Temporarily remove SSRC DCHECK in RTPSender::SendToNetwork.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 4a285ca2198e04ec4de86e742c34fa118ededd42..99cafb6615a0ecbb11e846a2ea50504cff8c0e99 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -882,7 +882,9 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
   // 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);
   }