71677452f9
|transport overhead| field is added to rtc::NetworkRoute structure. In PackTransportInternal: 1. network_route() is added which returns the current network route. 2. debug_name() is removed. 3. transport_name() is moved from DtlsTransportInternal and IceTransportInternal to PacketTransportInternal. When the selected candidate pair is changed, the P2PTransportChannel will fire the SignalNetworkRouteChanged instead of SignalSelectedCandidatePairChanged to upper layers. The Rtp/SrtpTransport takes the responsibility of calculating the transport overhead from the BaseChannel so that the BaseChannel doesn't need to depend on P2P layer transports. Bug: webrtc:7013 Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c Reviewed-on: https://webrtc-review.googlesource.com/13520 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20661}
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.
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