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webrtc / src / / b0acde349cee0b05b34d5db96ad31880385d6746
commitb0acde349cee0b05b34d5db96ad31880385d6746[log] [tgz]
authorVictor Boivie <boivie@webrtc.org>Tue Dec 03 09:15:50 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Dec 04 14:10:32 2024
tree802c75411117c1e86076acac6afe948e6c2895f5
parent1d2f30b8b9bc5e8e942f26246bdac55ba7350683 [diff]
dcsctp: Add handover test for interleaved streams

This test was missing, which made me believe that it wasn't supported as
the handover state only included SSN and not MID. But when adding tests,
I saw that the current implementation used the SSN field to handover the
MID information for ordered streams which is sufficient given the 32 bit
type used for that (SSNs are only 16 bits).

For unordered streams, there is no need to handover any state there are
no expected next MID for unordered streams (they can be received in any
order).

So, adding tests and removing the handover state I just added.

Bug: webrtc:41481008
Change-Id: If1799cb1def5bd9f585a87cff6d835f4a9053b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370121
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43495}
2 files changed
tree: 802c75411117c1e86076acac6afe948e6c2895f5
  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