b43e3bbd87
In non-BUNDLE use cases, it is possible for multiple RTP streams to have the same SSRC (as long as the SSRC is unique within the same transport). This CL adds support for "outbound-rtp" and "inbound-rtp" stream stats to have the same SSRC on different transports by adding the transport to the stats ID. This avoids multiple RTP stream stats having the same stats ID and fixes the problem. It's a stupid use case, but it should work. There could still be a stats ID collision in the event of multiple "remote-inbound-rtp" or "remote-outbound-rtp" reference the same SSRC but on separate transports for the same reason, and would require the same fix... but one bug at a time. Not addressed in this CL. Bug: webrtc:14443 Change-Id: I1a2ffd79fc67c2765e6dbd1ccc6828d4e91c4589 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38201}
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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation