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a743303211
Previously, RTP header extensions with encryption had been filtered if the encryption had been activated (not the other way around) which was likely an unintended logic inversion. In addition, it ensures that encrypted RTP header extensions are only negotiated if RTP header extension encryption is turned on. Formerly, which extensions had been negotiated depended on the order in which they were inserted, regardless of whether or not header encryption was actually enabled, leading to no extensions being sent on the wire. Further changes: - If RTP header encryption enabled, prefer encrypted extensions over non-encrypted extensions - Add most extensions to list of extensions supported for encryption - Discard encrypted extensions in a session description in case encryption is not supported for that extension Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte header extensions will prevent any RTP packets being sent/received. Bug: webrtc:11713 Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33723} |
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| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| data | ||
| docs | ||
| examples | ||
| g3doc | ||
| logging | ||
| media | ||
| modules | ||
| net/dcsctp | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| style-guide | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .vpython | ||
| abseil-in-webrtc.md | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| DEPS | ||
| DIR_METADATA | ||
| ENG_REVIEW_OWNERS | ||
| g3doc.lua | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| PATENTS | ||
| PRESUBMIT.py | ||
| presubmit_test.py | ||
| presubmit_test_mocks.py | ||
| pylintrc | ||
| README.chromium | ||
| README.md | ||
| style-guide.md | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| webrtc_lib_link_test.cc | ||
| whitespace.txt | ||
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs