Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track October 31, 2016
Expires: May 4, 2017
WebRTC Forward Error Correction Requirements
draft-ietf-rtcweb-fec-04
Abstract
This document provides information and requirements for how Forward
Error Correction (FEC) should be used by WebRTC applications.
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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This Internet-Draft will expire on May 4, 2017.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Types of FEC . . . . . . . . . . . . . . . . . . . . . . . . 2
3.1. Separate FEC Stream . . . . . . . . . . . . . . . . . . . 3
3.2. Redundant Encoding . . . . . . . . . . . . . . . . . . . 3
3.3. Codec-Specific In-band FEC . . . . . . . . . . . . . . . 3
4. FEC for Audio Content . . . . . . . . . . . . . . . . . . . . 4
4.1. Recommended Mechanism . . . . . . . . . . . . . . . . . . 4
4.2. Negotiating Support . . . . . . . . . . . . . . . . . . . 4
5. FEC for Video Content . . . . . . . . . . . . . . . . . . . . 5
5.1. Recommended Mechanism . . . . . . . . . . . . . . . . . . 5
5.2. Negotiating Support . . . . . . . . . . . . . . . . . . . 5
6. FEC for Application Content . . . . . . . . . . . . . . . . . 6
7. Implementation Requirements . . . . . . . . . . . . . . . . . 6
8. Adaptive Use of FEC . . . . . . . . . . . . . . . . . . . . . 6
9. Security Considerations . . . . . . . . . . . . . . . . . . . 7
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
12.1. Normative References . . . . . . . . . . . . . . . . . . 7
12.2. Informative References . . . . . . . . . . . . . . . . . 8
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 10
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 client
implementations.
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].
3. Types of FEC
By its name, FEC describes the sending of redundant information in an
outgoing packet stream so that information can still be recovered
even in the face of packet loss. There are multiple ways in which
this can be accomplished; this section enumerates the various
mechanisms and describes their tradeoffs.
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3.1. Separate FEC Stream
This approach, as described in [RFC5956], Section 4.3, sends FEC
packets as an independent SSRC-multiplexed stream, with its own SSRC
and payload type. While by far the most flexible, each FEC packet
will have its own IP+UDP+RTP+FEC header, leading to additional
overhead of the FEC stream.
3.2. Redundant Encoding
This approach, as descibed 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
packet, or for codecs that support variable-bitrate encodings,
possibly a smaller, lower-quality representation. In certain cases,
the redundant data could include multiple prior packets.
Since there is only a single set of packet headers, this approach
allows for a very efficient representation of primary + 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.
3.3. Codec-Specific In-band FEC
Some audio codecs, notably Opus [RFC6716] and AMR [RFC4867] support
their own in-band FEC mechanism, where redundant data is included in
the codec payload.
For Opus, packets deemed as important are re-encoded at a lower
bitrate and added to the subsequent packet, 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 about 20-30%, depending on the amount of protection needed. Note
that this mechanism can only carry redundancy information for the
immediately preceding packet; as such the decoder cannot fully
recover multiple consecutive lost packets, which can be a problem on
wireless networks. See [RFC6716], Section 2.1.7 for complete
details.
For AMR/AMR-WB, packets can contain copies or lower-quality encodings
of multiple prior audio frames. This mechanism is similar to the
[RFC2198] mechanism described above, but as it adds no additional
framing, it can be slightly more efficient. See [RFC4867],
Section 3.7.1 for details on this mechanism.
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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 the Opus codec, use of the built-in Opus FEC mechanism is
RECOMMENDED. This provides reasonable protection of the audio stream
against typical losses, with modest overhead. Note that as indicated
above the built-in Opus FEC only provides single-frame redundancy; if
multi-packet protection is needed, the built-in FEC should be
combined with [RFC2198] redundancy to protect the N-2th, N-3rd, etc.
packets.
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 [RFC2198].
When using variable-bitrate codecs without an internal FEC, [RFC2198]
redundant encoding with lower-fidelity version(s) of previous
packet(s) is RECOMMENDED. This provides reasonable protection of the
payload with moderate overhead.
When using constant-bitrate codecs, e.g. PCMU, use of [RFC2198]
redundant encoding MAY be used, but note that this will result in a
potentially significant bitrate increase, and that 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.
