Internet Engineering Task Force (IETF) S. Proust, Ed.
Request for Comments: 7875 Orange
Category: Informational May 2016
ISSN: 2070-1721
Additional WebRTC Audio Codecs for Interoperability
Abstract
To ensure a baseline of interoperability between WebRTC endpoints, a
minimum set of required codecs is specified. However, to maximize
the possibility of establishing the session without the need for
audio transcoding, it is also recommended to include in the offer
other suitable audio codecs that are available to the browser.
This document provides some guidelines on the suitable codecs to be
considered for WebRTC endpoints to address the use cases most
relevant to interoperability.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It has been approved for publication by the Internet
Engineering Steering Group (IESG). Not all documents approved by the
IESG are a candidate for any level of Internet Standard; see
Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7875.
1. Introduction
As indicated in [OVERVIEW], it has been anticipated that WebRTC will
not remain an isolated island and that some WebRTC endpoints will
need to communicate with devices used in other existing networks with
the help of a gateway. Therefore, in order to maximize the
possibility of establishing the session without the need for audio
transcoding, it is recommended in [RFC7874] to include in the offer
other suitable audio codecs beyond those that are mandatory to
implement. This document provides some guidelines on the suitable
codecs to be considered for WebRTC endpoints to address the use cases
most relevant to interoperability.
The codecs considered in this document are recommended to be
supported and included in the offer, only for WebRTC endpoints for
which interoperability with other non-WebRTC endpoints and non-
WebRTC-based services is relevant as described in Sections 4.1.2,
4.2.2, and 4.3.2. Other use cases may justify offering other
additional codecs to avoid transcoding.
2. Definitions and Abbreviations
o Legacy networks: In this document, legacy networks encompass the
conversational networks that are already deployed like the PSTN,
the PLMN, the IP/IMS networks offering VoIP services, including
3GPP "4G" Evolved Packet System [TS23.002] supporting voice over
LTE (VoLTE) radio access [IR.92].
o WebRTC endpoint: A WebRTC endpoint can be a WebRTC browser or a
WebRTC non-browser (also called "WebRTC device" or "WebRTC native
application") as defined in [OVERVIEW].
o AMR: Adaptive Multi-Rate
o AMR-WB: Adaptive Multi-Rate Wideband
o CAT-iq: Cordless Advanced Technology - internet and quality
o DECT: Digital Enhanced Cordless Telecommunications
o IMS: IP Multimedia Subsystems
o LTE: Long Term Evolution (3GPP "4G" wireless data transmission
standard)
o MOS: Mean Opinion Score, defined in ITU-T Recommendation P.800
[P.800]
o PSTN: Public Switched Telephone Network
o PLMN: Public Land Mobile Network
o VoLTE: Voice over LTE
3. Rationale for Additional WebRTC Codecs
The mandatory implementation of Opus [RFC6716] in WebRTC endpoints
can guarantee codec interoperability (without transcoding) at state-
of-the-art voice quality (better than narrow-band "PSTN" quality)
between WebRTC endpoints. The WebRTC technology is also expected to
be used to communicate with other types of endpoints using other
technologies. It can be used for instance as an access technology to
VoLTE services (Voice over LTE as specified in [IR.92]) or to
interoperate with fixed or mobile Circuit-Switched or VoIP services
like mobile Circuit-Switched voice over 3GPP 2G/3G mobile networks
[TS23.002] or DECT-based VoIP telephony [EN300175-1]. Consequently,
a significant number of calls are likely to occur between terminals
supporting WebRTC endpoints and other terminals like mobile handsets,
fixed VoIP terminals, and DECT terminals that do not support WebRTC
endpoints nor implement Opus. As a consequence, these calls are
likely to be either of low narrow-band PSTN quality using G.711
[G.711] at both ends or affected by transcoding operations. The
drawback of such transcoding operations are listed below:
o Degraded user experience with respect to voice quality: voice
quality is significantly degraded by transcoding. For instance,
the degradation is around 0.2 to 0.3 MOS for most of the
transcoding use cases with AMR-WB codec (Section 4.1) at 12.65
kbit/s and in the same range for other wideband transcoding cases.
It should be stressed that if G.711 is used as a fallback codec
for interoperation, wideband voice quality will be lost. Such
bandwidth reduction effect down to narrow band clearly degrades
the user-perceived quality of service leading to shorter and less
frequent calls. Such a switch to G.711 is a choice for customers.
If transcoding is performed between Opus and any other wideband
codec, wideband communication could be maintained but with
degraded quality (MOSs of transcoding between AMR-WB at 12.65
kbit/s and Opus at 16 kbit/s in both directions are significantly
lower than those of AMR-WB at 12.65 kbit/s or Opus at 16 kbit/s).
