dcsctp: Move last_assembled_tsn_watermark further

The ReassemblyQueue will need to track which messages that have already
been delivered to the client so that they are not re-delivered on e.g.
retransmissions. It does that by tracking which TSNs that those messages
were built from. It tracks that in two variables,
`last_assembled_tsn_watermark` and `delivered_tsns_`, where the first
one represent that all TSNs including and prior this one have been
delivered and `delivered_tsns` contain additional ones when there are
gaps.

When receiving a FORWARD-TSN and asked to forget about some partially
received messages, these two variables were updated correctly, but the
`delivered_tsns_` were left in a state where it could be adjacent to the
`last_assembled_tsn_watermark` - when `last_assembled_tsn_watermark`
could actually have been moved further.

Added consistency check (that would trigger in existing tests) and
fixing the issue.

This bug is quite benign, as any received chunk would've corrected the
problem, and even at this faulty state, the ReassemblyQueue would
function completely fine.

Bug: webrtc:13154
Change-Id: Iaa7c612999c9dc609fc6e2fb3be2d0bd04534c90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232124
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35013}
2 files changed
tree: a0fed0a5dca9438abf6cf4c6104924daa235233f
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. .vpython3
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. DEPS
  37. DIR_METADATA
  38. ENG_REVIEW_OWNERS
  39. g3doc.lua
  40. LICENSE
  41. license_template.txt
  42. native-api.md
  43. OWNERS
  44. PATENTS
  45. PRESUBMIT.py
  46. presubmit_test.py
  47. presubmit_test_mocks.py
  48. pylintrc
  49. README.chromium
  50. README.md
  51. WATCHLISTS
  52. webrtc.gni
  53. webrtc_lib_link_test.cc
  54. 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