Set send times in send time history via OnSentPacket.

webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 27 matching lines @@
     const std::vector<PacketInfo>& packet_feedback_vector) {
   rtc::CritScope cs(&lock_);
   for (PacketInfo info : packet_feedback_vector)
     OnPacketArrival(info.sequence_number, info.arrival_time_ms);
 }
 
 void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms,
                                           size_t payload_size,
                                           const RTPHeader& header,
                                           bool was_paced) {
-  RTC_DCHECK(header.extension.hasTransportSequenceNumber);
+  if (!header.extension.hasTransportSequenceNumber) {
+    LOG(LS_WARNING) << "RemoteEstimatorProxy: Incoming packet "
+                       "is missing the transport sequence number extension!";
+    return;
+  }
   rtc::CritScope cs(&lock_);
   media_ssrc_ = header.ssrc;
   OnPacketArrival(header.extension.transportSequenceNumber, arrival_time_ms);
 }
 
 void RemoteEstimatorProxy::RemoveStream(unsigned int ssrc) {}
 
 bool RemoteEstimatorProxy::LatestEstimate(std::vector<unsigned int>* ssrcs,
                                           unsigned int* bitrate_bps) const {
   return false;
@@ skipping 92 matching lines @@
     // they need to be re-sent after a reordering. Removal will be handled
     // by OnPacketArrival once packets are too old.
   }
   if (it == packet_arrival_times_.end())
     window_start_seq_ = -1;
 
   return true;
 }
 
 }  // namespace webrtc