rfc8854
Internet Engineering Task Force (IETF) J. Uberti
Request for Comments: 8854 Google
Category: Standards Track January 2021
ISSN: 2070-1721
WebRTC Forward Error Correction Requirements
Abstract
This document provides information and requirements for the use of
Forward Error Correction (FEC) by WebRTC implementations.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8854.
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/license-info) in effect on the date of
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
2. Terminology
3. Types of FEC
3.1. Separate FEC Stream
3.2. Redundant Encoding
3.3. Codec-Specific In-Band FEC
4. FEC for Audio Content
4.1. Recommended Mechanism
4.2. Negotiating Support
5. FEC for Video Content
5.1. Recommended Mechanism
5.2. Negotiating Support
6. FEC for Application Content
7. Implementation Requirements
8. Adaptive Use of FEC
9. Security Considerations
10. IANA Considerations
11. References
11.1. Normative References
11.2. Informative References
Acknowledgements
Author's Address
1. Introduction
In situations where packet loss is high, or perfect media quality is
essential, Forward Error Correction (FEC) can be used to proactively
recover from packet losses. This specification provides guidance on
which FEC mechanisms to use, and how to use them, for WebRTC
implementations.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
3. Types of FEC
FEC describes the sending of redundant information in an outgoing
packet stream so that information can still be recovered even in the
event of packet loss. There are multiple ways this can be
accomplished for RTP media streams [RFC3550]; this section enumerates
the various mechanisms available and describes their trade-offs.
3.1. Separate FEC Stream
This approach, as described in [RFC5956], Section 4.3, sends FEC
packets as an independent RTP stream with its own synchronization
source (SSRC) [RFC3550] and payload type, multiplexed with the
primary encoding. While this approach can protect multiple packets
of the primary encoding with a single FEC packet, each FEC packet
will have its own IP/UDP/RTP/FEC header, and this overhead can be
excessive in some cases, e.g., when protecting each primary packet
with a FEC packet.
This approach allows for recovery of entire RTP packets, including
the full RTP header.
3.2. Redundant Encoding
This approach, as described in [RFC2198], allows for redundant data
to be piggybacked on an existing primary encoding, all in a single
packet. This redundant data may be an exact copy of a previous
payload, or for codecs that support variable-bitrate encodings, the
redundant data may possibly be a smaller, lower-quality
representation. In certain cases, the redundant data could include
encodings of multiple prior audio frames.
Since there is only a single set of packet headers, this approach
allows for a very efficient representation of primary and redundant
data. However, this savings is only realized when the data all fits
into a single packet (i.e. the size is less than a MTU). As a
result, this approach is generally not useful for video content.
As described in [RFC2198], Section 4, this approach cannot recover
certain parts of the RTP header, including the marker bit,
contributing source (CSRC) information, and header extensions.
3.3. Codec-Specific In-Band FEC
Some audio codecs, notably Opus [RFC6716] and Adaptive Multi-Rate
(AMR) [RFC4867], support their own in-band FEC mechanism, where
redundant data is included in the codec payload. This is similar to
the redundant encoding mechanism described above, but as it adds no
additional framing, it can be slightly more efficient.
For Opus, audio frames deemed important are re-encoded at a lower
bitrate and appended to the next payload, allowing partial recovery
of a lost packet. This scheme is fairly efficient; experiments
performed indicate that when Opus FEC is used, the overhead imposed
is only about 20-30%, depending on the amount of protection needed.
Note that this mechanism can only carry redundancy information for
the immediately preceding audio frame; thus the decoder cannot fully
recover multiple consecutive lost packets, which can be a problem on
wireless networks. See [RFC6716], Section 2.1.7, and this Opus
mailing list post [OpusFEC] for more details.
For AMR and AMR-Wideband (AMR-WB), packets can contain copies or
lower-quality encodings of multiple prior audio frames. See
[RFC4867], Section 3.7.1, for details on this mechanism.
In-band FEC mechanisms cannot recover any of the RTP header.
4. FEC for Audio Content
The following section provides guidance on how to best use FEC for
transmitting audio data. As indicated in Section 8 below, FEC should
only be activated if network conditions warrant it, or upon explicit
application request.
4.1. Recommended Mechanism
When using variable-bitrate codecs without an internal FEC, redundant
encoding (as described in Section 3.2) with lower-fidelity version(s)
of the previous packet(s) is RECOMMENDED. This provides reasonable
protection of the payload with only moderate bitrate increase, as the
redundant encodings can be significantly smaller than the primary
encoding.
When using the Opus codec, use of the built-in Opus FEC mechanism is
RECOMMENDED. This provides reasonable protection of the audio stream
against individual losses, with minimal overhead. Note that, as
indicated above, the built-in Opus FEC only provides single-frame
redundancy; if multi-packet protection is needed, the aforementioned
redundant encoding with reduced-bitrate Opus encodings SHOULD be used
instead.
