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webrtc / src.git / ad5465b443cc5fda563dca4b40e52580e14cb621
commitad5465b443cc5fda563dca4b40e52580e14cb621[log] [tgz]
authorJeroen de Borst <jeroendb@webrtc.org>Mon Oct 01 22:02:12 2018
committerCommit Bot <commit-bot@chromium.org>Tue Oct 02 18:23:12 2018
tree9c8ebeec52c7bc35d4c176254e6fb1a4cc52f5ef
parent5fb245498ccd2c7c6be3db2492f0996d25b6cb20 [diff]
Deliver partial SCTP messages when the SID is (unexpectedly) changed.

We're receiving SCTP messages one chunk at a time. The sender is supposed to set the MSG_EOR bit on the last chunk of a message. A crash (RTC_CHECK) happened when the sender started a new message without indicating the end of the previous message. This is likely due to an SCTP implementation that doesn't (correctly) support the MSG_EOR bit.

This change, rather than calling RTC_CHECK, delivers partial messages when SID is (unexpectedly) changed, which is what happened before we paid attention to the MSG_EOR bit ourselves.

TBR=pthatcher@webrtc.org

Bug: chromium:884926
Change-Id: I18c773bb60ae724b735a83933eedd030f6360a3c
Reviewed-on: https://webrtc-review.googlesource.com/c/103023
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24935}
2 files changed
tree: 9c8ebeec52c7bc35d4c176254e6fb1a4cc52f5ef
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. abseil-in-webrtc.md
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. common_types.h
  37. DEPS
  38. ENG_REVIEW_OWNERS
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. WATCHLISTS
  52. webrtc.gni
  53. 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 http://www.webrtc.org/native-code/development 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