WebRTC extension

Abstract

WebRTC Extension

Authors

Walter Fan

Status

WIP as draft

Category

LearningNote

Updated

2024-08-21

Overview

根据 RFC3550 和 RFC8285, 我们可以定义若干 RTP 扩展头, 用于各种不同的用途。

除了一些标准的 RTP header, 比如 audio level , WebRTC 还定义了一些其他的扩展头

  • abs-send-time

  • abs-capture-time

  • color-space

  • playout-delay

  • transport-wide-cc-02

  • video-content-type

  • video-timing

  • inband-cn

  • video-layers-allocation00

refer to https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext

audio extension

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id

video extension

a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id

new extension

a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time

Video Layers Allocation

The goal of this extension is for a video sender to provide information about the target bitrate, resolution and frame rate of each scalability layer in order to aid a selective forwarding middlebox to decide which layer to relay.

In a conference scenario, a video from a single sender may be received by several recipients with different downlink bandwidth constraints and UI requirements.

To allow this, a sender can send video with several scalability layers and a middle box can choose a layer to relay for each receiver.

This extension support temporal layers, multiple spatial layers sent on a single rtp stream (SVC), or independent spatial layers sent on multiple rtp streams (simulcast).

refer to https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/video-layers-allocation00

Reference