WebRTC-HTTP ingestion protocol (WHIP)
draft-ietf-wish-whip-01

wish                                                          S. Murillo
Internet-Draft                                             A. Gouaillard
Intended status: Standards Track                          CoSMo Software
Expires: 23 April 2022                                   20 October 2021

                 WebRTC-HTTP ingestion protocol (WHIP)
                        draft-ietf-wish-whip-01

Abstract

   While WebRTC has been very sucessful 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 into a streaming service using WebRTC
   and so content providers still rely heavily on protocols like RTMP
   for it.

   These protocols are much older than WebRTC and by default lack some
   important security and resilience features provided by WebRTC with
   minimal overhead and additional latency.

   The media codecs used for ingestion in older protocols tend to be
   limited and not negotiated.  WebRTC includes support for negotiation
   of codecs, potentially alleviating transcoding on the ingest node
   (wich can introduce delay and degrade media quality).  Server side
   transcoding that has traditionally been done to present multiple
   renditions in Adaptive Bit Rate Streaming (ABR) implementations can
   be replaced with simulcasting and SVC codecs that are well supported
   by WebRTC clients.  In addition, WebRTC clients can adjust client-
   side encoding parameters based on RTCP feedback to maximize encoding
   quality.

   Encryption is mandatory in WebRTC, therefore secure transport of
   media is implicit.

   This document proposes a simple HTTP based protocol that will allow
   WebRTC based ingest of content into streaming servics and/or CDNs.

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/.

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   Internet-Drafts are draft documents valid for a maximum of six months
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   This Internet-Draft will expire on 23 April 2022.

Copyright Notice

   Copyright (c) 2021 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Overview  . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   4.  Protocol Operation  . . . . . . . . . . . . . . . . . . . . .   5
     4.1.  ICE and NAT support . . . . . . . . . . . . . . . . . . .   6
     4.2.  WebRTC constraints  . . . . . . . . . . . . . . . . . . .   8
     4.3.  Load balancing and redirections . . . . . . . . . . . . .   8
     4.4.  STUN/TURN server configuration  . . . . . . . . . . . . .   8
     4.5.  Authentication and authorization  . . . . . . . . . . . .   9
     4.6.  Simulcast and scalable video coding . . . . . . . . . . .  10
     4.7.  Protocol extensions . . . . . . . . . . . . . . . . . . .  10
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  11
     6.1.  Link Relation Type: urn:ietf:params:whip:ice-server . . .  11
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  11
   8.  Normative References  . . . . . . . . . . . . . . . . . . . .  11
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  13

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1.  Introduction

   RTCWEB standardized JSEP ([RFC8829]), a mechanishm used to control
   the setup, management, and teardown of a multimedia session, how to
   apply it using the SDP Offer/Answer model and all the formats for the
   data sent over the wire (media, codec, encryption, ...).  Also,
   WebRTC intentionally does not specify a signaling transport protocol
   at application level.  This flexibility has allowed the
   implementation of 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.

   In the broadcasting/streaming world, the usage of hardware encoders
   that make it very simple to plug in (SDI) cables carrying raw media,
   encode it in place, and push it to any streaming service or CDN
   ingest is already ubiquitous.  It is the adoption of a custom
   signaling transport protocol for each WebRTC service has hindered
   broader adoption as an ingestion protocol.

   While some standard signaling 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 may be
   the closest in terms of features to WebRTC, is not compatible with
   the WebRTC SDP offer/answer model.

   In the specific case of media ingestion into a streaming service,
   some assumptions 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
   media ingestion method which is:

   *  Easy to implement,

   *  As easy to use as current RTMP URIs.

   *  Fully compliant with WebRTC and RTCWEB specs.

   *  Allows for both ingest in traditional media platforms and ingest
      in WebRTC end-to-end platforms with the 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.

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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].

   *  WHIP client: WebRTC media encoder or producer that acts as a
      client of the WHIP protocol by encoding and delivering the media
      to a remote media server.

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

   *  WHIP endpoint URL: URL of the WHIP endpoint that will create the
      WHIP resource.

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

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

   *  WHIP resource URL: URL allocated to a specific media session by
      the WHIP endpoint which can be used to perform operations such as
      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 (WHIP client) and the
   broadcasting ingestion endpoint (media server).

