Minor Dtls-in-STUN-encapsulation cleanups

That unfortunately does not solve the existing bug :(

1) Add a new method GetPending that is used when retransmitting
unencapsulated, this only matters for PQC and means that we might
send 2 packets "back-to-back", which is how it's ordinarely done.

2) Add a new UpdateHandshakeTimeout, that is used after the initial
estimate and pipe that all the way to
rtc_base/openssl_stream_adapter.cc.

3) Move the RunBenchmark into the test fixture, that way it can easily
be used from other tests.

4) Modify the "with packetloss" test to run with 25% packetloss, from 50%. This might seem like a bummer, but at 50% the tests very super fragile, changing *anything* made them randomly fail. I.e. we had overfitted the settings *a lot*. At 25% they are stable.

5) Add a new test that verifies that using dtls-in-stun leads to short
connect time that not using it.

BUG=webrtc:367395350

Change-Id: I9cc207eab413dc0e7542464174e504a39f425466
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/435182
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46524}
12 files changed
tree: e85af64f902090d2c2c05874b0e20564705504fd
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. WATCHLISTS
  60. webrtc.gni
  61. webrtc_lib_link_test.cc
  62. 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