| Network Working Group | JM. Valin |
| Internet-Draft | Mozilla |
| Intended status: Standards Track | C. Bran |
| Expires: October 23, 2016 | Plantronics |
| April 21, 2016 |
WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-11
This document outlines the audio codec and processing requirements for WebRTC endpoints.
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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.
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 RFC 2119 [RFC2119].
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.
WebRTC endpoints are REQUIRED to implement the following audio codecs:
+------------+--------------------------------+-----------+
|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 |
+------------+--------------------------------+-----------+
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.
For additional information on implementing codecs other than the mandatory-to-implement codecs listed above, refer to [I-D.ietf-rtcweb-audio-codecs-for-interop].
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.
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.
The RECOMMENDED filter for normalizing the signal energy is a second-order Butterworth filter with a 300 Hz cutoff frequency.
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.
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.
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.
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.
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.
The codec requirements above will ensure, at a minimum, voice interoperability capabilities between WebRTC endpoints and legacy phone systems that support G.711.
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an RFC.
For security considerations regarding the codecs themselves please refer their specifications, including [RFC6716], [RFC7587], [RFC3551], [RFC3389], and [RFC4733]. Likewise, consult the RTP base specification for RTP-based security considerations. WebRTC security is further discussed in [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch] and [I-D.ietf-rtcweb-rtp-usage].
Implementers should consider whether the use of variable bitrate is appropriate for their application based on [RFC6562]. Encryption and authentication issues are beyond the scope of this document.
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.
| [I-D.ietf-rtcweb-security] | Rescorla, E., "Security Considerations for WebRTC", Internet-Draft draft-ietf-rtcweb-security-08, February 2015. |
| [I-D.ietf-rtcweb-security-arch] | Rescorla, E., "WebRTC Security Architecture", Internet-Draft draft-ietf-rtcweb-security-arch-11, March 2015. |
| [I-D.ietf-rtcweb-rtp-usage] | Perkins, C., Westerlund, M. and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", Internet-Draft draft-ietf-rtcweb-rtp-usage-25, June 2015. |
| [I-D.ietf-rtcweb-audio-codecs-for-interop] | Proust, S., "Additional WebRTC audio codecs for interoperability.", Internet-Draft draft-ietf-rtcweb-audio-codecs-for-interop-05, February 2016. |