   Once the ICE/DTLS session is set up, the media will flow
   unidirectionally from the encoder/media producer (WHIP client) to the
   broadcasting ingestion endpoint (media server).  In order to reduce
   complexity, no SDP renegotiation is supported, so no tracks or
   streams can be added or removed once the initial SDP offer/answer
   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

              Figure 1: 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 the body.  The WHIP endpoint will generate
   an SDP answer and return a 201 Created response with a content type
   of application/sdp and the SDP answer as the body and a Location
   header pointing to the newly created resource.

   The 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 in the Location
   header of the initial HTTP POST.  Upon receiving the HTTP DELETE
   request, the WHIP resource will be removed and the resources freed on
   the media server, terminating the ICE and DTLS sessions.

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   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

   The initial offer by the WHIP client MAY be sent after the full ICE
   gathering is complete with the full list of ICE candidates, or only
   contain local candidates or even an empty list of candidates.

   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.  The WHIP Endpoint SHALL gather all the ICE
   candidates for the media server before responding to the client
   request and the SDP answer SHALL contain the full list of ICE
   candidates 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 trickle
   ICE (i.e.  ICE lite media servers) or ICE restart, in that case, it
   MUST return a 405 Method Not Allowed response for any HTTP PATCH
   request.

   A WHIP resource receving a PATH request with new ICE candidates, but
   which does not perform an ICE restart, MUST return a 204 No content
   response without body.

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PATCH /resource/id HTTP/1.1
Host: whip.example.com
Content-Type: application/trickle-ice-sdpfrag
Content-Length: 548

a=ice-ufrag:EsAw
a=ice-pwd:P2uYro0UCOQ4zxjKXaWCBui1
m=audio RTP/AVP 0
a=mid:0
a=candidate:1387637174 1 udp 2122260223 192.0.2.1 61764 typ host generation 0 ufrag EsAw network-id 1
a=candidate:3471623853 1 udp 2122194687 198.51.100.1 61765 typ host generation 0 ufrag EsAw network-id 2
a=candidate:473322822 1 tcp 1518280447 192.0.2.1 9 typ host tcptype active generation 0 ufrag EsAw network-id 1
a=candidate:2154773085 1 tcp 1518214911 198.51.100.2 9 typ host tcptype active generation 0 ufrag EsAw network-id 2
a=end-of-candidates

HTTP/1.1 204 No Content

                    Figure 2: Trickle ICE request

   If the HTTP PATCH request results in an ICE restart, the WHIP
   resource SHALL return a 200 OK with an "application/trickle-ice-
   sdpfrag" body containing the new ICE username fragment and password
   and, optionaly, the new set of ICE candidates for the media server.

   PATCH /resource/id HTTP/1.1
   Host: whip.example.com
   Content-Type: application/trickle-ice-sdpfrag
   Content-Length: 54

   a=ice-ufrag:ysXw
   a=ice-pwd:vw5LmwG4y/e6dPP/zAP9Gp5k

   HTTP/1.1 200 OK
   Content-Type: application/trickle-ice-sdpfrag
   Content-Length: 102

   a=ice-lite
   a=ice-ufrag:289b31b754eaa438
   a=ice-pwd:0b66f472495ef0ccac7bda653ab6be49ea13114472a5d10a

                       Figure 3: ICE restart request

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   As the HTTP PATCH request sent by a WHIP client may be received out
   of order by the WHIP resource, the WHIP resource SHOULD keep track of
   the previous values of the ICE username fragment and client used by
   the WHIP client.  If an HTTP PATCH request is received with a
   previously used ICE username fragment and password by the client, the
   WHIP endpoint SHALL NOT perform and ICE restart but reject the
   request with a 409 Conflict response instead.

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 the 307 Temporary Redirect response code in the initial HTTP
   response to the WHIP endpoint URL.  The WHIP resource URL MUST be a
   final one, and redirections are not required to be supported for the
   PATCH and DELETE request sent to it.

   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

   The WHIP endpoint MAY return ICE server configuration urls and
   credentials usable by the client in the 201 Created response to the
   HTTP POST request to the WHIP endpoint url.

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   Each ICE server will be returned on a Link header with a "rel"
   attribribute value of "ice-server" where the Link target URI is the
   ICE server URL and the credentials are encoded in the Link target
   attributes as follows:

   *  username: If this the Link header represents a TURN server, and
      creadential-type is "password", then this attribute specifies the
      username to use with that TURN server.

   *  credential: If credentialType is "password", credential represents
      a long-term authentication password, as described in [RFC8489],
      Section 10.2.

   *  creadential-type: If this RTCIceServer object represents a TURN
      server, then this attribute specifies how credential should be
      used when that TURN server requests authorization.  The default
      value if the attribute is not present is "password".

