Internet DRAFT - draft-murillo-avtcore-multi-codec-payload-format
draft-murillo-avtcore-multi-codec-payload-format
AVTCORE S. Garcia Murillo
Internet-Draft CoSMo
Intended status: Standards Track Y. Fablet
Expires: 12 January 2022 Apple Inc.
A. Gouaillard
CoSMo
J. Uberti
Clubhouse
11 July 2021
Multi Codec RTP payload format
draft-murillo-avtcore-multi-codec-payload-format-01
Abstract
RTP Media Chains usually rely on piping encoder output directly to
packetizers. Media packetization formats often support a specific
codec format and optimize RTP packets generation accordingly. With
the development of Selective Forward Unit (SFU) solutions, RTP Media
Chains used in WebRTC solutions are increasingly relying on
application-specific transforms that sit between encoder and
packetizer on one end and between depacketizer and decoder on the
other end. These transforms are typically encrypting media content
so that the media content is not readable from the SFU, for instance
using [SFrame] or [WebRTCInsertableStreams]. In that context, RTP
packetizers can no longer expect to use packetization formats that
mandate media content to be in a specific codec format. This
document provides a solution to that problem by describing a RTP
packetization format that can be used for many media content, and how
to negotiate use of this format. This document also describes a
solution to allow SFUs to continue performing packet routing on top
of this RTP packetization format.
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
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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
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
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This Internet-Draft will expire on 12 January 2022.
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
Provisions Relating to IETF Documents (https://trustee.ietf.org/
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Please review these documents carefully, as they describe your rights
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provided without warranty as described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. RTP Packetization . . . . . . . . . . . . . . . . . . . . . . 6
4. Payload Multiplexing . . . . . . . . . . . . . . . . . . . . 7
5. SDP Negotiation . . . . . . . . . . . . . . . . . . . . . . . 8
6. SFU Packet Selection . . . . . . . . . . . . . . . . . . . . 9
7. Sender Processing Rules . . . . . . . . . . . . . . . . . . . 10
8. Redundancy Techniques Considerations . . . . . . . . . . . . 10
8.1. Retransmission Techniques . . . . . . . . . . . . . . . . 10
8.2. Forward Error Correction (FEC) Techniques . . . . . . . . 11
8.3. Redundant Audio Data Techniques . . . . . . . . . . . . . 11
9. Alternatives . . . . . . . . . . . . . . . . . . . . . . . . 11
9.1. Generic Packetization With In-Payload APT . . . . . . . . 12
9.2. A Payload Type for Generic Packetization AND Media
Format . . . . . . . . . . . . . . . . . . . . . . . . . 12
9.3. A RTP Header To Choose Packetization . . . . . . . . . . 13
10. Security Considerations . . . . . . . . . . . . . . . . . . . 14
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
11.1. Registration of audio/generic . . . . . . . . . . . . . 14
12. Registration of video/generic . . . . . . . . . . . . . . . . 15
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 15
13.1. Normative References . . . . . . . . . . . . . . . . . . 15
13.2. Informative References . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
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1. Introduction
As per Figure 1 of [RFC7656], a Media Packetizer transforms a single
Encoded Stream into one or several RTP packets. The Encoded Stream
is coming straight from the Media Encoder and is expected to follow
the format produced by the Media Encoder. A number of Media
Packetizer formats have been designed to process a specific format
produced by Media Encoder. For instance [RFC6184] is dedicated to
the processing of content produced by H.264 Media Encoders, and
generates packets following NALUs organization.
WebRTC applications are increasingly deploying end-to-end encryption
solutions on top of RTP Media Chains. End-to-end encryption is
implemented by inserting application-specific Media Transformers
between Media Encoder and Media Packetizer on the sending side, and
between Media Depacketizer and Media Decoder on the receiving side,
as described in Figure 1 and Figure 2. To support end-to-end
encryption, Media Transformers can use the [SFrame] format. In
browsers, Media Transformers are implemented using
[WebRTCInsertableStreams], for instance by injecting JavaScript code
provided by web pages.
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Physical Stimulus
|
V
+----------------------+
| Media Capture |
+----------------------+
|
Raw Stream
V
+----------------------+
| Media Source |<-- Synchronization Timing
+----------------------+
|
Source Stream
V
+----------------------+
| Media Encoder |
+----------------------+
|
Encoded Stream
V
+----------------------+
| Media Transformer |<-- NEW: application-specific transform
+----------------------+ (e.g. SFrame Encryption)
|
Transformed Stream +------------+
V | V
+----------------------+ | +----------------------+
| Media Packetizer | | | RTP-Based Redundancy |
+----------------------+ | +----------------------+
| | |
+-------------+ Redundancy RTP Stream
Source RTP Stream |
V V
+----------------------+ +----------------------+
| RTP-Based Security | | RTP-Based Security |
+----------------------+ +----------------------+
| |
Secured RTP Stream Secured Redundancy RTP Stream
V V
+----------------------+ +----------------------+
| Media Transport | | Media Transport |
+----------------------+ +----------------------+
Figure 1: Sender side concepts in the Media Chain with
application-level Media Transform
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These RTP packets are sent over the wire to a receiver media chain
matching the sender side, reaching the Media Depacketizer that will
reconstruct the Encoded Stream before passing it to the Media
Decoder.
+----------------------+ +----------------------+
| Media Transport | | Media Transport |
+----------------------+ +----------------------+
Received | Received | Secured
Secured RTP Stream Redundancy RTP Stream
V V
+----------------------+ +----------------------+
| RTP-Based Validation | | RTP-Based Validation |
+----------------------+ +----------------------+
| |
Received RTP Stream Received Redundancy RTP Stream
| |
| +--------------------+
V V
+----------------------+
| RTP-Based Repair |
+----------------------+
|
Repaired RTP Stream
V
+----------------------+
| Media Depacketizer |
+----------------------+
|
Received Transformed Stream
V
+----------------------+
| Media Transformer |<-- NEW: application-specific transform
+----------------------+ (e.g. SFrame Decryption)
|
Received Encoded Stream
V
+----------------------+
| Media Decoder |
+----------------------+
|
Received Source Stream
V
+----------------------+
| Media Sink |--> Synchronization Information
+----------------------+
|
Received Raw Stream
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V
+----------------------+
| Media Render |
+----------------------+
|
V
Physical Stimulus
Figure 2: Receiver side concepts in the Media Chain with
application-level Media Transform
This packetization does not change how the mapping between one or
several encoded or dependant streams are mapped to the RTP streams or
how the synchronization sources(s) (SSRC) are assigned.
Given the use of post-encoder application-specific transforms, the
whole Media Chain needs to be made aware of it. This includes the
sender post-transform Media Chain, Media Transport intermediaries
(SFUs typically) and receiver pre-transform Media Chain.
As these transforms can alter Encoded Streams in any possible way,
the use of codec-specific Media Packetizers like [RFC6184] on
Transformed Stream may be suboptimal on sender side. It may also be
problematic on the receiving side in case codec-specific processing
is done prior the Media Transformer. Media Transport intermediaries
are often looking at the Media Content itself to fuel their packet
selection algorithms.
2. Goals
The objective of this document is to support inserting any
application-specific transform between encoders and packetizers in
the Media Chain. For that purpose, this document will: 1. Provide a
packetization format that supports multiple media content used by
WebRTC applications (audio compressed by Opus, video compressed by
H264 or VP8, encrypted content...) that allows reuse of existing RTP
mechanisms in place in WebRTC applications such as RTX, RED or FEC.
2. Provide a way to negotiate use of this packetization format
between sender and receiver, with minimum impact on existing
negotiation approaches. 3. Provide a side-channel information so
that network intermediaries (SFU in particular) can do their existing
packet routing strategies without inspecting the media content.
3. RTP Packetization
This packetizer, by design, is not expected to understand the format
of the media to transmit. The unit used by the packetizer to do
processing is called a frame in the remainder of the document.
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It is the responsibility of the application using the packetizer to
group media content in meaningful frames. In the common case of a
video codec, the packetizer frame is the frame in byte format (h264
annex b for example) generated by the encoder.
If the application wants to transform encoded content, the
application needs to split the encoded content into frames prior the
transform. Each frame is then transformed independently, for
instance encrypted using [SFrame]. The content of each transformed
frame is then processed by the packetizer.
In the case of a video codec supporting spatial scalability, each
spatial layer MUST be split in its own frame by the application
before passing it to the packetizer.
