One document matched: draft-ietf-rtcweb-audio-11.xml
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<rfc category="std" docName="draft-ietf-rtcweb-audio-11" ipr="trust200902">
<front>
<title abbrev="WebRTC Audio">WebRTC Audio Codec and Processing
Requirements</title>
<author fullname="Jean-Marc Valin" initials="JM." surname="Valin">
<organization>Mozilla</organization>
<address>
<postal>
<street>331 E. Evelyn Avenue</street>
<city>Mountain View</city>
<region>CA</region>
<code>94041</code>
<country>USA</country>
</postal>
<email>jmvalin@jmvalin.ca</email>
</address>
</author>
<author fullname="Cary Bran" initials="C." surname="Bran">
<organization>Plantronics</organization>
<address>
<postal>
<street>345 Encinial Street</street>
<city>Santa Cruz</city>
<region>CA</region>
<code>95060</code>
<country>USA</country>
</postal>
<phone>+1 206 661-2398</phone>
<email>cary.bran@plantronics.com</email>
</address>
</author>
<date day="21" month="April" year="2016"/>
<abstract>
<t>This document outlines the audio codec and processing requirements
for WebRTC endpoints.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>An integral part of the success and adoption of the Web Real Time
Communications (WebRTC) will be the voice and video interoperability
between WebRTC applications. This specification will outline the audio
processing and codec requirements for WebRTC endpoints.</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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
<section title="Codec Requirements">
<t>To ensure a baseline level of interoperability between WebRTC
endpoints, a minimum set of required codecs are specified below.
If other suitable audio codecs are available for the WebRTC endpoint to use,
it is RECOMMENDED that they are also be included in the offer in order
to maximize the possibility to establish the session without the need
for audio transcoding.</t>
<t>WebRTC endpoints are REQUIRED to implement the following audio
codecs:</t>
<t><list style="symbols">
<t>Opus <xref target="RFC6716"/> with the payload format specified
in <xref target="RFC7587"/>.</t>
<t>G.711 PCMA and PCMU with the payload format specified in section
4.5.14 of <xref target="RFC3551"/>.</t>
<t><xref target="RFC3389"/> comfort noise (CN). WebRTC endpoints MUST
support RFC3389 CN for streams encoded with G.711 or any other supported
codec that does not provide its own CN. Since Opus provides its own CN mechanism,
the use of RFC3389 CN with Opus is NOT RECOMMENDED.
Use of DTX/CN by senders is OPTIONAL.</t>
<t>The audio/telephone-event media format as specified in <xref
target="RFC4733"/>. The endpoints MAY send DTMF events at any time
and SHOULD suppress in-band DTMF tones, if any.
DTMF events generated by a WebRTC endpoint MUST have a duration of no more than 8000 ms and no less than 40 ms. The recommended default duration is 100 ms for each tone. The gap between events MUST be no less than 30 ms; the recommended default gap duration is 70 ms. WebRTC endpoints are not required to do anything with RFC 4733 tones sent to them, except gracefully drop them. There is currently no API to inform JavaScript about the received DTMF or other RFC 4733 tones.
WebRTC endpoints are REQUIRED to be able to
generate and consume the following events:</t>
</list></t>
<t/>
<figure>
<artwork><![CDATA[
+------------+--------------------------------+-----------+
|Event Code | Event Name | Reference |
+------------+--------------------------------+-----------+
| 0 | DTMF digit "0" | RFC4733 |
| 1 | DTMF digit "1" | RFC4733 |
| 2 | DTMF digit "2" | RFC4733 |
| 3 | DTMF digit "3" | RFC4733 |
| 4 | DTMF digit "4" | RFC4733 |
| 5 | DTMF digit "5" | RFC4733 |
| 6 | DTMF digit "6" | RFC4733 |
| 7 | DTMF digit "7" | RFC4733 |
| 8 | DTMF digit "8" | RFC4733 |
| 9 | DTMF digit "9" | RFC4733 |
| 10 | DTMF digit "*" | RFC4733 |
| 11 | DTMF digit "#" | RFC4733 |
| 12 | DTMF digit "A" | RFC4733 |
| 13 | DTMF digit "B" | RFC4733 |
| 14 | DTMF digit "C" | RFC4733 |
| 15 | DTMF digit "D" | RFC4733 |
+------------+--------------------------------+-----------+
]]></artwork>
</figure>
<t/>
<t>For all cases where the endpoint is able to process audio at a sampling
rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before
PCMA/PCMU. For Opus, all modes MUST be supported on the decoder side.
