a9bbd8684936dedbd4ca8f6402644d7a2e152cbe

Adds a field |use_media_transport_for_data_channels| to RTCConfiguration. PeerConnection requires a media transport factory to be set if this bit is set. As with |use_media_transport|, the value may not be modified after setting the local or remote description. If either |use_media_transport| or |use_media_transport_for_data_channel| is set, PeerConnection uses its media transport factory when creating a JSEP transport controller. PeerConnection stops unconditionally using media transport in CreateVoiceChannel, as it may be present only for use in data channels. It uses the media transport if it is present and |use_media_transport| is set. Bug: webrtc:9719 Change-Id: I59d4ce8f7531fd19d9c17eefe033f063f663ebcc Reviewed-on: https://webrtc-review.googlesource.com/c/109041 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25507}
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
Description
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