commit | 6e1055bdc05cb1f86b7c19c9091865bef873c1ae | [log] [tgz] |
---|---|---|
author | Qingsi Wang <qingsi@webrtc.org> | Tue Aug 20 22:47:12 2019 |
committer | Qingsi Wang <qingsi@webrtc.org> | Tue Aug 20 22:50:37 2019 |
tree | b2609c1bd8251dc310599649859fcff0ba9554a8 | |
parent | c7065123bed355bfad6826267614a5c4001fb0b1 [diff] |
Merge to M77: Reland "Set the usage pattern bits for adding remote ICE candidates from SDP." This is a reland of 7c6f74ab0344e9c6201de711d54026e9990b8e6c Compared to the previous commit, new bits are added to log calls of AddIceCandidate, and the gathering and reception of IPv6 candidates. Original change's description: > Set the usage pattern bits for adding remote ICE candidates from SDP. > > Currently these bits are only set when a remote ICE candidate is > successfully added via addIceCandidate. For non-trickled sessions in > which the remote candidates are added via the remote description, these > bits are lost. This also happens for trickled sessions, though a rare > case, when addIceCandidate does not succeed because the peer connection > is not ready to add any remote candidate. > > Bug: webrtc:10868 > Change-Id: Ib2f199f9ffc936060473934d25ba397ef31131a3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148880 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28844} TBR=hta@webrtc.org (cherry picked from commit 1ba5dec7694562a095cd08a0008ab3d397345db1) Bug: webrtc:10868 Change-Id: Ifac0593dcfb64d88619fd24b4ab61c14a0810beb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149024 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#28904} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150026 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/branch-heads/m77@{#9} Cr-Branched-From: 2bac7da1349c75e5cf89612ab9619a1920d5d974-refs/heads/master@{#28685}
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.
Authoritative list of directories that contain the native API header files.