Furthermore, in degraded conditions, the addition of defects, like
(a) audio artifacts due to packet losses and (b) audio effects due
to the cascading of different packet loss recovery algorithms, may
result in a quality below the acceptable limit for the customers.
o Degraded user experience with respect to conversational
interactivity: the degradation of conversational interactivity is
due to the increase of end-to-end latency for both directions that
is introduced by the transcoding operations. Transcoding requires
full de-packetization for decoding of the media stream (including
mechanisms of de-jitter buffering and packet loss recovery) then
re-encoding, re-packetization, and resending. The delays produced
by all these operations are additive and may increase the end-to-
end delay up to 1 second, much beyond the acceptable limit.
o Additional cost in networks: transcoding places important
additional cost on network gateways mainly related to codec
implementation, codecs licenses, deployment, testing and
validation cost. It must be noted that transcoding of wideband to
wideband would require more CPU processing and be more costly than
transcoding between narrowband codecs.
4. Additional Suitable Codecs for WebRTC
The following are considered relevant codecs with respect to the
general purpose described in Section 3. This list reflects the
current status of foreseen use cases for WebRTC. It is not
limitative; it is open to inclusion of other codecs for which
relevant use cases can be identified. It's recommended to include
codecs (in addition to Opus and G.711) according to the foreseen
interoperability cases to be addressed.
4.1. AMR-WB
4.1.1. AMR-WB General Description
The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP-defined speech
codec that is mandatory to implement in any 3GPP terminal that
supports wideband speech communication. It is being used in circuit-
switched mobile telephony services and new multimedia telephony
services over IP/IMS. It is specially used for voice over LTE as
specified by GSMA in [IR.92]. More detailed information on AMR-WB
can be found in [IR.36]. References for AMR-WB-related
specifications including the detailed codec description and source
code are in [TS26.171], [TS26.173], [TS26.190], and [TS26.204].
4.1.2. WebRTC-Relevant Use Case for AMR-WB
The market of personal voice communication is driven by mobile
terminals. AMR-WB is now very widely implemented in devices and
networks offering "HD voice", where "HD" stands for "High
Definition". Consequently, a high number of calls are likely to
occur between WebRTC endpoints and mobile 3GPP terminals offering
AMR-WB. Thus, the use of AMR-WB by WebRTC endpoints would allow
transcoding-free interoperation with all mobile 3GPP wideband
terminals. Besides, WebRTC endpoints running on mobile terminals
(smartphones) may reuse the AMR-WB codec already implemented on those
devices.
4.1.3. Guidelines for AMR-WB Usage and Implementation with WebRTC
The payload format to be used for AMR-WB is described in [RFC4867]
with a bandwidth-efficient format and one speech frame encapsulated
in each RTP packet. Further guidelines for implementing and using
AMR-WB and ensuring interoperability with 3GPP mobile services can be
found in [TS26.114]. In order to ensure interoperability with 4G/
VoLTE as specified by GSMA, the more specific IMS profile for voice
derived from [TS26.114] should be considered in [IR.92]. In order to
maximize the possibility of successful call establishment for WebRTC
endpoints offering AMR-WB, it is important that the WebRTC endpoints:
o Offer AMR in addition to AMR-WB, with AMR-WB listed first (AMR-WB
being a wideband codec) as the preferred payload type with respect
to other narrow-band codecs (AMR, G.711, etc.) and with a
bandwidth-efficient payload format preferred.
o Be capable of operating AMR-WB with any subset of the nine codec
modes and source-controlled rate operation.
o Offer at least one AMR-WB configuration with parameter settings as
defined in Table 6.1 of [TS26.114]. In order to maximize
interoperability and quality, this offer does not restrict the
codec modes offered. Restrictions on the use of codec modes may
be included in the answer.
4.2. AMR
4.2.1. AMR General Description
Adaptive Multi-Rate (AMR) is a 3GPP-defined speech codec that is
mandatory to implement in any 3GPP terminal that supports voice
communication. This includes both mobile phone calls using GSM and
3G cellular systems as well as multimedia telephony services over IP/
IMS and 4G/VoLTE, such as the GSMA voice IMS profile for VoLTE in
[IR.92]. References for AMR-related specifications including
detailed codec description and source code are [TS26.071],
[TS26.073], [TS26.090], and [TS26.104].
4.2.2. WebRTC-Relevant Use Case for AMR
A user of a WebRTC endpoint on a device integrating an AMR module
wants to communicate with another user that can only be reached on a
mobile device that only supports AMR. Although more and more
terminal devices are now "HD voice" and support AMR-WB; there are
still a high number of legacy terminals supporting only AMR
(terminals with no wideband / HD voice capabilities) that are still
in use. The use of AMR by WebRTC endpoints would consequently allow
transcoding free interoperation with all mobile 3GPP terminals.
