"to be compliant with Web Real-Time Communication (WebRTC)."RFC8825より引用。
"The total WebRTC effort consists of two major parts ... A protocol specification ... A JavaScript API specification"RFC8825より引用。
'first three groups as forming a "media transport infrastructure"'RFC8825より引用。
"WebRTC endpoints MUST implement the transport protocols described in [RFC8835]."RFC8825より引用。
"The format for media transport is RTP ... Implementation of the Secure Real-time Transport Protocol (SRTP) [RFC3711] is REQUIRED for all implementations."RFC8825より引用。
"Considerations for the transfer of data that is not in RTP format are described in [RFC8831] ... WebRTC endpoints MUST implement these two specifications."RFC8825より引用。
"This document specifies a minimum baseline ... and leaves further codecs to be included"RFC8825より引用。
'the last three groups as forming a "media service".'RFC8825より引用。
"Local system support functions ... Examples in this category include echo cancellation ... local authentication and authorization mechanisms, OS access control, and the ability to do local recording of conversations."RFC8825より引用。
"WebRTC endpoints MUST implement the processing functions in [RFC7874]"RFC8825より引用。
"The following terms are used across the documents specifying the WebRTC suite, with the specific meanings given here."RFC8825より引用。
"Media where the generation and display of content are intended to occur closely together in time (on the order of no more than hundreds of milliseconds)."RFC8825より引用。
