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..a55ccc37b9b189dd7bb20bec06a57916e3672d27 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -882,7 +882,10 @@ 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): Uncomment the DCHECK line below when |ssrc_| cannot |
+ // change after the first packet has been sent. For more details, see |
+ // https://bugs.chromium.org/p/webrtc/issues/detail?id=6887. |
+ // RTC_DCHECK_EQ(ssrc, SSRC()); |
packet_history_.PutRtpPacket(std::move(packet), storage, true); |
} |