WebRTC Forward Error Correction
Requirements
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FEC
This document provides information and requirements for the use of Forward
Error Correction (FEC) by WebRTC implementations.
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
implementations.
Terminology
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
when, and only when, they appear in all capitals, as shown here.
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 event
of packet loss. There are multiple ways this can be accomplished
for RTP media streams ; this section enumerates
the various mechanisms available and describes their trade-offs.
Separate FEC Stream
This approach, as described in ,
sends FEC packets as an independent RTP stream with its own
synchronization source (SSRC) 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.
Redundant Encoding
This approach, as described in
, 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
, this approach cannot recover
certain parts of the RTP header, including the marker bit, contributing source (CSRC)
information, and header extensions.
Codec-Specific In-Band FEC
Some audio codecs, notably Opus
and Adaptive Multi-Rate (AMR)
, 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
,
and this Opus mailing list post
for more details.
For AMR and AMR-Wideband (AMR-WB), packets can contain copies or lower-quality
encodings of multiple prior audio frames. See
,
for details on this mechanism.
In-band FEC mechanisms cannot recover any of the RTP header.
FEC for Audio Content
The following section provides guidance on how to best use FEC for
transmitting audio data. As indicated in
below, FEC should only be activated if
network conditions warrant it, or upon explicit application request.
Recommended Mechanism
When using variable-bitrate codecs without an internal FEC,
redundant encoding
(as described in )
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 , 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
and
. Implementations SHOULD account for this and
attempt to approximate this information, using an approach similar to
those described in
, and
.
Negotiating Support
Support for redundant encoding of a given RTP stream SHOULD be
indicated by including audio/red
as an additional supported media type for the
associated "m=" section in the SDP offer
. 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
. 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
.
Receivers MUST include this
parameter, and set it to an appropriate value, as specified in
, Table 6.3.
FEC for Video Content
The following section provides guidance on how to best use FEC for
transmitting video data. As indicated in
below, FEC should only be activated if
network conditions warrant it, or upon explicit application request.
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 Flexible FEC mechanism
described in
is also capable
of protecting multiple RTP streams via a single FEC stream, including
all the streams that are part of a BUNDLE group
. 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.
Negotiating Support
Support for an SSRC-multiplexed Flexible FEC stream to protect a given RTP
stream SHOULD be indicated by including video/flexfec (described in
) as
an additional supported media type for the associated "m=" section in the
SDP offer
. 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
, 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).
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
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.
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
. Use of these formats when sending, as
mentioned above, is RECOMMENDED.
The general FEC mechanism described in
SHOULD also be
supported, as mentioned in
.
Implementations MAY support additional FEC mechanisms if desired, e.g.,
.
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
or Flexible FEC's retransmission mode (),
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
), 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.,
, 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.
Security Considerations
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)
decryption process. Therefore, as described
in , 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
, which are enumerated in
.
Additional security considerations for each individual FEC mechanism
are enumerated in their respective documents.
IANA Considerations
This document requires no actions from IANA.
References
Normative References
IP Multimedia Subsystem (IMS); Multimedia telephony; Media
handling and interaction
3GPP
Informative References
Negotiating Media Multiplexing Using the Session Description Protocol (SDP)
WebRTC Data Channels
Subject: Opus FEC
Xiph
message to the opus mailing list
Acknowledgements
Several people provided significant input into this document,
including , , , ,
, , and .