draft-ietf-rtcweb-fec.xml

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<?rfc toc="yes" ?>
<?rfc symrefs="yes" ?>
<?rfc iprnotified="no" ?>
<?rfc strict="yes" ?>
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<?rfc rfcedstyle="no" ?>
<?rfc tocdepth="4"?>
<rfc category="std" docName="draft-ietf-rtcweb-fec-09" ipr="trust200902">
  <front>
    <title abbrev="WebRTC FEC">WebRTC Forward Error Correction
    Requirements</title>
    <author fullname="Justin Uberti" initials="J." surname="Uberti">
      <organization>Google</organization>
      <address>
        <postal>
          <street>747 6th St S</street>
          <city>Kirkland</city>
          <region>WA</region>
          <code>98033</code>
          <country>USA</country>
        </postal>
        <email>justin@uberti.name</email>
      </address>
    </author>
    <date day="8" month="Feb" year="2019" />
    <area>RAI</area>
    <abstract>

      <t>This document provides information and requirements for how Forward
      Error Correction (FEC) should be used by WebRTC implementations.</t>
    </abstract>
  </front>
  <middle>
    <section title="Introduction">

      <t>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.</t>
    </section>
    <section title="Terminology">

      <t>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 <xref target="RFC2119"/> <xref target="RFC8174"/>
      when, and only when, they appear in all capitals, as shown here.</t>
    </section>
    <section title="Types of FEC">

      <t>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.</t>
      <section title="Separate FEC Stream">

        <t>This approach, as described in
        <xref target="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.</t>

        <t>This approach allows for recovery of entire RTP packets, including
        the full RTP header.</t>
      </section>
      <section title="Redundant Encoding">

        <t>This approach, as descibed in
        <xref target="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.</t>

        <t>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.</t>

        <t>As described in
        <xref target="RFC2198" />, Section 4, this approach cannot recover
        certain parts of the RTP header, including the marker bit, CSRC
        information, and header extensions.</t>
      </section>
      <section title="Codec-Specific In-band FEC">

        <t>Some audio codecs, notably Opus
        <xref target="RFC6716" /> and AMR
        <xref target="RFC4867" />, support their own in-band FEC mechanism,
        where redundant data is included in the codec payload.</t>

        <t>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
        <xref target="RFC6716" />, Section 2.1.7 for complete details.</t>

        <t>For AMR/AMR-WB, packets can contain copies or lower-quality
        encodings of multiple prior audio frames. This mechanism is similar to
        the
        <xref target="RFC2198" /> mechanism described above, but as it adds no
        additional framing, it can be slightly more efficient. See
        <xref target="RFC4867" />, Section 3.7.1 for details on this
        mechanism.</t>

        <t>In-band FEC mechanisms cannot recover any of the RTP header.</t>
      </section>
    </section>
    <section title="FEC for Audio Content" anchor="audio-fec">

      <t>The following section provides guidance on how to best use FEC for
      transmitting audio data. As indicated in
      <xref target="adaptive-fec" /> below, FEC should only be activated if
      network conditions warrant it, or upon explicit application request.</t>
      <section title="Recommended Mechanism">

        <t>When using variable-bitrate codecs without an internal FEC,
        <xref target="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.</t>

        <t>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
        <xref target="RFC2198" /> redundancy with reduced-bitrate Opus encodings
        SHOULD be used instead.</t>

        <t>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
        <xref target="RFC2198" />.</t>

        <t>When using constant-bitrate codecs, e.g. PCMU, use of
        <xref target="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.</t>

        <t>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.</t>

        <t>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
        <xref target="RFC6464" /> and
        <xref target="RFC6465" />. Implementations SHOULD account for this and
        attempt to approximate this information, using an approach similar to
        those described in
        <xref target="RFC2198" />, Section 4, and
        <xref target="RFC6464" />, Section 5.</t>
      </section>
      <section title="Negotiating Support">

        <t>Support for redundant encoding of a given RTP stream SHOULD be
        indicated by including audio/red
        <xref target="RFC2198" /> as an additional supported media type for the
        associated m= section in the SDP offer
        <xref target="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.</t>

        <t>Support for codec-specific FEC mechanisms are typically indicated
        via "a=fmtp" parameters.</t>

        <t>For Opus, a receiver MUST indicate that it is prepared to use
        incoming FEC data with the "useinbandfec=1" parameter, as specified in
        <xref target="RFC7587" />. This parameter is declarative and can be
        negotiated separately for either media direction.</t>

        <t>For AMR/AMR-WB, support for redundant encoding, and the maximum
        supported depth, are controlled by the 'max-red' parameter, as
        specified in
        <xref target="RFC4867" />, Section 8.1. Receivers MUST include this
        parameter, and set it to an appropriate value, as specified in
        <xref target="TS.26114" />, Table 6.3.</t>
      </section>
    </section>
    <section title="FEC for Video Content" anchor="video-fec">

      <t>The following section provides guidance on how to best use FEC for
      transmitting video data. As indicated in
      <xref target="adaptive-fec" /> below, FEC should only be activated if
      network conditions warrant it, or upon explicit application request.</t>
      <section title="Recommended Mechanism">

        <t>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
        <xref target="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
        <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" /> group. As a
        result, for video content, use of a separate FEC stream with the
        flexfec RTP payload format is RECOMMENDED.</t>

        <t>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
        packets.</t>
      </section>
      <section title="Negotiating Support">

        <t>Support for a SSRC-multiplexed flexfec stream to protect a given RTP
        stream SHOULD be indicated by including one of the formats described in

        <xref target="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
        <xref target="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.</t>

        <t>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.</t>

        <t>Use of FEC-only m-lines, and grouping using the SDP group mechanism
        as described in
        <xref target="RFC5956" />, Section 4.1 is not currently defined for
        WebRTC, and SHOULD NOT be offered.</t>

        <t>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).</t>
      </section>
    </section>
    <section title="FEC for Application Content">

      <t>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
      <xref target="I-D.ietf-rtcweb-data-channel" /> for details.</t>

      <t>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.</t>

      <t>As a result, this document makes no recommendations regarding FEC for
      the underlying data transport.</t>
    </section>
    <section title="Implementation Requirements">

      <t>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
      <xref target="audio-fec" />. Use of these formats when sending, as
      mentioned above, is RECOMMENDED.</t>

      <t>The general FEC mechanism described in
      <xref target="I-D.ietf-payload-flexible-fec-scheme" /> SHOULD also be
      supported, as mentioned in
      <xref target="video-fec" />.</t>

      <t>Implementations MAY support additional FEC mechanisms if desired, e.g.,

      <xref target="RFC5109" />.</t>
    </section>
    <section title="Adaptive Use of FEC" anchor="adaptive-fec">

      <t>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
      <xref target="RFC4588" /> or flexfec
      <xref target="I-D.ietf-payload-flexible-fec-scheme" /> retransmissions,
      which only transmit redundant data when necessary, at the cost of an
      extra roundtrip.</t>

      <t>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
      <xref target="RFC3550" />), unless the application indicates it is
      willing to pay a quality penalty to proactively avoid losses.</t>

      <t>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.</t>

      <t>When using FEC with layered codecs, e.g.,
      <xref target="RFC6386" />, where only base layer frames are critical to
      the decoding of future frames, implementations SHOULD only apply FEC to
      these base layer frames.</t>

      <t>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.</t>
    </section>

    <section title="Security Considerations">
      <t>In the WebRTC context, FEC is specifically concerned with recovering
      data from lost packets; any corrupted packets will be discarded by the
      SRTP <xref target="RFC3711" /> decryption process. Therefore, as described
      in <xref target="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
      <xref target="RFC5763" />, which are enumerated in
      <xref target="RFC5764" />, Section 4.1.2.</t>

      <t>Additional security considerations for each individual FEC mechanism
      are enumerated in their respective documents.</t>
    </section>
    <section title="IANA Considerations">

      <t>This document requires no actions from IANA.</t>
    </section>
    <section title="Acknowledgements">

