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Versions: (draft-uberti-rtcweb-fec) 00 01 02 03 04 05 06 07 08 Draft is active
In: Approved-announcement_to_be_sent
Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                            May 23, 2017
Expires: November 24, 2017


              WebRTC Forward Error Correction Requirements
                        draft-ietf-rtcweb-fec-05

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.

   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 http://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."

   This Internet-Draft will expire on November 24, 2017.

Copyright Notice

   Copyright (c) 2017 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
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   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 . . . . . . . . . . . . . . . . . . .   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-04 (work in
              progress), March 2017.

   [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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