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Versions: 00 01 02                                                      
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
   working documents as Internet-Drafts.  The list of current Internet-
   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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   In the broadcasting/streaming world, the usage of hardware encoders
   that would make it very simple to plug in (SDI) cables carrying raw
   media, encoding it in place, and pushing it to any streaming service
   or CDN ingest is ubiquitous.  Having to implement a custom signalling
   transport protocol for each different webrtc services has hindered
   adoption.

   While some standard signalling protocols are available that can be
   integrated with WebRTC, like SIP or XMPP, they are not designed to be
   used in broadcasting/streaming services, and there also is no sign of
   adoption in that industry.  RTSP, which is based on RTP and maybe the
   closest in terms of features to webrtc, is not compatible with WebRTC
   SDP offer/answer model.

   In the specific case of ingest into a platform, some assumption can
   be made about the server-side which simplifies the webrtc compliance
   burden, as detailled in webrtc-gateway document.
   https://tools.ietf.org/html/draft-ietf-rtcweb-gateways-02

   This document proposse a simple protocol for supporting WebRTC as
   ingest method which is: - Easy to implement, - As easy to use as
   current RTMP URI.  - Fully compliant with Webrtc and RTCWEB specs.  -
   Allow for both ingest in traditionnal media platforms for extention
   and ingest in webrtc end-to-end platform for lowest possible latency.
   - Lowers the requirements on both hardware encoders and broadcasting
   services to support webrtc.  - Usable both in web browsers and in
   native encoders.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Overview

   The WebRTC-HTTP ingest protocol (WHIP) uses an HTTP POST request to
   perform a single shot SDP offer/answer so an ICE/DTLS session can be
   established between the encoder/media producer and the broadcasting
   ingestion endpoint.

   Once the ICE/DTLS session is set up, the media will flow
   unidirectionally from the encoder/media producer broadcasting
   ingestion endpoint.  In order to reduce complexity, no SDP
   renegotiation is supported, so no tracks or streams can be added or
   removed once the initial SDP O/A over HTTP is completed.





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   +-----------------+         +---------------+ +--------------+
   | WebRTC Producer |         | WHIP endpoint | | Media Server |
   +---------+-------+         +-------+- -----+ +------+-------+
             |                         |                |
             |                         |                |
             |HTTP POST (SDP Offer)    |                |
             +-------------------------+                |
             |202 Accepted (SDP answer)|                |
             +<------------------------+                |
             |          ICE REQUEST                     |
             +----------------------------------------->+
             |          ICE RESPONSE                    |
             <------------------------------------------+
             |          DTLS SETUP                      |
             <==========================================>
             |          RTP FLOW                        |
             +------------------------------------------>

                            WHIP session setup

4.  Protocol Operation

   In order to setup an ingestion session, the WebRTC encoder/media
   producer will generate an SDP offer according the the JSEP rules and
   do an HTTP POST request to the WHIP endpoint configured URL.

   The HTTP POST request will have a content type of application/sdp and
   contain the SDP offer as body.  The WHIP ingestion endpoint will
   generate an SDP answer and return it on a 202 Accepted response with
   content type of application/sdp and the SDP answer as body.

   SDP offer SHOULD use the sendonly attribute and the SDP answer MUST
   use the recvonly attribute.

   Once session is setup ICE consent freshness [RFC7675] will be used to
   detect abrupt disconnection and DTLS teardown for session termination
   by either side.

4.1.  ICE and NAT support

   In order to simplify the protocol, there is no support of exchanging
   gathered tickle ICE candidates one the SDP offer or answer is sent.
   So in order to support encoders/media producers behind NAT, the WHIP
   media server MUST be publicly accessible.

   The initial offer by the encoder/media producer MAY be sent after the
   full ICE gathering is complete containing the full list of ICE
   candidates, or only contain local candidates or even an empty list of



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   candidates.  The WHIP endpoint SDP answer SHALL contain the full list
   of ICE candidates publicly accessible of the media server.  The media
   server MAY use ICE lite, while the encoder/media producer MUST
   implement full ICE.

   If the Encoder/Media producer gathers additional candidates (via
   STUN/TURN) after the SDP offer is sent, it will send directly a STUN
   request to the ICE candidates received from the media server as per
   [I-D.draft-ietf-ice-trickle-21].

4.2.  Webrtc contrains

   In order to reduce the complexity of implementing WHIP in both
   encoders and media servers, some restrictions regarding WebRTC usage
   are made.

   SDP bundle SHALL be used by both the encoder/media producer and the
   media server.  The SDP offer created by the encoder/media producer
   MUST include the bundle-only attribute in all m-lines as per
   [I-D.draft-ietf-mmusic-sdp-bundle-negotiation-54].  Also, RTCP muxing
   SHALL be supported by the both the encoder/media producer and the
   media server.

4.3.  Load balancing and redirections

   Encoders/media MAY not be colocated on the same server so it is
   possible to load balance incoming request to different media server.
   Encoders/media producers SHALL support HTTP redirection via 307
   Temporary Redirect response code.

   In case of high load, the WHIP endpoints may return a 503 (Service
   Unavailable) status code indicating that the server is currently
   unable to handle the request due to a temporary overload or scheduled
   maintenance, which will likely be alleviated after some delay.  The
   server MAY send a Retry-After header field indicating the minimum
   time that the user agent is asked to wait before issuing the
   redirected request.

4.4.  Authentication and authorization

   Authtentication and authorization is supported by the Authorization
   HTTP header with a bearear token as per [RFC6750].

4.5.  Simulcast and scalable video coding

   Both simulcast and scalable video coding (including K-SVC modes) MAY
   be supported by both media servers and encoders/media producers.




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5.  Security Considerations

   HTTPS SHALL be used in order to preserve WebRTC security model.

6.  IANA Considerations

7.  Acknowledgements

8.  Normative References

   [I-D.draft-ietf-ice-trickle-21]
              Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
              "Trickle ICE: Incremental Provisioning of Candidates for
              the Interactive Connectivity Establishment (ICE)
              Protocol", draft-ietf-ice-trickle-21 (work in progress),
              April 2018.

   [I-D.draft-ietf-mmusic-sdp-bundle-negotiation-54]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-54 (work in progress), December 2018.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

   [RFC6750]  Jones, M. and D. Hardt, "The OAuth 2.0 Authorization
              Framework: Bearer Token Usage", RFC 6750,
              DOI 10.17487/RFC6750, October 2012,
              <https://www.rfc-editor.org/info/rfc6750>.

   [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
              Thomson, "Session Traversal Utilities for NAT (STUN) Usage
              for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
              October 2015, <https://www.rfc-editor.org/info/rfc7675>.

Authors' Addresses

   Sergio Garcia Murillo
   CoSMo Software

   Email: sergio.garcia.murillo@cosmosoftware.io







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   Alexandre Gouaillard
   CoSMo Software

   Email: alex.gouaillard@cosmosoftware.io















































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