WebRTC术语(WebRTC常用名词)

WebRTC术语非常多，这里分析来自“给好奇者的WebRTC”给出的相关术语，欢迎大家继续补充。

ACK: Acknowledgment (确认报文)
AVP: Audio and Video profile (音频视频描述)
B-Frame: Bi-directional Predicted Frame. A partial picture, is a modification of previous and future pictures. (双向预测帧，存储图片的部分信息，存储的是相对前一张图片和后一张图片的差异信息)
DCEP: Data Channel Establishment Protocol defined in RFC 8832 (DataChannel 建立协议)
DeMux: Demultiplexer (解复用器)
DLSR: delay since last sender report (从最近一个 Sender Report 开始的时间延迟)
DTLS: Datagram Transport Layer Security defined in RFC 6347
E2E: end-to-end
FEC: Forward Error Correction (前向纠错)
FIR: Full INTRA-frame Request (完整 I 帧请求)
G.711: A narrowband audio codec (一个窄带音频编码器)
H.264: Advanced video coding for generic audiovisual services (面向通用视听服务的高级视频编码)
H.265: Conformance specification for ITU-T H.265 high efficiency video coding. (ITU-T H.265 高效视频编码的一致性规范)
HEVC: High Efficiency Video Coding (高效视频编码)
HTTP: Hypertext Transfer Protocol (超文本传输协议)
HTTPS: HTTP Over TLS defined in RFC 2818 (基于 TLS 的 HTTP)
I-Frame: Intra-coded Frame. A complete picture, can be decoded without anything else. (内部编码帧，保存完整图片信息，自解码，不依赖外部数据)
ICE: Interactive Connectivity Establishment defined in RFC 8445 (交互式连接建立协议)
INIT: Initiate (初始化)
IoT: Internet of Things (物联网)
IPv4: Internet Protocol, Version 4 (第四代因特网协议)
IPv6: Internet Protocol, Version 6 (第六代因特网协议)
ITU-T: International Telecommunication Union Telecommunication Standardization Sector (国际电信联盟电信标准分局)
JSEP: JavaScript Session Establishment Protocol defined in RFC 8829 (JavaScript 会话建立协议)
MCU: Multi-point Conferencing Unit (多点会话单元)
mDNS: Multicast DNS defined in RFC 6762 (组播 DNS)
MITM: Man-In-The-Middle
MTU: Maximum Transmission Unit (最大传输单元)
MUX: Multiplexing (复用，一般指把不同格式的数据合并存储或传输)
NACK: Negative Acknowledgment (逆确认报文，ACK 反馈收到报文 ,NACK 反馈未收到报文 )
NAT: Network Address Translation defined in RFC 4787 (网络地址转换，域名地址转换成 IP 地址)
Opus: A totally open, royalty-free, highly versatile audio codec (一个完全开放、免版税、高度通用的音频编解码器)
P-Frame: Predicted Frame. A partial picture, containing only changes from the previous picture. (前向预测帧，只保存相对于上一帧的差异信息)
P2P: peer-to-peer
PLI: Picture Loss Indication (图片丢失指示)
PPID: Payload Protocol Identifier (Payload 协议标识)
REMB: Receiver Estimated Maximum Bitrate (接收端估计的最大比特率)
RFC: Request for Comments (征求意见)
RMCAT: RTP Media Congestion Avoidance Techniques (RTP 媒体拥塞避免技术)
RR: Receiver Report (RCTP 接收者报告)
RTCP: RTP Control Protocol defined in RFC 3550 (RTP 控制协议)
RTP: Real-time transport protocol defined in RFC 3550 (实时传输协议)
RTT: Round-trip time (往返时间)
SACK: Selective Acknowledgment (选择性确认)
SCTP: Stream Control Transmission Protocol defined in RFC 4960 (流控传输协议)
SDP: Session Description Protocol defined in RFC 8866 (会话描述协议)
SFU: Selective Forwarding Unit (选择性转发单元)
SR: Sender Report (RCTP 发送者报告)
SRTP: Secure Real-time Transport Protocol defined in RFC 3711 (安全的 RTP)
SSRC: Synchronization Source (同步源)
STUN: Session Traversal Utilities for NAT defined in RFC 8489 (NAT 会话穿透)
TCP: Transmission Control Protocol (传输控制协议)
TLS: The Transport Layer Security defined in RFC 8446 (传输层安全)
TMMBN: Temporary Maximum Media Stream Bit Rate Notification (临时最大媒体流比特率通知)
TMMBR: Temporary Maximum Media Stream Bit Rate Request (临时最大媒体流比特率请求)
TSN: Transmission Sequence Number (传输序列号)
TURN: Traversal Using Relays around NAT defined in RFC 8656 (基于转发的 NAT 穿透)
TWCC: Transport Wide Congestion Control (传输拥塞控制)
UDP: User Datagram Protocol (数据报协议)
VP8, VP9: Highly-efficient video compression technologies (video “codecs”) developed by the WebM Project. Anyone may use these codecs royalty-free. (WebM 项目开发的高效视频压缩技术 (视频编解码)，完全免费)
WebM: An open media file format designed for the web. (一个开放的 Web 媒体文件格式)
WebRTC: Web Real-Time Communications. W3C WebRTC 1.0: Real-Time Communication Between Browsers (Web 实时通信)