WebRTC-HTTP ingestion protocol (WHIP)
draft-murillo-whip-00
Network Working Group S. Murillo
Internet-Draft A. Gouaillard
Intended status: Informational CoSMo Software
Expires: March 13, 2021 September 09, 2020
WebRTC-HTTP ingestion protocol (WHIP)
draft-murillo-whip-00
Abstract
While WebRTC has been very sucessfull in a wide range of scenarios,
its adption in the broadcasting/streaming industry is lagging behind.
Currently there is no standard protocol (like SIP or RTSP) designed
for ingesting media in a streaming service, and content providers
still rely heavily on protocols like RTMP for it.
These protocols are much older than webrtc and lack by default some
important security and resilience features provided by webrtc with
minimal delay.
The media codecs used in older protocols do not always match those
being used in WebRTC, mandating transcoding on the ingest node,
introducing delay and degrading media quality. This transcoding step
is always present in traditionnal streaming to support e.g. ABR, and
comes at no cost. However webrtc implements client-side ABR, also
called Network-Aware Encoding by e.g. Huavision, by means of
simulcast and SVC codecs, which otherwise alleviate the need for
server-side transcoding. Content protection and Privacy Enhancement
can be achieve with End-to-End Encryption, which preclude any server-
side media processing.
This document proposes a simple HTTP based protocol that will allow
WebRTC endpoings to ingest content into streaming servics and/or CDNs
to fill this gap and facilitate deployment.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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and may be updated, replaced, or obsoleted by other documents at any
Murillo & Gouaillard Expires March 13, 2021 [Page 1]
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time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on March 13, 2021.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Protocol Operation . . . . . . . . . . . . . . . . . . . . . 4
4.1. ICE and NAT support . . . . . . . . . . . . . . . . . . . 4
4.2. Webrtc contrains . . . . . . . . . . . . . . . . . . . . 5
4.3. Load balancing and redirections . . . . . . . . . . . . . 5
4.4. Authentication and authorization . . . . . . . . . . . . 5
4.5. Simulcast and scalable video coding . . . . . . . . . . . 5
5. Security Considerations . . . . . . . . . . . . . . . . . . . 6
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6
8. Normative References . . . . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6
1. Introduction
WebRTC intentionaly does not specify a signaling transport protocol
at application level, while RTCWEB standardized the signalling
protocol itself (JSEP, SDP O/A) and everything that was going over
the wire (media, codec, encryption, ...). This flexibility has
allowed for implementing a wide range of services. However, those
services are typically standalone silos which don't require
interoperability with other services or leverage the existence of
tools that can communicate with them.
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