| commit | 8c0f7a7a7041b2a50914297c4aa5953ccc51183d | [log] [tgz] |
|---|---|---|
| author | Steve Anton <steveanton@webrtc.org> | Tue Oct 03 10:03:10 2017 -0700 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Oct 03 23:26:28 2017 +0000 |
| tree | 2aabde09c5f99e7aa50cad972a954ebe2853a424 | |
| parent | 8a6d8e092864c71eb0191f597862acbacf5ca18a [diff] |
Add GetRemoteAudioSSLCertificate() to PeerConnection This method allows the client to get details about the SSL certificate sent by the remote side in the DTLS handshake. This functionality in this new method has been standardized in the RTCDtlsTransport, but until we have that implemented we wish to expose this functionality so clients do not need to depend on WebRtcSession. Bug: webrtc:8323 Change-Id: Ic964266dd7e734cec07289a147fd8d090d74ce6b Reviewed-on: https://webrtc-review.googlesource.com/5641 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20129}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.