The choice of encoder-side modes is left to the implementer. Endpoints MAY
use the offer/answer mechanism to signal a preference for a particular
mode or ptime.</t>
<t>For additional information on implementing codecs other than the
mandatory-to-implement codecs listed above, refer to <xref target="I-D.ietf-rtcweb-audio-codecs-for-interop"/>.</t>
</section>
<section anchor="level" title="Audio Level">
<t>It is desirable to standardize the "on the wire" audio level for
speech transmission to avoid users having to manually adjust the
playback and to facilitate mixing in conferencing applications. It is
also desirable to be consistent with ITU-T recommendations G.169 and
G.115, which recommend an active audio level of -19 dBm0. However,
unlike G.169 and G.115, the audio for WebRTC is not constrained to have
a passband specified by G.712 and can in fact be sampled at any sampling
rate from 8 kHz to 48 kHz and up. For this reason, the level SHOULD be
normalized by only considering frequencies above 300 Hz, regardless of
the sampling rate used. The level SHOULD also be adapted to avoid
clipping, either by lowering the gain to a level below -19 dBm0, or
through the use of a compressor.</t>
<t>Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to
a root mean square (RMS) level of 2600. Only active speech should be
considered in the RMS calculation. If the endpoint has control over the
entire audio capture path, as is typically the case for a regular phone,
then it is RECOMMENDED that the gain be adjusted in such a way that
active speech have a level of 2600 (-19 dBm0) for an average speaker. If
the endpoint does not have control over the entire audio capture, as is
typically the case for a software endpoint, then the endpoint SHOULD use
automatic gain control (AGC) to dynamically adjust the level to 2600
(-19 dBm0) +/- 6 dB. For music or desktop sharing applications, the
level SHOULD NOT be automatically adjusted and the endpoint SHOULD allow
the user to set the gain manually.</t>
<t>The RECOMMENDED filter for normalizing the signal energy is a
second-order Butterworth filter with a 300 Hz cutoff frequency.</t>
<t>It is common for the audio output on some devices to be "calibrated"
for playing back pre-recorded "commercial" music, which is typically
around 12 dB louder than the level recommended in this section. Because
of this, endpoints MAY increase the gain before playback.</t>
</section>
<section anchor="aec" title="Acoustic Echo Cancellation (AEC)">
<t>It is plausible that the dominant near to mid-term WebRTC usage model
will be people using the interactive audio and video capabilities to
communicate with each other via web browsers running on a notebook
computer that has built-in microphone and speakers. The notebook-as-
communication-device paradigm presents challenging echo cancellation
problems, the specific remedy of which will not be mandated here.
However, while no specific algorithm or standard will be required by
WebRTC-compatible endpoints, echo cancellation will improve the user
experience and should be implemented by the endpoint device.</t>
<t>WebRTC endpoints SHOULD include an AEC or some other form of echo
control. On general purpose platforms (e.g. PC), it is common for the
audio capture ADC and the audio playback DAC to use different clocks.
In these cases, such as when a webcam is used for capture and a separate
soundcard is used for playback, the sampling rates are likely to differ
slightly. Endpoint AECs SHOULD be robust to such conditions, unless they
are shipped along with hardware that guarantees capture and playback to
be sampled from the same clock.</t>
<t>Endpoints SHOULD allow the entire AEC and/or the non-linear processing
(NLP) to be turned off for applications, such as music, that do not
behave well with the spectral attenuation methods typically used in
NLPs. Similarly, endpoints SHOULD have the ability to detect the presence
of a headset and disable echo cancellation.</t>
<t>For some applications where the remote endpoint may not have an echo
canceller, the local endpoint MAY include a far-end echo canceller, but if
that is the case, it SHOULD be disabled by default.</t>
</section>
<section title="Legacy VoIP Interoperability">
<t>The codec requirements above will ensure, at a minimum, voice
interoperability capabilities between WebRTC endpoints and
legacy phone systems that support G.711.</t>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>For security considerations regarding the codecs themselves please
refer their specifications, including <xref target="RFC6716"/>,
<xref target="RFC7587"/>, <xref target="RFC3551"/>,
<xref target="RFC3389"/>, and <xref target="RFC4733"/>. Likewise, consult the RTP base specification
for RTP-based security considerations. WebRTC security is further
discussed in <xref target="I-D.ietf-rtcweb-security"/> and
<xref target="I-D.ietf-rtcweb-security-arch"/> and <xref target="I-D.ietf-rtcweb-rtp-usage"/>.</t>
<t>Implementers should consider whether the use of variable bitrate is appropriate
for their application based on <xref target="RFC6562"/>. Encryption and
authentication issues are beyond the scope of this document.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>This draft incorporates ideas and text from various other drafts. In
particular we would like to acknowledge, and say thanks for, work we
incorporated from Harald Alvestrand and Cullen Jennings.</t>
</section>
</middle>
<back>
<references title="Normative References">
&rfc2119;
&rfc3551;
&rfc3389;
&rfc4733;
&rfc6716;
&rfc6562;
&rfc7587;
</references>
<references title="Informative References">
&rtcweb-security;
&rtcweb-security-arch;
&rtcweb-rtp;
&draftlegacy;
</references>
</back>
</rfc>
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