d713e86058
Reason for revert: Broke the Chromium build by introducing static initializers. Original issue's description: > Accept all the media profiles required by JSEP. > > JSEP section 5.1.3 states that: > Any profile matching the following patterns MUST be accepted: > "RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]" > > NOTRY=True > BUG=webrtc:5638 > > Committed: https://crrev.com/b7f425ab68ec58e2a5beaaf5ef79f50f1982c6f9 > Cr-Commit-Position: refs/heads/master@{#12338} TBR=deadbeef@webrtc.org,pthatcher@webrtc.org,avi@chromium.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5638 Review URL: https://codereview.webrtc.org/1882923002 Cr-Commit-Position: refs/heads/master@{#12351}
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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