_batch;
@@ -294,8 +290,8 @@ void RtpSenderEgress::CompleteSendPacket(const Packet& compound_packet,
     RTC_DCHECK(packet->packet_type().has_value());
     RtpPacketMediaType packet_type = *packet->packet_type();
     RtpPacketCounter counter(*packet);
-    size_t size = packet->size();
-    UpdateRtpStats(now, packet_ssrc, packet_type, std::move(counter), size);
+    UpdateRtpStats(now, packet->Ssrc(), packet_type, std::move(counter),
+                   packet->size());
   }
 }
 
@@ -441,83 +437,6 @@ void RtpSenderEgress::AddPacketToTransportFeedback(
   }
 }
 
-void RtpSenderEgress::UpdateDelayStatistics(Timestamp capture_time,
-                                            Timestamp now,
-                                            uint32_t ssrc) {
-  RTC_DCHECK_RUN_ON(worker_queue_);
-  if (!send_side_delay_observer_ || capture_time.IsInfinite())
-    return;
-
-  TimeDelta avg_delay = TimeDelta::Zero();
-  TimeDelta max_delay = TimeDelta::Zero();
-  {
-    // Compute the max and average of the recent capture-to-send delays.
-    // The time complexity of the current approach depends on the distribution
-    // of the delay values. This could be done more efficiently.
-
-    // Remove elements older than kSendSideDelayWindowMs.
-    auto lower_bound = send_delays_.lower_bound(now - kSendSideDelayWindow);
-    for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
-      if (max_delay_it_ == it) {
-        max_delay_it_ = send_delays_.end();
-      }
-      sum_delays_ -= it->second;
-    }
-    send_delays_.erase(send_delays_.begin(), lower_bound);
-    if (max_delay_it_ == send_delays_.end()) {
-      // Removed the previous max. Need to recompute.
-      RecomputeMaxSendDelay();
-    }
-
-    // Add the new element.
-    TimeDelta new_send_delay = now - capture_time;
-    auto [it, inserted] = send_delays_.emplace(now, new_send_delay);
-    if (!inserted) {
-      // TODO(terelius): If we have multiple delay measurements during the same
-      // millisecond then we keep the most recent one. It is not clear that this
-      // is the right decision, but it preserves an earlier behavior.
-      TimeDelta previous_send_delay = it->second;
-      sum_delays_ -= previous_send_delay;
-      it->second = new_send_delay;
-      if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
-        RecomputeMaxSendDelay();
-      }
-    }
-    if (max_delay_it_ == send_delays_.end() ||
-        it->second >= max_delay_it_->second) {
-      max_delay_it_ = it;
-    }
-    sum_delays_ += new_send_delay;
-
-    size_t num_delays = send_delays_.size();
-    RTC_DCHECK(max_delay_it_ != send_delays_.end());
-    max_delay = max_delay_it_->second;
-    avg_delay = sum_delays_ / num_delays;
-  }
-  send_side_delay_observer_->SendSideDelayUpdated(avg_delay.ms(),
-                                                  max_delay.ms(), ssrc);
-}
-
-void RtpSenderEgress::RecomputeMaxSendDelay() {
-  RTC_DCHECK_RUN_ON(worker_queue_);
-  max_delay_it_ = send_delays_.begin();
-  for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
-    if (it->second >= max_delay_it_->second) {
-      max_delay_it_ = it;
-    }
-  }
-}
-
-void RtpSenderEgress::UpdateOnSendPacket(absl::optional<uint16_t> packet_id,
-                                         Timestamp capture_time,
-                                         uint32_t ssrc) {
-  if (!send_packet_observer_ || capture_time.IsInfinite()) {
-    return;
-  }
-
-  send_packet_observer_->OnSendPacket(packet_id, capture_time, ssrc);
-}
-
 bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
                                           const PacketOptions& options,
                                           const PacedPacketInfo& pacing_info) {
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index 7d32c8576f..42f3c96ff0 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -113,13 +113,7 @@ class RtpSenderEgress {
   void AddPacketToTransportFeedback(uint16_t packet_id,
                                     const RtpPacketToSend& packet,
                                     const PacedPacketInfo& pacing_info);
-  void UpdateDelayStatistics(Timestamp capture_time,
-                             Timestamp now,
-                             uint32_t ssrc);
-  void RecomputeMaxSendDelay();
-  void UpdateOnSendPacket(absl::optional<uint16_t> packet_id,
-                          Timestamp capture_time,
-                          uint32_t ssrc);
+
   // Sends packet on to `transport_`, leaving the RTP module.
   bool SendPacketToNetwork(const RtpPacketToSend& packet,
                            const PacketOptions& options,
@@ -153,7 +147,6 @@ class RtpSenderEgress {
   absl::optional<uint16_t> last_sent_rtx_seq_ RTC_GUARDED_BY(worker_queue_);
 
