[rtp_rtcp] Fix potentional time difference between rtp and rtcp packets.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  Copyright (c) 2012 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 273 matching lines @@
 }
 
 // Set SequenceNumber, default is a random number.
 void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
   rtp_sender_.SetSequenceNumber(seq_num);
 }
 
 bool ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc,
                                            const RtpState& rtp_state) {
   if (rtp_sender_.SSRC() == ssrc) {
+    SetStartTimestamp(rtp_state.start_timestamp);
     rtp_sender_.SetRtpState(rtp_state);
     return true;
   }
   if (rtp_sender_.RtxSsrc() == ssrc) {
     rtp_sender_.SetRtxRtpState(rtp_state);
     return true;
   }
   return false;
 }
 
@@ skipping 688 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc