WebRTC - VP9 Processing Use-After-Free

EDB-ID:

45443




Platform:

Multiple

Date:

2018-09-21


Become a Certified Penetration Tester

Enroll in Penetration Testing with Kali Linux and pass the exam to become an Offensive Security Certified Professional (OSCP). All new content for 2020.

GET CERTIFIED

There is a use-after-free in VP9 processing in WebRTC. In the method RtpFrameReferenceFinder::ManageFrameVp9 the following code occurs:

 auto gof_info_it = gof_info_.find((codec_header.temporal_idx == 0)
                                          ? codec_header.tl0_pic_idx - 1
                                          : codec_header.tl0_pic_idx);

   ... // snip

    info = &gof_info_it->second;
  }

  // Clean up info for base layers that are too old.
  uint8_t old_tl0_pic_idx = codec_header.tl0_pic_idx - kMaxGofSaved;
  auto clean_gof_info_to = gof_info_.lower_bound(old_tl0_pic_idx);
  gof_info_.erase(gof_info_.begin(), clean_gof_info_to);

  FrameReceivedVp9(frame->id.picture_id, info);

tl0_pic_idx is extracted from the incoming packet, and it if is higher than any picture id that exists in gof_info_, the entire vector will be erased, and info will be used in the call FrameReceivedVp9 even though it has been freed.

ASAN output:

==163231==ERROR: AddressSanitizer: heap-use-after-free on address 0x6060000031d0 at pc 0x0000014b0e1e bp 0x7ffe607dfd30 sp 0x7ffe607dfd28
READ of size 2 at 0x6060000031d0 thread T0
    #0 0x14b0e1d in webrtc::video_coding::RtpFrameReferenceFinder::FrameReceivedVp9(unsigned short, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo*) modules/video_coding/rtp_frame_reference_finder.cc:569:31
    #1 0x14ac2c5 in webrtc::video_coding::RtpFrameReferenceFinder::ManageFrameVp9(webrtc::video_coding::RtpFrameObject*) modules/video_coding/rtp_frame_reference_finder.cc:499:3
    #2 0x14a7849 in ManageFrameInternal modules/video_coding/rtp_frame_reference_finder.cc:89:14
    #3 0x14a7849 in webrtc::video_coding::RtpFrameReferenceFinder::ManageFrame(std::__1::unique_ptr<webrtc::video_coding::RtpFrameObject, std::__1::default_delete<webrtc::video_coding::RtpFrameObject> >) modules/video_coding/rtp_frame_reference_finder.cc:43
    #4 0x148a87e in non-virtual thunk to webrtc::RtpVideoStreamReceiver::OnReceivedFrame(std::__1::unique_ptr<webrtc::video_coding::RtpFrameObject, std::__1::default_delete<webrtc::video_coding::RtpFrameObject> >) video/rtp_video_stream_receiver.cc:336:22
    #5 0x1496f41 in webrtc::video_coding::PacketBuffer::InsertPacket(webrtc::VCMPacket*) modules/video_coding/packet_buffer.cc:130:31
    #6 0x1487e59 in webrtc::RtpVideoStreamReceiver::OnReceivedPayloadData(unsigned char const*, unsigned long, webrtc::WebRtcRTPHeader const*) video/rtp_video_stream_receiver.cc:231:19
    #7 0x12d9144 in webrtc::RTPReceiverVideo::ParseRtpPacket(webrtc::WebRtcRTPHeader*, webrtc::PayloadUnion const&, unsigned char const*, unsigned long, long) modules/rtp_rtcp/source/rtp_receiver_video.cc:109:26
    #8 0x12cc80d in webrtc::RtpReceiverImpl::IncomingRtpPacket(webrtc::RTPHeader const&, unsigned char const*, unsigned long, webrtc::PayloadUnion) modules/rtp_rtcp/source/rtp_receiver_impl.cc:181:42
    #9 0x1488e52 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:399:20
    #10 0x1488b03 in webrtc::RtpVideoStreamReceiver::OnRecoveredPacket(unsigned char const*, unsigned long) video/rtp_video_stream_receiver.cc:245:3
    #11 0x14b925c in webrtc::UlpfecReceiverImpl::ProcessReceivedFec() modules/rtp_rtcp/source/ulpfec_receiver_impl.cc:244:35
    #12 0x148bd42 in webrtc::RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:421:23
    #13 0x1488d51 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:390:5
    #14 0x14899f8 in webrtc::RtpVideoStreamReceiver::OnRtpPacket(webrtc::RtpPacketReceived const&) video/rtp_video_stream_receiver.cc:290:3
    #15 0x90c486 in webrtc::RtpDemuxer::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_demuxer.cc:157:11
    #16 0x9131bd in webrtc::RtpStreamReceiverController::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_stream_receiver_controller.cc:55:19
    #17 0x129940d in webrtc::internal::Call::DeliverRtp(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1321:36
    #18 0x129a8d5 in webrtc::internal::Call::DeliverPacket(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1361:10
    #19 0x61fe06 in webrtc::RtpReplay() video/replay.cc:279:31
    #20 0x62337d in main video/replay.cc:343:3
    #21 0x7f5ae03d82b0 in __libc_start_main (/lib/x86_64-linux-gnu/libc.so.6+0x202b0)

0x6060000031d0 is located 48 bytes inside of 56-byte region [0x6060000031a0,0x6060000031d8)
freed by thread T0 here:
    #0 0x61bbb2 in operator delete(void*) /b/build/slave/linux_upload_clang/build/src/third_party/llvm/compiler-rt/lib/asan/asan_new_delete.cc:150:3
    #1 0x14ac26c in __libcpp_deallocate buildtools/third_party/libc++/trunk/include/new:279:10
    #2 0x14ac26c in deallocate buildtools/third_party/libc++/trunk/include/memory:1802
    #3 0x14ac26c in deallocate buildtools/third_party/libc++/trunk/include/memory:1556
    #4 0x14ac26c in erase buildtools/third_party/libc++/trunk/include/__tree:2370
    #5 0x14ac26c in erase buildtools/third_party/libc++/trunk/include/__tree:2379
    #6 0x14ac26c in erase buildtools/third_party/libc++/trunk/include/map:1200
    #7 0x14ac26c in webrtc::video_coding::RtpFrameReferenceFinder::ManageFrameVp9(webrtc::video_coding::RtpFrameObject*) modules/video_coding/rtp_frame_reference_finder.cc:497
    #8 0x14a7849 in ManageFrameInternal modules/video_coding/rtp_frame_reference_finder.cc:89:14
    #9 0x14a7849 in webrtc::video_coding::RtpFrameReferenceFinder::ManageFrame(std::__1::unique_ptr<webrtc::video_coding::RtpFrameObject, std::__1::default_delete<webrtc::video_coding::RtpFrameObject> >) modules/video_coding/rtp_frame_reference_finder.cc:43
    #10 0x148a87e in non-virtual thunk to webrtc::RtpVideoStreamReceiver::OnReceivedFrame(std::__1::unique_ptr<webrtc::video_coding::RtpFrameObject, std::__1::default_delete<webrtc::video_coding::RtpFrameObject> >) video/rtp_video_stream_receiver.cc:336:22
    #11 0x1496f41 in webrtc::video_coding::PacketBuffer::InsertPacket(webrtc::VCMPacket*) modules/video_coding/packet_buffer.cc:130:31
    #12 0x1487e59 in webrtc::RtpVideoStreamReceiver::OnReceivedPayloadData(unsigned char const*, unsigned long, webrtc::WebRtcRTPHeader const*) video/rtp_video_stream_receiver.cc:231:19
    #13 0x12d9144 in webrtc::RTPReceiverVideo::ParseRtpPacket(webrtc::WebRtcRTPHeader*, webrtc::PayloadUnion const&, unsigned char const*, unsigned long, long) modules/rtp_rtcp/source/rtp_receiver_video.cc:109:26
    #14 0x12cc80d in webrtc::RtpReceiverImpl::IncomingRtpPacket(webrtc::RTPHeader const&, unsigned char const*, unsigned long, webrtc::PayloadUnion) modules/rtp_rtcp/source/rtp_receiver_impl.cc:181:42
    #15 0x1488e52 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:399:20
    #16 0x1488b03 in webrtc::RtpVideoStreamReceiver::OnRecoveredPacket(unsigned char const*, unsigned long) video/rtp_video_stream_receiver.cc:245:3
    #17 0x14b925c in webrtc::UlpfecReceiverImpl::ProcessReceivedFec() modules/rtp_rtcp/source/ulpfec_receiver_impl.cc:244:35
    #18 0x148bd42 in webrtc::RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:421:23
    #19 0x1488d51 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:390:5
    #20 0x14899f8 in webrtc::RtpVideoStreamReceiver::OnRtpPacket(webrtc::RtpPacketReceived const&) video/rtp_video_stream_receiver.cc:290:3
    #21 0x90c486 in webrtc::RtpDemuxer::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_demuxer.cc:157:11
    #22 0x9131bd in webrtc::RtpStreamReceiverController::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_stream_receiver_controller.cc:55:19
    #23 0x129940d in webrtc::internal::Call::DeliverRtp(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1321:36
    #24 0x129a8d5 in webrtc::internal::Call::DeliverPacket(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1361:10
    #25 0x61fe06 in webrtc::RtpReplay() video/replay.cc:279:31
    #26 0x62337d in main video/replay.cc:343:3
    #27 0x7f5ae03d82b0 in __libc_start_main (/lib/x86_64-linux-gnu/libc.so.6+0x202b0)

previously allocated by thread T0 here:
    #0 0x61af72 in operator new(unsigned long) /b/build/slave/linux_upload_clang/build/src/third_party/llvm/compiler-rt/lib/asan/asan_new_delete.cc:93:3
    #1 0x14b664f in __libcpp_allocate buildtools/third_party/libc++/trunk/include/new:259:10
    #2 0x14b664f in allocate buildtools/third_party/libc++/trunk/include/memory:1799
    #3 0x14b664f in allocate buildtools/third_party/libc++/trunk/include/memory:1548
    #4 0x14b664f in __construct_node<const short &, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo> buildtools/third_party/libc++/trunk/include/__tree:2191
    #5 0x14b664f in std::__1::pair<std::__1::__tree_iterator<std::__1::__value_type<unsigned char, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo>, std::__1::__tree_node<std::__1::__value_type<unsigned char, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo>, void*>*, long>, bool> std::__1::__tree<std::__1::__value_type<unsigned char, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo>, std::__1::__map_value_compare<unsigned char, std::__1::__value_type<unsigned char, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo>, webrtc::DescendingSeqNumComp<unsigned char, (unsigned char)0>, true>, std::__1::allocator<std::__1::__value_type<unsigned char, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo> > >::__emplace_unique_impl<short const&, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo>(short const&, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo&&) buildtools/third_party/libc++/trunk/include/__tree:2203
    #6 0x14ab9ca in __emplace_unique<const short &, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo> buildtools/third_party/libc++/trunk/include/__tree:1193:16
    #7 0x14ab9ca in emplace<const short &, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo> buildtools/third_party/libc++/trunk/include/map:1041
    #8 0x14ab9ca in webrtc::video_coding::RtpFrameReferenceFinder::ManageFrameVp9(webrtc::video_coding::RtpFrameObject*) modules/video_coding/rtp_frame_reference_finder.cc:445
    #9 0x14a7849 in ManageFrameInternal modules/video_coding/rtp_frame_reference_finder.cc:89:14
    #10 0x14a7849 in webrtc::video_coding::RtpFrameReferenceFinder::ManageFrame(std::__1::unique_ptr<webrtc::video_coding::RtpFrameObject, std::__1::default_delete<webrtc::video_coding::RtpFrameObject> >) modules/video_coding/rtp_frame_reference_finder.cc:43
    #11 0x148a87e in non-virtual thunk to webrtc::RtpVideoStreamReceiver::OnReceivedFrame(std::__1::unique_ptr<webrtc::video_coding::RtpFrameObject, std::__1::default_delete<webrtc::video_coding::RtpFrameObject> >) video/rtp_video_stream_receiver.cc:336:22
    #12 0x1496f41 in webrtc::video_coding::PacketBuffer::InsertPacket(webrtc::VCMPacket*) modules/video_coding/packet_buffer.cc:130:31
    #13 0x1487e59 in webrtc::RtpVideoStreamReceiver::OnReceivedPayloadData(unsigned char const*, unsigned long, webrtc::WebRtcRTPHeader const*) video/rtp_video_stream_receiver.cc:231:19
    #14 0x12d9144 in webrtc::RTPReceiverVideo::ParseRtpPacket(webrtc::WebRtcRTPHeader*, webrtc::PayloadUnion const&, unsigned char const*, unsigned long, long) modules/rtp_rtcp/source/rtp_receiver_video.cc:109:26
    #15 0x12cc80d in webrtc::RtpReceiverImpl::IncomingRtpPacket(webrtc::RTPHeader const&, unsigned char const*, unsigned long, webrtc::PayloadUnion) modules/rtp_rtcp/source/rtp_receiver_impl.cc:181:42
    #16 0x1488e52 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:399:20
    #17 0x1488b03 in webrtc::RtpVideoStreamReceiver::OnRecoveredPacket(unsigned char const*, unsigned long) video/rtp_video_stream_receiver.cc:245:3
    #18 0x14b925c in webrtc::UlpfecReceiverImpl::ProcessReceivedFec() modules/rtp_rtcp/source/ulpfec_receiver_impl.cc:244:35
    #19 0x148bd42 in webrtc::RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:421:23
    #20 0x1488d51 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:390:5
    #21 0x14899f8 in webrtc::RtpVideoStreamReceiver::OnRtpPacket(webrtc::RtpPacketReceived const&) video/rtp_video_stream_receiver.cc:290:3
    #22 0x90c486 in webrtc::RtpDemuxer::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_demuxer.cc:157:11
    #23 0x9131bd in webrtc::RtpStreamReceiverController::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_stream_receiver_controller.cc:55:19
    #24 0x129940d in webrtc::internal::Call::DeliverRtp(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1321:36
    #25 0x129a8d5 in webrtc::internal::Call::DeliverPacket(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1361:10
    #26 0x61fe06 in webrtc::RtpReplay() video/replay.cc:279:31
    #27 0x62337d in main video/replay.cc:343:3
    #28 0x7f5ae03d82b0 in __libc_start_main (/lib/x86_64-linux-gnu/libc.so.6+0x202b0)

SUMMARY: AddressSanitizer: heap-use-after-free modules/video_coding/rtp_frame_reference_finder.cc:569:31 in webrtc::video_coding::RtpFrameReferenceFinder::FrameReceivedVp9(unsigned short, webrtc::video_coding::RtpFrameReferenceFinder::GofInfo*)
Shadow bytes around the buggy address:
  0x0c0c7fff85e0: 00 00 00 00 00 00 00 fa fa fa fa fa fd fd fd fd
  0x0c0c7fff85f0: fd fd fd fd fa fa fa fa 00 00 00 00 00 00 00 00
  0x0c0c7fff8600: fa fa fa fa 00 00 00 00 00 00 00 00 fa fa fa fa
  0x0c0c7fff8610: 00 00 00 00 00 00 00 00 fa fa fa fa 00 00 00 00
  0x0c0c7fff8620: 00 00 00 00 fa fa fa fa fd fd fd fd fd fd fd fa
=>0x0c0c7fff8630: fa fa fa fa fd fd fd fd fd fd[fd]fa fa fa fa fa
  0x0c0c7fff8640: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
  0x0c0c7fff8650: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
  0x0c0c7fff8660: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
  0x0c0c7fff8670: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
  0x0c0c7fff8680: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
Shadow byte legend (one shadow byte represents 8 application bytes):
  Addressable:           00
  Partially addressable: 01 02 03 04 05 06 07 
  Heap left redzone:       fa
  Freed heap region:       fd
  Stack left redzone:      f1
  Stack mid redzone:       f2
  Stack right redzone:     f3
  Stack after return:      f5
  Stack use after scope:   f8
  Global redzone:          f9
  Global init order:       f6
  Poisoned by user:        f7
  Container overflow:      fc
  Array cookie:            ac
  Intra object redzone:    bb
  ASan internal:           fe
  Left alloca redzone:     ca
  Right alloca redzone:    cb
  Shadow gap:              cc
==163231==ABORTING

To reproduce the issue:

1) apply new.patch to your webrtc directory
2) build video_replay
3) download the attached filed into the same directory
4) run ./video_replay --input_file uaf


Proof of Concept:
https://github.com/offensive-security/exploitdb-bin-sploits/raw/master/bin-sploits/45443.zip