   TransportFeedbackObserver* const transport_feedback_observer_;
-  SendSideDelayObserver* const send_side_delay_observer_;
   SendPacketObserver* const send_packet_observer_;
   StreamDataCountersCallback* const rtp_stats_callback_;
   BitrateStatisticsObserver* const bitrate_callback_;
@@ -162,13 +155,6 @@ class RtpSenderEgress {
   bool force_part_of_allocation_ RTC_GUARDED_BY(worker_queue_);
   uint32_t timestamp_offset_ RTC_GUARDED_BY(worker_queue_);
 
-  // Maps capture time to send-side delay. Send-side delay is the difference
-  // between transmission time and capture time.
-  std::map<Timestamp, TimeDelta> send_delays_ RTC_GUARDED_BY(worker_queue_);
-  std::map<Timestamp, TimeDelta>::const_iterator max_delay_it_
-      RTC_GUARDED_BY(worker_queue_);
-  // The sum of delays over a kSendSideDelayWindowMs sliding window.
-  TimeDelta sum_delays_ RTC_GUARDED_BY(worker_queue_);
   StreamDataCounters rtp_stats_ RTC_GUARDED_BY(worker_queue_);
   StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(worker_queue_);
   // One element per value in RtpPacketMediaType, with index matching value.
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index a57293c43b..b278ea2f06 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -77,11 +77,6 @@ class MockStreamDataCountersCallback : public StreamDataCountersCallback {
               (override));
 };
 
-class MockSendSideDelayObserver : public SendSideDelayObserver {
- public:
-  MOCK_METHOD(void, SendSideDelayUpdated, (int, int, uint32_t), (override));
-};
-
 struct TransmittedPacket {
   TransmittedPacket(rtc::ArrayView<const uint8_t> data,
                     const PacketOptions& packet_options,
@@ -329,48 +324,6 @@ TEST_F(RtpSenderEgressTest,
   EXPECT_EQ(offset, 0);
 }
 
-TEST_F(RtpSenderEgressTest, OnSendSideDelayUpdated) {
-  StrictMock<MockSendSideDelayObserver> send_side_delay_observer;
-  RtpRtcpInterface::Configuration config = DefaultConfig();
-  config.send_side_delay_observer = &send_side_delay_observer;
-  auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
-
-  // Send packet with 10 ms send-side delay. The average, max and total should
-  // be 10 ms.
-  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(10, 10, kSsrc));
-  int64_t capture_time_ms = clock_->TimeInMilliseconds();
-  time_controller_.AdvanceTime(TimeDelta::Millis(10));
-  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms),
-                     PacedPacketInfo());
-
-  // Send another packet with 20 ms delay. The average, max and total should be
-  // 15, 20 and 30 ms respectively.
-  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(15, 20, kSsrc));
-  capture_time_ms = clock_->TimeInMilliseconds();
-  time_controller_.AdvanceTime(TimeDelta::Millis(20));
-  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms),
-                     PacedPacketInfo());
-
-  // Send another packet at the same time, which replaces the last packet.
-  // Since this packet has 0 ms delay, the average is now 5 ms and max is 10 ms.
-  // The total counter stays the same though.
-  // TODO(terelius): Is is not clear that this is the right behavior.
-  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(5, 10, kSsrc));
-  capture_time_ms = clock_->TimeInMilliseconds();
-  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms),
-                     PacedPacketInfo());
-
-  // Send a packet 1 second later. The earlier packets should have timed
-  // out, so both max and average should be the delay of this packet. The total
-  // keeps increasing.
-  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
-  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(1, 1, kSsrc));
-  capture_time_ms = clock_->TimeInMilliseconds();
-  time_controller_.AdvanceTime(TimeDelta::Millis(1));
-  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms),
-                     PacedPacketInfo());
-}
-
 TEST_F(RtpSenderEgressTest, WritesPacerExitToTimingExtension) {
   std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
   header_extensions_.RegisterByUri(kVideoTimingExtensionId,
@@ -852,7 +805,6 @@ TEST_F(RtpSenderEgressTest, SendPacketSetsPacketOptions) {
 
