Revert "srtp: add UseCryptex API to SrtpSession and SrtpTransport"

This reverts commit b9386f3f09ba0f38320f6a9e65dfe2992f433cb5.

Reason for revert: Downstream libSRTP is not updated yet

Original change's description:
> srtp: add UseCryptex API to SrtpSession and SrtpTransport
>
> Adds the UseCryptex(enable, require, sending_session) API on SrtpSession
> and UseCryptex(enable, require) on SrtpTransport. Wires libsrtp's
> srtp_set_stream_use_cryptex through DoSetKey, and adds a require-cryptex
> post-decrypt check via ParseRtpExtensionProfile (new helper in
> modules/rtp_rtcp/source/rtp_util) for the cases libsrtp does not yet
> enforce; see https://github.com/cisco/libsrtp/pull/805.
>
> No callers yet; the cryptex SDP negotiation wires this up in a follow-up.
>
> Bug: webrtc:455813732
> Change-Id: I407400d42912fb303c93c715e7110dafcc3d0cab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/470600
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#47707}

Bug: webrtc:455813732
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I496ebf60d95e1bbd3d117f5b30b4e4fe69e9a2ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/472440
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47708}
7 files changed
tree: 048fb21e68345862f648474584bc5d7130b563fc
  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. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  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