Integrate datagram feedback loop

This change removes RTCP Feedback loop if we are using datagram transport by removing transport sequence numbers from RTP packets and recreating RTCP Feedback from Datagram ACKs and Timestamps.

- For outgoing RTP packets, remove transport sequence number and store it with datagram_id. Note that removing transport sequence numbers does not only save 4-8 bytes per packet, but also prevents generation of feedback packets on the receiver side.

- When datagram ACKs, we re-created RTCP feedback with timestamp.

- Replacing previous assumption that datagram_id was the same as packet_id by storing incremental counter of datagram ids (I noticed some packets come without packet_id, which is a bit strange, but easy to support and it's also good not to rely on packet_ids being unique across multiple ssrcs).

Bug: webrtc:9719
Change-Id: Iecfe938ecea1a74e7c9e1484f0e985d72643d4a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145269
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28542}
6 files changed
tree: d821c957034fad11fd72a39d12d58910d462f7a4
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  6. AUTHORS
  7. BUILD.gn
  8. CODE_OF_CONDUCT.md
  9. DEPS
  10. ENG_REVIEW_OWNERS
  11. LICENSE
  12. OWNERS
  13. PATENTS
  14. PRESUBMIT.py
  15. README.chromium
  16. README.md
  17. WATCHLISTS
  18. abseil-in-webrtc.md
  19. api/
  20. audio/
  21. build_overrides/
  22. call/
  23. codereview.settings
  24. common_audio/
  25. common_types.h
  26. common_video/
  27. crypto/
  28. data/
  29. examples/
  30. license_template.txt
  31. logging/
  32. media/
  33. modules/
  34. native-api.md
  35. p2p/
  36. pc/
  37. presubmit_test.py
  38. presubmit_test_mocks.py
  39. pylintrc
  40. resources/
  41. rtc_base/
  42. rtc_tools/
  43. sdk/
  44. stats/
  45. style-guide.md
  46. style-guide/
  47. system_wrappers/
  48. test/
  49. tools_webrtc/
  50. video/
  51. webrtc.gni
  52. 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