     Link: stun:stun.example.net;
     Link: turn:turn.example.net?transport=udp; rel="ice-server"; username="user"; credential: "myPassword"; credential-type: "password";
     Link: turn:turn.example.net?transport=tcp; rel="ice-server"; username="user"; credential: "myPassword"; credential-type: "password";
     Link: turns:turn.example.net?transport=tcp; rel="ice-server"; username="user"; credential: "myPassword"; credential-type: "password";

              Figure 4: Example ICE server configuration

   There are some webrtc implementations that do not support updating
   the ICE server configuration after the local offer has been created.
   In order to support these clients, the WHIP endpoint MAY also include
   the ICE server configuration on the responses to an authenticated
   OPTIONS request sent to the WHIP endpoint URL sent before the POST
   requests.

   It COULD be also possible to configure the STUN/TURN server URLs with
   long term credentials provided by either the broadcasting service or
   an external TURN provider on the WHIP client overriding the values
   provided by the WHIP endpoint.

4.5.  Authentication and authorization

   WHIP endpoints and resources MAY require the HTTP request to be
   authenticated using an HTTP Authorization header with a Bearer token
   as specified in [RFC6750] section 2.1.  WHIP clients MUST implemenent
   this authentication and authorization mechanism and send the HTTP
   Authorization header in all HTTP request sent to either the WHIP
   endpoint or resource.

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   The nature, syntax and semantics of the bearer token as well as how
   to distribute it to the client is outside the scope of this document.
   Some examples ot the kind of tokens that could be used are, but are
   not limited to, JWT tokens as per [RFC6750] and [RFC8725] or a shared
   secret stored on a database.  The tokens are typically made available
   to the end user alongside the WHIP endpoint url and configured on the
   WHIP clients.

   WHIP endpoints and resources COULD perform the authentication and
   authorization by encoding an authentication token withing the urls
   for the WHIP endpoints or resources instead.  In case the WHIP client
   is not configured to use a bearer token the HTTP Authorization header
   must not be sent in any request.

4.6.  Simulcast and scalable video coding

   Both simulcast and scalable video coding (including K-SVC modes) MAY
   be supported by both the media servers and WHIP clients through
   negotiation in the SDP offer/answer.

   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 the initial HTTP
   POST request sent to the WHIP endpoint.  The WHIP endpoint MUST
   return 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 requests related to that extension.

   Protocol extensions are optional for both 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:".

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   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 and a "rel"
   attribute of "urn:ietf:params:whip:server-sent-events".

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

HTTP/1.1 201 Created
Content-Type: application/sdp
Location: https://whip.example.org/resource/id
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

   The link relation types below have been registered by IANA per
   Section 4.2 of [RFC8288].

6.1.  Link Relation Type: urn:ietf:params:whip:ice-server

   Relation Name: ice-server Description: Describe the STUN and TURN
   servers that can be used by the ICE Agent to establish a connection
   with a peer.  Reference: TBD

7.  Acknowledgements

8.  Normative References

   [I-D.draft-alvestrand-rtcweb-gateways]
              Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways",
              Work in Progress, Internet-Draft, draft-alvestrand-rtcweb-
              gateways-02, 9 March 2015,
              <https://www.ietf.org/archive/id/draft-alvestrand-rtcweb-
              gateways-02.txt>.

   [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>.

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   [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>.

   [RFC8288]  Nottingham, M., "Web Linking", RFC 8288,
              DOI 10.17487/RFC8288, October 2017,
              <https://www.rfc-editor.org/info/rfc8288>.

   [RFC8489]  Petit-Huguenin, M., Salgueiro, G., Rosenberg, J., Wing,
              D., Mahy, R., and P. Matthews, "Session Traversal
              Utilities for NAT (STUN)", RFC 8489, DOI 10.17487/RFC8489,
              February 2020, <https://www.rfc-editor.org/info/rfc8489>.

   [RFC8725]  Sheffer, Y., Hardt, D., and M. Jones, "JSON Web Token Best
              Current Practices", BCP 225, RFC 8725,
              DOI 10.17487/RFC8725, February 2020,
              <https://www.rfc-editor.org/info/rfc8725>.

   [RFC8829]  Uberti, J., Jennings, C., and E. Rescorla, Ed.,
              "JavaScript Session Establishment Protocol (JSEP)",
              RFC 8829, DOI 10.17487/RFC8829, January 2021,
              <https://www.rfc-editor.org/info/rfc8829>.

   [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>.

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   [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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