DtlsInStun: Modify termination by application data

This is a follow up to https://webrtc-review.git.corp.google.com/c/src/+/45152
which I think(?) address the comment in
https://webrtc-review.git.corp.google.com/c/src/+/451521/comment/0e8a1639_699fdc3a

I.e. encrypted srtp is only pass though by the dtls layer, meaning we
need to check if we are writable before concluding that both are
writable.

Update method name to ApplicationPacketReceived to highlight
that it doesn't have to be decrypted.

Bug: webrtc:367395350
Change-Id: I37f2d0c2fdeb6cb33c1a042b30d2afdfe1524032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452280
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#47000}
4 files changed
tree: da4634e809a2b949a78df14eb276e5aa7929dba7
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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