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webrtc / src / 4221e1bc2fc360448c1663f0bbae058ff9f65a48
commit4221e1bc2fc360448c1663f0bbae058ff9f65a48[log] [tgz]
authorJonas Oreland <jonaso@webrtc.org>Tue Aug 26 13:57:29 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Tue Aug 26 15:42:20 2025
tree2cb1a17a0c64eea86ccd7d9f3b93bbc8434a78f2
parentf9af67ca783cb88b38fa75c4758f2054c4159c1a [diff]
DTLS1.3: Fix missing retransmission due to failure to set timer.

DTLS 1.3 considers itself connected earlier than DTLS 1.2 did - when
second flight reaches the client. This CL fixes a bug that when
client is connected (state_ != SSL_CONNECTING), it would not
continue retransmitting. Continuous retransmission is needed
when the third flight is lost multiple times. Or really anytime that DTLS request it :)

This fixes the TODO in dtls_ice_integrationtest.cc in which dtls1.3
spuriously failed with certain (packet loss intensive) configurations.

CREDITS: sergeysu@ that found and fixed the problem!

Bug: chromium:441245658
Change-Id: I3302f6f384d7e4cda090184094a6fadaf7e4f129
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406320
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45439}
3 files changed
tree: 2cb1a17a0c64eea86ccd7d9f3b93bbc8434a78f2
  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