dcsctp: Reset synchronously with incoming request

When a sender has requested a stream to be reset, and the last sender
assigned TSN hasn't been received yet, the receiver will enter deferred
reset mode, where it will store any data chunks received after that
given TSN, and replay those later, when the stream has been reset.

Before this CL, leaving deferred mode was done as soon as the sender's
last assigned TSN was received. That's actually not how the RFC
describes the process[1], but was done that way to properly handle some
sequences of RE-CONFIG and FORWARD-TSN. But after having read the RFCs
again, and realizing that whenever RFC6525 mention "any data arriving",
this also applies to any FORWARD-TSN[2] - it's better to reset streams
synchronously with the incoming requests, and defer not just DATA past
the sender last assigned TSN, but also any FORWARD-TSN after that TSN.

This mostly simplifies the code and is mostly a refactoring, but most
importantly aligns it with how the resetting procedure is explained in
the RFC. It also fixes two bugs:

 * It defers FORWARD-TSN *as well as* DATA chunks with a TSN later
   than the sender's last assigned TSN - see test case. The old
   implementation tried to handle that by exiting the deferred reset
   processing as soon as it reached the sender's last assigned TSN, but
   it didn't manage to do that in all cases.
 * It only defers DATA chunks for streams that are to be reset, not
   all DATA chunks with a TSN > sender's last assigned TSN. This was
   missed in the old implementation, but as it's now implemented
   strictly according to the RFC, this was now done.

[1] https://datatracker.ietf.org/doc/html/rfc6525#section-5.2.2
[2] RFC6525 cover stream resetting, and RFC3758 cover FORWARD-TSN, and
    the combination of these is not covered in the RFCs.

Bug: webrtc:14600
Change-Id: Ief878b755291b9c923aa6fb4317b0f5c00231df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322623
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40889}
10 files changed
tree: 851b19ec285d422aeb6b80c0d2a9d8706895984e
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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