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webrtc / src / 0deb9d6d33111cbf2a5b248434870dd9d8b982fc
commit0deb9d6d33111cbf2a5b248434870dd9d8b982fc[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Thu Aug 28 09:34:08 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu Aug 28 12:11:06 2025
treee44a34bf6f21a2d61ca9efaa73b406316c650f10
parentf4370b1ca4c12ba2d3aa611f8d67c7fc999eaca1 [diff]
Make WebRTC-RTP-Lifetime enabled-by-default.

Ships spec-compliant RTP stats lifetimes as per Intent to Ship:
https://groups.google.com/a/chromium.org/g/blink-dev/c/GYqPzIUUZCQ

The TL;DR change is:
1. outbound-rtp creation is delayed until O/A has completed, but can
   exist prior to sending any packets.
2. inbound-rtp creation is delayed until first packet has been received,
   whether or not O/A has completed (allowing early media use case).

The flag is kept as a kill-switch, to be removed after this has reached
Chrome stable.

Bug: chromium:406585888
Change-Id: Ibb42d77eb156ba14d2f50e6521d51615551fe489
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45468}
2 files changed
tree: e44a34bf6f21a2d61ca9efaa73b406316c650f10
  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. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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