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Issue 2542233002: Relaxing DCHECK for packets sent before SRTP is enabled. (Closed)
Patch Set: Created 4 years ago
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1 /* 1 /*
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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720 GetRtcpType(data, len, &type); 720 GetRtcpType(data, len, &type);
721 LOG(LS_ERROR) << "Failed to protect " << content_name_ 721 LOG(LS_ERROR) << "Failed to protect " << content_name_
722 << " RTCP packet: size=" << len << ", type=" << type; 722 << " RTCP packet: size=" << len << ", type=" << type;
723 return false; 723 return false;
724 } 724 }
725 } 725 }
726 726
727 // Update the length of the packet now that we've added the auth tag. 727 // Update the length of the packet now that we've added the auth tag.
728 packet->SetSize(len); 728 packet->SetSize(len);
729 } else if (secure_required_) { 729 } else if (secure_required_) {
730 // This is a double check for something that supposedly can't happen. 730 // The audio/video engines may attempt to send RTCP packets as soon as the
731 LOG(LS_ERROR) << "Can't send outgoing " << PacketType(rtcp) 731 // streams are created, so don't treat this as an error for RTCP.
732 << " packet when SRTP is inactive and crypto is required"; 732 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
733 733 if (rtcp) {
734 return false;
735 }
736 // However, there shouldn't be any RTP packets sent before SRTP is set up
737 // (and SetSend(true) is called).
738 LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive"
739 << " and crypto is required";
734 RTC_DCHECK(false); 740 RTC_DCHECK(false);
735 return false; 741 return false;
736 } 742 }
737 743
738 // Bon voyage. 744 // Bon voyage.
739 int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL; 745 int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL;
740 int ret = channel->SendPacket(packet->data<char>(), packet->size(), 746 int ret = channel->SendPacket(packet->data<char>(), packet->size(),
741 updated_options, flags); 747 updated_options, flags);
742 if (ret != static_cast<int>(packet->size())) { 748 if (ret != static_cast<int>(packet->size())) {
743 if (channel->GetError() == ENOTCONN) { 749 if (channel->GetError() == ENOTCONN) {
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2457 } 2463 }
2458 2464
2459 void DataChannel::OnStreamClosedRemotely(uint32_t sid) { 2465 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
2460 rtc::TypedMessageData<uint32_t>* message = 2466 rtc::TypedMessageData<uint32_t>* message =
2461 new rtc::TypedMessageData<uint32_t>(sid); 2467 new rtc::TypedMessageData<uint32_t>(sid);
2462 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_STREAMCLOSEDREMOTELY, 2468 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_STREAMCLOSEDREMOTELY,
2463 message); 2469 message);
2464 } 2470 }
2465 2471
2466 } // namespace cricket 2472 } // namespace cricket
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