Only verify the certificate once.

WebRTC is currently using the SSL_CTX_set_verify callback. This
configures a callback for use with X509_STORE_CTX_set_verify_cb. See
https://www.openssl.org/docs/man1.0.2/crypto/X509_STORE_CTX_set_verify_cb.html

This callback does not override certificate verification. Rather, it
allows EACH failure in OpenSSL's built-in certificate verification, as
well as the final success, to be overridden (that's why there's an ok
parameter). It still runs the usual OpenSSL certificate verification
(which will never succeed).

The upshot is that the callback is called multiple times and
OpenSSLStreamAdapter does a ton of redundant work and checks the hash at
least twice, or more for certificates with other errors.

Instead, use SSL_CTX_set_cert_verify_callback. This short-circuits the
OpenSSL behavior entirely and uses a caller-supplied one.
https://commondatastorage.googleapis.com/chromium-boringssl-docs/ssl.h.html#SSL_CTX_set_cert_verify_callback
https://wiki.openssl.org/index.php/Manual:SSL_CTX_set_cert_verify_callback(3)

(This also removes the SSL_CTX_set_verify_depth call which is ignored
with SSL_CTX_set_cert_verify_callback. It didn't do anything before
either---it tells OpenSSL to reject chains that are too short, but the
rejection was overwritten by the callback anyway.)

(Later on, we'll need to switch this to the BoringSSL-only
SSL_CTX_set_custom_verify and CRYPTO_BUFFER APIs to fix WebRTC's
contribution to Chrome's binary size, but I've left that alone for the
time being.)

Bug: none
Change-Id: I9320a367d0961935836df63dc6f0868b069f0af0
Reviewed-on: https://webrtc-review.googlesource.com/4581
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20053}
2 files changed
tree: b11b0701878f37d66478f19750c9bababf62a970
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  53. 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.

More info