When the packetizer receives a frame from the application, it MUST
fragment the frame content in multiple RTP packets to ensure packets
do not exceed the network maximum transmission unit. The content of
the frame will be treated as a binary blob by the packetizer, so the
decision about the boundaries of each fragment is decided arbitrarily
by the packetizer. The packetizer or any relying server MUST NOT
modify the frame content and concatenating the RTP payload of the RTP
packets for each frame MUST produce the exact binary content of the
input frame content.
The marker bit of each RTP packet in a frame MUST be set according to
the audio and video profiles specified in [RFC3551].
The spatial layer frames are sent in ascending order, with the same
RTP timestamp, and only the last RTP packet of the last spatial layer
frame will have the marker bit set to 1.
4. Payload Multiplexing
In order to reduce the number of payload type in the SDP exchange, a
single payload type code for this multi-codec packetization can be
used for all negotiated media formats that the multi-codec
packetization supports. That requires to identify the original
payload type code of the frame negotiated media format, called the
associated payload type (APT) hereunder. The APT value is the
payload type code of the associated format passed to the multi-codec
Media Packetizer before any transformation is applied.
The APT value is sent in a dedicated header extension. The payload
of this header extension can be encoded using either the one-byte or
two-byte header defined in [RFC5285]. Figures 3 and 4 show examples
with each one of these examples.
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0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | len=0 |S| APT |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: Frame associated payload type encoding using the One-
Byte header format
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | len=1 |S| APT | 0 (pad) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: Frame associated payload type encoding using the Two-
Byte header format
The APT value is the associated payload type value. The S bit
indicates if the media stream can be forwarded safely starting from
this RTP packet. Typically, it will be set to 1 on the first RTP
packet of an intra video frame and in all RTP audio packets.
Receivers MUST be ready to receive RTP packets with different
associated payload types in the same way they would receive different
payload type codes on the RTP packets.
The URI for declaring this header extension in an extmap attribute is
"urn:ietf:params:rtp-hdrext:associated-payload-type".
5. SDP Negotiation
To use the multi-codec packetization, the SDP Offer/Answer exchange
MUST negotiate: - The payload type of the negotiated codec format -
The multi-codec payload type - The associated payload type header
extension
Only the negotiated payload types are allowed to be used as
associated payload types. Figure 5 illustrates a SDP that negotiates
exchange of video using either VP8 or VP9 codecs with the possibility
to use the multi-codec packetization. In this example, RTX is also
negotiated and will be applied normally on each associated payload
type.
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m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=setup:actpass
a=mid:1
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=extmap:4 urn:ietf:params:rtp-hdrext:associated-payload-type
a=sendrecv
a=rtpmap:96 vp9/90000
a=rtpmap:97 vp8/90000
a=rtpmap:98 generic/90000
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=96
a=rtpmap:100 rtx/90000
a=fmtp:100 apt=97
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=98
Figure 5: SDP example negotiating the multi-codec payload type
and related header extension for video
6. SFU Packet Selection
SFUs need to have a basic understanding of each frame they receive so
they can decide to forward it or not and to which endpoint. They
might need similar information to support media content recording.
This information is either generic to a group of frames (called a
stream hereafter) or specific to each frame.
The information is transmitted as a RTP header extension as the RTP
packet payload should be treated as opaque by the SFU. This is
especially necessary if the payload is end-to-end encrypted. The
amount of information should be limited to what is strictly necessary
to the SFU task since it is not always as trusted as individual
peers.
For audio, configuration information such as Opus TOC might be
useful. For video, configuration information might include: - Stream
configuration information: resolution, quality, frame rate... - Codec
specific configuration information: codec profile like profile_idc...
- Frame specific information: whether the stream is decodable when
starting from this frame, whether the frame is skippable...
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For video content, this information is sent using a Dependency
Descriptor header extension. In that case, the first RTP packet of
the frame will have its start_of_frame equal to 1 and the last packet
will have its end_of_frame equal to 1.
7. Sender Processing Rules
The sender identifies the use of the multi-codec payload format by
using the urn:ietf:params:rtp-hdrext:associated-payload-type
extension. When doing so, the sender follows these additional rules:
- For audio content, the associated payload type MUST reference an
audio codec in the supported audio codec list. The supported audio
codec list contains the audio codecs enumerated in [RFC7874]. This
list may be extended in future versions of this specification. - For
video content, H.264 and VP8 are supported as described in [RFC7742],
as well as VP9 and AV.1. In the case scalable video coding is used,
the sender MUST generate a Dependency Descriptor header extension.
