| commit | b7f425ab68ec58e2a5beaaf5ef79f50f1982c6f9 | [log] [tgz] |
|---|---|---|
| author | zhihuang <zhihuang@webrtc.org> | Wed Apr 13 01:32:30 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Apr 13 01:32:36 2016 |
| tree | d39cb7d6fe679891cc85e689b74470917a9aab2f | |
| parent | 79299afa3022560691068678bea711af3781eeb6 [diff] |
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 Review URL: https://codereview.webrtc.org/1880913002 Cr-Commit-Position: refs/heads/master@{#12338}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.