dtls-in-stun: Prepare for multi packet handshakes

1) Change format of ACK attribute to be hash of packet (crc32).
this is good since we anyway can't retransmit parts of DTLS packet
and it also works for the encrypted handshake packages.

This change is "somewhat upgrade" compatible,
version prior to this does not actually check/use the content of
the ack attribute, only the presence.

This version will automatically clear pending packets
when a new flight begins => It works, but retransmit of 1
packet in a multi placket flight does not unless both
ens has this version....But multi packet handshake flight
are not yet "in play"

2) Keep track of individually acked packets.
Clear array of pending when getting an ACK. This is "already today"
and improvment of existing code.

3) Limit size of ACK to 4 packets,
this limit *should* never be reached...but there is a test for it :)

4) Readd restrictions removed in https://webrtc-review.googlesource.com/c/src/+/381102. It only worked "by accident",
and the use case is of zero value. Even hard to explain!
Maybe I'll take a stab after PQC is done.

---

With this, PQC handshake _almost_ work with dtls-in-stun.

BUG=webrtc:367395350

Change-Id: I582397babb099896ba56ef26e174f47f7e98d1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385401
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44340}
7 files changed
tree: 7edfe1a3d343cf6c437c6963028e6eab8b3f28f9
  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. .rustfmt.toml
  34. .style.yapf
  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. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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