snap: fix race condition

With the SNAP out-of-band SCTP connect (WebRTC-Sctp-Snap), the dcSCTP
socket is created in DcSctpTransport::Start() but only connected later
in MaybeConnectSocket(), once the DTLS transport becomes writable. The
DTLS transport can decrypt and deliver an inbound packet as soon as its
handshake completes, which precedes writability (set_writable() is
deferred until ICE is writable). A packet arriving in that window is
delivered to the still-kClosed socket, which rejects it as having an
invalid verification tag.

End-to-end this is masked by SCTP retransmission, so it only surfaces as
an 'invalid verification tag' log line and a one-RTO delay opening the
data channel. The dcsctp_transport unit test observes the drop directly
via the mock socket.

Bug: webrtc:526609852
Change-Id: Iff17f2deb766ae465d68728d613e8c17d046263e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/484720
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Denise Tell <dct@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#48061}
3 files changed
tree: c1a6f6c337c73e7934978e83514e9db5e95312d1
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  40. .vpython3
  41. .yapfignore
  42. AUTHORS
  43. BUILD.gn
  44. CODE_OF_CONDUCT.md
  45. codereview.settings
  46. DEPS
  47. DIR_METADATA
  48. ENG_REVIEW_OWNERS
  49. GEMINI.md
  50. LICENSE
  51. license_template.txt
  52. native-api.md
  53. OWNERS
  54. OWNERS_INFRA
  55. PATENTS
  56. PRESUBMIT.py
  57. presubmit_test.py
  58. presubmit_test_mocks.py
  59. pylintrc
  60. pylintrc_old_style
  61. README.chromium
  62. README.md
  63. unsafe_buffers_paths.txt
  64. WATCHLISTS
  65. webrtc.gni
  66. webrtc_lib_link_test.cc
  67. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info