WebRTC-HTTP ingestion protocol (WHIP)
draft-ietf-wish-whip-02
The information below is for an old version of the document.
| Document | Type |
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|---|---|---|---|
| Authors | Sergio Garcia Murillo , Dr. Alex Gouaillard | ||
| Last updated | 2022-03-07 | ||
| Replaces | draft-murillo-whip | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
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draft-ietf-wish-whip-02
wish S. Murillo
Internet-Draft A. Gouaillard
Intended status: Standards Track CoSMo Software
Expires: 8 September 2022 7 March 2022
WebRTC-HTTP ingestion protocol (WHIP)
draft-ietf-wish-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 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
(which 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 services 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
and may be updated, replaced, or obsoleted by other documents at any
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 8 September 2022.
Copyright Notice
Copyright (c) 2022 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 Revised BSD License text as
described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Revised BSD License.
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 . . . . . . . . . . . . . . . . . . . 9
4.3. Load balancing and redirections . . . . . . . . . . . . . 9
4.4. STUN/TURN server configuration . . . . . . . . . . . . . 9
4.5. Authentication and authorization . . . . . . . . . . . . 10
4.6. Simulcast and scalable video coding . . . . . . . . . . . 11
4.7. Protocol extensions . . . . . . . . . . . . . . . . . . . 11
5. Security Considerations . . . . . . . . . . . . . . . . . . . 12
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
6.1. Link Relation Type: ice-server . . . . . . . . . . . . . 12
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 12
8. Normative References . . . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14
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1. Introduction
RTCWEB standardized JSEP ([RFC8829]), a mechanism 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.
As the HTTP PATCH request sent by a WHIP client may be received out
of order by the WHIP resource, the WHIP resource MUST generate a
unique strong entity-tag identifying the ICE session as per [RFC7232]
section 2.3. The initial value of the entity-tag identifying the
initial ICE session MUST be returned in an ETag header in the 201
response to the initial POST request to the WHIP endpoint and in the
200 OK of a PATCH request that triggers an ICE restart.
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POST /whip/endpoint HTTP/1.1
Host: whip.example.com
Content-Type: application/sdp
<SDP Offer>
HTTP/1.1 201 Created
ETag: "38sdf4fdsf54:EsAw"
Content-Type: application/sdp
Location: https://whip.example.org/resource/id
<SDP answer>
A WHIP client sending a PATCH request for performing trickle ICE MUST
contain an If-Match header with the latest known entity-tag as per
[RFC7232] section 3.1. When the PATCH request is received by the
WHIP resource, it MUST compare the entity-tag value requested with
the current entinty-tag of the resource as per [RFC7232] section 3.1
and return a 412 Precondition Failed response if they do not match.
Entity-tag validation MUST only be used for HTTP requests requiring
to match a known ICE session and SHOULD NOT be used otherwise, for
example in the HTTP DELETE request to terminate the session.
A WHIP resource receiving a PATCH request with new ICE candidates,
but which does not perform an ICE restart, MUST return a 204 No
content response without body. If the media server does not support
a candidate transport or is not able to resolve the connection
address it MUST accept the HTTP request with the 204 response and
silently discard the candidate.
PATCH /resource/id HTTP/1.1
Host: whip.example.com
If-Match: "38sdf4fdsf54:EsAw"
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
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Figure 2: Trickle ICE request
A WHIP client sending a PATCH request for performing ICE restart MUST
contain an If-Match header with a field-value "*" as per [RFC7232]
section 3.1.
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, optionally, the new set of ICE candidates for the media server
and the new entity-tag correspond to the new ICE session in an ETag
response header.
If the ICE request can not be performed by the WHIP resource it MUST
return an appropriate HTTP error code but MUST NOT terminate the
session immediately. The WHIP client COULD try again to perform a
new ICE restart or terminate the session issuing a HTTP DELETE
request instead. In any case the session MUST be terminated if the
ICE consent expires as a consequence of the failed ICE restart.
PATCH /resource/id HTTP/1.1
Host: whip.example.com
If-Match: "*"
Content-Type: application/trickle-ice-sdpfrag
Content-Length: 54
a=ice-ufrag:ysXw
a=ice-pwd:vw5LmwG4y/e6dPP/zAP9Gp5k
HTTP/1.1 200 OK
ETag: "289b31b754eaa438:ysXw"
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
Given that in order to send new ICE candidates to the WHIP resource,
the WHIP client needs to know the entity-tag associated to the ICE
session, it MUST buffer any gathered candidates before the HTTP
response to the initial PUT request or the PATCH request with the new
entity-tag value is received. Once the entity-tag value is known the
WHIP client SHOULD send a single aggregated HTTP PATCH request with
all the ICE candidates it has buffered so far.
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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 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.
Each ICE server will be returned on a Link header with a "rel"
attribute 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:
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* username: If the Link header represents a TURN server, and
credential-type is "password", then this attribute specifies the
username to use with that TURN server.
* credential: If credential-type attribute is missing or has a
"password" value, the credential attribute represents a long-term
authentication password, as described in [RFC8489], Section 10.2.
* credential-type: If the Link header represents a TURN server, then
this attribute specifies how the credential attribute value 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; rel="ice-server";
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 implement
this authentication and authorization mechanism and send the HTTP
Authorization header in all HTTP requests sent to either the WHIP
endpoint or resource.
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 of 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.
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WHIP endpoints and resources COULD perform the authentication and
authorization by encoding an authentication token within 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 according 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 a 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 and 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.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: 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>.
[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>.
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[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>.
[RFC7232] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Conditional Requests", RFC 7232,
DOI 10.17487/RFC7232, June 2014,
<https://www.rfc-editor.org/info/rfc7232>.
[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>.
Murillo & Gouaillard Expires 8 September 2022 [Page 13]
Internet-Draft whip March 2022
[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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