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WebRTC is making progress! WebRTC has the goal of enabling browser-to-browser realtime communications, allowing popular applications such as audio and video chatting to be implemented in Web apps. This goal requires both browser APIs (for accessing WebRTC capabilities from the Web app) and network APIs (for browsers to communicate media streams across the Web). WebRTC works on all these aspects, and has made some progress today:
"Identifiers for WebRTC's Statistics API" went from WD to CR. It "defines a set of WebIDL objects that allow access to the statistical information about a RTCPeerConnection. These objects are returned from the getStats API that is specified in WebRTC."
"WebRTC DSCP Control API" was published as a first WD. "This API defines a control surface for manipulating the network control bits (DSCP bits) of outgoing WebRTC packets."
"MediaStreamTrack Content Hints" was published as a first WD. "This specification extends MediaStreamTrack to provide a media-content hint attribute. This optional hint permits MediaStreamTrack consumers such as PeerConnection (defined in WebRTC) or MediaRecorder (defined in MediaStream Recording) to encode or process track media with methods more appropriate to the type of content that is being consumed. Adding a media-content hint provides a way for a web application to help track consumers make more informed decision of what encoder parameters and processing algorithms to use on the consumed content."
An expert in protocol design and structured data, Erik Wilde consults with organizations to help them get the most out of APIs and microservices. Erik has been involved in the development of innovative technologies since the advent of the Web and is active in the IETF and W3C communities. He obtained his PhD from ETH Zurich and served as Associate Adjunct Professor at Berkeley before working at EMC, Siemens and now CA Technologies.