WebRTC – FEC Processing Overflow

  • 作者: Google Security Research
    日期: 2018-08-01
  • 类别:
    平台:
  • 来源:https://www.exploit-db.com/exploits/45122/
  • There are several calls to memcpy that can overflow the destination buffer in webrtc::UlpfecReceiverImpl::AddReceivedRedPacket. The method takes a parameter incoming_rtp_packet, which is an RTP packet with a mac length that is defined by the transport (2048 bytes for DTLS in Chrome). This packet is then copied to the received_packet in several locations in the method, depending on packet properties, using the lenth of the incoming_rtp_packet as the copy length. The received_packet is a ForwardErrorCorrection::ReceivedPacket, which has a max size of 1500. Therefore, the memcpy calls in this method can overflow this buffer.
    
    ==204614==ERROR: AddressSanitizer: heap-buffer-overflow on address 0x61b000046670 at pc 0x00000059d958 bp 0x7ffcac5716f0 sp 0x7ffcac570ea0
    WRITE of size 2316 at 0x61b000046670 thread T0
    #0 0x59d957 in __asan_memcpy /b/build/slave/linux_upload_clang/build/src/third_party/llvm/compiler-rt/lib/asan/asan_interceptors_memintrinsics.cc:23:3
    #1 0x1b6aacc in webrtc::UlpfecReceiverImpl::AddReceivedRedPacket(webrtc::RTPHeader const&, unsigned char const*, unsigned long, unsigned char) modules/rtp_rtcp/source/ulpfec_receiver_impl.cc:173:5
    #2 0x1b3cd5c in webrtc::RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:426:27
    #3 0x1b39a31 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:402:5
    #4 0x1b3a895 in webrtc::RtpVideoStreamReceiver::OnRtpPacket(webrtc::RtpPacketReceived const&) video/rtp_video_stream_receiver.cc:301:3
    #5 0x8c7a26 in webrtc::RtpDemuxer::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_demuxer.cc:157:11
    #6 0x8cec3d in webrtc::RtpStreamReceiverController::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_stream_receiver_controller.cc:55:19
    #7 0x12e8507 in webrtc::internal::Call::DeliverRtp(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1291:36
    #8 0x12e92a0 in webrtc::internal::Call::DeliverPacket(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1316:10
    #9 0x5da2a6 in webrtc::RtpReplay() video/replay.cc:635:31
    #10 0x5dd5fe in main video/replay.cc:700:3
    #11 0x7feaa1ee92b0 in __libc_start_main (/lib/x86_64-linux-gnu/libc.so.6+0x202b0)
    
    0x61b000046670 is located 0 bytes to the right of 1520-byte region [0x61b000046080,0x61b000046670)
    allocated by thread T0 here:
    #0 0x5c9362 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 0x1b6a8c8 in webrtc::UlpfecReceiverImpl::AddReceivedRedPacket(webrtc::RTPHeader const&, unsigned char const*, unsigned long, unsigned char) modules/rtp_rtcp/source/ulpfec_receiver_impl.cc:165:35
    #2 0x1b3cd5c in webrtc::RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:426:27
    #3 0x1b39a31 in webrtc::RtpVideoStreamReceiver::ReceivePacket(unsigned char const*, unsigned long, webrtc::RTPHeader const&) video/rtp_video_stream_receiver.cc:402:5
    #4 0x1b3a895 in webrtc::RtpVideoStreamReceiver::OnRtpPacket(webrtc::RtpPacketReceived const&) video/rtp_video_stream_receiver.cc:301:3
    #5 0x8c7a26 in webrtc::RtpDemuxer::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_demuxer.cc:157:11
    #6 0x8cec3d in webrtc::RtpStreamReceiverController::OnRtpPacket(webrtc::RtpPacketReceived const&) call/rtp_stream_receiver_controller.cc:55:19
    #7 0x12e8507 in webrtc::internal::Call::DeliverRtp(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1291:36
    #8 0x12e92a0 in webrtc::internal::Call::DeliverPacket(webrtc::MediaType, rtc::CopyOnWriteBuffer, webrtc::PacketTime const&) call/call.cc:1316:10
    #9 0x5da2a6 in webrtc::RtpReplay() video/replay.cc:635:31
    #10 0x5dd5fe in main video/replay.cc:700:3
    #11 0x7feaa1ee92b0 in __libc_start_main (/lib/x86_64-linux-gnu/libc.so.6+0x202b0)
    
    SUMMARY: AddressSanitizer: heap-buffer-overflow /b/build/slave/linux_upload_clang/build/src/third_party/llvm/compiler-rt/lib/asan/asan_interceptors_memintrinsics.cc:23:3 in __asan_memcpy
    Shadow bytes around the buggy address:
    0x0c3680000c70: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
    0x0c3680000c80: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
    0x0c3680000c90: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
    0x0c3680000ca0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
    0x0c3680000cb0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
    =>0x0c3680000cc0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00[fa]fa
    0x0c3680000cd0: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
    0x0c3680000ce0: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
    0x0c3680000cf0: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
    0x0c3680000d00: fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa fa
    0x0c3680000d10: 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
    
    
    To reproduce this issue:
    
    1) replace video/replay.cc with the attached version, and build it with asan (ninja -C out/asan video_replay). Note that this file adds the ability to load a full receiver config to the video replay tool, I'm hoping to eventually get this change committed to WebRTC.
    
    2) Download the attached files config4.txt and fallbackoob1
    
    3) run video_replay --input_filefallbackoob1--config_file config4.txt
    
    Proof of Concept:
    https://gitlab.com/exploit-database/exploitdb-bin-sploits/-/raw/main/bin-sploits/45122.zip