This document provides information and requirements for the use of Forward Error Correction (FEC) by WebRTC implementations.

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.

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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.

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 [RFC 2119] [RFC 8174] when, and only when, they appear in all capitals, as shown here.

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 [RFC 3550]; this section enumerates the various mechanisms available and describes their trade-offs.

This approach, as described in RFC 5956, Section 4.3, sends FEC packets as an independent RTP stream with its own synchronization source (SSRC) [RFC 3550] 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.

This approach, as described in [RFC 2198], 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 RFC 2198, Section 4, this approach cannot recover certain parts of the RTP header, including the marker bit, contributing source (CSRC) information, and header extensions.

Some audio codecs, notably Opus [RFC 6716] and Adaptive Multi-Rate (AMR) [RFC 4867], 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 RFC 6716, Section 2.1.7, and [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 RFC 4867, Section 3.7.1, for details on this mechanism. In-band FEC mechanisms cannot recover any of the RTP header.

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.

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 [RFC 5391], 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 [RFC 6464] and [RFC 6465]. Implementations SHOULD account for this and attempt to approximate this information, using an approach similar to those described in RFC 2198, Section 4, and RFC 6464, Section 5.

Support for redundant encoding of a given RTP stream SHOULD be indicated by including audio/red [RFC 2198] as an additional supported media type for the associated "m=" section in the SDP offer [RFC 3264]. 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 [RFC 7587]. 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 RFC 4867, Section 8.1. Receivers MUST include this parameter, and set it to an appropriate value, as specified in [TS.26114], Table 6.3.

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.

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 [RFC 8627] is also capable of protecting multiple RTP streams via a single FEC stream, including all the streams that are part of a BUNDLE group [RFC 8843]. 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.

Support for an SSRC-multiplexed Flexible FEC stream to protect a given RTP stream SHOULD be indicated by including video/flexfec (described in RFC 8627, Section 5.1.2) as an additional supported media type for the associated "m=" section in the SDP offer [RFC 3264]. 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 RFC 5956, 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).

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 [RFC 8831] 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.

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 [RFC 8627] SHOULD also be supported, as mentioned in Section 5. Implementations MAY support additional FEC mechanisms if desired, e.g., [RFC 5109].

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 [RFC 4588] or Flexible FEC's retransmission mode (RFC 8627, 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 [RFC 3550]), 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., [RFC 6386], 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.

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) [RFC 3711] decryption process. Therefore, as described in RFC 3711, 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 [RFC 5763], which are enumerated in RFC 5764, Section 4.1.2. Additional security considerations for each individual FEC mechanism are enumerated in their respective documents.

This document requires no actions from IANA.

[RFC2119]

S. Bradner, "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]

C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C. Bolot, A. Vega-Garcia, 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]

J. Rosenberg, 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]

J. Sjoberg, M. Westerlund, A. Lakaniemi, 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]

A. Begen, "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]

J. Spittka, K. Vos, 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]

B. Leiba, "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]

M. Zanaty, V. Singh, A. Begen, 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, September 2017,
<http://www.3gpp.org/ftp/Specs/html-info/26114.htm>.

[OpusFEC]

Xiph, T. Terriberry, "Subject: Opus FEC", January 2013,
<http://lists.xiph.org/pipermail/opus/2013-January/001904.html>.

[RFC3550]

H. Schulzrinne, S. Casner, R. Frederick, 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]

M. Baugher, D. McGrew, M. Naslund, E. Carrara, 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]

J. Rey, D. Leon, A. Miyazaki, V. Varsa, and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>.

[RFC5109]

A. Li, "RTP Payload Format for Generic Forward Error Correction", RFC 5109, DOI 10.17487/RFC5109, December 2007,
<https://www.rfc-editor.org/info/rfc5109>.

[RFC5391]

A. Sollaud, "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]

J. Fischl, H. Tschofenig, 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]

D. McGrew, 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]

J. Bankoski, J. Koleszar, L. Quillio, J. Salonen, P. Wilkins, 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]

J. Lennox, E. Ivov, 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]

E. Ivov, E. Marocco, 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]

JM. Valin, K. Vos, 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]

R Jesup, S Loreto, and M Tüxen, "WebRTC Data Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
<https://www.rfc-editor.org/info/rfc8831>.

[RFC8843]

C Holmberg, H Alvestrand, 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>.

Several people provided significant input into this document, including Bernard Aboba, Jonathan Lennox, Giri Mandyam, Varun Singh, Tim Terriberry, Magnus Westerlund, and Mo Zanaty.

Justin Uberti