Enable SSL logging per default

Done in order to simplify connection debuging.

Example log:

openssl_adapter.cc:829): connect_loop TLS client read_server_hello
(openssl_adapter.cc:829): connect_loop TLS client read_server_certificate
(openssl_adapter.cc:829): connect_loop TLS client read_certificate_status
(openssl_adapter.cc:829): connect_loop TLS client verify_server_certificate
(openssl_stream_adapter.cc:1128): Accepted peer certificate.
(openssl_adapter.cc:829): connect_loop TLS client read_server_key_exchange
(openssl_adapter.cc:829): connect_loop TLS client read_certificate_request
(openssl_adapter.cc:829): connect_loop TLS client read_server_hello_done
(openssl_adapter.cc:829): connect_loop TLS client send_client_certificate
(openssl_adapter.cc:829): connect_loop TLS client send_client_key_exchange
(openssl_adapter.cc:829): connect_loop TLS client send_client_certificate_verify
(openssl_adapter.cc:829): connect_loop TLS client send_client_finished
(openssl_adapter.cc:829): connect_loop TLS client finish_flight
(openssl_adapter.cc:829): connect_loop TLS client read_session_ticket
(openssl_adapter.cc:829): connect_exit TLS client read_session_ticket
(openssl_adapter.cc:829): accept_loop TLS server verify_client_certificate
(openssl_stream_adapter.cc:1128): Accepted peer certificate.
(openssl_adapter.cc:829): accept_loop TLS server read_client_key_exchange
(peer_connection.cc:1952): Changing IceConnectionState 0 => 1
(openssl_adapter.cc:829): accept_loop TLS server read_client_certificate_verify
(peer_connection.cc:1971): Changing standardized IceConnectionState 0 => 1
(peer_connection.cc:1971): Changing standardized IceConnectionState 0 => 1
(peer_connection.cc:1971): Changing standardized IceConnectionState 1 => 2
(peer_connection.cc:1971): Changing standardized IceConnectionState 1 => 2
(openssl_adapter.cc:829): accept_loop TLS server read_change_cipher_spec
(openssl_adapter.cc:829): accept_loop TLS server process_change_cipher_spec
(openssl_adapter.cc:829): accept_loop TLS server read_next_proto
(openssl_adapter.cc:829): accept_loop TLS server read_channel_id
(openssl_adapter.cc:829): accept_loop TLS server read_client_finished
(openssl_adapter.cc:829): accept_loop TLS server send_server_finished
(openssl_adapter.cc:829): accept_loop TLS server finish_server_handshake
(openssl_adapter.cc:829): accept_loop TLS server done
(openssl_adapter.cc:829): handshake_done TLS server done
(openssl_adapter.cc:829): accept_exit TLS server done
(dtls_transport.cc:688): DtlsTransport[0|1|__]: DTLS handshake complete.

Bug: b/275671043
Change-Id: Ib8d394aa74c5665c489b485bb44152aff67d3b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302300
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39955}
3 files changed
tree: 92465ea784850cce42911e46eea9719280be6d6c
  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. .vpython
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  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