Besides, WebRTC endpoints running on mobile terminals (smartphones)
may reuse the AMR codec already implemented on these devices.
4.2.3. Guidelines for AMR Usage and Implementation with WebRTC
The payload format to be used for AMR is described in [RFC4867] with
bandwidth efficient format and one speech frame encapsulated in each
RTP packet. Further guidelines for implementing and using AMR with
purpose to ensure interoperability with 3GPP mobile services can be
found in [TS26.114]. In order to ensure interoperability with 4G/
VoLTE as specified by GSMA, the more specific IMS profile for voice
derived from [TS26.114] should be considered in [IR.92]. In order to
maximize the possibility of successful call establishment for WebRTC
endpoints offering AMR, it is important that the WebRTC endpoints:
o Be capable of operating AMR with any subset of the eight codec
modes and source-controlled rate operation.
o Offer at least one configuration with parameter settings as
defined in Tables 6.1 and 6.2 of [TS26.114]. In order to maximize
the interoperability and quality, this offer shall not restrict
AMR codec modes offered. Restrictions on the use of codec modes
may be included in the answer.
4.3. G.722
4.3.1. G.722 General Description
G.722 [G.722] is an ITU-T-defined wideband speech codec. G.722 was
approved by the ITU-T in 1988. It is a royalty-free codec that is
common in a wide range of terminals and endpoints supporting wideband
speech and requiring low complexity. The complexity of G.722 is
estimated to 10 MIPS [EN300175-8], which is 2.5 to 3 times lower than
AMR-WB. In particular, G.722 has been chosen by ETSI DECT as the
mandatory wideband codec for New Generation DECT in order to greatly
increase the voice quality by extending the bandwidth from narrow
band to wideband. G.722 is the wideband codec required for terminals
that are certified as CAT-iq DECT, and version 2.0 of the CAT-iq
specifications have been approved by GSMA as the minimum requirements
for the "HD voice" logo usage on "fixed" devices, i.e., broadband
connections using the G.722 codec.
4.3.2. WebRTC-Relevant Use Case for G.722
G.722 is the wideband codec required for DECT CAT-iq terminals. DECT
cordless phones are still widely used to offer short-range wireless
connection to PSTN or VoIP services. G.722 has also been specified
by ETSI in [TS181005] as the mandatory wideband codec for IMS
multimedia telephony communication service and supplementary services
using fixed broadband access. The support of G.722 would
consequently allow transcoding-free IP interoperation between WebRTC
endpoints and fixed VoIP terminals including DECT CAT-iq terminals
supporting G.722. Besides, WebRTC endpoints running on fixed
terminals that implement G.722 may reuse the G.722 codec already
implemented on these devices.
4.3.3. Guidelines for G.722 Usage and Implementation with WebRTC
The payload format to be used for G.722 is defined in [RFC3551] with
each octet of the stream of octets produced by the codec to be octet-
aligned in an RTP packet. The sampling frequency for the G.722 codec
is 16 kHz, but the RTP clock rate is set to 8000 Hz in SDP to stay
backward compatible with an erroneous definition in the original
version of the RTP audio/video profile. Further guidelines for
implementing and using G.722 to ensure interoperability with
multimedia telephony services over IMS can be found in Section 7 of
[TS26.114]. Additional information about the G.722 implementation in
DECT can be found in [EN300175-8], and the full codec description and
C source code are in [G.722].
5. Security Considerations
Relevant security considerations can be found in [RFC7874], "WebRTC
Audio Codec and Processing Requirements". Implementers making use of
the additional codecs considered in this document are advised to also
refer more specifically to the "Security Considerations" sections of
[RFC4867] (for AMR and AMR-WB) and [RFC3551] (for the RTP audio/video
profile).
Acknowledgements
We would like to thank Magnus Westerlund, Barry Dingle, and Sanjay
Mishra who carefully reviewed the document and helped to improve it.
Contributors
The following individuals contributed significantly to this document:
o Stephane Proust, Orange, stephane.proust@orange.com
o Espen Berger, Cisco, espeberg@cisco.com
o Bernhard Feiten, Deutsche Telekom, Bernhard.Feiten@telekom.de
o Bo Burman, Ericsson, bo.burman@ericsson.com
o Kalyani Bogineni, Verizon Wireless,
Kalyani.Bogineni@VerizonWireless.com
o Mia Lei, Huawei, lei.miao@huawei.com
o Enrico Marocco, Telecom Italia, enrico.marocco@telecomitalia.it
though only the editor is listed on the front page.
Author's Address
Stephane Proust (editor)
Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: stephane.proust@orange.com