4.2. Negotiating Support
Support for redundant encoding MUST be indicated by offering "red" as
a supported payload type in the offer. Answerers can reject the use
of redundant encoding by not including "red" as a supported payload
type in the 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
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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
[3GPP.26.114], 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
For video content, use of a separate FEC stream with the RTP payload
format described in [I-D.ietf-payload-flexible-fec-scheme] is
RECOMMENDED. The receiver can demultiplex the incoming FEC stream by
SSRC and correlate it with the primary stream via the SSRC field
present in the FEC header.
Support for protecting multiple primary streams with a single FEC
stream is complicated by WebRTC's 1-m-line-per-stream policy, which
does not allow for a m-line dedicated specifically to FEC.
5.2. Negotiating Support
To offer support for a SSRC-multiplexed FEC stream that is associated
with a given primary stream, the offerer MUST offer the formats
supported for the primary stream, as well as one of the formats
described in [I-D.ietf-payload-flexible-fec-scheme], Section 5.1.
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 can reject the use of SSRC-multiplexed FEC, by not
including FEC formats in the answer.
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).
This ensures that implementations will not malfunction when said
future version of WebRTC enables offers of FEC-only m-lines.
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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
[I-D.ietf-rtcweb-data-channel] 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
[I-D.ietf-payload-flexible-fec-scheme] 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
Since use of FEC always causes redundant data to be transmitted, this
will lead to less bandwidth available for the primary encoding when
in a bandwidth-constrained environment. This is in contrast to
methods like RTX [RFC4588], which only transmits redundant data when
necessary, at the cost of an extra roundtrip.
Given this, WebRTC implementations SHOULD consider using RTX instead
of FEC when RTT is low, and 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 RTCP
Receiver Reports [RFC3550]), unless the application indicates it is
willing to pay a quality penalty to proactively avoid losses.
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.
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9. Security Considerations
This document makes recommendations regarding the use of FEC.
Generally, 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.
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. 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.
12. References
12.1. Normative References
[I-D.ietf-payload-flexible-fec-scheme]
Singh, V., Begen, A., Zanaty, M., and G. Mandyam, "RTP
Payload Format for Flexible Forward Error Correction
(FEC)", draft-ietf-payload-flexible-fec-scheme-03 (work in
progress), October 2016.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
DOI 10.17487/RFC2198, September 1997,
<http://www.rfc-editor.org/info/rfc2198>.
[RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in
the Session Description Protocol", RFC 5956,
DOI 10.17487/RFC5956, September 2010,
<http://www.rfc-editor.org/info/rfc5956>.
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12.2. Informative References
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), January 2015.
[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, <http://www.rfc-editor.org/info/rfc3550>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>.
[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, <http://www.rfc-editor.org/info/rfc4867>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <http://www.rfc-editor.org/info/rfc5109>.
[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,
<http://www.rfc-editor.org/info/rfc6386>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <http://www.rfc-editor.org/info/rfc6716>.
[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,
<http://www.rfc-editor.org/info/rfc7587>.
Appendix A. Change log
Changes in draft -04:
o Discussion of layered codecs.
o Discussion of RTX.
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o Clarified implementation requirements.
o FlexFEC MUST -> SHOULD.
o Clarified AMR max-red handling.
o Updated references.
Changes in draft -03:
o Added overhead stats for Opus.
o Expanded discussion of multi-packet FEC for Opus.
o Added discussion of AMR/AMR-WB.
o Removed discussion of ssrc-group.
o Referenced the data channel doc.
o Referenced the RTP/RTCP RFC.
o Several small edits based on feedback from Magnus.
Changes in draft -02:
o Expanded discussion of FEC-only m-lines, and how they should be
handled in offers and answers.
Changes in draft -01:
o Tweaked abstract/intro text that was ambiguously normative.
o Removed text on FEC for Opus in CELT mode.
o Changed RFC 2198 recommendation for PCMU to be MAY instead of NOT
RECOMMENDED, based on list feedback.
o Explicitly called out application data as something not addressed
in this document.
o Updated flexible-fec reference.
Changes in draft -00:
o Initial version, from sidebar conversation at IETF 90.
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Author's Address
Justin Uberti
Google
747 6th St S
Kirkland, WA 98033
USA
Email: justin@uberti.name
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