This requires the associated payload type to reference a video codec
that can be described using the Dependency Descriptor header
extension. This also requires the sender to split the video encoder
output in frames that can each be described using the Dependency
Descriptor header extension.
These rules apply to both the originator of the content as well as
SFUs that might route the content to end receivers.
8. Redundancy Techniques Considerations
The solution described in this document is expected to integrate well
with the existing RTP ecosystem. This section describes how the
multi-codec packetizer can be used jointly with existing techniques
that allow to mitigate unreliable transports.
8.1. Retransmission Techniques
[RFC4588] defines a retransmission payload format (RTX) that can be
used in case of packet loss. As defined in [RFC4588], RTX is able to
handle any payload format, including the format described in this
document. Given RTX preserves both RTP packet payload and headers,
the receiver will be able to identify the payload type of the
recovered packet and whether multi-codec packetization is used. RTX
will also allow recovering RTP header extensions that convey
information on the media content itself.
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8.2. Forward Error Correction (FEC) Techniques
FEC is another technique used in RTP Media Chains to protect media
content against packet loss. [RFC5109] defines such a payload format
used to transmit FEC for specific packets protection.
FEC may protect some parts of the media content more than others.
For instance, intra video frame encoded data or important network
abstraction layer units (NALUs) like SPS/PPS may be more protected.
With a post-encoder transform and the use of a multi-codec
packetization, the granularity of the recovery mechanism is no longer
at the NALU level but at the level of the frame generated by the
post-encoder transform. In case a SVC codec is used, each spatial
layer will be processed as an independent frame. In that case, base
layers can be protected more heavily than higher resolution layers.
8.3. Redundant Audio Data Techniques
As defined in [RFC7656] RTP-based redundancy is defined here as a
transformation that generates redundant or repair packets sent out as
a Redundancy RTP Stream to mitigate Network Transport impairments,
like packet loss and delay.
[RFC2198] defines a payload format for sending the same audio data
encoded multiple times at different quality levels. This allows to
use a lower quality encoding of the audio data, should the higher
quality encoding of the audio data is lost during the transmission.
If a Media Transformation is in use, both the primary and redundant
encoding must be transformed independently and the redundant packet
created normally. As the RTP headers present in the redundant packet
are only applicable to the primary encoding, if the payload type for
a redundant encoding block is mapped to the multi-codec packetizer,
the value of the associated payload type for the primary encoding is
applied to the redundant encoding block as well.
9. Alternatives
Various alternatives can be used to implement and negotiate multi-
codec packetization. This section describes a few additional
alternatives. This section is to be removed before finalization of
the document.
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9.1. Generic Packetization With In-Payload APT
Instead of using a RTP header extension to convey the APT value, it
is prepended in the RTP payload itself. As the value cannot change
for a whole frame, its value is prepended to the first packet
generated of the frame only. This removes the need to negotiate a
dedicated header extension, but may require the SFU to update the
payload when sending or recording content.
9.2. A Payload Type for Generic Packetization AND Media Format
The payload type is negotiated in the SDP so as to identify both the
negotiated codec format and the multi-codec packetization use. There
is no network cost but this increases the number of payload types
used in the SDP.
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=setup:actpass
a=mid:1
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=rtpmap:96 vp9/90000
a=rtpmap:97 generic/90000
a=fmtp:97 apt=96
a=rtpmap:98 vp8/90000
a=rtpmap:99 generic/90000
a=fmtp:99 apt=98
a=rtpmap:100 rtx/90000
a=fmtp:100 apt=96
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=97
a=rtpmap:102 rtx/90000
a=fmtp:102 apt=98
a=rtpmap:103 rtx/90000
a=fmtp:103 apt=99
Figure 6: SDP example negotiating a payload type for format and
multi-codec packetization
A variation of this approach is to consider defining several multi-
codec payload types, each of them having an identified codec format.