When using the AMR/AMR-WB codecs, use of their built-in FEC mechanism
is RECOMMENDED. This provides slightly more efficient protection of
the audio stream than redundant encoding does.
When using constant-bitrate codecs, e.g., PCMU [RFC5391], redundant
encoding MAY be used, but this will result in a potentially
significant bitrate increase, and suddenly increasing bitrate to deal
with losses from congestion may actually make things worse.
Because of the lower packet rate of audio encodings, usually a single
packet per frame, use of a separate FEC stream comes with a higher
overhead than other mechanisms, and therefore is NOT RECOMMENDED.
As mentioned above, the recommended mechanisms do not allow recovery
of parts of the RTP header that may be important in certain audio
applications, e.g., CSRCs and RTP header extensions like those
specified in [RFC6464] and [RFC6465]. Implementations SHOULD account
for this and attempt to approximate this information, using an
approach similar to those described in [RFC2198], Section 4, and
[RFC6464], Section 5.
4.2. Negotiating Support
Support for redundant encoding of a given RTP stream SHOULD be
indicated by including audio/red [RFC2198] as an additional supported
media type for the associated "m=" section in the SDP offer
[RFC3264]. Answerers can reject the use of redundant encoding by not
including the audio/red media type in the corresponding "m=" section
in the SDP answer.
Support for codec-specific FEC mechanisms are typically indicated via
"a=fmtp" parameters.
For Opus, a receiver MUST indicate that it is prepared to use
incoming FEC data with the "useinbandfec=1" parameter, as specified
in [RFC7587]. This parameter is declarative and can be negotiated
separately for either media direction.
For AMR/AMR-WB, support for redundant encoding, and the maximum
supported depth, are controlled by the "max-red" parameter, as
specified in [RFC4867], Section 8.1. Receivers MUST include this
parameter, and set it to an appropriate value, as specified in
[TS.26114], Table 6.3.
5. FEC for Video Content
The following section provides guidance on how to best use FEC for
transmitting video data. As indicated in Section 8 below, FEC should
only be activated if network conditions warrant it, or upon explicit
application request.
5.1. Recommended Mechanism
Video frames, due to their size, often require multiple RTP packets.
As discussed above, a separate FEC stream can protect multiple
packets with a single FEC packet. In addition, the Flexible FEC
mechanism described in [RFC8627] is also capable of protecting
multiple RTP streams via a single FEC stream, including all the
streams that are part of a BUNDLE group [RFC8843]. As a result, for
video content, use of a separate FEC stream with the Flexible FEC RTP
payload format is RECOMMENDED.
To process the incoming FEC stream, the receiver can demultiplex it
by SSRC, and then correlate it with the appropriate primary stream(s)
via the CSRC(s) present in the RTP header of Flexible FEC repair
packets, or the SSRC field present in the FEC header of Flexible FEC
retransmission packets.
5.2. Negotiating Support
Support for an SSRC-multiplexed Flexible FEC stream to protect a
given RTP stream SHOULD be indicated by including video/flexfec
(described in [RFC8627], Section 5.1.2) as an additional supported
media type for the associated "m=" section in the SDP offer
[RFC3264]. As mentioned above, when BUNDLE is used, only a single
Flexible FEC repair stream will be created for each BUNDLE group,
even if Flexible FEC is negotiated for each primary stream.
Answerers can reject the use of SSRC-multiplexed FEC by not including
the video/flexfec media type in the corresponding "m=" section in the
SDP answer.
Use of FEC-only "m=" lines, and grouping using the SDP group
mechanism as described in [RFC5956], Section 4.1, is not currently
defined for WebRTC, and SHOULD NOT be offered.
Answerers SHOULD reject any FEC-only "m=" lines, unless they
specifically know how to handle such a thing in a WebRTC context
(perhaps defined by a future version of the WebRTC specifications).
6. FEC for Application Content
WebRTC also supports the ability to send generic application data,
and provides transport-level retransmission mechanisms to support
full and partial (e.g., timed) reliability. See [RFC8831] for
details.
Because the application can control exactly what data to send, it has
the ability to monitor packet statistics and perform its own
application-level FEC if necessary.
As a result, this document makes no recommendations regarding FEC for
the underlying data transport.
7. Implementation Requirements
To support the functionality recommended above, implementations MUST
be able to receive and make use of the relevant FEC formats for their
supported audio codecs, and MUST indicate this support, as described
in Section 4. Use of these formats when sending, as mentioned above,
is RECOMMENDED.
The general FEC mechanism described in [RFC8627] SHOULD also be
supported, as mentioned in Section 5.
Implementations MAY support additional FEC mechanisms if desired,
e.g., [RFC5109].
