WebRTC-HTTP ingestion protocol (WHIP)
draft-murillo-whip-02

Document Type Active Internet-Draft (individual)
Authors Sergio Garcia Murillo  , Alex Gouaillard 
Last updated 2021-06-09
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Network Working Group                                         S. Murillo
Internet-Draft                                             A. Gouaillard
Intended status: Informational                            CoSMo Software
Expires: December 11, 2021                                 June 09, 2021

                 WebRTC-HTTP ingestion protocol (WHIP)
                         draft-murillo-whip-02

Abstract

   While WebRTC has been very successful in a wide range of scenarios,
   its adoption 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 traditional 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 achieved with End-to-End Encryption, which preclude any
   server-side media processing.

   This document proposes a simple HTTP based protocol that will allow
   WebRTC endpoints to ingest content into streaming services 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
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   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 December 11, 2021.

Copyright Notice

   Copyright (c) 2021 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
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   include Simplified BSD License text as described in Section 4.e of
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   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Overview  . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   4.  Protocol Operation  . . . . . . . . . . . . . . . . . . . . .   5
     4.1.  ICE and NAT support . . . . . . . . . . . . . . . . . . .   6
     4.2.  Webrtc constraints  . . . . . . . . . . . . . . . . . . .   7
     4.3.  Load balancing and redirections . . . . . . . . . . . . .   7
     4.4.  STUN/TURN server configuration  . . . . . . . . . . . . .   7
     4.5.  Authentication and authorization  . . . . . . . . . . . .   8
     4.6.  Simulcast and scalable video coding . . . . . . . . . . .   8
     4.7.  Protocol extensions . . . . . . . . . . . . . . . . . . .   8
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .   9
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   9
   8.  Normative References  . . . . . . . . . . . . . . . . . . . .   9
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  10

1.  Introduction

   WebRTC intentionally 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

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   interoperability with other services or leverage the existence of
   tools that can communicate with them.

   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 detailed in webrtc-gateway document
   [I-D.draft-alvestrand-rtcweb-gateways].

   This document proposes a simple protocol for supporting WebRTC as
   ingest method which is:

   o  Easy to implement,

   o  As easy to use as current RTMP URIs.

   o  Fully compliant with Webrtc and RTCWEB specs.

   o  Allow for both ingest in traditional media platforms for extension
      and ingest in webrtc end-to-end platform for lowest possible
      latency.

   o  Lowers the requirements on both hardware encoders and broadcasting
      services to support webrtc.

   o  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].

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   o  WHIP client: WebRTC Media encoder or producer that acts as client
      on the WHIP protocol and encodes and delivers the media to a
      remote media server.

   o  WHIP endpoint: Ingest server receiving the initial WHIP request.

   o  WHIP endpoint URL: URL of the WHIP endpoint that will create the
      WHIP resource

   o  Media Server: WebRTC media server that establishes the media
      session with the WHIP client and receives the media produced by
      it.

   o  WHIP resource: Allocated resource by the WHIP endpoint for an
      ongoing ingest session that the WHIP client can send request for
      altering the session (ICE operations or termination, for example).

   o  WHIP resource URL: URL allocated to a specific media session by
      the WHIP endpoint which can be used to perform operations such
      terminating the session or ICE restarts.

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 to the 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 | | WHIP Resource  |
 +---------+-------+         +-------+- -----+ +------+-------+ +--------|-------+
           |                         |                |                  |
           |                         |                |                  |
           |HTTP POST (SDP Offer)    |                |                  |
           +------------------------>+                |                  |
           |201 Created (SDP answer) |                |                  |
           +<------------------------+                |                  |
           |          ICE REQUEST                     |                  |
           +----------------------------------------->+                  |
           |          ICE RESPONSE                    |                  |
           <------------------------------------------+                  |
           |          DTLS SETUP                      |                  |
           <==========================================>                  |
           |          RTP/RTCP FLOW                   |                  |
           +------------------------------------------>                  |
           | HTTP DELETE                                                 |
           +------------------------------------------------------------>+
           | 200 OK                                                      |
           <-------------------------------------------------------------x

                      WHIP session setup and teardown

4.  Protocol Operation

   In order to setup an ingestion session, the WHIP client will generate
   an SDP offer according to 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 endpoint will generate an
   SDP answer and return it on a 201 Accepted response with content type
   of application/sdp and the SDP answer as body and a Location header
   pointing to the newly created resource.

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

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

   To explicitly terminate the session, the WHIP client MUST perform an
   HTTP DELETE request to the resource url returned on the Location
   header of the initial HTTP POST.  Upon receiving the HTTP DELETE

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   request, the WHIP resource will be removed and the resources freed on
   the media server, terminating the ICE and DTLS sessions.

   A media server terminating a session MUST follow the procedures in
   [RFC7675] section 5.2 for immediate revocation of consent.

   The WHIP endpoints MUST return an HTTP 405 response for any HTTP GET,
   HEAD or PUT requests on the resource URL in order to reserve its
   usage for future versions of this protocol specification.

   The WHIP resources MUST return an HTTP 405 response for any HTTP GET,
   HEAD, POST or PUT requests on the resource URL in order to reserve
   its usage for future versions of this protocol specification.

4.1.  ICE and NAT support

   In order to simplify the protocol, there is no support for exchanging
   gathered trickle candidates from media server ICE candidates once the
   SDP answer is sent.  So in order to support the WHIP client behind
   NAT, the WHIP media server SHOULD be publicly accessible.

   The initial offer by the WHIP client 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 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 WHIP client MUST implement full ICE.

   The WHIP client MAY perform trickle ICE or an ICE restarts [RFC8863]
   by sending a HTTP PATCH request to the WHIP resource URL with a body
   containing a SDP fragment with mime type "application/trickle-ice-
   sdpfrag" as specified in [RFC8840] with the new ice candidate or ice
   ufrag/pwd for ice restarts.  A WHIP resource MAY not support either
   trickle ICE (i.e.  ICE lite media servers) or ICE restart, and it
   MUST return a 405 Method Not Allowed for any HTTP PATCH request.

   A WHIP client receiving a 405 response for an HTTP PATCH request
   SHALL not send further request for ICE trickle or restart.  If the
   WHIP client gathers additional candidates (via STUN/TURN) after the
   SDP offer is sent, it MUST send STUN request to the ICE candidates
   received from the media server as per [RFC8838] regardless if the
   HTTP PATCH is supported by either the WHIP client or the WHIP
   resource.

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4.2.  Webrtc constraints

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

   SDP bundle SHALL be used by both the WHIP client and the media
   server.  The SDP offer created by the WHIP client MUST include the
   bundle-only attribute in all m-lines as per [RFC8843].  Also, RTCP
   muxing SHALL be supported by both the WHIP client and the media
   server.

   Unlike [RFC5763] a WHIP client MAY use a setup attribute value of
   setup:active in the SDP offer, in which case the WHIP endpoint MUST
   use a setup attribute value of setup:passive in the SDP answer.

4.3.  Load balancing and redirections

   WHIP endpoints and media servers MAY not be colocated on the same
   server so it is possible to load balance incoming requests to
   different media servers.  WHIP clients 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 WHIP endpoint 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.  STUN/TURN server configuration

   Configuration of the TURN or STUN servers used by the WHIP client is
   out of the scope of this document.

   It is RECOMMENDED that broadcasting server provides an HTTP interface
   for provisioning the TUNR/STUN servers url and short term credentiasl
   as in [I-D.draft-uberti-behave-turn-rest-00].  Note that the
   authentication information or the url of this API are not related to
   the WHIP enpoint URLs or authentication.

   It could also be possilble to configure the STUN/TURN server URLS and
   long term credentials provided by the either broadcasting service or
   an external TURN provider.

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4.5.  Authentication and authorization

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

4.6.  Simulcast and scalable video coding

   Both simulcast and scalable video coding (including K-SVC modes) MAY
   be supported by both media servers and WHIP clients and negotiated in
   the SDP O/A.

