WebRTC-HTTP ingestion protocol (WHIP)
draft-murillo-whip-00

Document Type Active Internet-Draft (individual)
Authors Sergio Garcia Murillo  , Alexandre Gouaillard 
Last updated 2020-09-09
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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.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at https://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any

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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.

Copyright Notice

   Copyright (c) 2020 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (https://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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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