WebRTC标准，协议和规范

• 相关的标准和协议

• 相关的扩展协议

• 新标准和规范

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:

RFC Reader - An online reader for IETF RFCs

• Datagram Transport Layer Security Version 1.2

RFC Reader - An online reader for IETF RFCs

• RTCWeb Offer/Answer Protocol (ROAP)

draft-jennings-rtcweb-signaling-01

• Javascript Session Establishment Protocol (JSEP)

RFC 8829 - JavaScript Session Establishment Protocol (JSEP)

• Session Traversal Utilities for NAT (STUN)

RFC 5389 - Session Traversal Utilities for NAT (STUN)

• Traversal Using Relays around NAT (TURN)

RFC 5766 - Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)

• Interactive Connectivity Establishment (ICE)

RFC 8445 - Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal

相关的扩展协议¶

• Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE)

RFC 8839 - Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE)

• TCP Candidates with Interactive Connectivity Establishment (ICE)

RFC 6544 - TCP Candidates with Interactive Connectivity Establishment (ICE)

• Trickling ICE

draft-ivov-mmusic-trickle-ice-sip-02

• Datagram Transport Layer Security for SRTP (DTLS-SRTP)

RFC Reader - An online reader for IETF RFCs

• Connection-Oriented Media Transport over TLS in SDP

RFC Reader - An online reader for IETF RFCs

• TCP-Based Media Transport in SDP

RFC Reader - An online reader for IETF RFCs

• Web Real-Time Communication (WebRTC): Media Transport and Use of RTP

RFC 8834 - Media Transport and Use of RTP in WebRTC

• Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)

RFC 5104 - Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)

• Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF)

RFC 4585 - Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)

• REMB - RTCP message for Receiver Estimated Maximum Bitrate

draft-alvestrand-rmcat-remb-03

• Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)

RFC 5104 - Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)

• A Google Congestion Control Algorithm for Real-Time Communication

draft-ietf-rmcat-gcc-02

• Framing RTP and RTCP Packets over Connection-Oriented Transport

RFC 4571 - Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Transport

• Source-Specific Media Attributes in the Session Description Protocol (SDP)

RFC5576: RFC 5576 - Source-Specific Media Attributes in the Session Description Protocol (SDP)

• Using Simulcast in Session Description Protocol (SDP) and RTP Sessions

RFC8853: RFC 8853 - Using Simulcast in Session Description Protocol (SDP) and RTP Sessions

• (RTP) Header Extension for Client-to-Mixer Audio Level Indication

RFC6464: RFC 6464 - A Real-time Transport Protocol (RTP) Header Extension for Client-to-Mixer Audio Level Indication

• RTP Retransmission Payload Format

RFC 4588 - RTP Retransmission Payload Format

• Guidelines for Using the Multiplexing Features of RTP to Support Multiple Media Streams

RFC 8872 - Guidelines for Using the Multiplexing Features of RTP to Support Multiple Media Streams

• Negotiating Media Multiplexing Using SDP

RFC 8843 - Negotiating Media Multiplexing Using the Session Description Protocol (SDP)

• RTP Stream Identifier Source Description (SDES)

draft-ietf-avtext-rid-09

• WebRTC MediaStream Identification in SDP

RFC 8830 - WebRTC MediaStream Identification in the Session Description Protocol

• RTP Extensions for Transport-wide Congestion Control

draft-ietf-avtext-rid-09

• RTP Header Extension for the RTCP Source Description Items

RFC 7941 - RTP Header Extension for the RTP Control Protocol (RTCP) Source Description Items

• RTP Extensions for Transport-wide Congestion Control (draft-holmer-rmcat-transport-wide-cc-extensions-01)

draft-holmer-rmcat-transport-wide-cc-extensions-01

• A Framework for SDP Attributes when Multiplexing

RFC 8859 - A Framework for Session Description Protocol (SDP) Attributes When Multiplexing

• ULPFEC - RTP Payload Format for Generic Forward Error Correction

RFC 5109 - RTP Payload Format for Generic Forward Error Correction

• RED - RTP Payload for Redundant Audio Data

RFC 2198 - RTP Payload for Redundant Audio Data

• RTP Payload Format for H.264 Video

RFC 6184 - RTP Payload Format for H.264 Video

• RTP Payload Format for Scalable Video Coding

RFC 6190 - RTP Payload Format for Scalable Video Coding

• Definition of the Opus Audio Codec

RFC 6716 - Definition of the Opus Audio Codec

• WebRTC Data Channels

RFC 8831 - WebRTC Data Channels

• Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets

RFC 8261 - Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets

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.

1. 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.

1. The extensions to Capture,

• MediaStreamTrack Insertable Streams

• Media Capture and Streams Extensions

• MediaCapture Depth Stream Extensions

1. 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