Update how the DTLS role is queried and reported.

Update DataChannelController to query the DtlsTransport transport
directly for the SSL role instead of sending the request through
PeerConnection, JsepTransportController, JsepTransport and eventually
DtlsTransport. This reduces unnecessary coupling and threading checks
for several of the classes (most importantly PeerConnection and
JsepTransportController) and simplifies the PeerConnection interface by
removing the redundant method, associated mocks and fakes.

While making this change, also update SctpTransport to query its
internal DTLS transport directly for the role.

Lastly, DtlsTransport is adjusted to ensure role information is
correctly propagated during state changes, not only for the
`DtlsTransportState::kConnected` state so that Information() returns up
to date information.

The change also clarifies threading requirements in the
JsepTransportController.

Bug: none
Change-Id: If925416c589c52ceef1d20ffe58b72fa59f7d496
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/463540
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47420}
12 files changed
tree: feab7fe97a2b9a818515f561c93271bda5e43025
  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. sdk/
  25. stats/
  26. system_wrappers/
  27. test/
  28. tools_webrtc/
  29. video/
  30. .clang-format
  31. .clang-tidy
  32. .git-blame-ignore-revs
  33. .gitignore
  34. .gn
  35. .mailmap
  36. .rustfmt.toml
  37. .style.mdformat
  38. .style.yapf
  39. .vpython3
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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