| commit | 5f0b83b7fb6de95f33f20656e969e0280d72c315 | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Mar 18 15:02:07 2016 -0700 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Mar 18 22:02:13 2016 +0000 |
| tree | ee545b54eb50fdeba78e5136eb2b733a4b149ab6 | |
| parent | 1300caa3fea840572614772f96755ce7f4080ef6 [diff] |
Enabling rtcp-rsize negotiation and fixing some issues with it. Sending of reduced size RTCP packets should be enabled only if it's enabled in the send parameters (which corresponds to the remote description). Since the RTCPReceiver's RtcpMode isn't used at all, I removed it to ease confusion. BUG=webrtc:4868 R=pbos@webrtc.org, pthatcher@google.com, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1713493003 . Cr-Commit-Position: refs/heads/master@{#12057}
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.