Temporarily remove SSRC DCHECK in RTPSender::SendToNetwork.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 864 matching lines @@
     {
       rtc::CritScope lock(&send_critsect_);
       media_has_been_sent_ = true;
     }
     UpdateRtpStats(*packet, false, false);
   }
 
   // To support retransmissions, we store the media packet as sent in the
   // packet history (even if send failed).
   if (storage == kAllowRetransmission) {
-    RTC_DCHECK_EQ(ssrc, SSRC());
+    // TODO(brandtr): Here we should RTC_DCHECK_EQ(ssrc, SSRC()), but that is

[danilchap, 2017/01/09 12:08:59]

Better to phrase TODO by stating what and when should be done. e.g.:
Uncomment RTC_DCHECK_EQ(ssrc, SSRC()) when ssrc will not change after a packet
is send. For details see
// bug-link as it is now.

[brandtr, 2017/01/09 12:16:08]

Done.

+    // currently not possible, as detailed by
+    // https://bugs.chromium.org/p/webrtc/issues/detail?id=6887.
     packet_history_.PutRtpPacket(std::move(packet), storage, true);
   }
 
   return sent;
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
@@ skipping 397 matching lines @@
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet =
         rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc