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https://github.com/leaysgur/webrtc-rfcs
WebRTC related RFCs that I have read through.
https://github.com/leaysgur/webrtc-rfcs
Last synced: 24 days ago
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WebRTC related RFCs that I have read through.
- Host: GitHub
- URL: https://github.com/leaysgur/webrtc-rfcs
- Owner: leaysgur
- Created: 2019-01-24T10:35:31.000Z (almost 6 years ago)
- Default Branch: master
- Last Pushed: 2019-11-05T02:28:51.000Z (about 5 years ago)
- Last Synced: 2024-04-15T07:12:07.882Z (7 months ago)
- Language: JavaScript
- Homepage:
- Size: 832 KB
- Stars: 13
- Watchers: 1
- Forks: 1
- Open Issues: 0
-
Metadata Files:
- Readme: README.md
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# webrtc-rfcs
WebRTC related RFCs, Dradts, and Specs.
- See [`/markdown`](./markdown) for Markdown formatted RFC.
- See [`/summary`](./summary) for Japanese summary of each RFC.
Linked means I've already read through.
## Overview
- [draft-ietf-rtcweb-overview-19](./summary/draft-ietf-rtcweb-overview-19.md)
- Overview: Real Time Protocols for Browser-based Applications
- そもそもの思想や全体像について
- [RFC7478](./summary/rfc7478.md)
- Web Real-Time Communication Use Cases and Requirements
- WebRTCのユースケースと、その実現に必要なコンポーネント、実装に対する要求について
- [draft-ietf-rtcweb-transports-17](./summary/draft-ietf-rtcweb-transports-17.md)
- Transports for WebRTC
- WebRTCで使用しているトランスポートや実装すべきプロトコルについて
- [webrtc-nv-use-cases](https://w3c.github.io/webrtc-nv-use-cases/)
- WebRTC Next Version Use Cases
- 過去を踏まえたこれからのWebRTCのユースケースについて
- [RFC7742](./summary/rfc7742.md)
- WebRTC Video Processing and Codec Requirements
- WebRTCで扱う映像の概要について
- [RFC7874](./summary/rfc7874.md)
- WebRTC Audio Processing and Codec Requirements
- WebRTCで扱う音声の概要について
## W3C JavaScript API
- [draft-ietf-rtcweb-jsep-26](./summary/draft-ietf-rtcweb-jsep-26.md)
- JavaScript Session Establishment Protocol
- JavaScriptのAPI、各RFCとAPIがどう関係してるかなど実践的な全体像について
- [webrtc-pc](https://w3c.github.io/webrtc-pc/)
- WebRTC 1.0: Real-time Communication Between Browsers
- `RTCPeerConnection`について
- [webrtc-stats](https://w3c.github.io/webrtc-stats/)
- Identifiers for WebRTC's Statistics API
- `getStats()`について
- [mediacapture-main](https://w3c.github.io/mediacapture-main/)
- Media Capture and Streams
- `MediaStream`と`getUserMedia()`について
- [mediacapture-image](https://w3c.github.io/mediacapture-image/)
- MediaStream Image Capture
- `ImageCapture`について
- [mediacapture-fromelement](https://w3c.github.io/mediacapture-fromelement/)
- Media Capture from DOM Elements
- `captureStream()`について
- [mediacapture-record](https://w3c.github.io/mediacapture-record/)
- MediaStream Recording
- `MediaRecorder`について
- [mediacapture-output](https://w3c.github.io/mediacapture-output/)
- Audio Output Devices API
- `sinkId`と`setSinkId()`について
- [mediacapture-screen-share](https://w3c.github.io/mediacapture-screen-share/)
- Screen Capture
- `getDisplayMedia()`について
- [webrtc-svc](https://w3c.github.io/webrtc-svc/)
- Scalable Video Coding (SVC) Extension for WebRTC
- WebRTCでSVCを実現するための拡張について
- [webrtc-ice](https://w3c.github.io/webrtc-ice/)
- IceTransport Extensions for WebRTC
- `RTCIceTransport`について
- [webrtc-quic](https://w3c.github.io/webrtc-quic/)
- QUIC API for Peer-to-peer Connections
- `RTCQuicTransport`について
- [webrtc-quic/cs](https://w3c.github.io/webrtc-quic/cs.html)
- QUIC API for Client-to-Server Connections
- `RTCQuicStream`について
## ICE(STUN/TURN)
- [RFC8445](./summary/rfc8445.md)
- ICE: A Protocol for Network Address Translator (NAT) Traversal
- NATを越えて通信するためのプロトコルであるICEについて
- [RFC7675](./summary/rfc7675.md)
- STUN Usage for Consent Freshness
- ICEのKeepAliveの代わりに行うConsentFreshnessという手順について
- [draft-ietf-ice-trickle-21](./summary/draft-ietf-ice-trickle-21.md)
- Trickle ICE: Incremental Provisioning of Candidates for the ICE Protocol
- ICEのcandidateを送るタイミングを早めるTrickleというデファクトスタンダードな手法について
- [RFC8421](./summary/rfc8421.md)
- Guidelines for Multihomed and IPv4/IPv6 Dual-Stack ICE
- candidateの優先度計算でIPv4とIPv6の違いをどう考慮するかについて
- [draft-thatcher-ice-renomination-01](./summary/draft-thatcher-ice-renomination-01.md)
- ICE Renomination: Dynamically selecting ICE candidate pairs
- ノミネーションを動的に行い通信経路をいつでも変えられるようにするICE Renominationについて
- [RFC5389](./summary/rfc5389.md)
- Session Traversal Utilities for NAT
- ICEの過程で使われるSTUNプロトコルについて
- [RFC7064](./summary/rfc7064.md)
- STUN URI Scheme for the STUN Protocol
- STUNサーバーをURIで表すための`stun`と`stuns`について
- [draft-ietf-ice-pac-03](./summary/draft-ietf-ice-pac-03.md)
- ICE Patiently Awaiting Connectivity
- ICEで`prflx`な候補を見つけるためのタイマーについて
- RFC6544
- TCP Candidates with ICE
- draft-ietf-tram-turnbis-21
- Traversal Using Relays around NAT (TURN): Relay Extensions to STUN
- [draft-uberti-rtcweb-turn-rest-00](./summary/draft-uberti-rtcweb-turn-rest-00.md)
- A REST API For Access To TURN Services
- 秘密鍵を事前に共有することで、RESTのAPI経由でTURNを利用できる仕組みについて
## SDP
- [draft-ietf-rtcweb-sdp-11](./summary/draft-ietf-rtcweb-sdp-11.md)
- Annotated Example SDP for WebRTC
- WebRTCで実際に使われるSDPのサンプルについて
- [RFC4566](./summary/rfc4556.md)
- SDP: Session Description Protocol
- WebRTC以前からあるSDPの基本について
- [RFC3264](./summary/rfc3264.md)
- An Offer/Answer Model with the SDP
- WebRTCのオファー・アンサーモデルでSDPをどう使うかについて
- [draft-ietf-mmusic-ice-sip-sdp-39](./summary/draft-ietf-mmusic-ice-sip-sdp-39.md)
- SDP Offer/Answer procedures for ICE
- ICEの情報をSDPにどう載せて使うかについて
- [RFC4145](./summary/rfc4145.md)
- TCP-Based Media Transport in the SDP
- 元々はTCPのコネクションを確立するための`a=setup`行の成り立ちについて
- [RFC8122](./summary/rfc8122.md)
- Connection-Oriented Media Transport over the TLS Protocol in the SDP
- セキュアにメディアを転送するための`a=fingerprint`行の成り立ちについて
- [RFC5763](./summary/rfc5763.md)
- Framework for Establishing a SRTP Security Context Using DTLS