      <t>Several people provided significant input into this document,
      including Bernard Aboba, Jonathan Lennox, Giri Mandyam, Varun Singh, Tim
      Terriberry, Magnus Westerlund, and Mo Zanaty.</t>
    </section>
  </middle>
  <back>
    <references title="Normative References">
      <?rfc include='reference.RFC.2119.xml'?>
      <?rfc include='reference.RFC.2198.xml'?>
      <?rfc include='reference.RFC.3264.xml'?>
      <?rfc include='reference.RFC.4867.xml'?>
      <?rfc include='reference.RFC.5956.xml'?>
      <?rfc include='reference.RFC.7587.xml'?>
      <?rfc include='reference.RFC.8174.xml'?>
      <?rfc include='reference.I-D.ietf-payload-flexible-fec-scheme'?>
      <reference anchor="TS.26114">
        <front>
          <title>IP Multimedia Subsystem (IMS); Multimedia telephony; Media
          handling and interaction</title>
          <author>
            <organization>3GPP</organization>
          </author>
          <date day="22" month="September" year="2017" />
        </front>
        <seriesInfo name="3GPP TS" value="26.114 15.0.0" />
        <format type="HTML"
        target="http://www.3gpp.org/ftp/Specs/html-info/26114.htm" />
      </reference>
      </references>
    <references title="Informative References">
      <?rfc include='reference.RFC.3550.xml'?>
      <?rfc include='reference.RFC.3711.xml'?>
      <?rfc include='reference.RFC.4588.xml'?>
      <?rfc include='reference.RFC.5109.xml'?>
      <?rfc include='reference.RFC.5763.xml'?>
      <?rfc include='reference.RFC.5764.xml'?>
      <?rfc include='reference.RFC.6386.xml'?>
      <?rfc include='reference.RFC.6464.xml'?>
      <?rfc include='reference.RFC.6465.xml'?>
      <?rfc include='reference.RFC.6716.xml'?>
      <?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
      <?rfc include='reference.I-D.ietf-rtcweb-data-channel'?>
    </references>
    <section title="Change log">
       <t>Changes in draft -09:
      <list style="symbols">
        <t>Editorial changes.</t>
      </list></t>

       <t>Changes in draft -08:
      <list style="symbols">
        <t>Switch to RFC 8174 boilerplate.</t>
      </list></t>

      <t>Changes in draft -07:
      <list style="symbols">
        <t>Clarify how bandwidth management interacts with FEC.</t>

        <t>Make 3GPP reference normative.</t>
      </list></t>

      <t>Changes in draft -06:
      <list style="symbols">

        <t>Discuss how multiple streams can be protected by a single FlexFEC
        stream.</t>

        <t>Discuss FEC for bandwidth probing.</t>

        <t>Add note about recovery of RTP headers and header extensions.</t>

        <t>Add note about FEC/SRTP ordering.</t>

        <t>Clarify flexfec demux text, and mention retransmits.</t>

        <t>Clarify text regarding offers/answers.</t>

        <t>Make RFC2198 support SHOULD strength.</t>

        <t>Clean up references.</t>
      </list></t>

      <t>Changes in draft -05:
      <list style="symbols">

        <t>No changes.</t>
      </list></t>

      <t>Changes in draft -04:
      <list style="symbols">

        <t>Discussion of layered codecs.</t>

        <t>Discussion of RTX.</t>

        <t>Clarified implementation requirements.</t>

        <t>FlexFEC MUST -&gt; SHOULD.</t>

        <t>Clarified AMR max-red handling.</t>

        <t>Updated references.</t>
      </list></t>

      <t>Changes in draft -03:
      <list style="symbols">

        <t>Added overhead stats for Opus.</t>

        <t>Expanded discussion of multi-packet FEC for Opus.</t>

        <t>Added discussion of AMR/AMR-WB.</t>

        <t>Removed discussion of ssrc-group.</t>

        <t>Referenced the data channel doc.</t>

        <t>Referenced the RTP/RTCP RFC.</t>

        <t>Several small edits based on feedback from Magnus.</t>
      </list></t>

      <t>Changes in draft -02:
      <list style="symbols">

        <t>Expanded discussion of FEC-only m-lines, and how they should be
        handled in offers and answers.</t>
      </list></t>

      <t>Changes in draft -01:
      <list style="symbols">

        <t>Tweaked abstract/intro text that was ambiguously normative.</t>

        <t>Removed text on FEC for Opus in CELT mode.</t>

        <t>Changed RFC 2198 recommendation for PCMU to be MAY instead of NOT
        RECOMMENDED, based on list feedback.</t>

        <t>Explicitly called out application data as something not addressed in
        this document.</t>

        <t>Updated flexible-fec reference.</t>
      </list></t>

      <t>Changes in draft -00:
      <list style="symbols">

        <t>Initial version, from sidebar conversation at IETF 90.</t>
      </list></t>
    </section>
  </back>
</rfc>