8. Adaptive Use of FEC
Because use of FEC always causes redundant data to be transmitted,
and the total amount of data must remain within any bandwidth limits
indicated by congestion control and the receiver, this will lead to
less bandwidth available for the primary encoding, even when the
redundant data is not being used. This is in contrast to methods
like RTX [RFC4588] or Flexible FEC's retransmission mode ([RFC8627],
Section 1.1.7), which only transmit redundant data when necessary, at
the cost of an extra round trip and thereby increased media latency.
Given this, WebRTC implementations SHOULD prefer using RTX or
Flexible FEC retransmissions instead of FEC when the connection RTT
is within the application's latency budget, and otherwise SHOULD only
transmit the amount of FEC needed to protect against the observed
packet loss (which can be determined, e.g., by monitoring transmit
packet loss data from RTP Control Protocol (RTCP) receiver reports
[RFC3550]), unless the application indicates it is willing to pay a
quality penalty to proactively avoid losses.
Note that when probing bandwidth, i.e., speculatively sending extra
data to determine if additional link capacity exists, FEC data SHOULD
be used as the additional data. Given that extra data is going to be
sent regardless, it makes sense to have that data protect the primary
payload; in addition, FEC can typically be applied in a way that
increases bandwidth only modestly, which is necessary when probing.
When using FEC with layered codecs, e.g., [RFC6386], where only base
layer frames are critical to the decoding of future frames,
implementations SHOULD only apply FEC to these base layer frames.
Finally, it should be noted that, although applying redundancy is
often useful in protecting a stream against packet loss, if the loss
is caused by network congestion, the additional bandwidth used by the
redundant data may actually make the situation worse and can lead to
significant degradation of the network.
9. Security Considerations
In the WebRTC context, FEC is specifically concerned with recovering
data from lost packets; any corrupted packets will be discarded by
the Secure Real-Time Transport Protocol (SRTP) [RFC3711] decryption
process. Therefore, as described in [RFC3711], Section 10, the
default processing when using FEC with SRTP is to perform FEC
followed by SRTP at the sender, and SRTP followed by FEC at the
receiver. This ordering is used for all the SRTP protection profiles
used in DTLS-SRTP [RFC5763], which are enumerated in [RFC5764],
Section 4.1.2.
Additional security considerations for each individual FEC mechanism
are enumerated in their respective documents.
10. IANA Considerations
This document requires no actions from IANA.
11. References
11.1. 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>.
[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>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867,
April 2007, <https://www.rfc-editor.org/info/rfc4867>.
[RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in
the Session Description Protocol", RFC 5956,
DOI 10.17487/RFC5956, September 2010,
<https://www.rfc-editor.org/info/rfc5956>.
[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for the Opus Speech and Audio Codec", RFC 7587,
DOI 10.17487/RFC7587, June 2015,
<https://www.rfc-editor.org/info/rfc7587>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8627] Zanaty, M., Singh, V., Begen, A., and G. Mandyam, "RTP
Payload Format for Flexible Forward Error Correction
(FEC)", RFC 8627, DOI 10.17487/RFC8627, July 2019,
<https://www.rfc-editor.org/info/rfc8627>.
[TS.26114] 3GPP, "IP Multimedia Subsystem (IMS); Multimedia
telephony; Media handling and interaction", 3GPP TS 26.114
15.0.0, 22 September 2017,
<http://www.3gpp.org/ftp/Specs/html-info/26114.htm>.
11.2. Informative References
[OpusFEC] Terriberry, T., "Subject: Opus FEC", message to the opus
mailing list, 28 January 2013,
<http://lists.xiph.org/pipermail/
opus/2013-January/001904.html>.
[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>.
[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>.
[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>.
[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>.
[RFC5391] Sollaud, A., "RTP Payload Format for ITU-T Recommendation
G.711.1", RFC 5391, DOI 10.17487/RFC5391, November 2008,
<https://www.rfc-editor.org/info/rfc5391>.
[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>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6386] Bankoski, J., Koleszar, J., Quillio, L., Salonen, J.,
Wilkins, P., and Y. Xu, "VP8 Data Format and Decoding
Guide", RFC 6386, DOI 10.17487/RFC6386, November 2011,
<https://www.rfc-editor.org/info/rfc6386>.
[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>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <https://www.rfc-editor.org/info/rfc6716>.
[RFC8831] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
<https://www.rfc-editor.org/info/rfc8831>.
[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>.
Acknowledgements
Several people provided significant input into this document,
including Bernard Aboba, Jonathan Lennox, Giri Mandyam, Varun Singh,
Tim Terriberry, Magnus Westerlund, and Mo Zanaty.
Author's Address
Justin Uberti
Google
747 6th St S
Kirkland, WA 98033
United States of America
Email: justin@uberti.name
ERRATA
