dtls-in-stun: Only read IceConfig.dtls_handshake_in_stun in 1 place.

This patch fixes 2 problems, that both stem from the same root,
the the code had scattered checking IceConfig.dtls_handshake_in_stun
but that value was not updated when we discovered that remote peer
does not support piggy backing, "restart".

the patch modifies the code so that the IceConfig.dtls_handshake_in_stun is only checked during DtlsTransport::SetupDtls.

The problems fixed are:
1) P2PTransportChannel correctly (set/does not set) Connection::RegisterDtlsPiggyback based on the existing
dtls_stun_piggyback_callbacks_.empty() (that is reset when we detect
that peer does not support piggybacking) rather than on config which
is unchanged.

2) The timeout was not set properly during "restart",
properly == the value based upon ice rtt, but was still using the
"infinitely high" value for piggybacking.
This is tested with the DtlsTransportVersionTest which now runs
(optionally) with dtls-in-stun piggybacking.

BUG=webrtc:367395350

Change-Id: Ib511bcd1d3371a2132cefe26a3c49372208735ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/379760
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44044}
9 files changed
tree: 17dda6de8d440b09974e11137e7a4dc052f332a6
  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