   If the client supports simulcast and wants to enable it for
   publishing, it MUST negotiate the support in the SDP offer according
   to the procedures in [RFC8853] section 5.3.  A server accepting a
   simulcast offer MUST create an answer accoding to the procedures
   [RFC8853] section 5.3.2.

4.7.  Protocol extensions

   In order to support future extensions to be defined for the WHIP
   protocol, a common procedure for registering and announcing the new
   extensions is defined.

   Protocol extensions supported by the WHIP server MUST be advertised
   to the WHIP client on the 201 created response to initial HTTP POST
   request to the WHIP enpoint by inserting one Link header for each
   extension with the extension "rel" type attribute and the uri for the
   HTTP resource that will be available for receiving request related to
   that extension.

   Protocol extensions are optionasl for bot WHIP clients and servers.
   WHIP clients MUST ignore any Link attribute with an unknown "rel"
   attribute value and WHIP servers MUST not require the usage of any of
   the extensions.

   Each protocol extension MUST register an unique "rel" attribute
   values at IANA starting with the prefix: "urn:ietf:params:whip:".

   For example, taking a potential extension of server to client
   communication using server sent events as specified in
   https://html.spec.whatwg.org/multipage/server-sent-
   events.html#server-sent-events, the url for connecting to the server
   side event resource for the published stream will be returned in the
   initial HTTP "201 Created" response with a "Link" header an a "rel"
   attribute of "urn:ietf:params:whip:server-sent-events".

   The HTTP 201 response to the HTTP POST request would look like:

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   "HTTP/1.1 201 Created Content-Type: application/sdp Location:
   https://whip.ietf.org/publications/213786HF Link:
   <https://whip.ietf.org/publications/213786HF/
   sse>;rel="urn:ietf:params:whip:server-side-events " "

5.  Security Considerations

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

6.  IANA Considerations

7.  Acknowledgements

8.  Normative References

   [I-D.draft-alvestrand-rtcweb-gateways]
              Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways",
              draft-alvestrand-rtcweb-gateways-02 (work in progress),
              March 2015.

   [I-D.draft-uberti-behave-turn-rest-00]
              Uberti, J., "A REST API For Access To TURN Services",
              draft-uberti-behave-turn-rest-00 (work in progress), July
              2013.

   [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>.

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
              2010, <https://www.rfc-editor.org/info/rfc5763>.

   [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>.

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   [RFC8838]  Ivov, E., Uberti, J., and P. Saint-Andre, "Trickle ICE:
              Incremental Provisioning of Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol", RFC 8838,
              DOI 10.17487/RFC8838, January 2021,
              <https://www.rfc-editor.org/info/rfc8838>.

   [RFC8840]  Ivov, E., Stach, T., Marocco, E., and C. Holmberg, "A
              Session Initiation Protocol (SIP) Usage for Incremental
              Provisioning of Candidates for the Interactive
              Connectivity Establishment (Trickle ICE)", RFC 8840,
              DOI 10.17487/RFC8840, January 2021,
              <https://www.rfc-editor.org/info/rfc8840>.

   [RFC8843]  Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8843,
              DOI 10.17487/RFC8843, January 2021,
              <https://www.rfc-editor.org/info/rfc8843>.

   [RFC8853]  Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
              "Using Simulcast in Session Description Protocol (SDP) and
              RTP Sessions", RFC 8853, DOI 10.17487/RFC8853, January
              2021, <https://www.rfc-editor.org/info/rfc8853>.

   [RFC8863]  Holmberg, C. and J. Uberti, "Interactive Connectivity
              Establishment Patiently Awaiting Connectivity (ICE PAC)",
              RFC 8863, DOI 10.17487/RFC8863, January 2021,
              <https://www.rfc-editor.org/info/rfc8863>.

Authors' Addresses

   Sergio Garcia Murillo
   CoSMo Software

   Email: sergio.garcia.murillo@cosmosoftware.io

   Alexandre Gouaillard
   CoSMo Software

   Email: alex.gouaillard@cosmosoftware.io

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