| commit | 586b19bdb615dde34cdcf107272d8857fe2f5631 | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Fri Sep 18 09:14:31 2015 |
| committer | Stefan Holmer <stefan@webrtc.org> | Fri Sep 18 09:14:42 2015 |
| tree | f75f8c7ab59cda4dadc0039a63fd90d6be43ee2d | |
| parent | 71df77bba01c86bc3bd359a61573698ea064f35f [diff] |
Enable probing with repeated payload packets by default. To make this possible padding only packets will have the same timestamp as the previously sent media packet, as long as RTX is not enabled. This has the side effect that if we send only padding for a long time without sending media, a receive-side jitter buffer could potentially overflow. In practice this shouldn't be an issue, partly because RTX is recommended and used by default, but also because padding typically is terminated before being received by a client. It is also not an issue for bandwidth estimation as long as abs-send-time is used instead of toffset. BUG=chromium:425925 R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1327933003 . Cr-Commit-Position: refs/heads/master@{#9984}
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.