commit | 8e1343aeda4b1a3ad1b34864b39a5458123e82fa | [log] [tgz] |
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author | Bjorn A Mellem <mellem@webrtc.org> | Mon Sep 30 22:12:47 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Sep 30 23:10:34 2019 |
tree | b82f913e6c1bb2e53ce49aa13fb732a6f665e392 | |
parent | 894eb8b0c9ba8aa304eca33465dad3754afdc7e5 [diff] |
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
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.
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.