c816ffc0a5
The original test did not properly test the bugs it fixed (SAN vs. CN) and violated BoringSSL invariants: - That SSL_get_peer_certificate works on the pending session before the handshake is a weird OpenSSL quirk that may later get fixed in BoringSSL. Calling code should not rely on this. - SSL_SESSION is a private struct and may not be accessed directly by callers. - Caller especially may not mutate private structs. The tests did not keep the SSL_SESSION's X509 and CRYPTO_BUFFER fields in sync. Instead, make an actual connected SSL object and better test the SAN vs. CN case. Bug: webrtc:8888 Change-Id: I773508c676e47be12e52a1bd6bd71562f474e09c Reviewed-on: https://webrtc-review.googlesource.com/73900 Commit-Queue: David Benjamin <davidben@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23129}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct