######################## WebRTC 标准，协议和规范 ######################## .. include:: ../links.ref .. include:: ../tags.ref .. include:: ../abbrs.ref ============ ========================== **Abstract** WebRTC Spec **Authors** Walter Fan **Status** WIP as draft **Category** LearningNote **Updated** |date| ============ ========================== .. contents:: :local: 相关的标准和协议 =================== * WebRTC standard: https://www.w3.org/TR/webrtc * SDP 参见 `RFC4566`_ Session Decscription Protocol * RTP 参见 `RFC3550`_ Realtime Transport Protocols * SRTP 参见 `RFC3711`_ Secure Realtime Transport Protocols * RTP Profile: https://www.rfcreader.com/#rfc3551 * Datagram Transport Layer Security Version 1.2 https://www.rfcreader.com/#rfc6347 * RTCWeb Offer/Answer Protocol (ROAP) https://tools.ietf.org/html/draft-jennings-rtcweb-signaling-01 * Javascript Session Establishment Protocol (JSEP) https://tools.ietf.org/html/rfc8829 * Session Traversal Utilities for NAT (STUN) https://tools.ietf.org/html/rfc5389 * Traversal Using Relays around NAT (TURN) https://tools.ietf.org/html/rfc5766 * Interactive Connectivity Establishment (ICE) https://tools.ietf.org/html/rfc8445 相关的扩展协议 =================== * Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE) https://tools.ietf.org/html/rfc8839 * TCP Candidates with Interactive Connectivity Establishment (ICE) https://tools.ietf.org/html/rfc6544 * Trickling ICE https://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-02 * Datagram Transport Layer Security for SRTP (DTLS-SRTP) https://www.rfcreader.com/#rfc5764 * Connection-Oriented Media Transport over TLS in SDP https://www.rfcreader.com/#rfc4572 * TCP-Based Media Transport in SDP https://www.rfcreader.com/#rfc4145 * Web Real-Time Communication (WebRTC): Media Transport and Use of RTP https://tools.ietf.org/html/rfc8834 * Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF) https://tools.ietf.org/html/rfc5104 * Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF) https://tools.ietf.org/html/rfc4585 * REMB - RTCP message for Receiver Estimated Maximum Bitrate https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 * Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF) https://tools.ietf.org/html/rfc5104 * A Google Congestion Control Algorithm for Real-Time Communication https://tools.ietf.org/html/draft-ietf-rmcat-gcc-02 * Framing RTP and RTCP Packets over Connection-Oriented Transport https://tools.ietf.org/html/rfc4571 * Source-Specific Media Attributes in the Session Description Protocol (SDP) RFC5576: https://datatracker.ietf.org/doc/html/rfc5576 * Using Simulcast in Session Description Protocol (SDP) and RTP Sessions RFC8853: https://datatracker.ietf.org/doc/html/rfc8853 * (RTP) Header Extension for Client-to-Mixer Audio Level Indication RFC6464: https://tools.ietf.org/html/rfc6464 * RTP Retransmission Payload Format https://tools.ietf.org/html/rfc4588 * Guidelines for Using the Multiplexing Features of RTP to Support Multiple Media Streams https://tools.ietf.org/html/rfc8872 * Negotiating Media Multiplexing Using SDP https://tools.ietf.org/html/rfc8843 * RTP Stream Identifier Source Description (SDES) https://tools.ietf.org/html/draft-ietf-avtext-rid-09 * WebRTC MediaStream Identification in SDP https://tools.ietf.org/html/rfc8830 * RTP Extensions for Transport-wide Congestion Control https://tools.ietf.org/html/draft-ietf-avtext-rid-09 * RTP Header Extension for the RTCP Source Description Items https://datatracker.ietf.org/doc/html/rfc7941 * RTP Extensions for Transport-wide Congestion Control (draft-holmer-rmcat-transport-wide-cc-extensions-01) https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 * A Framework for SDP Attributes when Multiplexing https://tools.ietf.org/html/rfc8859 * ULPFEC - RTP Payload Format for Generic Forward Error Correction https://tools.ietf.org/html/rfc5109 * RED - RTP Payload for Redundant Audio Data https://tools.ietf.org/html/rfc2198 * RTP Payload Format for H.264 Video https://tools.ietf.org/html/rfc6184 * RTP Payload Format for Scalable Video Coding https://tools.ietf.org/html/rfc6190 * Definition of the Opus Audio Codec https://tools.ietf.org/html/rfc6716 * WebRTC Data Channels https://datatracker.ietf.org/doc/html/rfc8831 * Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets https://datatracker.ietf.org/doc/html/rfc8261 新标准和规范 ========================= 1. The extensions to WebRTC PeerConnection * `WebRTC Extensions`_: defines a set of ECMAScript APIs in WebIDL to extend the WebRTC 1.0 API * WebRTC-SVC * `Insertable Streams`_: defines an API surface for manipulating the bits on MediaStreamTracks being sent via an RTCPeerConnection. 2. Some involves features which did not meet the implementation or maturity requirements for inclusion in the WebRTC-PC Proposed Recommendation * WebRTC Identity * WebRTC Priority Control * WebRTC DSCP. 3. The extensions to Capture, * MediaStreamTrack Insertable Streams * Media Capture and Streams Extensions * MediaCapture Depth Stream Extensions 4. standalone specifications, which are not necessarily dependent on either `RTCPeerConnection` or the existing Media Capture specifications. * WebRTC-ICE (which so far has been implemented as a standalone specification) * WebTransport (in the W3C WebTransport WG), * WebRTC-QUIC (in the ORTC CG) and * `Web Codecs`_ (in the WICG): provide JavaScript interfaces to implementations of existing codec technology developed elsewhere