commit | ed09dc6f56231f55eb7bd96abdb9abbf941a217f | [log] [tgz] |
---|---|---|
author | Steve Anton <steveanton@webrtc.org> | Thu Mar 29 19:59:17 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 29 20:36:08 2018 |
tree | 731f6086201988160f9a864b0216df5b8b639b21 | |
parent | 003930a3ce40c8d0642403d5c91b7b8d2714e732 [diff] |
Don't check MIDs when demuxing RTP packets in Call The MID header extension is handled by the RtpTransport which lives above Call and takes care of demuxing to SSRC. Bug: webrtc:4050 Change-Id: I27135e296ae9c7b15e926ba17547c26c75684ad6 Reviewed-on: https://webrtc-review.googlesource.com/65025 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22682}
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.