- SRTPの鍵交換にDTLSを使うための`a=setup`行とその後の手順について
- [RFC3605](./summary/rfc3605.md)
- RTCP attribute in SDP
- RTCPのポート番号を定義する`a=rtcp`行について
- [RFC5506](./summary/rfc5506.md)
- Support for Reduced-Size RTCP: Opportunities and Consequences
- 通常のRTCPの複合パケットとは別で送られるReduced-Size RTCPの使用を意味する`a=rtcp-rsize`行について
- [RFC5761](./summary/rfc5761.md)
- Multiplexing RTP Data and Control Packets on a Single Port
- RTPとRTCPを多重化するメディアレベルの`a=rtcp-mux`行について
- [draft-ietf-mmusic-mux-exclusive-12](./summary/draft-ietf-mmusic-mux-exclusive-12.md)
- Indicating Exclusive Support of RTP/RTCP Multiplexing using SDP
- RTPとRTCPの多重化を強制するメディアレベルの`a=rtcp-mux-only`行について
- [draft-roach-mmusic-unified-plan-00](./summary/draft-roach-mmusic-unified-plan-00.md)
- A Unified Plan for Using SDP with Large Numbers of Media Flows
- 複数のメディアをSDPにどう記述するかを定めたUnifiedPlanの基本の考え方について
- [RFC5576](./summary/rfc5576.md)
- Source-Specific Media Attributes in the SDP
- RTPのSSRCとの関連を表すソースレベルの`a=ssrc`行について
- [RFC5888](./summary/rfc5888.md)
- The Session Description Protocol (SDP) Grouping Framework
- メディアをグルーピングできるセッションレベルの`a=group`属性とメディアレベルの`a=mid`属性について
- [draft-ietf-mmusic-sdp-bundle-negotiation-54](./summary/draft-ietf-mmusic-sdp-bundle-negotiation-54.md)
- Negotiating Media Multiplexing Using the SDP
- 1つのポートで複数のメディアを多重化するための`a=group:BUNDLE`について
- [draft-ietf-mmusic-msid-17](./summary/draft-ietf-mmusic-msid-17.md)
- WebRTC MediaStream Identification in the Session Description Protocol
- SDPとMediaStream(Track)を紐付けるメディアレベルの`a=msid`行について
- [draft-ietf-mmusic-sdp-mux-attributes-17](./summary/draft-ietf-mmusic-sdp-mux-attributes-17.md)
- A Framework for SDP Attributes when Multiplexing
- 多重化を実施する場合に、それぞれのSDPの属性がどのように影響を受けるかについて
- RFC5956
- Forward Error Correction Grouping Semantics in the SDP
- RFC3556
- SDP Bandwidth Modifiers for RTCP Bandwidth
## DTLS
- [RFC6347](./summary/rfc6347.md)
- Datagram Transport Layer Security Version 1.2
- UDPでTLSを使用するための、DTLS 1.2について
- [RFC5764](./summary/rfc5764.md)
- DTLS Extension to Establish Keys for the SRTP
- SRTPのための鍵を取得するためのDLTSの`use_srtp`拡張について
- [RFC7983](./summary/rfc7983.md)
- Multiplexing Scheme Updates for SRTP Extension for DTLS
- STUN/DTLS/SRTPがすべて多重化された場合の識別方法について
- RFC6083
- DTLS for SCTP
## SRTP
- RFC3711
- The Secure Real-time Transport Protocol (SRTP)
- RFC6904
- Encryption of Header Extensions in the SRTP
- RFC6562
- Guidelines for the Use of Variable Bit Rate Audio with Secure RTP
## RTP/RTCP
- [draft-ietf-rtcweb-rtp-usage-26](./summary/draft-ietf-rtcweb-rtp-usage-26.md)
- WebRTC Media Transport and Use of RTP
- WebRTCにおけるRTPの関係とその使い方について
- RFC3550
- RTP: A Transport Protocol for Real-Time Applications
- RFC8108
- Sending Multiple RTP Streams in a Single RTP Session
- RFC7160
- Support for Multiple Clock Rates in an RTP Session
- RFC7164
- RTP and Leap Seconds
- RFC7667
- RTP Topologies
- RFC4588
- RTP Retransmission Payload Format
- RFC8083
- Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
- RFC3611
- RTP Control Protocol Extended Reports (RTCP XR)
- RFC6792
- Guidelines for Use of the RTP Monitoring Framework
- RFC2736
- Guidelines for Writers of RTP Payload Format Specifications
- RFC5968
- Guidelines for Extending the RTCP
- RFC5285
- A General Mechanism for RTP Header Extensions
- RFC6051
- Rapid Synchronisation of RTP Flow
- RFC3551
- RTP Profile for Audio and Video Conferences with Minimal Control
- RFC5104
- Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)
- RFC5761
- Multiplexing RTP Data and Control Packets on a Single Port
- RFC4961
- Symmetric RTP / RTCP
- RFC7022
- Guidelines for Choosing RTCP Canonical Names (CNAMEs)
- RFC4585
- Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF)
- RFC5124
- Extended Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF)
- RFC6464
- A RTP Header Extension for Client-to-Mixer Audio Level Indication
- RFC6465
- A RTP Header Extension for Mixer-to-Client Audio Level Indication
- RFC7941
- A RTP Header Extension for the RTCP Source Description Items
- draft-ietf-rtcweb-fec-10
- WebRTC Forward Error Correction Requirements
- draft-ietf-rmcat-cc-requirements-09
- Congestion Control Requirements for Interactive Real-Time Media
## SCTP
- RFC4960
- Stream Control Transmission Protocol
- RFC3758
- SCTP Partial Reliability Extension
- RFC6525
- SCTP Stream Reconfiguration
- RFC5061
- SCTP Dynamic Address Reconfiguration
- RFC6951
- UDP Encapsulation of SCTP Packets for End-Host to End-Host Communication
## DataChannel
- [draft-ietf-rtcweb-data-channel-13](./summary/draft-ietf-rtcweb-data-channel-13.md)
- WebRTC Data Channels
- WebRTCのData Channelについて
- draft-ietf-rtcweb-data-protocol-09
- WebRTC Data Channel Establishment Protocol
## Security
- [draft-ietf-rtcweb-ip-handling-12](./summary/draft-ietf-rtcweb-ip-handling-12.md)
- WebRTC IP Address Handling Requirements
- P2Pするために集めたIPアドレスの扱いについて
- [draft-ietf-rtcweb-mdns-ice-candidates-04](./summary/draft-ietf-rtcweb-mdns-ice-candidates-04.md)
- Using Multicast DNS to protect privacy when exposing ICE candidates
- プライバシー保護のためにプライベートIPの代わりにmDNSの登録名を使う提案について
- [draft-ietf-rtcweb-security-12](./summary/draft-ietf-rtcweb-security-12.md)
- Security Considerations for WebRTC
- WebRTCにおけるセキュリティについて
- draft-ietf-rtcweb-security-arch-17
- WebRTC Security Architecture
## Not yet maintained
---
## Simulcast / SVC
- draft-ietf-mmusic-sdp-simulcast-14
- Using Simulcast in SDP and RTP Sessions
- draft-ietf-mmusic-rid-15
- RTP Payload Format Restrictions
- draft-ietf-avtext-rid-09
- RTP Stream Identifier Source Description (SDES)
## WebTransport
- draft-vvv-webtransport-overview-00
- The WebTransport Protocol Framework
- draft-vvv-webtransport-http3-00
- WebTransport over HTTP/3
- draft-vvv-webtransport-quic-00
- WebTransport over QUIC
---