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webrtc/src/refs/heads/main
commit
6c3c1fe605015581560866db99056e46f210bcaa
[log]
author
Philipp Hancke <philipp.hancke@googlemail.com>
Fri Sep 11 05:48:22 2026
committer
webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>
Wed Sep 16 13:46:43 2026
tree
613ea218168e3f312211c998f55b0356b6a5f469
parent
7314d7980362fd1f22635d309636adc8c45d8714 [diff]
cryptex consistency fixes

* ValidateCryptex: identify RTP m= sections by protocol instead of media
  type so an unsupported m= section can not become the reference.
* JsepTransportController: take cryptex from the first RTP-based m=
  section of the bundle group. The BUNDLE-tagged one may be the data
  channel which does not have a cryptex attribute.
* CreateMediaContentAnswer: declare cryptex at media level when the
  answer is not declaring it at session level.

Bug: webrtc:455813732
Change-Id: Ifd8fc72560cbcb4567c199e35d11114522a3b9a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/502140
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48613}
4 files changed
tree: 613ea218168e3f312211c998f55b0356b6a5f469
  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.mdformat
  39. .style.yapf
  40. .vpython3
  41. .yapfignore
  42. AUTHORS
  43. BUILD.gn
  44. CODE_OF_CONDUCT.md
  45. codereview.settings
  46. DEPS
  47. DIR_METADATA
  48. ENG_REVIEW_OWNERS
  49. GEMINI.md
  50. LICENSE
  51. license_template.txt
  52. native-api.md
  53. OWNERS
  54. OWNERS_INFRA
  55. PATENTS
  56. PRESUBMIT.py
  57. presubmit_test.py
  58. presubmit_test_mocks.py
  59. pylintrc
  60. pylintrc_old_style
  61. README.chromium
  62. README.md
  63. unsafe_buffers_paths.txt
  64. WATCHLISTS
  65. webrtc.gni
  66. webrtc_lib_link_test.cc
  67. 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