6cacef2402
If the SSRC of an RTP module is changed at runtime, we may get conflicts with packets already there. Eg: * Put seq# 123 in the history for SSRC 1. * Change the SSRC to 2. * Send a NACK for seq# 123 from SSRC 2. Currently, we will respond with the packet belonging to SSRC 1 (and not if the NACK specifies SSRC 1, to boot). We can gen similar issues if the sequence number is changed, where half frame are left in the buffer. In these cases, the stream is likely being reset so we should just clear the packet history too. Bug: webrtc:10794 Change-Id: I28147c2532cf1c78840d4808c4366d4a647541f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145729 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28658}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- 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
- Coding style guide
- Code of conduct