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WebRTC Forward Error Correction Requirements

The information below is for an old version of the document.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 8854.
Author Justin Uberti
Last updated 2019-02-21 (Latest revision 2018-03-02)
Replaces draft-uberti-rtcweb-fec
RFC stream Internet Engineering Task Force (IETF)
Additional resources Mailing list discussion
Stream WG state Submitted to IESG for Publication
Document shepherd Ted Hardie
Shepherd write-up Show Last changed 2018-01-08
IESG IESG state Became RFC 8854 (Proposed Standard)
Consensus boilerplate Yes
Telechat date (None)
Responsible AD Adam Roach
Send notices to "Ted Hardie" <>
IANA IANA review state IANA OK - No Actions Needed
Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                           March 2, 2018
Expires: September 3, 2018

              WebRTC Forward Error Correction Requirements


   This document provides information and requirements for how Forward
   Error Correction (FEC) should be used by WebRTC implementations.

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

   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 September 3, 2018.

Copyright Notice

   Copyright (c) 2018 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
   ( 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 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.

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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 . . . . . . . . . . . . . . . . . . .   5
   5.  FEC for Video Content . . . . . . . . . . . . . . . . . . . .   5
     5.1.  Recommended Mechanism . . . . . . . . . . . . . . . . . .   5
     5.2.  Negotiating Support . . . . . . . . . . . . . . . . . . .   6
   6.  FEC for Application Content . . . . . . . . . . . . . . . . .   6
   7.  Implementation Requirements . . . . . . . . . . . . . . . . .   7
   8.  Adaptive Use of FEC . . . . . . . . . . . . . . . . . . . . .   7
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   11. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   8
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .   8
     12.1.  Normative References . . . . . . . . . . . . . . . . . .   8
     12.2.  Informative References . . . . . . . . . . . . . . . . .   9
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .  11
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  12

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

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "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
   face of packet loss.  There are multiple ways in which this can be

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   accomplished; this section enumerates the various mechanisms and
   describes their tradeoffs.

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 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 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.

   As described in [RFC2198], Section 4, this approach cannot recover
   certain parts of the RTP header, including the marker bit, CSRC
   information, and header extensions.

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

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   recover multiple consecutive lost packets, which can be a problem on
   wireless networks.  See [RFC6716], Section 2.1.7 for complete

   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.

   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, [RFC2198]
   redundant encoding 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
   [RFC2198] redundancy 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 [RFC2198].

   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.

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   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

   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 "flexfec" FEC
   mechanism described in [I-D.ietf-payload-flexible-fec-scheme] is also
   capable of protecting multiple RTP streams via a single FEC stream,
   including all the streams that are part of a BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation] group.  As a result, for

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   video content, use of a separate FEC stream with the flexfec 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 flexfec repair packets,
   or the SSRC field present in the FEC header of flexfec retransmission

5.2.  Negotiating Support

   Support for a SSRC-multiplexed flexfec stream to protect a given RTP
   stream SHOULD be indicated by including one of the formats described
   in [I-D.ietf-payload-flexible-fec-scheme], Section 5.1, 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 flexfec repair stream will be created for each BUNDLE group,
   even if flexfec is negotiated for each primary stream.

   Answerers can reject the use of SSRC-multiplexed FEC, by not
   including the offered FEC formats 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
   [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.

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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,

   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

   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 flexfec [I-D.ietf-payload-flexible-fec-scheme]
   retransmissions, which only transmit redundant data when necessary,
   at the cost of an extra roundtrip.

   Given this, WebRTC implementations SHOULD consider using RTX or
   flexfec retransmissions 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.

   Note that when probing bandwidth, i.e., speculatively sending extra
   data to determine if additional link capacity exists, FEC SHOULD be
   used in all cases.  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 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.

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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.

   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], as
   described 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.  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

              Singh, V., Begen, A., Zanaty, M., and G. Mandyam, "RTP
              Payload Format for Flexible Forward Error Correction
              (FEC)", draft-ietf-payload-flexible-fec-scheme-05 (work in
              progress), July 2017.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [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,

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   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,

   [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, <>.

   [RFC5956]  Begen, A., "Forward Error Correction Grouping Semantics in
              the Session Description Protocol", RFC 5956,
              DOI 10.17487/RFC5956, September 2010,

   [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,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

              3GPP, "IP Multimedia Subsystem (IMS); Multimedia
              telephony; Media handling and interaction", 3GPP TS 26.114
              15.0.0, September 2017.

12.2.  Informative References

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-48 (work in progress), January 2018.

              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, <>.

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   [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,

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <>.

   [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, <>.

   [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,

   [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,

   [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,

   [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,

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
              September 2012, <>.

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Appendix A.  Change log

   Changes in draft -08:

   o  Switch to RFC 8174 boilerplate.

   Changes in draft -07:

   o  Clarify how bandwidth management interacts with FEC.

   o  Make 3GPP reference normative.

   Changes in draft -06:

   o  Discuss how multiple streams can be protected by a single FlexFEC

   o  Discuss FEC for bandwidth probing.

   o  Add note about recovery of RTP headers and header extensions.

   o  Add note about FEC/SRTP ordering.

   o  Clarify flexfec demux text, and mention retransmits.

   o  Clarify text regarding offers/answers.

   o  Make RFC2198 support SHOULD strength.

   o  Clean up references.

   Changes in draft -05:

   o  No changes.

   Changes in draft -04:

   o  Discussion of layered codecs.

   o  Discussion of RTX.

   o  Clarified implementation requirements.

   o  FlexFEC MUST -> SHOULD.

   o  Clarified AMR max-red handling.

   o  Updated references.

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   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.

Author's Address

   Justin Uberti
   747 6th St S
   Kirkland, WA  98033


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