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webrtc / src / adb224b3e9e0df3dbd5c5a49a413cfe66e3ea7d4
commitadb224b3e9e0df3dbd5c5a49a413cfe66e3ea7d4[log] [tgz]
authorVictor Boivie <boivie@webrtc.org>Fri Aug 16 13:50:16 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Sun Aug 18 15:20:35 2024
treee54e248626d67a30e9fc31745bc485dbe0f29abe
parentde972c1126eafdd34c673006f5ba17bf67bd0031 [diff]
dcsctp: Simplify congestion control algorithm

There was a feature in the retransmission queue to avoid fragmenting
large messages when the congestion window was large. This was a feature
that intended to improve data channel performance under Chrome, where
communication with the network process (over MOJO) was lossy and losing
messages with small fragments could result in unnecessary
retransmissions. But back then, the implementation for fast retransmit
wasn't implemented correctly, so the benchmarking result don't
reproduce any longer.

So just improve the algorithm by removing this code. This aligns it with
the RFC and makes it easier to implement pluggable congestion control
algorithms (that wouldn't want this feature).

Bug: webrtc:42223116
Change-Id: Ifaaa82dac4b8fe2f55418158ae8b3da97199212f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359681
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42797}
5 files changed
tree: e54e248626d67a30e9fc31745bc485dbe0f29abe
  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. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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