dcsctp: Add Retransmission Queue

The Retransmission Queue contain all message fragments (DATA chunks)
that have once been sent, but not yet ACKed by the receiver. It will
process incoming SACK chunks, which informs it which chunks that the
receiver has seen (ACKed) and which that are lost (NACKed), and will
retransmit chunks when it's time.

If a message has been sent with partial reliability, e.g. to have a
limited number of retransmissions or a limited lifetime, the
Retransmission Queue may discard a partially sent and expired message
and will instruct the receiver that "don't expect this message - it's
expired" by sending a FORWARD-TSN chunk.

This currently also includes the congestion control algorithm as it's
tightly coupled with the state of the retransmission queue. This is
a fairly complicated piece of logic which decides how much data that
can be in-flight, depending on the available bandwidth. This is not done
by any bandwidth estimation, but similar to TCP, where data is sent
until it's lost, and then "we dial down a knob" and take it more
carefully from here on.

Future refactoring will try to separate the logic regarding fragment
retransmission and the congestion control algorithm.

Bug: webrtc:12614
Change-Id: I8678250abb766e567c3450634686919936ea077b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214046
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33833}
5 files changed
tree: c2da015295290e17c8b4addf6304112899dc1eae
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. style-guide/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. abseil-in-webrtc.md
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. style-guide.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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