 TEST_F(RtpSenderEgressTest, SendPacketUpdatesStats) {
   const size_t kPayloadSize = 1000;
-  StrictMock<MockSendSideDelayObserver> send_side_delay_observer;
 
   const rtc::ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
   FlexfecSender flexfec(kFlexfectPayloadType, kFlexFecSsrc, kSsrc, /*mid=*/"",
@@ -860,7 +812,6 @@ TEST_F(RtpSenderEgressTest, SendPacketUpdatesStats) {
                         /*rtp_state=*/nullptr, time_controller_.GetClock());
   RtpRtcpInterface::Configuration config = DefaultConfig();
   config.fec_generator = &flexfec;
-  config.send_side_delay_observer = &send_side_delay_observer;
   auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
 
   header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
@@ -889,11 +840,6 @@ TEST_F(RtpSenderEgressTest, SendPacketUpdatesStats) {
   const int64_t kDiffMs = 25;
   time_controller_.AdvanceTime(TimeDelta::Millis(kDiffMs));
 
-  EXPECT_CALL(send_side_delay_observer,
-              SendSideDelayUpdated(kDiffMs, kDiffMs, kSsrc));
-  EXPECT_CALL(send_side_delay_observer,
-              SendSideDelayUpdated(kDiffMs, kDiffMs, kFlexFecSsrc));
-
   EXPECT_CALL(send_packet_observer_, OnSendPacket(Eq(1), capture_time, kSsrc));
 
   sender->SendPacket(std::move(video_packet), PacedPacketInfo());
diff --git a/video/send_statistics_proxy.h b/video/send_statistics_proxy.h
index b7c8d7c5a4..e9ac24391e 100644
--- a/video/send_statistics_proxy.h
+++ b/video/send_statistics_proxy.h
@@ -43,8 +43,7 @@ class SendStatisticsProxy : public VideoStreamEncoderObserver,
                             public RtcpPacketTypeCounterObserver,
                             public StreamDataCountersCallback,
                             public BitrateStatisticsObserver,
-                            public FrameCountObserver,
-                            public SendSideDelayObserver {
+                            public FrameCountObserver {
  public:
   static constexpr TimeDelta kStatsTimeout = TimeDelta::Seconds(5);
   // Number of required samples to be collected before a metric is added
@@ -129,11 +128,6 @@ class SendStatisticsProxy : public VideoStreamEncoderObserver,
   void FrameCountUpdated(const FrameCounts& frame_counts,
                          uint32_t ssrc) override;
 
-  // From SendSideDelayObserver.
-  void SendSideDelayUpdated(int avg_delay_ms,
-                            int max_delay_ms,
-                            uint32_t ssrc) override {}
-
  private:
   class SampleCounter {
    public:
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 571eb62c6b..b99b08eefb 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -101,7 +101,6 @@ RtpSenderObservers CreateObservers(RtcpRttStats* call_stats,
   observers.bitrate_observer = stats_proxy;
   observers.frame_count_observer = stats_proxy;
   observers.rtcp_type_observer = stats_proxy;
-  observers.send_delay_observer = nullptr;
   observers.send_packet_observer = send_packet_observer;
   return observers;
 }