Network Working Group S. Murillo
Internet-Draft A. Gouaillard
Intended status: Informational CoSMo Software
Expires: November 20, 2021 May 19, 2021
WebRTC-HTTP ingestion protocol (WHIP)
draft-murillo-whip-01
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.
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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 November 20, 2021.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Protocol Operation . . . . . . . . . . . . . . . . . . . . . 4
4.1. ICE and NAT support . . . . . . . . . . . . . . . . . . . 5
4.2. Webrtc constraints . . . . . . . . . . . . . . . . . . . 6
4.3. Load balancing and redirections . . . . . . . . . . . . . 6
4.4. Authentication and authorization . . . . . . . . . . . . 6
4.5. Simulcast and scalable video coding . . . . . . . . . . . 6
5. Security Considerations . . . . . . . . . . . . . . . . . . . 7
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
8. Normative References . . . . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8
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
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 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: - Easy to implement, - As easy to use as
current RTMP URIs. - Fully compliant with Webrtc and RTCWEB specs. -
Allow for both ingest in traditional media platforms for extension
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].
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 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).
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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.
+-----------------+ +---------------+ +--------------+ +----------------+
| 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
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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
request, the WHIP resource will be removed and the resources freed on
the media server, terminating the ICE and DTLS sessions.
The media server may terminate the session by using the Immediate
Revocation of Consent as defined in [RFC7675] section 5.2.
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
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HTTP PATCH is supported by either the WHIP client or the WHIP
resource.
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. Authentication and authorization
Authentication and authorization is supported by the Authorization
HTTP header with a bearer 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 WHIP clients.
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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.
[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>.
[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>.
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[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>.
[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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