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m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=setup:actpass
a=mid:1
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=rtpmap:96 generic/90000
a=fmtp:96 codec=vp9
a=rtpmap:97 generic/90000
a=fmtp:97 codec=vp8
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=96
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=97
Figure 7: Alternative SDP example negotiating a payload type for
format and multi-codec packetization
9.3. A RTP Header To Choose Packetization
A RTP header extension can be used to flag content as opaque so that
the receiver knows whether to use or not the multi-codec
packetization. As for the API header extension, the RTP header
extension may not need to be sent for every packet, it could for
instance be sent for the first packet of every intra video frame.
The main advantage of this approach is the reduced impact on SDP
negotiation.
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=setup:actpass
a=mid:1
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=extmap:4 urn:ietf:params:rtp-hdrext:multi-codec-packetization-use
a=sendrecv
a=rtpmap:96 vp9/90000
a=rtpmap:97 vp8/90000
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=96
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=97
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Figure 8: SDP example negotiating multi-codec packetization as
RTP header extension
10. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in
[RFC3550]. It is not expected that the proposed solution presented
in this document can create new security threats. The use and
implementation of RTP Media Chains containing Media Transformers
needs to be done carefully. It is important to refer to the security
considerations discussed in [SFrame] and [WebRTCInsertableStreams].
In particular Media Transformers on the receiver side need to be
prepared to receive arbitrary content, like decoders already do.
Similarly, since Media Transformers can be implemented as JavaScript
in browsers, RTP Packetizers should be prepared to receive arbitrary
content.
11. IANA Considerations
Two new media subtypes have been registered with IANA, as described
in this section.
11.1. Registration of audio/generic
Type name: audio
Subtype name: generic
Required parameters: none
Optional parameters: none
Encoding considerations: This format is framed (see Section 4.8 in
the template document) and contains binary data.
Security considerations: TBD.
Interoperability considerations: TBD
Published specification: TBD.
Applications that use this media type: TBD.
Additional information: none
Intended usage: COMMON
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Restrictions on usage: TBD
Author:
Change controller:
12. Registration of video/generic
Type name: video
Subtype name: generic
Required parameters: none
Optional parameters: none
Encoding considerations: This format is framed (see Section 4.8 in
the template document) and contains binary data.
Security considerations: TBD.
Interoperability considerations: TBD
Published specification: TBD.
Applications that use this media type: TBD.
Additional information: none
Intended usage: COMMON
Restrictions on usage: TBD
Author:
Change controller:
13. References
13.1. Normative References
[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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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <https://www.rfc-editor.org/info/rfc5285>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/info/rfc7656>.
[RFC8285] Singer, D., Desineni, H., and R. Even, Ed., "A General
Mechanism for RTP Header Extensions", RFC 8285,
DOI 10.17487/RFC8285, October 2017,
<https://www.rfc-editor.org/info/rfc8285>.
13.2. Informative References
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J.C., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
DOI 10.17487/RFC2198, September 1997,
<https://www.rfc-editor.org/info/rfc2198>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>.
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[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <https://www.rfc-editor.org/info/rfc5109>.
[RFC6184] Wang, Y.-K., Even, R., Kristensen, T., and R. Jesup, "RTP
Payload Format for H.264 Video", RFC 6184,
DOI 10.17487/RFC6184, May 2011,
<https://www.rfc-editor.org/info/rfc6184>.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011,
<https://www.rfc-editor.org/info/rfc6464>.
[RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
time Transport Protocol (RTP) Header Extension for Mixer-
to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011,
<https://www.rfc-editor.org/info/rfc6465>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904,
DOI 10.17487/RFC6904, April 2013,
<https://www.rfc-editor.org/info/rfc6904>.
[RFC7742] Roach, A.B., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<https://www.rfc-editor.org/info/rfc7742>.
[RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
<https://www.rfc-editor.org/info/rfc7874>.
[SFrame] "Secure Frame (SFrame)", n.d.,
<https://tools.ietf.org/html/draft-omara-sframe>.
[WebRTCInsertableStreams]
"WebRTC Insertable Media using Streams", n.d.,
<https://w3c.github.io/webrtc-insertable-streams>.
Authors' Addresses
Sergio Garcia Murillo
CoSMo
Email: sergio.garcia.murillo@cosmosoftware.io
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Youenn Fablet
Apple Inc.
Email: youenn@apple.com
Alex Gouaillard
CoSMo
Email: alex.gouaillard@cosmosoftware.io
Justin Uberti
Clubhouse
Email: justin@uberti.name
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